diff --git a/.gitignore b/.gitignore index a559c3e4..a7708cab 100644 --- a/.gitignore +++ b/.gitignore @@ -1,6 +1,7 @@ /shairport-sync /*.o /*~ +*.xml~ /config.mk /config.h Makefile diff --git a/Makefile.am b/Makefile.am index eced4e63..fcf68059 100644 --- a/Makefile.am +++ b/Makefile.am @@ -1,7 +1,7 @@ SUBDIRS = man bin_PROGRAMS = shairport-sync -shairport_sync_SOURCES = shairport.c rtsp.c mdns.c mdns_external.c common.c rtp.c player.c alac.c audio.c audio_dummy.c audio_pipe.c +shairport_sync_SOURCES = shairport.c rtsp.c mdns.c mdns_external.c common.c rtp.c player.c alac.c audio.c AM_CFLAGS = -Wno-multichar @@ -26,6 +26,18 @@ if USE_SNDIO shairport_sync_SOURCES += audio_sndio.c endif +if USE_STDOUT +shairport_sync_SOURCES += audio_stdout.c +endif + +if USE_PIPE +shairport_sync_SOURCES += audio_pipe.c +endif + +if USE_DUMMY +shairport_sync_SOURCES += audio_dummy.c +endif + if USE_AO shairport_sync_SOURCES += audio_ao.c endif @@ -39,5 +51,16 @@ shairport_sync_SOURCES += mdns_dns_sd.c endif install-exec-hook: - [ -f /etc/init.d/shairport-sync ] || cp scripts/shairport-sync /etc/init.d/ - update-rc.d shairport-sync defaults 90 10 +if INSTALL_CONFIG_FILES + [ -e $(DESTDIR)/etc ] || mkdir $(DESTDIR)/etc + cp scripts/shairport-sync.conf $(DESTDIR)/etc/shairport-sync.conf.sample + [ -f $(DESTDIR)/etc/shairport-sync.conf ] || cp scripts/shairport-sync.conf $(DESTDIR)/etc/shairport-sync.conf +endif +if INSTALL_SYSTEMV + [ -e $(DESTDIR)/etc/init.d ] || mkdir -p $(DESTDIR)/etc/init.d + [ -f $(DESTDIR)/etc/init.d/shairport-sync ] || cp scripts/shairport-sync $(DESTDIR)/etc/init.d/ +endif +if INSTALL_SYSTEMD + [ -e $(DESTDIR)/usr/lib/systemd/system ] || mkdir -p $(DESTDIR)/usr/lib/systemd/system + cp scripts/shairport-sync.service $(DESTDIR)/usr/lib/systemd/system +endif diff --git a/README.md b/README.md index 228ebde8..add485d1 100644 --- a/README.md +++ b/README.md @@ -1,63 +1,89 @@ Shairport Sync ============= Shairport Sync emulates an AirPort Express for the purpose of streaming audio from iTunes, iPods, iPhones, iPads and AppleTVs. -Audio played by a Shairport Sync-powered device stays synchronised with the source and hence with similar devices playing the same source. Thus, for example, synchronised multi-room audio is possible without difficulty. (Hence the name Shairport Sync, BTW.) +Audio played by a Shairport Sync-powered device stays synchronised with the source and hence with similar devices playing the same source. In this way, synchronised multi-room audio is possible without difficulty. (Hence the name Shairport Sync, BTW.) Shairport Sync does not support AirPlay video or photo streaming. This branch -- "master" -- is the stable branch of Shairport Sync. To access the development version, please switch to the "development" branch. More Information -------- -Shairport Sync works by using timing information present in the audio data stream to keep in step with the source. It does this by monitoring and controlling the "latency" -- the time between when a sound is time-stamped at the source and when it is played by the audio output device. To measure latency precisely, it keeps its own clock synchronised with the clock used by the source, usually to within a fraction of a millisecond, using a variant of NTP synchronisation protocols. +---------- +Shairport Sync works by using timing information and timestamps present in data coming from the audio source (e.g. an iPhone) to "play" audio at exactly the right time. It does this by monitoring and controlling the *latency* — the time between a sound frame is supposed to be played, as specified by its `timestamp`, and the time when it is actually played by the audio output device, usually a Digital to Audio Converter (DAC). Timestamps are measured relative to the source computer's clocks, the `source clock`, but timing must be done relative to the clock of the computer running Shairport Sync, the `local clock`. The source and local clocks are synchronised, usually to within a fraction of a millisecond, using a variant of NTP synchronisation protocols. -To maintain the exact latency required, if an output device is running slow relative to the source, Shairport Sync will delete frames of audio to allow the device to keep up; if the device is running fast, Shairport Sync will insert frames to keep time. The number of frames inserted or deleted is so small as to be almost inaudible. Frames are inserted or deleted as necessary at pseudorandom intervals. Alternatively, with `libsoxr` support, Shairport Sync can resample the audio feed to ensure the output device can keep up. This is even less obtrusive than insertion and deletion but requires a good deal of processing power -- most embedded devices probably can't support it. +To maintain the exact latency required, if an output device is running slow relative to the source, Shairport Sync will delete frames of audio to allow the device to keep up. If the output device is running fast, Shairport Sync will insert frames to keep time. The number of frames inserted or deleted is so small as to be almost inaudible on normal audio material. Frames are inserted or deleted as necessary at pseudorandom intervals. Alternatively, with `libsoxr` support, Shairport Sync can resample the audio feed to ensure the output device can keep up. This is less obtrusive than insertion and deletion but requires a good deal of processing power — most embedded devices probably can't support it. The process of insertion/deletion or resampling is rather inelegantly called “stuffing”. -There are three default latency settings, chosen automatically. One latency (the "iTunes" latency) matches the latency used by recent versions of iTunes when playing audio. The "AirPlay" latency is used by older version of iTunes, by iOS devices and by iTunes and Quicktime Player when playing video. Finally, the "forkedDaapd" latency is used for forked-daapd sources. +There are four default latency settings, chosen automatically. One latency matches the latency used by recent versions of iTunes when playing audio and another matches the latency used by so-called "AirPlay" devices — iOS devices and iTunes and Quicktime Player when they are playing video. A third latency is used when the audio source is `forked-daapd`. The fourth latency is the default if no other latency is chosen and is used for older versions of iTunes. -Shairport Sync is a pretty substantial rewrite of Shairport 1.0 by James Laird and others -- please see https://github.com/abrasive/shairport/blob/master/README.md#contributors-to-version-1x for a list of the contributors to Shairport 1.x and Shairport 0.x. From a "heritage" point of view, Shairport Sync is a fork of Shairport 1.0. +Shairport Sync is a pretty substantial rewrite of the fantastic work done in Shairport 1.0 by James Laird and others — please see https://github.com/abrasive/shairport/blob/master/README.md#contributors-to-version-1x for a list of the contributors to Shairport 1.x and Shairport 0.x. From a "heritage" point of view, Shairport Sync is a fork of Shairport 1.0. -Shairport Sync works only with Linux and ALSA. The sound card you use must be capable of working with 44,100 samples per second interleaved PCM stereo (you'll get a message in the logfile if there's a problem). +Shairport Sync is designed for Linux and ALSA. It must have direct access to the output device, which must be a real sound card capable of working with 44,100 samples per second interleaved PCM stereo (you'll get a message in the logfile if there's a problem). For more about the motivation behind Shairport Sync, please see the wiki at https://github.com/mikebrady/shairport-sync/wiki. What else? -------------- -* Shairport Sync offers finer control at very top and very bottom of the volume range. See http://tangentsoft.net/audio/atten.html for a good discussion of audio "attenuators", upon which volume control in Shairport Sync is modelled. See also the diagram of the volume transfer function in the documents folder. -* Shairport Sync will mute properly if the hardware supports it. -* If Shairport Sync has to use software volume and mute controls, the response time is shorter than before -- now it responds in 0.15 seconds. -* Shairport Sync sends back a "busy" signal if it's already playing audio from another source, so other sources can't disrupt an existing Shairport Sync session. (If a source disappears without warning, the session automatically terminates after two minutes and the device becomes available again.) +* Better Volume Control — Shairport Sync offers finer control at very top and very bottom of the volume range. See http://tangentsoft.net/audio/atten.html for a good discussion of audio "attenuators", upon which volume control in Shairport Sync is modelled. See also the diagram of the volume transfer function in the documents folder. +* Hardware Mute — Shairport Sync will mute properly if the hardware supports it. +* Fast Response — With hardware volume control, response is instantaneous; otherwise the response time is 0.15 seconds. +* Non-Interruptible — Shairport Sync sends back a "busy" signal if it's already playing audio from another source, so other sources can't disrupt an existing Shairport Sync session. (If a source disappears without warning, the session automatically terminates after two minutes and the device becomes available again.) +* Metadata — Shairport Sync can be configured to deliver metadata, such as Album Name, Artist Name, Cover Art, etc. through a pipe to a recipient application program — see https://github.com/mikebrady/shairport-sync-metadata-reader for a sample recipient. +* Raw Audio — Shairport Sync can deliver raw PCM audio to standard output or to a pipe. This output is delivered synchronously with the source after the appropriate latency and is not interpolated or "stuffed" on its way through Shairport Sync. +* Autotools and Libtool Support — One important difference between Shairport Sync and other versions of Shairport is that the Shairport Sync build process uses GNU autotools and libtool to examine and configure the build environment — very important for cross compilation. Previous versions of Shairport looked at the current system to determine which packages were available, instead of looking at the target system for what packages were available. Status ------ -Shairport Sync works on standard Ubuntu laptops, on the Raspberry Pi with Raspian and with OpenWrt, and it runs on a Linksys NSLU2 using OpenWrt. It works with built-in audio and with a variety of USB-connected audio amplifiers and DACs, including a cheapo USB "3D Sound" dongle, a first generation iMic and a Topping TP30 amplifier with a USB DAC input. +Shairport Sync works on a wide variety of Linux devices. It works on standard Ubuntu laptops, on the Raspberry Pi with Raspian, Arch Linux and OpenWrt, and it runs on a Linksys NSLU2 and a TP-Link 710N using OpenWrt. It works with built-in audio and with a variety of USB-connected audio amplifiers and DACs, including a cheapo USB "3D Sound" dongle, a first generation iMic and a Topping TP30 amplifier with a USB DAC input. It will not work properly – if at all – with a PulseAudio (pseudo-)output device. -Shairport Sync runs well on the Raspberry Pi. It can drive the built-in sound card, and though you wouldn't mistake the output for HiFi, it's really not too shabby. USB-connected sound cards work well on the latest version of Raspian; however older versions of Raspian appear to suffer from a problem -- see http://www.raspberrypi.org/forums/viewtopic.php?t=23544, so it is wise to update. Shairport Sync works very well with the IQAudIO Pi-DAC -- see http://www.iqaudio.com. +Shairport Sync runs well on the Raspberry Pi. It can drive the built-in sound card, though the audio out of the card is of poor quality. USB-connected sound cards work well on recent versions of Raspian; however older versions of Raspian appear to suffer from a problem — see http://www.raspberrypi.org/forums/viewtopic.php?t=23544, so it is wise to update. Shairport Sync works well with the IQAudIO Pi-DAC — see http://www.iqaudio.com. At the time of writing, OpenWrt trunk does not support USB audio well on the Raspberry Pi. -Shairport Sync runs on Ubuntu and Debian inside VMWare Fusion 7 on a Mac, but synchronisation does not work -- possibly because the soundcard is being emulated. +Shairport Sync runs on Ubuntu, OpenWrt, Debian, Arch Linux and Fedora inside VMWare Fusion on a Mac, but synchronisation in inaccurate — possibly because the soundcard is being emulated. -Shairport Sync works only with the ALSA back end. You can try compiling the other back ends in as you wish, but it definitely will not work properly with them. Maybe someday... - -One other difference from other versions of Shairport is that the Shairport Sync build process uses GNU autotools and libtool to examine and configure the build environment -- very important for cross compilation. All previous versions looked in the current system to determine which packages were available, instead of looking at what packages were available in the target system. +Shairport Sync will output to alsa cards, to standard output and to pipes using appropriate backends. You can try compiling additional backends in as you wish, but it definitely will not work properly with them. Maybe someday... For information about changes and updates, please refer to the RELEASENOTES.md file in the distribution. +Note: Historically, Shairport Sync has taken its settings from command line arguments. While this is still the case, it does not always work well across distributions. Accordingly, from version 2.4 onwards, Shairport Sync reads settings from the file `/etc/shairport-sync.conf`. Access to new features will only be provided via the settings file. + Building And Installing --------------------- -Shairport Sync packages exist for a number of popular Linux distributions: -* OpenWrt: Shairport Sync is available as an optional package in trunk OpenWrt. -Alternatively, there's an OpenWrt package at https://github.com/mikebrady/shairport-sync-for-openwrt. +If you wish to install Shairport Sync on OpenWrt, Arch or Fedora platforms, please follow the appropriate instructions below. Otherwise follow the General Build Instructions. Then, when the progam has been installed, refer to the section on Configuring Shairport Sync that follows. -* Buildroot: Shairport Sync is available as an optional package in Buildroot. +**OpenWrt:** +If you're interested in Shairport Sync for OpenWrt, there's an OpenWrt package at https://github.com/mikebrady/shairport-sync-for-openwrt. OpenWrt doesn't support the IQaudIO Pi-DAC. -* Arch Linux: An Arch Linux installation package suitable for compilation on any architecture is available at [EliaCereda/shairport-sync-PKGBUILD](https://github.com/EliaCereda/shairport-sync-PKGBUILD). -Arch Linux installation packages, precompiled for the Intel 64-bit and Intel 32 bit architectures, are available in the official repo: https://www.archlinux.org/packages/community/x86_64/shairport-sync/. +**Arch Linux:** +An Arch Linux installation package is available at [EliaCereda/shairport-sync-PKGBUILD](https://github.com/EliaCereda/shairport-sync-PKGBUILD). -(Thanks to all package contributors!) +**Fedora:** +Install the toolchain and pre-requisites, if necessary: +``` +% sudo yum install make automake gcc gcc-c++ kernel-devel +% sudo yum install alsa-lib-devel autoconf automake avahi-devel libconfig-devel libdaemon-devel openssl-devel popt-devel soxr-devel +``` +Download the tarball from the "releases" tab on github or use `wget` and then use `rpmbuild`. This example is for version 2.4: +``` +% wget -O shairport-sync-2.4.tar.gz https://github.com/mikebrady/shairport-sync/archive/2.4.tar.gz +% rpmbuild -ta shairport-sync-2.4.tar.gz +``` +The `-ta` means "build all from this tarball". -Otherwise, follow these instructions. +The RPM will be built in a directory and will have a pathname like, for example, `~/rpmbuild/RPMS/i686/shairport-sync-2.4-1.fc22.i686.rpm` You should then install it with (for this example): +``` +%sudo rpm -i ~/rpmbuild/RPMS/i686/shairport-sync-2.4-1.fc22.i686.rpm +``` +You may have to manually create the directory `/var/shairport-sync` for the installation to succeed. Having edited the configuration file `/etc/shairport-sync.conf` as appropriate (see "Configuring Shairport Sync" below), enable and start the service with: +``` +%sudo systemctl enable shairport-sync.service +%sudo systemctl start shairport-sync.service +``` +Sincere thanks to all package contributors! + +**General Build Instructions** + +To build Shairport Sync from sources on Debian, Ubuntu, Raspian, etc. follow these instructions. The following libraries are required: * OpenSSL or PolarSSL @@ -67,19 +93,20 @@ The following libraries are required: * autoconf * libtool * libpopt +* libconfig Optional: * libsoxr -Many linux distributions have Avahi and OpenSSL already in place, so normally it probably makes sense to choose those options rather than tinysvcmdns or PolarSSL. Libsoxr is available in recent linux distributions, but it requires lots of processor power -- chances are an embedded processor won't be able to keep up. +Many Linux distributions have Avahi and OpenSSL already in place, so normally it probably makes sense to choose those options rather than tinysvcmdns or PolarSSL. Libsoxr is available in recent Linux distributions, but it requires lots of processor power — chances are an embedded processor won't be able to keep up. Assuming the usual build essentials and git, Debian, Ubuntu and Raspian users can get the basics with: -- `apt-get install autoconf libtool libdaemon-dev libasound2-dev libpopt-dev` +- `apt-get install autoconf libtool libdaemon-dev libasound2-dev libpopt-dev libconfig-dev` - `apt-get install avahi-daemon libavahi-client-dev` if you want to use Avahi (recommended). - `apt-get install libssl-dev` if you want to use OpenSSL and libcrypto, or use PolarSSL otherwise. - `apt-get install libpolarssl-dev` if you want to use PolarSSL, or use OpenSSL/libcrypto otherwise. -- `apt-get install libsoxr-dev` if you want support for libsoxr-based resampling. This library is not yet part of Raspian; instructions for how to build it from source are available at [LIBSOXR.md](https://github.com/mikebrady/shairport-sync/blob/2.2/LIBSOXR.md). +- `apt-get install libsoxr-dev` if you want support for libsoxr-based resampling. This library is not yet part of Raspian; instructions for how to build it from source are available at [LIBSOXR.md](https://github.com/mikebrady/shairport-sync/blob/2.3/LIBSOXR.md). Download Shairport Sync: @@ -94,141 +121,226 @@ $ autoreconf -i -f Choose the appropriate `--with-*` options: - `--with-alsa` for the ALSA audio back end. This is required. -- `--with-avahi` or `--with-tinysvcmdns` for mdns support. Avahi is a widely-used system-wide zero-configuration networking (zeroconf) service -- it may already be in your system. (There seem to be problems with the `--with-tinysvcmdns` option right now, so read the following with caution.) If you don't have Avahi, or similar, then consider including tinysvcmdns, which is a tiny zeroconf service embedded inside the shairport-sync application itself. To enable multicast for `tinysvcmdns`, you may have to add a default route with the following command: `route add -net 224.0.0.0 netmask 224.0.0.0 eth0` (substitute the correct network port for `eth0`). You should not have more than one zeroconf service on the same system -- bad things may happen, according to RFC 6762, §15. +- `--with-stdout` include an optional backend module to enable raw audio to be output through standard output (stdout). +- `--with-pipe` include an optional backend module to enable raw audio to be output through a unix pipe. +- `--with-avahi` or `--with-tinysvcmdns` for mdns support. Avahi is a widely-used system-wide zero-configuration networking (zeroconf) service — it may already be in your system. If you don't have Avahi, or similar, then consider including tinysvcmdns, which is a tiny zeroconf service embedded inside the shairport-sync application itself. To enable multicast for `tinysvcmdns`, you may have to add a default route with the following command: `route add -net 224.0.0.0 netmask 224.0.0.0 eth0` (substitute the correct network port for `eth0`). You should not have more than one zeroconf service on the same system — bad things may happen, according to RFC 6762, §15. - `--with-ssl=openssl` or `--with-ssl=polarssl` for encryption and related utilities using either OpenSSL or PolarSSL. - `--with-soxr` for libsoxr-based resampling. - `--with-piddir` for specifying where the PID file should be stored. This directory is normally chosen automatically. The directory must be writable. If you use this option, you may have to edit the init script to search for the PID file in your new location. +- `--with-metadata` to add support for Shairport Sync to pipe metadata to a compatible application of your choice. See https://github.com/mikebrady/shairport-sync-metadata-reader for a sample metadata reader. +- `--with-systemv` to install a System V init script at the `make install` stage. Default is not to to install. +- `--with-systemd` to install a systemd service description at the `make install` stage. Default is not to to install. +- `--with-configfile` to install a configuration file and a separate sample file at the `make install` stage. Default is to install. An existing `/etc/shairport-sync.conf` will not be overwritten. +- `--with-pkg-config` to use pkg-config to find libraries. Default is to use pkg-config — this option is for special purpose use. -Here is an example, suitable for most installations: +Here is an example, suitable for installations such as Ubuntu and Raspbian: -`$ ./configure --with-alsa --with-avahi --with-ssl=openssl --with-soxr` +`$ ./configure --with-alsa --with-avahi --with-ssl=openssl --with-metadata --with-soxr --with-systemv` -Omit the `--with-soxr` if the libsoxr library is not available. +Omit the `--with-soxr` if the libsoxr library is not available. For installation into a `systemd` system, replace the `--with-systemv` with `--with-systemd`. + +Enter: `$ make` -Run `$sudo make install` to install `shairport-sync` along with an init script which will automatically launch it at startup. The settings in the init script are the most basic defaults, so you will want to edit it -- the file is `/etc/init.d/shairport-sync` -- to give the service a name, use a different card, use the hardware mixer and volume control, etc. -- there are some examples in the script file. +to build the application. Next, run: -Man Page --------- -You can see a web version of the man page here: http://htmlpreview.github.io/?https://github.com/mikebrady/shairport-sync/blob/master/man/shairport-sync.html +``` +$sudo make install +$sudo update-rc.d shairport-sync defaults 90 10 +``` +to install `shairport-sync` along with a `man` page, a default configuration file and a System V startup script to launch it automatically at system startup. +The settings are the most basic defaults, so you will want to edit the configuration — the file is `/etc/shairport-sync.conf` — to give the service a name, +use a specific sound card and mixer control, etc. — there are some examples in the sample configuration file. + +*Man Page* + +You can view the man page here: http://htmlpreview.github.io/?https://github.com/mikebrady/shairport-sync/blob/master/man/shairport-sync.html Configuring Shairport Sync -------- -Shairport Sync installs a default configuration at `/etc/init.d/shairport-sync` (it won't replace an existing one) which should work in almost any system with a sound card. If there is a problem, it will be noted in the logfile, normally `/etc/log/syslog`. However, to get the most out of your software and hardware, you need to adjust some of the settings. +There are two logically distinct parts to getting Shairport Sync to run properly on your machine — (1) starting and stopping it and (2) ensuring it has the right settings. -To understand what follows, note that settings and parameters are passed to Shairport Sync through command line arguments. The purpose of the init script at `/etc/init.d/shairport-sync` is to launch or terminate Shairport Sync while passing the correct arguments to it. You are perfectly free to remove the init script and launch and terminate Shairport Sync yourself directly; indeed it is useful when you are troubleshooting the program. If you do launch it directly, make sure it isn't running already! +Starting and stopping automatically is taken care of differently in different versions of Linux. In the example above, when you run `$sudo make install`, a System V startup script is placed at `/etc/init.d/shairport-sync`. This will not be appropriate in Linuxes that use `systemd` such as Arch Linux, so please look at the separate installation scripts for those. -As well as the man page, don't forget you can launch Shairport Sync with the `-h` option to get some help on the options available. +To get the best from Shairport Sync, you’ll need to (1) give Shairport Sync a service name by which it will be seen in iTunes etc., (2) specify the output device to use and (3) specify the name of the mixer volume control to use to control the output level. To get values for (2) and (3) you might need to explore the ALSA output devices with a program like `alsamixer` or similar. +Shairport Sync reads settings from a configuration file at `/etc/shairport-sync.conf`. While it can also take configuration settings from command line options, it is recommended that you use the configuration file method. When you run `$sudo make install`, a default configuration is installed at `/etc/shairport-sync.conf` (it won't replace an existing one) which should work in almost any system with a sound card. -These are the important options: +A sample configuration file is installed (or updated) at `/etc/shairport-sync.conf.sample`. This contains all the setting groups and all the settings available, but they all are commented out (comments begin with `//`) so that default values are used. The file contains explanations of the settings, useful hints and suggestions. -The `-a` option allows you to specify the service name Shairport Sync will use on the network. If you don't specify a service name, the name `Shairport Sync on ...your computer's hostname...` will be used. +Settings in the configuration file are grouped. For instance, there is a `general` group within which you can use the `name` tag to set the service name. Suppose you wanted to set the name of the service to `Front Room`, give the service the password `secret` and used `libsoxr` interpolation, then you should do the following: -The `-S` option allows you to specify the kind of "stuffing" or interpolation to be used -- `basic` (default) for simple insertion/deletion or `soxr` for smoother resampling-based interpolation. +``` +general = +{ + name = "Front Room"; + password = "secret"; + interpolation = "soxr"; + // ... other general settings +}; +``` +The `alsa` group is used to specify properties of the output device. The most obvious setting is the name of the output device which you can set using the `output_device` tag. -The `--password` option allows you to password-protect access to the service provided by Shairport Sync. +The following `alsa` group settings are very important for maximum performance. If your audio device has a mixer that can be use to control the volume, then Shairport Sync can use it to give instant response to volume and mute commands and it can offload some work from the processor. +* The `mixer_control_name` tag allows you to specify the name of the mixer volume control. +* The `mixer_device` tag allows you specify where the mixer is. By default, the mixer is on the `output_device`, so you only need to use the `mixer_device` tag if the mixer is elsewhere. This can happen if you specify a *device* rather than a *card* with the `output_device` tag, because normally a mixer is associated with a *card* rather than a device. Suppose you wish to use the output device `5` of card `hw:0` and the mixer volume-control named `PCM`: -These may be also of interest: +``` +alsa = +{ + output_device = "hw:0,5"; + mixer_device = "hw:0"; + mixer_control_name = "PCM"; + // ... other alsa settings +}; +``` -The `-B`, `-E` and `-w` options allow you to specify a program to execute before (`-B`) and after (`-E`) Shairport Sync plays. This is to facilitate situations where something has to be done before and after playing, e.g. switching on an amplifier beforehand and switching it off afterwards. Use the `-w` option for Shairport Sync to wait until the respective commands have been completed before continuing. Please note that the full path to the programs must be specified, and script files will not be executed unless they are marked as executable and have the standard `#!/bin/...` first line. (This behaviour may be different from other Shairports.) +Shairport Sync can run programs just before it starts to play an audio stream and just after it finishes. You specify them using the `sessioncontrol` group settings `run_this_before_play_begins` and `run_this_after_play_ends`. This is to facilitate situations where something has to be done before and after playing, e.g. switching on an amplifier beforehand and switching it off afterwards. Set the `wait_for_completion` value to `"yes"` for Shairport Sync to wait until the respective commands have been completed before continuing. +Please note that the full path to the programs must be specified, and script files will not be executed unless they are marked as executable and have the standard `#!/bin/...` first line. (This behaviour may be different from other Shairports.) + +*Command Line Arguments* + +You can use command line arguments to provide settings to Shairport Sync as before. For full information, please read the Shairport Sync `man` page, also available at http://htmlpreview.github.io/?https://github.com/mikebrady/shairport-sync/blob/update-documentation/man/shairport-sync.html. + +Apart from the following options, all command line options can be replaced by settings in the configuration file. Here is a brief description of command line options that are not replicated by settings in the settings file. + +* The `-c` option allows you to specify the location of the configuration file — default is `/etc/shairport-sync.conf`. * The `-V` option gives you version information about Shairport Sync and then quits. -* The `-k` option causes Shairport Sync to kill an existing Shairport Sync daemon and then quit. You need to have sudo privileges for this. -* The `-v` option causes Shairport Sync to print some information and debug messages. * The `-d` option causes Shairport Sync to properly daemonise itself, that is, to run in the background. You may need sudo privileges for this. +* The `-k` option causes Shairport Sync to kill an existing Shairport Sync daemon. You may need to have sudo privileges for this. -Apart from arguments of Shairport Sync, there are also arguments for controlling the ALSA audio system. ALSA arguments follow a `--` on the command line -- see the examples below for layout of command line. -These are important because you use them to specify the actual audio device you wish to use and you give Shairport Sync important information about the capabilities of the device. The important ALSA arguments are: +The System V init script at `/etc/init.d/shairport-sync` has a bare minimum : +`-d`. Basically all it does is put the program in daemon mode. The program will read its settings from the configuration file, normally `/etc/shairport-sync.conf`. -* The `-d` option which allows you to specify the audio device to use. Typical values would be `default` (default), `hw:0`, `hw:1`, etc. Those examples are specifying which *soundcard* to use; the actual output device used is the card's default, typically output device 0. You could specify, for example, device 5 on card hw:0 with `-d hw:0,5`. +Examples +-------- -The following settings are very important for maximum performance. If your audio device has a hardware mixer and volume control, then Shairport Sync can use it to give faster response to volume and mute commands and it can offload some work from the processor. -* The `-m` option allows you specify where the mixer is. By default, the mixer is to be found where you specify with the `-d` option, so you only need to use the `-m` option if the mixer is elsewhere. This can happen if you specify a *device* rather than a *card* with the `-d` option, because normally a mixer is associated with a *card* rather than a device. For example, if you specified that the output device was device 5 of card hw:0 and if the mixer was associated with the card, you would write `-d hw:0,5 -m hw:0`. -* The `-t` option allows you to specify the type of audio mixer -- `software` (default) or `hardware`. -* The `-c` option allows you to specify the name of the volume control on the hardware mixer. +Here are some examples of complete configuration files. -The init script at `/etc/init.d/shairport-sync` has a bare minimum set of options (see line 60): +``` +general = { + name = "Joe's Stereo"; +}; -`-d` +alsa = { + output_device = "hw:0"; +}; +``` -Basically all it does is put the program in daemon mode, selects the default output device and uses a software volume control. +This gives the service a particular name — "Joe's Stereo" and specifies that audio device hw:0 be used. -*Examples* +For best results — including getting true mute and instant response to volume control and pause commands — you should access the hardware volume controls. Use `amixer` or `alsamixer` or similar to discover the name of the volume controller to be used after the `-c` option. -Here are some examples of complete commands. If you are modifying the init script, you don't need the `shairport-sync` at the start, but you should include the `-d` option, as it puts the program into daemon mode. There are some commented-out examples in the init script -- see lines 61--63. +Here is an example for for a Raspberry Pi using its internal soundcard — device hw:0 — that drives the headphone jack: +``` +general = { + name = "Mike's Boombox"; +}; -- `shairport-sync -d -a "Joe's Stereo" -- -d hw:0` - -This gives the service a particular name -- "Joe's Stereo" and specifies that audio device hw:0 be used. - - -For best results -- including getting true mute and instant response to volume control and pause commands -- you should access the hardware volume controls. Use `amixer` or `alsamixer` or similar to discover the name of the volume controller to be used after the `-c` option. - -Here is an example for for a Raspberry Pi using its internal soundcard -- device hw:0 -- that drives the headphone jack: - -- `shairport-sync -d -a "Mike's Boombox" -- -d hw:0 -t hardware -c PCM` +alsa = { + output_device = "hw:0"; + mixer_control_name = "PCM"; +}; +``` Here is an example of using soxr-based resampling and driving a Topping TP30 Digital Amplifier, which has an integrated USB DAC and which is connected as audio device `hw:1`: +``` +general = { + name = "Kitchen"; + interpolation = "soxr"; +}; -- `shairport-sync -d -a Kitchen -S soxr -- -d hw:1 -t hardware -c PCM` +alsa = { + output_device = "hw:1"; + mixer_control_name = "PCM"; +}; +``` For a cheapo "3D Sound" USB card (Stereo output and input only) on a Raspberry Pi: +``` +general = { + name = "Front Room"; +}; -- `shairport-sync -d -a "Front Room" -- -d hw:1 -t hardware -c Speaker` +alsa = { + output_device = "hw:1"; + mixer_control_name = "Speaker"; +}; +``` For a first generation Griffin iMic on a Raspberry Pi: +``` +general = { + name = "Attic"; +}; -- `shairport-sync -d -a "Attic" -- -d hw:1 -t hardware -c PCM` +alsa = { + output_device = "hw:1"; + mixer_control_name = "PCM"; +}; +``` -For an NSLU2, which has no internal soundcard, there appears to be a bug in ALSA -- you can not specify a device other than "default". Thus: +For an NSLU2, which has no internal soundcard, there appears to be a bug in ALSA — you can not specify a device other than "default". Thus: On an NSLU2, to drive a first generation Griffin iMic: +``` +general = { + name = "Den"; +}; -- `shairport-sync -d -a "Den" -- -t hardware -c PCM` +alsa = { + mixer_control_name = "PCM"; +}; +``` On an NSLU2, to drive the "3D Sound" USB card: +``` +general = { + name = "TV Room"; +}; -- `shairport-sync -d -a "TV Room" -- -t hardware -c Speaker` +alsa = { + mixer_control_name = "Speaker"; +}; +``` Latency ------- -Latency is the exact time from a sound signal's original timestamp until that signal actually "appears" on the output of the DAC, irrespective of any internal delays, processing times, etc. in the computer. From listening tests, it seems that there are two latencies in current use: +Latency is the exact time from a sound signal's original timestamp until that signal actually "appears" on the output of the audio output device, usually a Digital to Audio Converter (DAC), irrespective of any internal delays, processing times, etc. in the computer. From listening tests, it seems that there are three latencies in current use: * If the source is iTunes 10 or later, a latency of 99,400 frames seems to bring Shairport Sync into exact synchronisation both with the speakers on the iTunes computer itself and with AirPort Express receivers. * If the source is an AirPlay device, the latency seems to be exactly 88,200 frames. AirPlay devices include AppleTV, iPod, iPad and iPhone and Quicktime Player on Mac. * If the source is a `forked-daapd`-powered device, the latency seems to be exactly 99,400 frames. * If the source cannot be identified as AirPlay or as iTunes 10 or later, then the default latency of 88,200 frames seems to work in general. Note that some third party programs masquerade as older versions of iTunes. -Shairport Sync uses the latencies described above as defaults. You shouldn't need to change them, but occasionally problems arise when you are trying to synchronise with speaker systems -- typically surround-sound home theatre systems -- that have their own inherent delays. You can set the default latency with the `-L` or `--latency` option (e.g. `-L 99400` or `--latency=99400`). You can set your own iTunes 10 (or later) latency with the `-i` or `--iTunesLatency` option. Similarly you can set an AirPlay latency with the `-A` or `--AirPlayLatency` option and the forked-daapd latency with the `--forkedDaapdLatency` option. +Shairport Sync uses the latencies described above as defaults. You shouldn't need to change them. + +Problems can arise when you are trying to synchronise with speaker systems — typically surround-sound home theatre systems — that have their own inherent delays. You can compensate for an inherent delay using the `alsa` group `audio_backend_latency_offset`. Set this offset (in frames) to compensate for a fixed delay in the audio back end, for example, if the output device delays by 100 ms, set this to -4410. Resynchronisation ------------- Shairport Sync actively maintains synchronisation with the source. -If synchronisation is lost -- say due to a busy source or a congested network -- Shairport Sync will mute its output and resynchronise. The loss-of-sync threshold is a very conservative 50 ms -- i.e. the actual time and the expected time must differ by more than 50 ms to trigger a resynchronisation. Smaller disparities are corrected by insertions or deletions, as described above. -* You can vary the resync threshold, or turn resync off completely, with the `-r` opton. +If synchronisation is lost — say due to a busy source or a congested network — Shairport Sync will mute its output and resynchronise. The loss-of-sync threshold is a very conservative 50 ms — i.e. the actual time and the expected time must differ by more than 50 ms to trigger a resynchronisation. Smaller disparities are corrected by insertions or deletions, as described above. +* You can vary the resync threshold, or turn resync off completely, with the `general` `resync_threshold` setting. Tolerance --------- -Playback synchronisation is allowed to wander a small amount before attempting to correct it. The default is 88 frames, i.e. 2 ms. The smaller the tolerance, the more likely it is that overcorrection will occur. Overcorrection is when more corrections (insertions and deletions) are made than are strictly necessary to keep the stream in sync. Use the --statistics option to monitor correction levels. Corrections should not greatly exceed net corrections. -* You can vary the tolerance with the `--tolerance` option. +Playback synchronisation is allowed to wander a small amount before attempting to correct it. The default is 88 frames, i.e. 2 ms. The smaller the tolerance, the more likely it is that overcorrection will occur. Overcorrection is when more corrections (insertions and deletions) are made than are strictly necessary to keep the stream in sync. Use the statistics setting to monitor correction levels. Corrections should not greatly exceed net corrections. +* You can vary the tolerance with the `general` `drift` setting. Some Statistics --------------- -If you add the option `--statistics`, e.g. as follows for the Raspberry Pi with "3D Sound" card: - -`shairport-sync -a "Shairport Sync" --statistics -- -d hw:1 -t hardware -c Speaker` - -it will print statistics like this occasionally on the console (or in the logfile if running in daemon mode): +If you turn on the `general` `statistics` setting, statistics like this will be printed at intervals on the console (or in the logfile if running in daemon mode): `Sync error: -35.4 (frames); net correction: 24.2 (ppm); corrections: 24.2 (ppm); missing packets 0; late packets 5; too late packets 0; resend requests 6; min DAC queue size 4430.` -"Sync error" is the average deviation from exact synchronisation. The example above indicates that the output is on average 35.4 frames ahead of exact synchronisation. Sync is allowed to wander by the tolerance -- 88 frames (± 2 milliseconds) by default -- before a correction will be made. +"Sync error" is the average deviation from exact synchronisation. The example above indicates that the output is on average 35.4 frames ahead of exact synchronisation. Sync is allowed to wander by the tolerance — 88 frames (± 2 milliseconds) by default — before a correction will be made. -"Net correction" is actually the net sum of corrections -- the number of frame insertions less the number of frame deletions -- given as a moving average in parts per million. After an initial settling period, it represents the divergence between the source clock and the sound device's clock. The example above indicates that the output DAC's clock is running 24.2 ppm faster than the source's clock. +"Net correction" is actually the net sum of corrections — the number of frame insertions less the number of frame deletions — given as a moving average in parts per million. After an initial settling period, it represents the drift — the divergence between the rate at which frames are generated at the source and the rate at which the output device consumes them. The example above indicates that the output device is consuming frames 24.2 ppm faster than the source is generating them. -"Corrections" is the number of frame insertions plus the number of frame deletions (i.e. the total number of corrections), given as a moving average in parts per million. The closer this is to the absolute value of the drift, the fewer "unnecessary" corrections that are being made. Third party programs tend to have much larger levels of corrections. +"Corrections" is the number of frame insertions plus the number of frame deletions (i.e. the total number of corrections), given as a moving average in parts per million. The closer this is to the net corrections, the fewer "unnecessary" corrections that are being made. Third party programs tend to have much larger levels of corrections. For reference, a drift of one second per day is approximately 11.57 ppm. Left uncorrected, even a drift this small between two audio outputs will be audible after a short time. The above sample is from a second-generation iPod driving the Raspberry Pi which is connected over Ethernet. diff --git a/RELEASENOTES.md b/RELEASENOTES.md index 0bd26daf..0612da14 100644 --- a/RELEASENOTES.md +++ b/RELEASENOTES.md @@ -1,3 +1,179 @@ +Version 2.4 +---- +**Stable release** + +This stable release is the culmination of the 2.3.X sequence of development releases. + +**Change Summary** + +Changes from the previous stable version -- 2.2.5 -- are summarised here: + * Settings are now read from a configuration file. Command-line settings are supported but discouraged. + * Metadata is now supported -- it can be delivered to a unix pipe for processing by a helper application. See https://github.com/mikebrady/shairport-sync-metadata-reader for a sample metadata reader. + * Raw PCM audio can be delivered to standard output ("stdout") or to a unix pipe. The internal architecture has changed considerably to support this. + * Support for compilation on OpenWrt back to Attitude Adjustment. + * Version 2.4 uses the libconfig library. + * Runs on a wider range of platforms, including Arch Linux and Fedora. + * Bug fixes. + +Please note that building instructions have changed slightly from the previous version. +Also, the `-t hardware/software` option has been deprecated in the alsa back end. + +Version 2.3.13 +---- +**Note** +* We're getting ready to release the development branch as the new, stable, master branch at 2.4. If you're packaging Shairport Sync, you might prefer to wait a short while as we add a little polish before the release. + +**Changes** +* Harmonise version numbers on the release and on the `shairport.spec` file used in Fedora. + +Version 2.3.12 +---- +**Note** +* We're getting ready to release the development branch as the new, stable, master branch at 2.4. If you're packaging Shairport Sync, you might prefer to wait a short while as we add a little polish before the release. + +**Changes** +* `update-rc.d` has been removed from the installation script for System V because it causes problems for package makers. It's now noted in the user installation instructions. +* The `alsa` group `mixer_type` setting is deprecated and you should stop using it. Its functionality has been subsumed into `mixer_name` – when you specify a `mixer_name` it automatically chooses the `hardware` mixer type. + + +**Enhancements** +* Larger range of interpolation. Shairport Sync has previously constrained not to make interpolations ("corrections") of more than about 1 per 1000 real frames. This contraint has been relaxed, and it is now able to make corrections of up to 1 in 352 real frames. This might result in a faster and undesirably sudden correction early during a play session, so a number of further changes have been made. The full set of these changes is as follows: + * No corrections happen for the first five seconds. + * Corrections of up to about 1 in 1000 for the next 25 seconds. + * Corrections of up to 1 in 352 thereafter. + +**Documentation Update** +* Nearly there with updates concerning the configuration file. + +Version 2.3.11 +---- +Documentation Update +* Beginning to update the `man` document to include information about the configuration file. It's pretty sparse, but it's a start. + +Version 2.3.10 +---- +Bug fix +* The "pipe" backend used output code that would block if the pipe didn't have a reader. This has been replaced by non-blocking code. Here are some implications: + * When the pipe is created, Shairport Sync will not block if a reader isn't present. + * If the pipe doesn't have a reader when Shairport Sync wants to output to it, the output will be discarded. + * If a reader disappears while writing is occuring, the write will time out after five seconds. + * Shairport Sync will only close the pipe on termination. + +Version 2.3.9 +---- +* Bug fix + * Specifying the configuration file using a *relative* file path now works properly. + * The debug verbosity requested with `-v`, `-vv`, etc. is now honoured before the configuration file is read. It is read and honoured from when the command line arguments are scanned the first time to get a possible configuration file path. + +Version 2.3.8 +---- +* Annoying changes you must make + * You probably need to change your `./configure` arguments. The flag `with-initscript` has changed to `with-systemv`. It was previously enabled by default; now you must enable it explicitly. + +* Changes + * Added limited support for installing into `systemd` and Fedora systems. For `systemd` support, use the configuration flag `--with-systemd` in place of `--with-systemv`. The installation does not do everything needed, such as defining special users and groups. + * Renamed `with-initscript` configuration flag to `with-systemv` to describe its role more accurately. + * A System V startup script is no longer installed by default; if you want it, ask for it with the `--with-systemv` configuration flag. + * Added limited support for FreeBSD. You must specify `LDFLAGS='-I/usr/local/lib'` and `CPPFLAGS='-L/usr/local/include'` before running `./configure --with-foo etc.` + * Removed the `-configfile` annotation from the version string because it's no longer optional; it's always there. + * Removed the `dummy`, `pipe` and `stdout` backends from the standard build – they are now optional and are no longer automatically included in the build. + +* Bug fixes + * Allow more stack space to prevent a segfault in certain configurations (thanks to https://github.com/joerg-krause). + * Add missing header files(thanks to https://github.com/joerg-krause). + * Removed some (hopefully) mostly silent bugs from the configure.ac file. + +Version 2.3.7 +---- +* Changes + * Removed the two different buffer lengths for the alsa back end that made a brief appearance in 2.3.5. +* Enhancements + * Command line arguments are now given precedence over config file settings. This conforms to standard unix practice. + * A `–without-pkg-config` configuration argument now allows for build systems, e.g. for older OpenWrt builds, that haven't fully implemented it. There is still some unhappiness in arch linux builds. +* More + * Quite a bit of extra diagnostic code was written to investigate clock drift, DAC timings and so on. It was useful but has been commented out. If might be useful in the future. + +Version 2.3.5 +---- +* Changes + * The metadata item 'sndr' is no longer sent in metadata. It's been replaced by 'snam' and 'snua' -- see below. +* Enhancements + * When a play session is initiated by a source, it attempts to reserve the player by sending an "ANNOUNCE" packet. Typically, a source device name and/or a source "user agent" is sent as part of the packet. The "user agent" is usually the name of the sending application along with some more information. If metadata is enabled, the source name, if provided, is emitted as a metadata item with the type `ssnc` and code `snam` and similarly the user agent, if provided, is sent with the type `ssnc` and code `snua`. + * Two default buffer lengths for ALSA -- default 6615 frames if a software volume control is used, to minimise the response time to pause and volume control changes; default 22050 frames if a hardware volume control is used, to give more resilience to timing problems, sudden processor loading, etc. This is especially useful if you are processing metadata and artwork on the same machine. + * Extra metadata: when a play session starts, the "Active-Remote" and "DACP-ID" fields -- information that can be used to identify the source -- are provided as metadata, with the type `ssnc` and the codes `acre` and `daid` respectively. The IDs are provided as strings. + * Unencrypted audio data. The iOS player "Whaale" attempts to send unencrypted audio, presumably to save processing effort; if unsuccessful, it will send encrypted audio as normal. Shairport Sync now recognises and handles unencrypted audio data. (Apparently it always advertised that it could process unencrypted audio!) + * Handle retransmitted audio in the control channel. When a packet of audio is missed, Shairport Sync will ask for it to be retransmitted. Normally the retransmitted audio comes back the audio channel, but "Whaale" sends it back in the control channel. (I think this is a bug in "Whaale".) Shairport Sync will now correctly handle retransmitted audio packets coming back in the control channel. +* Bugfixes + * Generate properly-formed `..` items of information. + +Version 2.3.4 +---- +* Enhancement + * When a play session starts, Shairport Sync opens three UDP ports to communicate with the source. Until now, those ports could be any high numbered port. Now, they are located within a range of 100 port locations starting at port 6001. The starting port and the port range are settable by two new general settings in `/etc/shairport-sync.conf` -- `udp_port_base` (default 6001) and `udp_port_range` (default 100). To retain the previous behaviour, set the `udp_port_base` to `0`. +* Bugfixes + * Fix an out-of-stack-space error that can occur in certain cases (thanks to https://github.com/joerg-krause). + * Fix a couple of compiler warnings (thanks to https://github.com/joerg-krause). + * Tidy up a couple of debug messages that were emitting misleading information. + +Version 2.3.3.2 +---- +* Bugfix -- fixed an error in the sample configuration file. + +Version 2.3.3.1 +---- +* Enhancement + * Metadata format has changed slightly -- the format of each item is now `........`, where the `..` part is present if the length is non-zero. The change is that everything is now enclosed in an `..` pair. + +Version 2.3.2 and 2.3.3 +---- +These releases were faulty and have been deleted. + +Version 2.3.1 +----- +Some big changes "under the hood" have been made, leading to limited support for unsynchronised output to `stdout` or to a named pipe and continuation of defacto support for unsynchronised PulseAudio. Also, support for a configuration file in preference to command line options, an option to ignore volume control and other improvements are provided. + +In this release, Shairport Sync gains the ability to read settings from `/etc/shairport-sync.conf`. +This gives more flexibility in adding features gives better compatability across different versions of Linux. +Existing command-line options continue to work, but some will be deprecated and may disappear in a future version of Shairport Sync. New settings will only be available via the configuration file. + +Note that, for the present, settings in the configuration will have priority over command line options for Shairport Sync itself, in contravention of the normal unix convention. Audio back end command line options, i.e. those after the `--`, have priority over configuration file settings for the audio backends. + +In moving to the the use of a configuration file, some "housekeeping" is being done -- some logical corrections and other small changes are being made to option names and modes of operations, so the settings in the configuration file do not exactly match command line options. + +When `make install` is executed, a sample configuration is installed or updated at `/etc/shairport-sync.conf.sample`. The same file is also installed as `/etc/shairport-sync.conf` if that file doesn't already exist. To prevent the configuration files being installed, use the configuration option `--without-configfiles`. + +* Pesky Change You Must Do Something About + +If you are using metadata, please note that the option has changed somewhat. The option `-M` has a new long name equivalent: `--metadata-pipename` and the argument you provide must now be the full name of the metadata pipe, e.g. `-M /tmp/shairport-sync-metadata`. + +* Enhancements + * Shairport Sync now reads settings from the configuration file `/etc/shairport-sync.conf`. This has settings for most command-line options and it's where any new settings will go. A default configuration file will be installed if one doesn't exist, and a sample file configuration file is always installed or updated. Details of settings are provided in the sample file. Shairport Sync relies on the `libconfig` library to read configuration files. For the present, you can disable the new feature (and save the space taken up by `libconfig`) by using the configure option `--without-configfile-support`. + * New command-line option `-c ` or `--configfile=` allows you to specify a configuration file other than `/etc/shairport-sync.conf`. + * Session Timeout and Allow Session Interruption can now be set independently. This is really some "housekeeping" as referred to above -- it's a kind of a bug fix, where the bug in question is an inappropriate connection of the setting of two parameters. To explain: (1) By default, when a source such as iTunes starts playing to the Shairport Sync device, any other source attempting to start a play session receives a "busy" signal. If a source disappears without warning, Shairport Sync will wait for 120 seconds before dropping the session and allowing another source to start a play session. (2) The command-line option `-t` or `--timeout` allows you to set the wait time before dropping the session. If you set this parameter to `0`, Shairport Sync will not send a "busy" signal, thus allowing another source to interrupt an existing one. (3) The problem is that if you set the parameter to `0`, a session will never be dropped if the source disappears without warning. + The (obvious) fix for this is to separate the setting of the two parameters, and this is now done in the configuration file `/etc/shairport-sync.conf` -- please see the settings `allow_session_interruption` and `session_timeout`. The behaviour of the `-t` and `--timeout` command-line options is unchanged but deprecated. + * New Option -- "Ignore Volume Control" ('ignore_volume_control'). If you set this to "yes", the output from Shairport Sync is always set at 100%. This is useful when you want to set the volume locally. Available via the settings file only. + * Statistics option correctly reports when no frames are received in a sampling interval and when output is not being synchronised. + * A new, supported audio back end called `stdout` provides raw 16-bit 44.1kHz stereo PCM output. To activate, set `output_backend = "stdout"` in the general section of the configuration file. Output is provided synchronously with the source feed. No stuffing or stripping is done. If you are feeding it to an output device that runs slower or faster, you'll eventually get buffer overflow or underflow in that device. To include support for this back end, use the configuration option `--with-stdout`. + * Support for the `pipe` back end has been enhanced to provide raw 16-bit 44.1kHz stereo PCM output to a named pipe. To activate, set `output_backend = "pipe"` in the general section of the configuration and give the fully-specified pathname to the pipe in the pipe section of the configuration file -- see `etc/shairport-sync.conf.sample` for an example. No stuffing or stripping is done. If you are feeding it to an output device that runs slower or faster, you'll eventually get buffer overflow or underflow in that device. To include support for this back end, use the configuration option `--with-pipe`. + * Support for the `dummy` audio backend device continues. To activate, set `output_backend = "dummy"` in in the general section of the configuration. To include support for this back end, use the configuration option `--with-dummy`. + * Limited support for the PulseAudio audio backend continues. To activate, set `output_backend = "pulse"` in in the general section of the configuration. You must still enter its settings via the command line, after the `--` as before. Note that no stuffing or stripping is done: if the PulseAudio sink runs slower or faster, you'll eventually get buffer overflow or underflow. + * New backend-specific settings are provided for setting the size of the backend's buffer and for adding or removing a fixed offset to the overall latency. The `audio_backend_buffer_desired_length` default is 6615 frames, or 0.15 seconds. On some slower machines, particularly with metadata processing going on, the DAC buffer can underflow on this setting, so it might be worth making the buffer larger. A problem on software mixers only is that changes to volume control settings have to propagate through the buffer to be heard, so the larger the buffer, the longer the response time. If you're using an alsa back end and are using a hardware mixers, this isn't a problem. The `audio_backend_latency_offset` allows you emit frames to the audio back end some time before or after the synchronised time. This would be useful, for example, if you are outputting to a device that takes 20 ms to process audio; yoou would specify a `audio_backend_latency_offset = -882`, where 882 is the number of frames in 20 ms, to compensate for the device delay. + +Version 2.3 +----- +* Enhancements + * Adding the System V startup script (the "initscript") is now a configuration option. The default is to include it, so if you want to omit the installation of the initscript, add the configuration option `--without-initscript`. + * Metadata support is now a compile-time option: `--with-metadata`. + * A metadata feed has been added. Use the option `-M `, e.g. `-M /tmp`. Shairport Sync will provide metadata in a pipe called `/shairport-sync-metadata`. (This is changed in 2.3.1.) There's a sample metadata reader at https://github.com/mikebrady/shairport-sync-metadata-reader. The format of the metadata is a mixture of XML-style tags, 4-character codes and base64 data. Please look at `rtsp.c` and `player.c` for examples. Please note that the format of the metadata may change. +Beware: there appears to be a serious bug in iTunes before 12.1.2, such that it may stall for a long period when sending large (more than a few hundred kilobytes) coverart images. + +* Bugfix + * Fix a bug when compiling for Arch Linux on Raspberry Pi 2 (thanks to https://github.com/joaodriessen). + * Fix a bug whereby if the ANNOUNCE and/or SETUP method fails, the play_lock mutex is never unlocked, thus blocking other clients from connecting. This can affect all types of users, but particularly Pulseaudio users. (Thanks to https://github.com/jclehner.) + * Modify the init script to start after all services are ready. Add in a commented-out sleep command if users find it necessary (thanks to https://github.com/BNoiZe). + * Two memory leaks fixed (thanks to https://github.com/pdgendt). + * An error handling time specifications for flushes was causing an audible glitch when pausing and resuming some tracks. This has been fixed (thanks to https://github.com/Hamster128). + Version 2.2.5 ----- * Bugfixes @@ -16,9 +192,6 @@ Version 2.2.3 * Bugfixes * Fix a bug when compiling for Arch Linux on Raspberry Pi 2 (thanks to https://github.com/joaodriessen). * Fix a compiler warning (thanks to https://github.com/sdigit). - * Modify the init script to start after all services are ready. Add in a commented-out sleep command if users find it necessary (thanks to https://github.com/BNoiZe). - * Two memory leaks fixed (thanks to https://github.com/pdgendt). - * An error handling time specifications for flushes was causing an audible glitch when pausing and resuming some tracks. This has been fixed (thanks to https://github.com/Hamster128). Version 2.2.2 ----- diff --git a/audio.c b/audio.c index 95bedc2c..6c2c6fce 100644 --- a/audio.c +++ b/audio.c @@ -41,7 +41,15 @@ extern audio_output audio_pulse; #ifdef CONFIG_ALSA extern audio_output audio_alsa; #endif -extern audio_output audio_dummy, audio_pipe; +#ifdef CONFIG_DUMMY +extern audio_output audio_dummy; +#endif +#ifdef CONFIG_PIPE +extern audio_output audio_pipe; +#endif +#ifdef CONFIG_STDOUT +extern audio_output audio_stdout; +#endif static audio_output *outputs[] = { #ifdef CONFIG_SNDIO @@ -56,36 +64,41 @@ static audio_output *outputs[] = { #ifdef CONFIG_AO &audio_ao, #endif +#ifdef CONFIG_DUMMY &audio_dummy, +#endif +#ifdef CONFIG_PIPE &audio_pipe, - NULL -}; - +#endif +#ifdef CONFIG_STDOUT + &audio_stdout, +#endif + NULL}; audio_output *audio_get_output(char *name) { - audio_output **out; + audio_output **out; - // default to the first - if (!name) - return outputs[0]; + // default to the first + if (!name) + return outputs[0]; - for (out=outputs; *out; out++) - if (!strcasecmp(name, (*out)->name)) - return *out; + for (out = outputs; *out; out++) + if (!strcasecmp(name, (*out)->name)) + return *out; - return NULL; + return NULL; } void audio_ls_outputs(void) { - audio_output **out; + audio_output **out; - printf("Available audio outputs:\n"); - for (out=outputs; *out; out++) - printf(" %s%s\n", (*out)->name, out==outputs ? " (default)" : ""); + printf("Available audio outputs:\n"); + for (out = outputs; *out; out++) + printf(" %s%s\n", (*out)->name, out == outputs ? " (default)" : ""); - for (out=outputs; *out; out++) { - printf("\n"); - printf("Options for output %s:\n", (*out)->name); - (*out)->help(); - } + for (out = outputs; *out; out++) { + printf("\n"); + printf("Options for output %s:\n", (*out)->name); + (*out)->help(); + } } diff --git a/audio.h b/audio.h index 96b56d61..4738a071 100644 --- a/audio.h +++ b/audio.h @@ -2,6 +2,7 @@ #define _AUDIO_H #include +#include typedef struct { double airplay_volume; @@ -14,33 +15,33 @@ typedef struct { } audio_parameters; typedef struct { - void (*help)(void); - char *name; + void (*help)(void); + char *name; - // start of program - int (*init)(int argc, char **argv); - // at end of program - void (*deinit)(void); + // start of program + int (*init)(int argc, char **argv); + // at end of program + void (*deinit)(void); - void (*start)(int sample_rate); - - // block of samples - void (*play)(short buf[], int samples); - void (*stop)(void); - - // may be null if not implemented - void (*flush)(void); - - // returns the delay before the next frame to be sent to the device would actually be audible. - // almost certainly wrong if the buffer is empty, so put silent buffers into it to make it busy. - // will change dynamically, so keep watching it. Implemented in ALSA only. - uint32_t (*delay)(); + void (*start)(int sample_rate); - // may be NULL, in which case soft volume is applied - void (*volume)(double vol); - - // may be NULL, in which case soft volume parameters are used - void (*parameters)(audio_parameters* info); + // block of samples + void (*play)(short buf[], int samples); + void (*stop)(void); + + // may be null if not implemented + void (*flush)(void); + + // returns the delay before the next frame to be sent to the device would actually be audible. + // almost certainly wrong if the buffer is empty, so put silent buffers into it to make it busy. + // will change dynamically, so keep watching it. Implemented in ALSA only. + uint32_t (*delay)(); + + // may be NULL, in which case soft volume is applied + void (*volume)(double vol); + + // may be NULL, in which case soft volume parameters are used + void (*parameters)(audio_parameters *info); } audio_output; audio_output *audio_get_output(char *name); diff --git a/audio_alsa.c b/audio_alsa.c index 9d29381d..9b423f9e 100644 --- a/audio_alsa.c +++ b/audio_alsa.c @@ -43,9 +43,9 @@ static void stop(void); static void flush(void); static uint32_t delay(void); static void volume(double vol); -static void parameters(audio_parameters* info); -static int has_mute=0; -static int has_db_vol=0; +static void parameters(audio_parameters *info); +static int has_mute = 0; +static int has_db_vol = 0; static double set_volume; audio_output audio_alsa = { @@ -58,7 +58,7 @@ audio_output audio_alsa = { .flush = &flush, .delay = &delay, .play = &play, - .volume = NULL, // to be set later on... + .volume = NULL, // to be set later on... .parameters = NULL // to be set later on... }; @@ -81,58 +81,114 @@ static char *alsa_mix_ctrl = "Master"; static int alsa_mix_index = 0; static int play_number; -static int64_t accumulated_delay,accumulated_da_delay; +static int64_t accumulated_delay, accumulated_da_delay; static void help(void) { - printf(" -d output-device set the output device [default*|...]\n" - " -t mixer-type set the mixer type [software*|hardware]\n" - " -m mixer-device set the mixer device ['output-device'*|...]\n" - " -c mixer-control set the mixer control [Master*|...]\n" - " -i mixer-index set the mixer index [0*|...]\n" - " *) default option\n" - ); + printf(" -d output-device set the output device [default*|...]\n" + " -m mixer-device set the mixer device ['output-device'*|...]\n" + " -c mixer-control set the mixer control [Master*|...]\n" + " -i mixer-index set the mixer index [0*|...]\n" + " *) default option\n"); } static int init(int argc, char **argv) { + const char *str; + int value; + int hardware_mixer = 0; + config.audio_backend_latency_offset = 0; // this is the default for ALSA + config.audio_backend_buffer_desired_length = 6615; // default for alsa with a software mixer + + + + // get settings from settings file first, allow them to be overridden by command line options + + if (config.cfg != NULL) { + /* Get the desired buffer size setting. */ + if (config_lookup_int(config.cfg, "alsa.audio_backend_buffer_desired_length_software", &value)) { + if ((value < 0) || (value > 66150)) + die("Invalid alsa audio backend buffer desired length (software) \"%d\". It should be between 0 and " + "66150, default is 6615", + value); + else { + config.audio_backend_buffer_desired_length = value; + } + } + + /* Get the latency offset. */ + if (config_lookup_int(config.cfg, "alsa.audio_backend_latency_offset", &value)) { + if ((value < -66150) || (value > 66150)) + die("Invalid alsa audio backend buffer latency offset \"%d\". It should be between -66150 and +66150, default is 0", + value); + else + config.audio_backend_latency_offset = value; + } + + /* Get the Output Device Name. */ + if (config_lookup_string(config.cfg, "alsa.output_device", &str)) { + alsa_out_dev = (char *)str; + } + + + /* Get the Mixer Type setting. */ + + if (config_lookup_string(config.cfg, "alsa.mixer_type", &str)) { + inform("The alsa mixer_type setting is deprecated and has been ignored. FYI, using the \"mixer_control_name\" setting automatically chooses a hardware mixer."); + } + + + /* Get the Mixer Device Name. */ + if (config_lookup_string(config.cfg, "alsa.mixer_device", &str)) { + alsa_mix_dev = (char *)str; + } + + /* Get the Mixer Control Name. */ + if (config_lookup_string(config.cfg, "alsa.mixer_control_name", &str)) { + alsa_mix_ctrl = (char *)str; + hardware_mixer = 1; + } + } + optind = 1; // optind=0 is equivalent to optind=1 plus special behaviour argv--; // so we shift the arguments to satisfy getopt() argc++; // some platforms apparently require optreset = 1; - which? int opt; while ((opt = getopt(argc, argv, "d:t:m:c:i:")) > 0) { - switch (opt) { - case 'd': - alsa_out_dev = optarg; - break; - case 't': - if (strcmp(optarg, "hardware") == 0) - hardware_mixer = 1; - break; - case 'm': - alsa_mix_dev = optarg; - break; - case 'c': - alsa_mix_ctrl = optarg; - break; - case 'i': - alsa_mix_index = strtol(optarg, NULL, 10); - break; - default: - help(); - die("Invalid audio option -%c specified", opt); - } + switch (opt) { + case 'd': + alsa_out_dev = optarg; + break; + + case 't': + inform("The alsa backend -t option is deprecated and has been ignored. FYI, using the -c option automatically chooses a hardware mixer."); + break; + + case 'm': + alsa_mix_dev = optarg; + break; + case 'c': + alsa_mix_ctrl = optarg; + hardware_mixer = 1; + break; + case 'i': + alsa_mix_index = strtol(optarg, NULL, 10); + break; + default: + help(); + die("Invalid audio option -%c specified", opt); + } } if (optind < argc) - die("Invalid audio argument: %s", argv[optind]); + die("Invalid audio argument: %s", argv[optind]); if (!hardware_mixer) - return 0; + return 0; if (alsa_mix_dev == NULL) - alsa_mix_dev = alsa_out_dev; + alsa_mix_dev = alsa_out_dev; int ret = 0; @@ -141,39 +197,41 @@ static int init(int argc, char **argv) { snd_mixer_selem_id_set_name(alsa_mix_sid, alsa_mix_ctrl); if ((snd_mixer_open(&alsa_mix_handle, 0)) < 0) - die ("Failed to open mixer"); + die("Failed to open mixer"); if ((snd_mixer_attach(alsa_mix_handle, alsa_mix_dev)) < 0) - die ("Failed to attach mixer"); + die("Failed to attach mixer"); if ((snd_mixer_selem_register(alsa_mix_handle, NULL, NULL)) < 0) - die ("Failed to register mixer element"); + die("Failed to register mixer element"); ret = snd_mixer_load(alsa_mix_handle); if (ret < 0) - die ("Failed to load mixer element"); + die("Failed to load mixer element"); alsa_mix_elem = snd_mixer_find_selem(alsa_mix_handle, alsa_mix_sid); if (!alsa_mix_elem) - die ("Failed to find mixer element"); - if (snd_mixer_selem_get_playback_volume_range(alsa_mix_elem, &alsa_mix_minv, &alsa_mix_maxv)<0) - debug(1,"Can't read mixer's [linear] min and max volumes."); + die("Failed to find mixer element"); + if (snd_mixer_selem_get_playback_volume_range(alsa_mix_elem, &alsa_mix_minv, &alsa_mix_maxv) < 0) + debug(1, "Can't read mixer's [linear] min and max volumes."); else { - if ((snd_mixer_selem_ask_playback_vol_dB(alsa_mix_elem,alsa_mix_minv,&alsa_mix_mindb)==0) && - (snd_mixer_selem_ask_playback_vol_dB(alsa_mix_elem,alsa_mix_maxv,&alsa_mix_maxdb)==0)) { - has_db_vol=1; + if ((snd_mixer_selem_ask_playback_vol_dB(alsa_mix_elem, alsa_mix_minv, &alsa_mix_mindb) == 0) && + (snd_mixer_selem_ask_playback_vol_dB(alsa_mix_elem, alsa_mix_maxv, &alsa_mix_maxdb) == 0)) { + has_db_vol = 1; audio_alsa.volume = &volume; // insert the volume function now we know it can do dB stuff audio_alsa.parameters = ¶meters; // likewise the parameters stuff - if (alsa_mix_mindb==-9999999) { + if (alsa_mix_mindb == -9999999) { // trying to say that the lowest vol is mute, maybe? Raspberry Pi does this - if (snd_mixer_selem_ask_playback_vol_dB(alsa_mix_elem,alsa_mix_minv+1,&alsa_mix_mindb)==0) - debug(1,"Can't get dB value corresponding to a \"volume\" of 1."); + if (snd_mixer_selem_ask_playback_vol_dB(alsa_mix_elem, alsa_mix_minv + 1, + &alsa_mix_mindb) == 0) + debug(1, "Can't get dB value corresponding to a \"volume\" of 1."); } - debug(1,"Hardware mixer has dB volume from %f to %f.",(1.0*alsa_mix_mindb)/100.0,(1.0*alsa_mix_maxdb)/100.0); + debug(1, "Hardware mixer has dB volume from %f to %f.", (1.0 * alsa_mix_mindb) / 100.0, + (1.0 * alsa_mix_maxdb) / 100.0); } else { - debug(1,"Hardware mixer does not have dB volume -- not used."); + debug(1, "Hardware mixer does not have dB volume -- not used."); } } if (snd_mixer_selem_has_playback_switch(alsa_mix_elem)) { - has_mute=1; - debug(1,"Has mute ability."); + has_mute = 1; + debug(1, "Has mute ability."); } return 0; } @@ -181,20 +239,20 @@ static int init(int argc, char **argv) { static void deinit(void) { stop(); if (alsa_mix_handle) { - snd_mixer_close(alsa_mix_handle); + snd_mixer_close(alsa_mix_handle); } } -int open_alsa_device(void) { +int open_alsa_device(void) { int ret, dir = 0; unsigned int my_sample_rate = desired_sample_rate; - snd_pcm_uframes_t frames = 441*10; - snd_pcm_uframes_t buffer_size = frames*4; + snd_pcm_uframes_t frames = 441 * 10; + snd_pcm_uframes_t buffer_size = frames * 4; ret = snd_pcm_open(&alsa_handle, alsa_out_dev, SND_PCM_STREAM_PLAYBACK, 0); if (ret < 0) - return(ret); - // die("Alsa initialization failed: unable to open pcm device: %s.", snd_strerror(ret)); + return (ret); + // die("Alsa initialization failed: unable to open pcm device: %s.", snd_strerror(ret)); snd_pcm_hw_params_alloca(&alsa_params); snd_pcm_hw_params_any(alsa_handle, alsa_params); @@ -208,75 +266,79 @@ int open_alsa_device(void) { if (ret < 0) { die("unable to set hw parameters: %s.", snd_strerror(ret)); } - if (my_sample_rate!=desired_sample_rate) { - die("Can't set the D/A converter to %d -- set to %d instead./n",desired_sample_rate,my_sample_rate); + if (my_sample_rate != desired_sample_rate) { + die("Can't set the D/A converter to %d -- set to %d instead./n", desired_sample_rate, + my_sample_rate); } - return(0); + return (0); } static void start(int sample_rate) { if (sample_rate != 44100) - die("Unexpected sample rate %d -- only 44,100 supported!",sample_rate); + die("Unexpected sample rate %d -- only 44,100 supported!", sample_rate); desired_sample_rate = sample_rate; // must be a variable } static uint32_t delay() { - if (alsa_handle==NULL) { - return 0; - } else { - pthread_mutex_lock(&alsa_mutex); - snd_pcm_sframes_t current_avail,current_delay = 0; - int derr,ignore; - if (snd_pcm_state(alsa_handle)==SND_PCM_STATE_RUNNING) { - derr = snd_pcm_avail_delay(alsa_handle,¤t_avail,¤t_delay); - // current_avail not used - if (derr != 0) { - ignore = snd_pcm_recover(alsa_handle, derr, 0); - debug(1,"Error %d in delay(): %s. Delay reported is %d frames.", derr, snd_strerror(derr),current_delay); - current_delay=-1; - } - } else if (snd_pcm_state(alsa_handle)==SND_PCM_STATE_PREPARED) { - current_delay=0; - } else { - if (snd_pcm_state(alsa_handle)==SND_PCM_STATE_XRUN) - current_delay=0; - else { - current_delay=-1; - debug(1,"Error -- ALSA delay(): bad state: %d.",snd_pcm_state(alsa_handle)); - } - if (derr = snd_pcm_prepare(alsa_handle)) { - ignore = snd_pcm_recover(alsa_handle, derr, 0); - debug(1,"Error preparing after delay error: %s.", snd_strerror(derr)); - current_delay = -1; - } - } - pthread_mutex_unlock(&alsa_mutex); - return current_delay; + if (alsa_handle == NULL) { + return 0; + } else { + pthread_mutex_lock(&alsa_mutex); + snd_pcm_sframes_t current_avail, current_delay = 0; + int derr, ignore; + if (snd_pcm_state(alsa_handle) == SND_PCM_STATE_RUNNING) { + derr = snd_pcm_avail_delay(alsa_handle, ¤t_avail, ¤t_delay); + // current_avail not used + if (derr != 0) { + ignore = snd_pcm_recover(alsa_handle, derr, 0); + debug(1, "Error %d in delay(): %s. Delay reported is %d frames.", derr, snd_strerror(derr), + current_delay); + current_delay = -1; + } + } else if (snd_pcm_state(alsa_handle) == SND_PCM_STATE_PREPARED) { + current_delay = 0; + } else { + if (snd_pcm_state(alsa_handle) == SND_PCM_STATE_XRUN) + current_delay = 0; + else { + current_delay = -1; + debug(1, "Error -- ALSA delay(): bad state: %d.", snd_pcm_state(alsa_handle)); + } + if (derr = snd_pcm_prepare(alsa_handle)) { + ignore = snd_pcm_recover(alsa_handle, derr, 0); + debug(1, "Error preparing after delay error: %s.", snd_strerror(derr)); + current_delay = -1; + } + } + pthread_mutex_unlock(&alsa_mutex); + return current_delay; } } static void play(short buf[], int samples) { int ret = 0; - if (alsa_handle==NULL) { - ret = open_alsa_device(); - if ((ret==0) && (audio_alsa.volume)) - volume(set_volume); - } - if (ret==0) { + if (alsa_handle == NULL) { + ret = open_alsa_device(); + if ((ret == 0) && (audio_alsa.volume)) + volume(set_volume); + } + if (ret == 0) { pthread_mutex_lock(&alsa_mutex); snd_pcm_sframes_t current_delay = 0; - int err,ignore; - if ((snd_pcm_state(alsa_handle)==SND_PCM_STATE_PREPARED) || (snd_pcm_state(alsa_handle)==SND_PCM_STATE_RUNNING)) { - err = snd_pcm_writei(alsa_handle, (char*)buf, samples); + int err, ignore; + if ((snd_pcm_state(alsa_handle) == SND_PCM_STATE_PREPARED) || + (snd_pcm_state(alsa_handle) == SND_PCM_STATE_RUNNING)) { + err = snd_pcm_writei(alsa_handle, (char *)buf, samples); if (err < 0) { ignore = snd_pcm_recover(alsa_handle, err, 0); - debug(1,"Error %d writing %d samples in play() %s.",err,samples, snd_strerror(err)); + debug(1, "Error %d writing %d samples in play() %s.", err, samples, snd_strerror(err)); } } else { - debug(1,"Error -- ALSA device in incorrect state (%d) for play.",snd_pcm_state(alsa_handle)); + debug(1, "Error -- ALSA device in incorrect state (%d) for play.", + snd_pcm_state(alsa_handle)); if (err = snd_pcm_prepare(alsa_handle)) { ignore = snd_pcm_recover(alsa_handle, err, 0); - debug(1,"Error preparing after play error: %s.", snd_strerror(err)); + debug(1, "Error preparing after play error: %s.", snd_strerror(err)); } } pthread_mutex_unlock(&alsa_mutex); @@ -288,20 +350,21 @@ static void flush(void) { if (alsa_handle) { // debug(1,"Dropping frames for flush..."); if (derr = snd_pcm_drop(alsa_handle)) - debug(1,"Error dropping frames: %s.", snd_strerror(derr)); + debug(1, "Error dropping frames: %s.", snd_strerror(derr)); // debug(1,"Dropped frames ok. State is %d.",snd_pcm_state(alsa_handle)); if (derr = snd_pcm_prepare(alsa_handle)) - debug(1,"Error preparing after flush: %s.", snd_strerror(derr)); + debug(1, "Error preparing after flush: %s.", snd_strerror(derr)); // debug(1,"Frames successfully dropped."); /* if (snd_pcm_state(alsa_handle)==SND_PCM_STATE_PREPARED) - debug(1,"Flush returns to SND_PCM_STATE_PREPARED state."); + debug(1,"Flush returns to SND_PCM_STATE_PREPARED state."); if (snd_pcm_state(alsa_handle)==SND_PCM_STATE_RUNNING) debug(1,"Flush returns to SND_PCM_STATE_RUNNING state."); - */ - if (!((snd_pcm_state(alsa_handle)==SND_PCM_STATE_PREPARED) || (snd_pcm_state(alsa_handle)==SND_PCM_STATE_RUNNING))) - debug(1,"Flush returning unexpected state -- %d.",snd_pcm_state(alsa_handle)); - + */ + if (!((snd_pcm_state(alsa_handle) == SND_PCM_STATE_PREPARED) || + (snd_pcm_state(alsa_handle) == SND_PCM_STATE_RUNNING))) + debug(1, "Flush returning unexpected state -- %d.", snd_pcm_state(alsa_handle)); + // flush also closes the device snd_pcm_close(alsa_handle); alsa_handle = NULL; @@ -309,29 +372,29 @@ static void flush(void) { } static void stop(void) { - if (alsa_handle!=0) - // when we want to stop, we want the alsa device - // to be closed immediately -- we may even be killing the thread, so we don't wish to wait - // so we should flush first - flush(); // flush will also close the device - // close_alsa_device(); + if (alsa_handle != 0) + // when we want to stop, we want the alsa device + // to be closed immediately -- we may even be killing the thread, so we don't wish to wait + // so we should flush first + flush(); // flush will also close the device + // close_alsa_device(); } -static void parameters(audio_parameters* info) { - info->has_true_mute=has_mute; +static void parameters(audio_parameters *info) { + info->has_true_mute = has_mute; info->is_muted = ((has_mute) && (set_volume == -144.0)); - info->minimum_volume_dB=alsa_mix_mindb; - info->maximum_volume_dB=alsa_mix_maxdb; - info->airplay_volume=set_volume; - info->current_volume_dB=vol2attn(set_volume,alsa_mix_maxdb,alsa_mix_mindb); + info->minimum_volume_dB = alsa_mix_mindb; + info->maximum_volume_dB = alsa_mix_maxdb; + info->airplay_volume = set_volume; + info->current_volume_dB = vol2attn(set_volume, alsa_mix_maxdb, alsa_mix_mindb); } static void volume(double vol) { - set_volume=vol; - double vol_setting = vol2attn(vol,alsa_mix_maxdb,alsa_mix_mindb); + set_volume = vol; + double vol_setting = vol2attn(vol, alsa_mix_maxdb, alsa_mix_mindb); // debug(1,"Setting volume db to %f, for volume input of %f.",vol_setting/100,vol); if (snd_mixer_selem_set_playback_dB_all(alsa_mix_elem, vol_setting, -1) != 0) - die ("Failed to set playback dB volume"); - if (has_mute) - snd_mixer_selem_set_playback_switch_all(alsa_mix_elem, (vol!=-144.0)); + die("Failed to set playback dB volume"); + if (has_mute) + snd_mixer_selem_set_playback_switch_all(alsa_mix_elem, (vol != -144.0)); } diff --git a/audio_ao.c b/audio_ao.c index d8b0cbd5..0acc8114 100644 --- a/audio_ao.c +++ b/audio_ao.c @@ -24,7 +24,6 @@ * OTHER DEALINGS IN THE SOFTWARE. */ - #include #include #include @@ -35,98 +34,96 @@ ao_device *dev = NULL; static void help(void) { - printf(" -d driver set the output driver\n" - " -o name=value set an arbitrary ao option\n" - " -i id shorthand for -o id=\n" - " -n name shorthand for -o dev= -o dsp=\n" - ); + printf(" -d driver set the output driver\n" + " -o name=value set an arbitrary ao option\n" + " -i id shorthand for -o id=\n" + " -n name shorthand for -o dev= -o dsp=\n"); } static int init(int argc, char **argv) { - ao_initialize(); - int driver = ao_default_driver_id(); - ao_option *ao_opts = NULL; + ao_initialize(); + int driver = ao_default_driver_id(); + ao_option *ao_opts = NULL; - optind = 1; // optind=0 is equivalent to optind=1 plus special behaviour - argv--; // so we shift the arguments to satisfy getopt() - argc++; - // some platforms apparently require optreset = 1; - which? - int opt; - char *mid; - while ((opt = getopt(argc, argv, "d:i:n:o:")) > 0) { - switch (opt) { - case 'd': - driver = ao_driver_id(optarg); - if (driver < 0) - die("could not find ao driver %s", optarg); - break; - case 'i': - ao_append_option(&ao_opts, "id", optarg); - break; - case 'n': - ao_append_option(&ao_opts, "dev", optarg); - // Old libao versions (for example, 0.8.8) only support - // "dsp" instead of "dev". - ao_append_option(&ao_opts, "dsp", optarg); - break; - case 'o': - mid = strchr(optarg, '='); - if (!mid) - die("Expected an = in audio option %s", optarg); - *mid = 0; - ao_append_option(&ao_opts, optarg, mid+1); - break; - default: - help(); - die("Invalid audio option -%c specified", opt); - } + optind = 1; // optind=0 is equivalent to optind=1 plus special behaviour + argv--; // so we shift the arguments to satisfy getopt() + argc++; + + config.audio_backend_buffer_desired_length = 44100; // one second. + config.audio_backend_latency_offset = 0; + + // some platforms apparently require optreset = 1; - which? + int opt; + char *mid; + while ((opt = getopt(argc, argv, "d:i:n:o:")) > 0) { + switch (opt) { + case 'd': + driver = ao_driver_id(optarg); + if (driver < 0) + die("could not find ao driver %s", optarg); + break; + case 'i': + ao_append_option(&ao_opts, "id", optarg); + break; + case 'n': + ao_append_option(&ao_opts, "dev", optarg); + // Old libao versions (for example, 0.8.8) only support + // "dsp" instead of "dev". + ao_append_option(&ao_opts, "dsp", optarg); + break; + case 'o': + mid = strchr(optarg, '='); + if (!mid) + die("Expected an = in audio option %s", optarg); + *mid = 0; + ao_append_option(&ao_opts, optarg, mid + 1); + break; + default: + help(); + die("Invalid audio option -%c specified", opt); } + } - if (optind < argc) - die("Invalid audio argument: %s", argv[optind]); + if (optind < argc) + die("Invalid audio argument: %s", argv[optind]); - ao_sample_format fmt; - memset(&fmt, 0, sizeof(fmt)); + ao_sample_format fmt; + memset(&fmt, 0, sizeof(fmt)); - fmt.bits = 16; - fmt.rate = 44100; - fmt.channels = 2; - fmt.byte_format = AO_FMT_NATIVE; + fmt.bits = 16; + fmt.rate = 44100; + fmt.channels = 2; + fmt.byte_format = AO_FMT_NATIVE; - dev = ao_open_live(driver, &fmt, ao_opts); + dev = ao_open_live(driver, &fmt, ao_opts); - return dev ? 0 : 1; + return dev ? 0 : 1; } static void deinit(void) { - if (dev) - ao_close(dev); - dev = NULL; - ao_shutdown(); + if (dev) + ao_close(dev); + dev = NULL; + ao_shutdown(); } static void start(int sample_rate) { - if (sample_rate != 44100) - die("unexpected sample rate!"); + if (sample_rate != 44100) + die("unexpected sample rate!"); } -static void play(short buf[], int samples) { - ao_play(dev, (char*)buf, samples*4); -} +static void play(short buf[], int samples) { ao_play(dev, (char *)buf, samples * 4); } -static void stop(void) { -} +static void stop(void) {} -audio_output audio_ao = { - .name = "ao", - .help = &help, - .init = &init, - .deinit = &deinit, - .start = &start, - .stop = &stop, - .flush = NULL, - .delay = NULL, - .play = &play, - .volume = NULL, - .parameters = NULL -}; +audio_output audio_ao = {.name = "ao", + .help = &help, + .init = &init, + .deinit = &deinit, + .start = &start, + .stop = &stop, + .flush = NULL, + .delay = NULL, + .play = &play, + .volume = NULL, + .parameters = NULL}; diff --git a/audio_dummy.c b/audio_dummy.c index 92129404..9742c58c 100644 --- a/audio_dummy.c +++ b/audio_dummy.c @@ -27,60 +27,37 @@ #include #include #include +#include "common.h" #include "audio.h" int Fs; long long starttime, samples_played; -static int init(int argc, char **argv) { - return 0; -} +static int init(int argc, char **argv) { return 0; } -static void deinit(void) { -} +static void deinit(void) {} static void start(int sample_rate) { - Fs = sample_rate; - starttime = 0; - samples_played = 0; - printf("dummy audio output started at Fs=%d Hz\n", sample_rate); + Fs = sample_rate; + starttime = 0; + samples_played = 0; + debug(1, "dummy audio output started at Fs=%d Hz\n", sample_rate); } -static void play(short buf[], int samples) { - struct timeval tv; +static void play(short buf[], int samples) {} - // this is all a bit expensive but it's long-term stable. - gettimeofday(&tv, NULL); +static void stop(void) { debug(1, "dummy audio stopped\n"); } - long long nowtime = tv.tv_usec + 1e6*tv.tv_sec; +static void help(void) { printf(" There are no options for dummy audio.\n"); } - if (!starttime) - starttime = nowtime; - - samples_played += samples; - - long long finishtime = starttime + samples_played * 1e6 / Fs; - usleep(finishtime - nowtime); -} - -static void stop(void) { - printf("dummy audio stopped\n"); -} - -static void help(void) { - printf(" There are no options for dummy audio.\n"); -} - -audio_output audio_dummy = { - .name = "dummy", - .help = &help, - .init = &init, - .deinit = &deinit, - .start = &start, - .stop = &stop, - .flush = NULL, - .delay = NULL, - .play = &play, - .volume = NULL, - .parameters = NULL -}; +audio_output audio_dummy = {.name = "dummy", + .help = &help, + .init = &init, + .deinit = &deinit, + .start = &start, + .stop = &stop, + .flush = NULL, + .delay = NULL, + .play = &play, + .volume = NULL, + .parameters = NULL}; diff --git a/audio_pipe.c b/audio_pipe.c index 46527611..d38cd8f8 100644 --- a/audio_pipe.c +++ b/audio_pipe.c @@ -29,6 +29,9 @@ #include #include #include +#include +#include +#include #include "common.h" #include "audio.h" @@ -37,55 +40,97 @@ static int fd = -1; char *pipename = NULL; static void start(int sample_rate) { - fd = open(pipename, O_WRONLY); - if (fd < 0) { - perror("open"); - die("could not open specified pipe for writing"); - } + // this will leave fd as -1 if a reader hasn't been attached + fd = open(pipename, O_WRONLY | O_NONBLOCK); } static void play(short buf[], int samples) { - int ignore = write(fd, buf, samples*4); + // if the file is not open, try to open it. + if (fd == -1) { + fd = open(pipename, O_WRONLY | O_NONBLOCK); + } + // if it's got a reader, write to it. + if (fd != -1) { + int ignore = non_blocking_write(fd, buf, samples * 4); + } } static void stop(void) { - close(fd); +// Don't close the pipe just because a play session has stopped. +// if (fd > 0) +// close(fd); } static int init(int argc, char **argv) { - if (argc != 1) - die("bad argument(s) to pipe"); + debug(1, "pipe init"); + const char *str; + int value; + + config.audio_backend_buffer_desired_length = 44100; // one second. + config.audio_backend_latency_offset = 0; + if (config.cfg != NULL) { + /* Get the Output Pipename. */ + const char *str; + if (config_lookup_string(config.cfg, "pipe.name", &str)) { + pipename = (char *)str; + } + + if ((pipename) && (strcasecmp(pipename, "STDOUT") == 0)) + die("Can't use \"pipe\" backend for STDOUT. Use the \"stdout\" backend instead."); + + /* Get the desired buffer size setting. */ + if (config_lookup_int(config.cfg, "pipe.audio_backend_buffer_desired_length", &value)) { + if ((value < 0) || (value > 132300)) + die("Invalid pipe audio backend buffer desired length \"%d\". It should be between 0 and 132300, default is 44100", + value); + else + config.audio_backend_buffer_desired_length = value; + } + + /* Get the latency offset. */ + if (config_lookup_int(config.cfg, "pipe.audio_backend_latency_offset", &value)) { + if ((value < -66150) || (value > 66150)) + die("Invalid pipe audio backend buffer latency offset \"%d\". It should be between -66150 and +66150, default is 0", + value); + else + config.audio_backend_latency_offset = value; + } + } + + if ((pipename == NULL) && (argc != 1)) + die("bad or missing argument(s) to pipe"); + + if (argc == 1) pipename = strdup(argv[0]); + + + // here, create the pipe + if (mkfifo(pipename, 0644) && errno != EEXIST) + die("Could not create output pipe \"%s\"", pipename); - // test open pipe so we error on startup if it's going to fail - start(44100); - stop(); + debug(1, "Pipename is \"%s\"", pipename); - return 0; + return 0; } static void deinit(void) { - if (fd > 0) - close(fd); - if (pipename) - free(pipename); + if (fd > 0) + close(fd); } static void help(void) { - printf(" pipe takes 1 argument: the name of the FIFO to write to.\n"); + printf(" pipe takes 1 argument: the name of the FIFO to write to.\n"); } -audio_output audio_pipe = { - .name = "pipe", - .help = &help, - .init = &init, - .deinit = &deinit, - .start = &start, - .stop = &stop, - .flush = NULL, - .delay = NULL, - .play = &play, - .volume = NULL, - .parameters = NULL -}; +audio_output audio_pipe = {.name = "pipe", + .help = &help, + .init = &init, + .deinit = &deinit, + .start = &start, + .stop = &stop, + .flush = NULL, + .delay = NULL, + .play = &play, + .volume = NULL, + .parameters = NULL}; diff --git a/audio_pulse.c b/audio_pulse.c index 993130c1..29feed88 100644 --- a/audio_pulse.c +++ b/audio_pulse.c @@ -24,7 +24,6 @@ * OTHER DEALINGS IN THE SOFTWARE. */ - #include #include #include @@ -37,117 +36,105 @@ static pa_simple *pa_dev = NULL; static struct { - char *server; - char *sink; - char *apname; -} pulse_options = { - .server = NULL, - .sink = NULL, - .apname = NULL -}; + char *server; + char *sink; + char *apname; +} pulse_options = {.server = NULL, .sink = NULL, .apname = NULL}; static int pa_error; static void pulse_connect(void); static void help(void) { - printf(" -a server set the server name\n" - " -s sink set the output sink\n" - " -n name set the application name, as seen by PulseAudio\n" - " defaults to the access point name\n" - ); + printf(" -a server set the server name\n" + " -s sink set the output sink\n" + " -n name set the application name, as seen by PulseAudio\n" + " defaults to the access point name\n"); } static int init(int argc, char **argv) { - pulse_options.apname = config.apname; - optind = 1; // optind=0 is equivalent to optind=1 plus special behaviour - argv--; // so we shift the arguments to satisfy getopt() - argc++; + pulse_options.apname = config.apname; - // some platforms apparently require optreset = 1; - which? - int opt; - while ((opt = getopt(argc, argv, "a:s:n:")) > 0) { - switch (opt) { - case 'a': - pulse_options.server = optarg; - break; - case 's': - pulse_options.sink = optarg; - break; - case 'n': - pulse_options.apname = optarg; - break; - default: - help(); - die("Invalid audio option -%c specified", opt); - } + config.audio_backend_buffer_desired_length = 44100; // one second. + config.audio_backend_latency_offset = 0; + + optind = 1; // optind=0 is equivalent to optind=1 plus special behaviour + argv--; // so we shift the arguments to satisfy getopt() + argc++; + + // some platforms apparently require optreset = 1; - which? + int opt; + while ((opt = getopt(argc, argv, "a:s:n:")) > 0) { + switch (opt) { + case 'a': + pulse_options.server = optarg; + break; + case 's': + pulse_options.sink = optarg; + break; + case 'n': + pulse_options.apname = optarg; + break; + default: + help(); + die("Invalid audio option -%c specified", opt); } + } - if (optind < argc) - die("Invalid audio argument: %s", argv[optind]); + if (optind < argc) + die("Invalid audio argument: %s", argv[optind]); - pulse_connect(); + pulse_connect(); - return 0; + return 0; } static void pulse_connect(void) { - static const pa_sample_spec ss = { - .format = PA_SAMPLE_S16LE, - .rate = 44100, - .channels = 2 - }; + static const pa_sample_spec ss = {.format = PA_SAMPLE_S16LE, .rate = 44100, .channels = 2}; - pa_dev = pa_simple_new(pulse_options.server, - pulse_options.apname, - PA_STREAM_PLAYBACK, - pulse_options.sink, - "Shairport Stream", - &ss, NULL, NULL, - &pa_error); + pa_dev = pa_simple_new(pulse_options.server, pulse_options.apname, PA_STREAM_PLAYBACK, + pulse_options.sink, "Shairport Stream", &ss, NULL, NULL, &pa_error); - if (!pa_dev) - die("Could not connect to pulseaudio server: %s", pa_strerror(pa_error)); + if (!pa_dev) + die("Could not connect to pulseaudio server: %s", pa_strerror(pa_error)); } static void deinit(void) { - if (pa_dev) - pa_simple_free(pa_dev); - pa_dev = NULL; + if (pa_dev) + pa_simple_free(pa_dev); + pa_dev = NULL; } static void start(int sample_rate) { - if (sample_rate != 44100) - die("unexpected sample rate!"); + if (sample_rate != 44100) + die("unexpected sample rate!"); } static void play(short buf[], int samples) { - if( pa_simple_write(pa_dev, (char *)buf, (size_t)samples * 4, &pa_error) < 0 ) { - fprintf(stderr, __FILE__": pa_simple_write() failed: %s\n", pa_strerror(pa_error)); - if (pa_error == PA_ERR_CONNECTIONTERMINATED) { - fprintf(stderr, __FILE__": reconnecting."); - deinit(); - pulse_connect(); - } + if (pa_simple_write(pa_dev, (char *)buf, (size_t)samples * 4, &pa_error) < 0) { + fprintf(stderr, __FILE__ ": pa_simple_write() failed: %s\n", pa_strerror(pa_error)); + if (pa_error == PA_ERR_CONNECTIONTERMINATED) { + fprintf(stderr, __FILE__ ": reconnecting."); + deinit(); + pulse_connect(); } + } } static void stop(void) { - if (pa_simple_drain(pa_dev, &pa_error) < 0) - fprintf(stderr, __FILE__": pa_simple_drain() failed: %s\n", pa_strerror(pa_error)); + if (pa_simple_drain(pa_dev, &pa_error) < 0) + fprintf(stderr, __FILE__ ": pa_simple_drain() failed: %s\n", pa_strerror(pa_error)); } -audio_output audio_pulse = { - .name = "pulse", - .help = &help, - .init = &init, - .deinit = &deinit, - .start = &start, - .stop = &stop, - .flush = NULL, - .delay = NULL, - .play = &play, - .volume = NULL, - .parameters = NULL -}; +audio_output audio_pulse = {.name = "pulse", + .help = &help, + .init = &init, + .deinit = &deinit, + .start = &start, + .stop = &stop, + .flush = NULL, + .delay = NULL, + .play = &play, + .volume = NULL, + .parameters = NULL}; diff --git a/audio_sndio.c b/audio_sndio.c index 564c7939..e59dbb85 100644 --- a/audio_sndio.c +++ b/audio_sndio.c @@ -24,64 +24,61 @@ static struct sio_hdl *sio; static struct sio_par par; static int init(int argc, char **argv) { - sio = sio_open(SIO_DEVANY, SIO_PLAY, 0); - if (!sio) - die("sndio: cannot connect to sound server"); + sio = sio_open(SIO_DEVANY, SIO_PLAY, 0); + if (!sio) + die("sndio: cannot connect to sound server"); - sio_initpar(&par); + sio_initpar(&par); - par.bits = 16; - par.rate = 44100; - par.pchan = 2; - par.le = SIO_LE_NATIVE; - par.sig = 1; + par.bits = 16; + par.rate = 44100; + par.pchan = 2; + par.le = SIO_LE_NATIVE; + par.sig = 1; - if (!sio_setpar(sio, &par)) - die("sndio: failed to set audio parameters"); - if (!sio_getpar(sio, &par)) - die("sndio: failed to get audio parameters"); + if (!sio_setpar(sio, &par)) + die("sndio: failed to set audio parameters"); + if (!sio_getpar(sio, &par)) + die("sndio: failed to get audio parameters"); - return 0; + config.audio_backend_buffer_desired_length = 44100; // one second. + config.audio_backend_latency_offset = 0; + + return 0; } -static void deinit(void) { - sio_close(sio); -} +static void deinit(void) { sio_close(sio); } static void start(int sample_rate) { - if (sample_rate != par.rate) - die("unexpected sample rate!"); - sio_start(sio); + if (sample_rate != par.rate) + die("unexpected sample rate!"); + sio_start(sio); } static void play(short buf[], int samples) { - sio_write(sio, (char *)buf, samples * par.bps * par.pchan); + sio_write(sio, (char *)buf, samples * par.bps * par.pchan); } -static void stop(void) { - sio_stop(sio); -} +static void stop(void) { sio_stop(sio); } static void help(void) { - printf(" There are no options for sndio audio.\n"); - printf(" Use AUDIODEVICE environment variable.\n"); + printf(" There are no options for sndio audio.\n"); + printf(" Use AUDIODEVICE environment variable.\n"); } static void volume(double vol) { - unsigned int v = vol * SIO_MAXVOL; - sio_setvol(sio, v); + unsigned int v = vol * SIO_MAXVOL; + sio_setvol(sio, v); } -audio_output audio_sndio = { - .name = "sndio", - .help = &help, - .init = &init, - .deinit = &deinit, - .start = &start, - .stop = &stop, - .flush = NULL, - .delay = NULL, - .play = &play, - .volume = &volume, - .parameters = NULL -}; +audio_output audio_sndio = {.name = "sndio", + .help = &help, + .init = &init, + .deinit = &deinit, + .start = &start, + .stop = &stop, + .flush = NULL, + .delay = NULL, + .play = &play, + .volume = &volume, + .parameters = NULL}; diff --git a/audio_stdout.c b/audio_stdout.c new file mode 100644 index 00000000..1637c825 --- /dev/null +++ b/audio_stdout.c @@ -0,0 +1,99 @@ +/* + * stdout output driver. This file is part of Shairport Sync. + * Copyright (c) Mike Brady 2015 + * Based on pipe output driver + * Copyright (c) James Laird 2013 + * All rights reserved. + * + * Permission is hereby granted, free of charge, to any person + * obtaining a copy of this software and associated documentation + * files (the "Software"), to deal in the Software without + * restriction, including without limitation the rights to use, + * copy, modify, merge, publish, distribute, sublicense, and/or + * sell copies of the Software, and to permit persons to whom the + * Software is furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be + * included in all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, + * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES + * OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND + * NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT + * HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, + * WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING + * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR + * OTHER DEALINGS IN THE SOFTWARE. + */ + +#include +#include +#include +#include +#include +#include +#include "common.h" +#include "audio.h" + +static int fd = -1; + +static void start(int sample_rate) { + fd = STDOUT_FILENO; +} + +static void play(short buf[], int samples) { + int ignore = write(fd, buf, samples * 4); +} + +static void stop(void) { + // don't close stdout +} + +static int init(int argc, char **argv) { + const char *str; + int value; + + config.audio_backend_buffer_desired_length = 44100; // one second. + config.audio_backend_latency_offset = 0; + + if (config.cfg != NULL) { + /* Get the desired buffer size setting. */ + if (config_lookup_int(config.cfg, "stdout.audio_backend_buffer_desired_length", &value)) { + if ((value < 0) || (value > 132300)) + die("Invalid stdout audio backend buffer desired length \"%d\". It should be between 0 and 132300, default is 44100", + value); + else + config.audio_backend_buffer_desired_length = value; + } + + /* Get the latency offset. */ + if (config_lookup_int(config.cfg, "stdout.audio_backend_latency_offset", &value)) { + if ((value < -66150) || (value > 66150)) + die("Invalid stdout audio backend buffer latency offset \"%d\". It should be between -66150 and +66150, default is 0", + value); + else + config.audio_backend_latency_offset = value; + } + } + return 0; +} + +static void deinit(void) { + // don't close stdout +} + +static void help(void) { + printf(" stdout takes no arguments\n"); +} + +audio_output audio_stdout = {.name = "stdout", + .help = &help, + .init = &init, + .deinit = &deinit, + .start = &start, + .stop = &stop, + .flush = NULL, + .delay = NULL, + .play = &play, + .volume = NULL, + .parameters = NULL}; diff --git a/common.c b/common.c index e8a67497..bb6dafa2 100644 --- a/common.c +++ b/common.c @@ -33,6 +33,7 @@ #include #include #include +#include #include #include @@ -66,12 +67,10 @@ #endif #endif - - #include "common.h" #include -//true if Shairport Sync is supposed to be sending output to the output device, false otherwise +// true if Shairport Sync is supposed to be sending output to the output device, false otherwise static volatile int requested_connection_state_to_output = 1; @@ -79,99 +78,98 @@ shairport_cfg config; int debuglev = 0; -int get_requested_connection_state_to_output() { - return requested_connection_state_to_output; -} - -void set_requested_connection_state_to_output(int v) { - requested_connection_state_to_output = v; -} +int get_requested_connection_state_to_output() { return requested_connection_state_to_output; } +void set_requested_connection_state_to_output(int v) { requested_connection_state_to_output = v; } void die(char *format, ...) { - char s[1024]; - s[0]=0; - va_list args; - va_start(args, format); - vsprintf(s,format,args); - va_end(args); - daemon_log(LOG_EMERG,"%s", s); - shairport_shutdown(); - exit(1); + char s[1024]; + s[0] = 0; + va_list args; + va_start(args, format); + vsprintf(s, format, args); + va_end(args); + daemon_log(LOG_EMERG, "%s", s); + shairport_shutdown(); + exit(1); } void warn(char *format, ...) { - char s[1024]; - s[0]=0; - va_list args; - va_start(args, format); - vsprintf(s,format,args); - va_end(args); - daemon_log(LOG_WARNING,"%s", s); + char s[1024]; + s[0] = 0; + va_list args; + va_start(args, format); + vsprintf(s, format, args); + va_end(args); + daemon_log(LOG_WARNING, "%s", s); } void debug(int level, char *format, ...) { - if (level > debuglev) - return; - char s[1024]; - s[0]=0; - va_list args; - va_start(args, format); - vsprintf(s,format,args); - va_end(args); - daemon_log(LOG_DEBUG,"%s", s); + if (level > debuglev) + return; + char s[1024]; + s[0] = 0; + va_list args; + va_start(args, format); + vsprintf(s, format, args); + va_end(args); + daemon_log(LOG_DEBUG, "%s", s); } void inform(char *format, ...) { - char s[1024]; - s[0]=0; - va_list args; - va_start(args, format); - vsprintf(s,format,args); - va_end(args); - daemon_log(LOG_INFO,"%s", s); + char s[1024]; + s[0] = 0; + va_list args; + va_start(args, format); + vsprintf(s, format, args); + va_end(args); + daemon_log(LOG_INFO, "%s", s); } #ifdef HAVE_LIBPOLARSSL char *base64_enc(uint8_t *input, int length) { char *buf = NULL; size_t dlen = 0; - int rc = base64_encode(NULL,&dlen,input,length); - if (rc && (rc!=POLARSSL_ERR_BASE64_BUFFER_TOO_SMALL)) - debug(1,"Error %d getting length of base64 encode.",rc); + int rc = base64_encode(NULL, &dlen, input, length); + if (rc && (rc != POLARSSL_ERR_BASE64_BUFFER_TOO_SMALL)) + debug(1, "Error %d getting length of base64 encode.", rc); else { buf = (char *)malloc(dlen); - rc = base64_encode((unsigned char *)buf,&dlen,input,length); - if (rc!=0) - debug(1,"Error %d encoding base64.",rc); + rc = base64_encode((unsigned char *)buf, &dlen, input, length); + if (rc != 0) + debug(1, "Error %d encoding base64.", rc); } return buf; } uint8_t *base64_dec(char *input, int *outlen) { - // slight problem here is that Apple cut the padding off their challenges. We must restore it before passing it in to the decoder, it seems + // slight problem here is that Apple cut the padding off their challenges. We must restore it + // before passing it in to the decoder, it seems uint8_t *buf = NULL; size_t dlen = 0; - int inbufsize = ((strlen(input)+3)/4)*4; // this is the size of the input buffer we will send to the decoder, but we need space for 3 extra "="s and a NULL - char *inbuf = malloc(inbufsize+4); - if (inbuf==0) - debug(1,"Can't malloc memory for inbuf in base64_decode."); + int inbufsize = ((strlen(input) + 3) / 4) * 4; // this is the size of the input buffer we will + // send to the decoder, but we need space for 3 + // extra "="s and a NULL + char *inbuf = malloc(inbufsize + 4); + if (inbuf == 0) + debug(1, "Can't malloc memory for inbuf in base64_decode."); else { - strcpy(inbuf,input); - strcat(inbuf,"==="); - // debug(1,"base64_dec called with string \"%s\", length %d, filled string: \"%s\", length %d.",input,strlen(input),inbuf,inbufsize); - int rc = base64_decode(buf,&dlen,(unsigned char *)inbuf,inbufsize); - if (rc && (rc!=POLARSSL_ERR_BASE64_BUFFER_TOO_SMALL)) - debug(1,"Error %d getting decode length, result is %d.",rc,dlen); + strcpy(inbuf, input); + strcat(inbuf, "==="); + // debug(1,"base64_dec called with string \"%s\", length %d, filled string: \"%s\", length + // %d.",input,strlen(input),inbuf,inbufsize); + int rc = base64_decode(buf, &dlen, (unsigned char *)inbuf, inbufsize); + if (rc && (rc != POLARSSL_ERR_BASE64_BUFFER_TOO_SMALL)) + debug(1, "Error %d getting decode length, result is %d.", rc, dlen); else { // debug(1,"Decode size is %d.",dlen); buf = malloc(dlen); - if (buf==0) - debug(1,"Can't allocate memory in base64_dec."); + if (buf == 0) + debug(1, "Can't allocate memory in base64_dec."); else { - rc = base64_decode(buf,&dlen,(unsigned char *)inbuf,inbufsize); - if (rc!=0) - debug(1,"Error %d in base64_dec.",rc); + rc = base64_decode(buf, &dlen, (unsigned char *)inbuf, inbufsize); + if (rc != 0) + debug(1, "Error %d in base64_dec.", rc); } } free(inbuf); @@ -183,221 +181,226 @@ uint8_t *base64_dec(char *input, int *outlen) { #ifdef HAVE_LIBSSL char *base64_enc(uint8_t *input, int length) { - BIO *bmem, *b64; - BUF_MEM *bptr; - b64 = BIO_new(BIO_f_base64()); - bmem = BIO_new(BIO_s_mem()); - b64 = BIO_push(b64, bmem); - BIO_set_flags(b64, BIO_FLAGS_BASE64_NO_NL); - BIO_write(b64, input, length); - BIO_flush(b64); - BIO_get_mem_ptr(b64, &bptr); + BIO *bmem, *b64; + BUF_MEM *bptr; + b64 = BIO_new(BIO_f_base64()); + bmem = BIO_new(BIO_s_mem()); + b64 = BIO_push(b64, bmem); + BIO_set_flags(b64, BIO_FLAGS_BASE64_NO_NL); + BIO_write(b64, input, length); + BIO_flush(b64); + BIO_get_mem_ptr(b64, &bptr); - char *buf = (char *)malloc(bptr->length); - if (bptr->length) { - memcpy(buf, bptr->data, bptr->length-1); - buf[bptr->length-1] = 0; - } + char *buf = (char *)malloc(bptr->length); + if (bptr->length) { + memcpy(buf, bptr->data, bptr->length - 1); + buf[bptr->length - 1] = 0; + } - BIO_free_all(bmem); + BIO_free_all(bmem); - return buf; + return buf; } uint8_t *base64_dec(char *input, int *outlen) { - BIO *bmem, *b64; - int inlen = strlen(input); + BIO *bmem, *b64; + int inlen = strlen(input); - b64 = BIO_new(BIO_f_base64()); - BIO_set_flags(b64, BIO_FLAGS_BASE64_NO_NL); - bmem = BIO_new(BIO_s_mem()); - b64 = BIO_push(b64, bmem); + b64 = BIO_new(BIO_f_base64()); + BIO_set_flags(b64, BIO_FLAGS_BASE64_NO_NL); + bmem = BIO_new(BIO_s_mem()); + b64 = BIO_push(b64, bmem); - // Apple cut the padding off their challenges; restore it - BIO_write(bmem, input, inlen); - while (inlen++ & 3) - BIO_write(bmem, "=", 1); - BIO_flush(bmem); + // Apple cut the padding off their challenges; restore it + BIO_write(bmem, input, inlen); + while (inlen++ & 3) + BIO_write(bmem, "=", 1); + BIO_flush(bmem); - int bufsize = strlen(input)*3/4 + 1; - uint8_t *buf = malloc(bufsize); - int nread; + int bufsize = strlen(input) * 3 / 4 + 1; + uint8_t *buf = malloc(bufsize); + int nread; - nread = BIO_read(b64, buf, bufsize); + nread = BIO_read(b64, buf, bufsize); - BIO_free_all(bmem); + BIO_free_all(bmem); - *outlen = nread; - return buf; + *outlen = nread; + return buf; } #endif static char super_secret_key[] = -"-----BEGIN RSA PRIVATE KEY-----\n" -"MIIEpQIBAAKCAQEA59dE8qLieItsH1WgjrcFRKj6eUWqi+bGLOX1HL3U3GhC/j0Qg90u3sG/1CUt\n" -"wC5vOYvfDmFI6oSFXi5ELabWJmT2dKHzBJKa3k9ok+8t9ucRqMd6DZHJ2YCCLlDRKSKv6kDqnw4U\n" -"wPdpOMXziC/AMj3Z/lUVX1G7WSHCAWKf1zNS1eLvqr+boEjXuBOitnZ/bDzPHrTOZz0Dew0uowxf\n" -"/+sG+NCK3eQJVxqcaJ/vEHKIVd2M+5qL71yJQ+87X6oV3eaYvt3zWZYD6z5vYTcrtij2VZ9Zmni/\n" -"UAaHqn9JdsBWLUEpVviYnhimNVvYFZeCXg/IdTQ+x4IRdiXNv5hEewIDAQABAoIBAQDl8Axy9XfW\n" -"BLmkzkEiqoSwF0PsmVrPzH9KsnwLGH+QZlvjWd8SWYGN7u1507HvhF5N3drJoVU3O14nDY4TFQAa\n" -"LlJ9VM35AApXaLyY1ERrN7u9ALKd2LUwYhM7Km539O4yUFYikE2nIPscEsA5ltpxOgUGCY7b7ez5\n" -"NtD6nL1ZKauw7aNXmVAvmJTcuPxWmoktF3gDJKK2wxZuNGcJE0uFQEG4Z3BrWP7yoNuSK3dii2jm\n" -"lpPHr0O/KnPQtzI3eguhe0TwUem/eYSdyzMyVx/YpwkzwtYL3sR5k0o9rKQLtvLzfAqdBxBurciz\n" -"aaA/L0HIgAmOit1GJA2saMxTVPNhAoGBAPfgv1oeZxgxmotiCcMXFEQEWflzhWYTsXrhUIuz5jFu\n" -"a39GLS99ZEErhLdrwj8rDDViRVJ5skOp9zFvlYAHs0xh92ji1E7V/ysnKBfsMrPkk5KSKPrnjndM\n" -"oPdevWnVkgJ5jxFuNgxkOLMuG9i53B4yMvDTCRiIPMQ++N2iLDaRAoGBAO9v//mU8eVkQaoANf0Z\n" -"oMjW8CN4xwWA2cSEIHkd9AfFkftuv8oyLDCG3ZAf0vrhrrtkrfa7ef+AUb69DNggq4mHQAYBp7L+\n" -"k5DKzJrKuO0r+R0YbY9pZD1+/g9dVt91d6LQNepUE/yY2PP5CNoFmjedpLHMOPFdVgqDzDFxU8hL\n" -"AoGBANDrr7xAJbqBjHVwIzQ4To9pb4BNeqDndk5Qe7fT3+/H1njGaC0/rXE0Qb7q5ySgnsCb3DvA\n" -"cJyRM9SJ7OKlGt0FMSdJD5KG0XPIpAVNwgpXXH5MDJg09KHeh0kXo+QA6viFBi21y340NonnEfdf\n" -"54PX4ZGS/Xac1UK+pLkBB+zRAoGAf0AY3H3qKS2lMEI4bzEFoHeK3G895pDaK3TFBVmD7fV0Zhov\n" -"17fegFPMwOII8MisYm9ZfT2Z0s5Ro3s5rkt+nvLAdfC/PYPKzTLalpGSwomSNYJcB9HNMlmhkGzc\n" -"1JnLYT4iyUyx6pcZBmCd8bD0iwY/FzcgNDaUmbX9+XDvRA0CgYEAkE7pIPlE71qvfJQgoA9em0gI\n" -"LAuE4Pu13aKiJnfft7hIjbK+5kyb3TysZvoyDnb3HOKvInK7vXbKuU4ISgxB2bB3HcYzQMGsz1qJ\n" -"2gG0N5hvJpzwwhbhXqFKA4zaaSrw622wDniAK5MlIE0tIAKKP4yxNGjoD2QYjhBGuhvkWKY=\n" -"-----END RSA PRIVATE KEY-----\0"; + "-----BEGIN RSA PRIVATE KEY-----\n" + "MIIEpQIBAAKCAQEA59dE8qLieItsH1WgjrcFRKj6eUWqi+bGLOX1HL3U3GhC/j0Qg90u3sG/1CUt\n" + "wC5vOYvfDmFI6oSFXi5ELabWJmT2dKHzBJKa3k9ok+8t9ucRqMd6DZHJ2YCCLlDRKSKv6kDqnw4U\n" + "wPdpOMXziC/AMj3Z/lUVX1G7WSHCAWKf1zNS1eLvqr+boEjXuBOitnZ/bDzPHrTOZz0Dew0uowxf\n" + "/+sG+NCK3eQJVxqcaJ/vEHKIVd2M+5qL71yJQ+87X6oV3eaYvt3zWZYD6z5vYTcrtij2VZ9Zmni/\n" + "UAaHqn9JdsBWLUEpVviYnhimNVvYFZeCXg/IdTQ+x4IRdiXNv5hEewIDAQABAoIBAQDl8Axy9XfW\n" + "BLmkzkEiqoSwF0PsmVrPzH9KsnwLGH+QZlvjWd8SWYGN7u1507HvhF5N3drJoVU3O14nDY4TFQAa\n" + "LlJ9VM35AApXaLyY1ERrN7u9ALKd2LUwYhM7Km539O4yUFYikE2nIPscEsA5ltpxOgUGCY7b7ez5\n" + "NtD6nL1ZKauw7aNXmVAvmJTcuPxWmoktF3gDJKK2wxZuNGcJE0uFQEG4Z3BrWP7yoNuSK3dii2jm\n" + "lpPHr0O/KnPQtzI3eguhe0TwUem/eYSdyzMyVx/YpwkzwtYL3sR5k0o9rKQLtvLzfAqdBxBurciz\n" + "aaA/L0HIgAmOit1GJA2saMxTVPNhAoGBAPfgv1oeZxgxmotiCcMXFEQEWflzhWYTsXrhUIuz5jFu\n" + "a39GLS99ZEErhLdrwj8rDDViRVJ5skOp9zFvlYAHs0xh92ji1E7V/ysnKBfsMrPkk5KSKPrnjndM\n" + "oPdevWnVkgJ5jxFuNgxkOLMuG9i53B4yMvDTCRiIPMQ++N2iLDaRAoGBAO9v//mU8eVkQaoANf0Z\n" + "oMjW8CN4xwWA2cSEIHkd9AfFkftuv8oyLDCG3ZAf0vrhrrtkrfa7ef+AUb69DNggq4mHQAYBp7L+\n" + "k5DKzJrKuO0r+R0YbY9pZD1+/g9dVt91d6LQNepUE/yY2PP5CNoFmjedpLHMOPFdVgqDzDFxU8hL\n" + "AoGBANDrr7xAJbqBjHVwIzQ4To9pb4BNeqDndk5Qe7fT3+/H1njGaC0/rXE0Qb7q5ySgnsCb3DvA\n" + "cJyRM9SJ7OKlGt0FMSdJD5KG0XPIpAVNwgpXXH5MDJg09KHeh0kXo+QA6viFBi21y340NonnEfdf\n" + "54PX4ZGS/Xac1UK+pLkBB+zRAoGAf0AY3H3qKS2lMEI4bzEFoHeK3G895pDaK3TFBVmD7fV0Zhov\n" + "17fegFPMwOII8MisYm9ZfT2Z0s5Ro3s5rkt+nvLAdfC/PYPKzTLalpGSwomSNYJcB9HNMlmhkGzc\n" + "1JnLYT4iyUyx6pcZBmCd8bD0iwY/FzcgNDaUmbX9+XDvRA0CgYEAkE7pIPlE71qvfJQgoA9em0gI\n" + "LAuE4Pu13aKiJnfft7hIjbK+5kyb3TysZvoyDnb3HOKvInK7vXbKuU4ISgxB2bB3HcYzQMGsz1qJ\n" + "2gG0N5hvJpzwwhbhXqFKA4zaaSrw622wDniAK5MlIE0tIAKKP4yxNGjoD2QYjhBGuhvkWKY=\n" + "-----END RSA PRIVATE KEY-----\0"; #ifdef HAVE_LIBSSL uint8_t *rsa_apply(uint8_t *input, int inlen, int *outlen, int mode) { -static RSA *rsa = NULL; + static RSA *rsa = NULL; - if (!rsa) { - BIO *bmem = BIO_new_mem_buf(super_secret_key, -1); - rsa = PEM_read_bio_RSAPrivateKey(bmem, NULL, NULL, NULL); - BIO_free(bmem); - } + if (!rsa) { + BIO *bmem = BIO_new_mem_buf(super_secret_key, -1); + rsa = PEM_read_bio_RSAPrivateKey(bmem, NULL, NULL, NULL); + BIO_free(bmem); + } - uint8_t *out = malloc(RSA_size(rsa)); - switch (mode) { - case RSA_MODE_AUTH: - *outlen = RSA_private_encrypt(inlen, input, out, rsa, - RSA_PKCS1_PADDING); - break; - case RSA_MODE_KEY: - *outlen = RSA_private_decrypt(inlen, input, out, rsa, - RSA_PKCS1_OAEP_PADDING); - break; - default: - die("bad rsa mode"); - } - return out; + uint8_t *out = malloc(RSA_size(rsa)); + switch (mode) { + case RSA_MODE_AUTH: + *outlen = RSA_private_encrypt(inlen, input, out, rsa, RSA_PKCS1_PADDING); + break; + case RSA_MODE_KEY: + *outlen = RSA_private_decrypt(inlen, input, out, rsa, RSA_PKCS1_OAEP_PADDING); + break; + default: + die("bad rsa mode"); + } + return out; } #endif #ifdef HAVE_LIBPOLARSSL uint8_t *rsa_apply(uint8_t *input, int inlen, int *outlen, int mode) { - rsa_context trsa; - const char *pers = "rsa_encrypt"; - int rc; - - entropy_context entropy; - ctr_drbg_context ctr_drbg; - entropy_init( &entropy ); - if( ( rc = ctr_drbg_init( &ctr_drbg, entropy_func, &entropy,(const unsigned char *) pers,strlen( pers ) ) ) != 0 ) - debug(1, "ctr_drbg_init returned %d\n", rc ); + rsa_context trsa; + const char *pers = "rsa_encrypt"; + int rc; - rsa_init(&trsa,RSA_PKCS_V21,POLARSSL_MD_SHA1); // padding and hash id get overwritten - // BTW, this seems to reset a lot of parameters in the rsa_context - rc = x509parse_key(&trsa,(unsigned char *)super_secret_key,strlen(super_secret_key),NULL,0); - if (rc!=0) - debug(1,"Error %d reading the private key."); - - uint8_t *out = NULL; + entropy_context entropy; + ctr_drbg_context ctr_drbg; + entropy_init(&entropy); + if ((rc = ctr_drbg_init(&ctr_drbg, entropy_func, &entropy, (const unsigned char *)pers, + strlen(pers))) != 0) + debug(1, "ctr_drbg_init returned %d\n", rc); - switch (mode) { - case RSA_MODE_AUTH: - trsa.padding = RSA_PKCS_V15; - trsa.hash_id = POLARSSL_MD_NONE; - debug(2,"rsa_apply encrypt"); - out = malloc(trsa.len); - rc = rsa_pkcs1_encrypt( &trsa, ctr_drbg_random, &ctr_drbg, RSA_PRIVATE, inlen, input, out ); - if (rc!=0) - debug(1,"rsa_pkcs1_encrypt error %d.",rc); - *outlen=trsa.len; - break; - case RSA_MODE_KEY: - debug(2,"rsa_apply decrypt"); - trsa.padding=RSA_PKCS_V21; - trsa.hash_id=POLARSSL_MD_SHA1; - out = malloc(trsa.len); + rsa_init(&trsa, RSA_PKCS_V21, POLARSSL_MD_SHA1); // padding and hash id get overwritten + // BTW, this seems to reset a lot of parameters in the rsa_context + rc = x509parse_key(&trsa, (unsigned char *)super_secret_key, strlen(super_secret_key), NULL, 0); + if (rc != 0) + debug(1, "Error %d reading the private key."); + + uint8_t *out = NULL; + + switch (mode) { + case RSA_MODE_AUTH: + trsa.padding = RSA_PKCS_V15; + trsa.hash_id = POLARSSL_MD_NONE; + debug(2, "rsa_apply encrypt"); + out = malloc(trsa.len); + rc = rsa_pkcs1_encrypt(&trsa, ctr_drbg_random, &ctr_drbg, RSA_PRIVATE, inlen, input, out); + if (rc != 0) + debug(1, "rsa_pkcs1_encrypt error %d.", rc); + *outlen = trsa.len; + break; + case RSA_MODE_KEY: + debug(2, "rsa_apply decrypt"); + trsa.padding = RSA_PKCS_V21; + trsa.hash_id = POLARSSL_MD_SHA1; + out = malloc(trsa.len); #if POLARSSL_VERSION_NUMBER >= 0x01020900 - rc = rsa_pkcs1_decrypt(&trsa, ctr_drbg_random, &ctr_drbg, RSA_PRIVATE, (size_t *)outlen, input, out, trsa.len); + rc = rsa_pkcs1_decrypt(&trsa, ctr_drbg_random, &ctr_drbg, RSA_PRIVATE, (size_t *)outlen, input, + out, trsa.len); #else - rc = rsa_pkcs1_decrypt(&trsa, RSA_PRIVATE, outlen, input, out, trsa.len); + rc = rsa_pkcs1_decrypt(&trsa, RSA_PRIVATE, outlen, input, out, trsa.len); #endif - if (rc!=0) - debug(1,"decrypt error %d.",rc); - break; - default: - die("bad rsa mode"); - } - rsa_free(&trsa); - debug(2,"rsa_apply exit"); - return out; + if (rc != 0) + debug(1, "decrypt error %d.", rc); + break; + default: + die("bad rsa mode"); + } + rsa_free(&trsa); + debug(2, "rsa_apply exit"); + return out; } #endif void command_start(void) { - if (config.cmd_start) { - /*Spawn a child to run the program.*/ - pid_t pid=fork(); - if (pid==0) { /* child process */ - int argC; - char **argV; - // debug(1,"on-start command found."); - if (poptParseArgvString(config.cmd_start,&argC,(const char ***)&argV)!=0) // note that argV should be free()'d after use, but we expect this fork to exit eventually. - debug(1,"Can't decipher on-start command arguments"); - else { - // debug(1,"Executing on-start command %s with %d arguments.",argV[0],argC); - execv(argV[0],argV); - warn("Execution of on-start command failed to start"); - debug(1,"Error executing on-start command %s",config.cmd_start); - exit(127); /* only if execv fails */ - } - } else { - if (config.cmd_blocking) { /* pid!=0 means parent process and if blocking is true, wait for process to finish */ - pid_t rc = waitpid(pid,0,0); /* wait for child to exit */ - if (rc!=pid) { - warn("Execution of on-start command returned an error."); - debug(1,"on-start command %s finished with error %d",config.cmd_start,errno); - } - } - // debug(1,"Continue after on-start command"); - } - } + if (config.cmd_start) { + /*Spawn a child to run the program.*/ + pid_t pid = fork(); + if (pid == 0) { /* child process */ + int argC; + char **argV; + // debug(1,"on-start command found."); + if (poptParseArgvString(config.cmd_start, &argC, (const char ***)&argV) != + 0) // note that argV should be free()'d after use, but we expect this fork to exit + // eventually. + debug(1, "Can't decipher on-start command arguments"); + else { + // debug(1,"Executing on-start command %s with %d arguments.",argV[0],argC); + execv(argV[0], argV); + warn("Execution of on-start command failed to start"); + debug(1, "Error executing on-start command %s", config.cmd_start); + exit(127); /* only if execv fails */ + } + } else { + if (config.cmd_blocking) { /* pid!=0 means parent process and if blocking is true, wait for + process to finish */ + pid_t rc = waitpid(pid, 0, 0); /* wait for child to exit */ + if (rc != pid) { + warn("Execution of on-start command returned an error."); + debug(1, "on-start command %s finished with error %d", config.cmd_start, errno); + } + } + // debug(1,"Continue after on-start command"); + } + } } void command_stop(void) { - if (config.cmd_stop) { - /*Spawn a child to run the program.*/ - pid_t pid=fork(); - if (pid==0) { /* child process */ - int argC; - char **argV; - // debug(1,"on-stop command found."); - if (poptParseArgvString(config.cmd_stop,&argC,(const char ***)&argV)!=0) // note that argV should be free()'d after use, but we expect this fork to exit eventually. - debug(1,"Can't decipher on-stop command arguments"); + if (config.cmd_stop) { + /*Spawn a child to run the program.*/ + pid_t pid = fork(); + if (pid == 0) { /* child process */ + int argC; + char **argV; + // debug(1,"on-stop command found."); + if (poptParseArgvString(config.cmd_stop, &argC, (const char ***)&argV) != + 0) // note that argV should be free()'d after use, but we expect this fork to exit + // eventually. + debug(1, "Can't decipher on-stop command arguments"); else { - // debug(1,"Executing on-stop command %s",config.cmd_stop); - execv(argV[0],argV); - warn("Execution of on-stop command failed to start"); - debug(1,"Error executing on-stop command %s",config.cmd_stop); - exit(127); /* only if execv fails */ - } - } else { - if (config.cmd_blocking) { /* pid!=0 means parent process and if blocking is true, wait for process to finish */ - pid_t rc = waitpid(pid,0,0); /* wait for child to exit */ - if (rc!=pid) { - warn("Execution of on-stop command returned an error."); - debug(1,"Stop command %s finished with error %d",config.cmd_stop,errno); - } - } - // debug(1,"Continue after on-stop command"); - } - } + // debug(1,"Executing on-stop command %s",config.cmd_stop); + execv(argV[0], argV); + warn("Execution of on-stop command failed to start"); + debug(1, "Error executing on-stop command %s", config.cmd_stop); + exit(127); /* only if execv fails */ + } + } else { + if (config.cmd_blocking) { /* pid!=0 means parent process and if blocking is true, wait for + process to finish */ + pid_t rc = waitpid(pid, 0, 0); /* wait for child to exit */ + if (rc != pid) { + warn("Execution of on-stop command returned an error."); + debug(1, "Stop command %s finished with error %d", config.cmd_stop, errno); + } + } + // debug(1,"Continue after on-stop command"); + } + } } - // this is for reading an unsigned 32 bit number, such as an RTP timestamp uint32_t uatoi(const char *nptr) { @@ -406,19 +409,25 @@ uint32_t uatoi(const char *nptr) { return r; } -// Given a volume (0 to -30) and high and low attenuations available in the mixer in dB, return an attenuation depending on the volume and the function's transfer function +// Given a volume (0 to -30) and high and low attenuations available in the mixer in dB, return an +// attenuation depending on the volume and the function's transfer function // See http://tangentsoft.net/audio/atten.html for data on good attenuators. -// We want a smooth attenuation function, like, for example, the ALPS RK27 Potentiometer transfer functions referred to at the link above. +// We want a smooth attenuation function, like, for example, the ALPS RK27 Potentiometer transfer +// functions referred to at the link above. // Note that the max_db and min_db are given as dB*100 -double vol2attn(double vol, long max_db, long min_db) { +double vol2attn(double vol, long max_db, long min_db) { -// We use a little coordinate geometry to build a transfer function from the volume passed in to the device's dynamic range. +// We use a little coordinate geometry to build a transfer function from the volume passed in to the +// device's dynamic range. // (See the diagram in the documents folder.) -// The x axis is the "volume in" which will be from -30 to 0. The y axis will be the "volume out" which will be from the bottom of the range to the top. -// We build the transfer function from one or more lines. We characterise each line with two numbers: -// the first is where on x the line starts when y=0 (x can be from 0 to -30); the second is where on y the line stops when when x is -30. +// The x axis is the "volume in" which will be from -30 to 0. The y axis will be the "volume out" +// which will be from the bottom of the range to the top. +// We build the transfer function from one or more lines. We characterise each line with two +// numbers: +// the first is where on x the line starts when y=0 (x can be from 0 to -30); the second is where on +// y the line stops when when x is -30. // thus, if the line was characterised as {0,-30}, it would be an identity transfer. // Assuming, for example, a dynamic range of lv=-60 to hv=0 // Typically we'll use three lines -- a three order transfer function @@ -429,27 +438,33 @@ double vol2attn(double vol, long max_db, long min_db) { #define order 3 double vol_setting = max_db; - - if ((vol<=0.0) && (vol>=-30.0)) { - long range_db = max_db-min_db; // this will be a positive nunmber + + if ((vol <= 0.0) && (vol >= -30.0)) { + long range_db = max_db - min_db; // this will be a positive nunmber // debug(1,"Volume min %ddB, max %ddB, range %ddB.",min_db,max_db,range_db); - //double first_slope = -3000.0; // this is the slope of the attenuation at the high end -- 30dB for the full rotation. - double first_slope = -range_db/2; // this is the slope of the attenuation at the high end -- 30dB for the full rotation. - if (-range_db>first_slope) + // double first_slope = -3000.0; // this is the slope of the attenuation at the high end -- 30dB + // for the full rotation. + double first_slope = + -range_db / + 2; // this is the slope of the attenuation at the high end -- 30dB for the full rotation. + if (-range_db > first_slope) first_slope = range_db; - double lines[order][2] = {{0,first_slope},{-5,first_slope-(range_db+first_slope)/2},{-17,-range_db}}; + double lines[order][2] = { + {0, first_slope}, {-5, first_slope - (range_db + first_slope) / 2}, {-17, -range_db}}; int i; - for (i=0;i #include +#include + #include "config.h" #include "audio.h" #include "mdns.h" #if defined(__APPLE__) && defined(__MACH__) - /* Apple OSX and iOS (Darwin). ------------------------------ */ +/* Apple OSX and iOS (Darwin). ------------------------------ */ #include #if TARGET_OS_MAC == 1 - /* OSX */ - #define COMPILE_FOR_OSX 1 +/* OSX */ +#define COMPILE_FOR_OSX 1 #endif #endif -#if defined(__linux__) - /* Linux. --------------------------------------------------- */ - #define COMPILE_FOR_LINUX 1 +#if defined(__linux__) || defined(__FreeBSD__) +/* Linux and FreeBSD */ +#define COMPILE_FOR_LINUX_AND_FREEBSD 1 #endif // struct sockaddr_in6 is bigger than struct sockaddr. derp #ifdef AF_INET6 - #define SOCKADDR struct sockaddr_storage - #define SAFAMILY ss_family +#define SOCKADDR struct sockaddr_storage +#define SAFAMILY ss_family #else - #define SOCKADDR struct sockaddr - #define SAFAMILY sa_family +#define SOCKADDR struct sockaddr +#define SAFAMILY sa_family #endif - enum stuffing_type { - ST_basic = 0, + ST_basic = 0, ST_soxr, } type; - - typedef struct { - char *password; - char *apname; - uint8_t hw_addr[6]; - int port; - int resyncthreshold; // if it get's out of whack my more than this, resync. Zero means never resync. - int timeout; // while in play mode, exit if no packets of audio come in for more than this number of seconds . Zero means never exit. - char *output_name; - audio_output *output; - char *mdns_name; - mdns_backend *mdns; - int buffer_start_fill; - uint32_t latency; - uint32_t userSuppliedLatency; // overrides all other latencies -- use with caution - uint32_t iTunesLatency; // supplied with --iTunesLatency option - uint32_t AirPlayLatency; //supplied with --AirPlayLatency option - uint32_t ForkedDaapdLatency; //supplied with --ForkedDaapdLatency option - int daemonise; - int statistics_requested; - char *cmd_start, *cmd_stop; - int tolerance; // allow this much drift before attempting to correct it - int cmd_blocking; - enum stuffing_type packet_stuffing; - char *pidfile; - char *logfile; - char *errfile; + config_t *cfg; + char *password; + char *apname; +#ifdef CONFIG_METADATA + int metadata_enabled; + char *metadata_pipename; + int get_coverart; +#endif + uint8_t hw_addr[6]; + int port; + int udp_port_base; + int udp_port_range; + int ignore_volume_control; + int resyncthreshold; // if it get's out of whack my more than this, resync. Zero means never + // resync. + int allow_session_interruption; + int timeout; // while in play mode, exit if no packets of audio come in for more than this number + // of seconds . Zero means never exit. + int dont_check_timeout; // this is used to maintain backward compatability with the old -t option + // behaviour; only set by -t 0, cleared by everything else + char *output_name; + audio_output *output; + char *mdns_name; + mdns_backend *mdns; + int buffer_start_fill; + uint32_t latency; + uint32_t userSuppliedLatency; // overrides all other latencies -- use with caution + uint32_t iTunesLatency; // supplied with --iTunesLatency option + uint32_t AirPlayLatency; // supplied with --AirPlayLatency option + uint32_t ForkedDaapdLatency; // supplied with --ForkedDaapdLatency option + int daemonise; + int statistics_requested; + char *cmd_start, *cmd_stop; + int cmd_blocking; + int tolerance; // allow this much drift before attempting to correct it + enum stuffing_type packet_stuffing; + char *pidfile; + char *logfile; + char *errfile; + char *configfile; + uint32_t audio_backend_buffer_desired_length; // this will be the desired number of frames in the + // audio backend buffer -- the DAC buffer for ALSA + uint32_t audio_backend_latency_offset; // this will be the offset to compensate for any fixed latency + // there might be in the audio } shairport_cfg; -//true if Shairport Sync is supposed to be sending output to the output device, false otherwise +// true if Shairport Sync is supposed to be sending output to the output device, false otherwise int get_requested_connection_state_to_output(); void set_requested_connection_state_to_output(int v); +ssize_t non_blocking_write(int fd, const void *buf, size_t count); // used in a few places + int debuglev; void die(char *format, ...); void warn(char *format, ...); @@ -82,10 +102,11 @@ uint8_t *base64_dec(char *input, int *outlen); char *base64_enc(uint8_t *input, int length); #define RSA_MODE_AUTH (0) -#define RSA_MODE_KEY (1) +#define RSA_MODE_KEY (1) uint8_t *rsa_apply(uint8_t *input, int inlen, int *outlen, int mode); -// given a volume (0 to -30) and high and low attenuations in dB*100 (e.g. 0 to -6000 for 0 to -60 dB), return an attenuation depending on the transfer function +// given a volume (0 to -30) and high and low attenuations in dB*100 (e.g. 0 to -6000 for 0 to -60 +// dB), return an attenuation depending on the transfer function double vol2attn(double vol, long max_db, long min_db); // return a monolithic (always increasing) time in nanoseconds @@ -97,9 +118,12 @@ uint64_t get_absolute_time_in_fp(void); uint32_t uatoi(const char *nptr); shairport_cfg config; -char sender_name[1024]; -char sender_ip[1024]; -char album_name[1024]; // we might need this for picture diagnostics +config_t config_file_stuff; + +uint32_t buffer_occupancy; +int64_t session_corrections; +uint32_t play_segment_reference_frame; +uint64_t play_segment_reference_frame_remote_time; void command_start(void); void command_stop(void); diff --git a/configure.ac b/configure.ac index f52049ca..4c3d3afd 100644 --- a/configure.ac +++ b/configure.ac @@ -2,7 +2,7 @@ # Process this file with autoconf to produce a configure script. AC_PREREQ([2.50]) -AC_INIT([shairport-sync], [2.2.5], [mikebrady@eircom.net]) +AC_INIT([shairport-sync], [2.4], [mikebrady@eircom.net]) AM_INIT_AUTOMAKE AC_CONFIG_SRCDIR([shairport.c]) AC_CONFIG_HEADERS([config.h]) @@ -11,7 +11,7 @@ AC_CONFIG_HEADERS([config.h]) # # Specifying the OS type, defaulting to linux. # -AC_ARG_WITH(os_type, AS_HELP_STRING([--with-os-type=OSType],[Specify the distribution to target: One of linux darwin])) +AC_ARG_WITH(os_type, AS_HELP_STRING([--with-os-type=OSType],[Specify the distribution to target: One of linux freebsd or darwin])) if test "z$with_os_type" = "z"; then with_os_type="linux" fi @@ -24,18 +24,67 @@ AC_PROG_INSTALL PKG_PROG_PKG_CONFIG([0.9.0]) # Checks for libraries. -if test "x${with_os_type}" = xlinux; then +if test "x${with_os_type}" = xlinux -o "x${with_os_type}" = xfreebsd ; then AC_CHECK_LIB([rt],[clock_gettime], , AC_MSG_ERROR(librt needed)) fi +##### Some build systems are not fully using pkg-config, so we can use the flag ${with_pkg_config} on a case-by-case basis +##### to control how to deal with them + +AC_ARG_WITH([pkg_config], +[ --with-pkg-config = use pkg-config to find libraries], ,[with_pkg_config=yes]) + AC_CHECK_LIB([daemon],[daemon_log], , AC_MSG_ERROR(libdaemon needed)) AC_CHECK_LIB([pthread],[pthread_create], , AC_MSG_ERROR(pthread library needed)) AC_CHECK_LIB([m],[exp], , AC_MSG_ERROR(maths library needed)) -PKG_CHECK_MODULES( - [POPT], [popt], - [LIBS="${POPT_LIBS} ${LIBS}" - AC_DEFINE([HAVE_LIBPOPT],[1],[Define to 1 if you have popt])]) +AC_MSG_RESULT(>>Including libpopt) +if test "x${with_pkg_config}" = xyes ; then + PKG_CHECK_MODULES( + [POPT], [popt], + [LIBS="${POPT_LIBS} ${LIBS}" + AC_DEFINE([HAVE_LIBPOPT],[1],[Define to 1 if you have popt])]) +else + AC_CHECK_LIB([popt],[poptGetContext], , AC_MSG_ERROR(libpopt needed)) + AC_DEFINE([HAVE_LIBPOPT],[1],[Define to 1 if you have popt]) +fi + +AC_ARG_WITH([dummy],[ --with-dummy = include the dummy audio back end ],[AC_MSG_RESULT(>>Including the dummy audio back end) AC_DEFINE([CONFIG_DUMMY], 1, [Needed by the compiler.]) ], ) +AM_CONDITIONAL([USE_DUMMY], [test "x$with_dummy" = "xyes" ]) + +AC_ARG_WITH([stdout],[ --with-stdout = include the stdout audio back end ],[ AC_MSG_RESULT(>>Including the stdout audio back end) AC_DEFINE([CONFIG_STDOUT], 1, [Needed by the compiler.]) ], ) + +AM_CONDITIONAL([USE_STDOUT], [test "x$with_stdout" = "xyes" ]) + +AC_ARG_WITH([pipe],[ --with-pipe = include the pipe audio back end ],[ AC_MSG_RESULT(>>Including the pipe audio back end) AC_DEFINE([CONFIG_PIPE], 1, [Needed by the compiler.]) ], ) +AM_CONDITIONAL([USE_PIPE], [test "x$with_pipe" = "xyes" ]) + +# Check to see if we should include the System V initscript + +AC_ARG_WITH([systemv], +[ --with-systemv = install a System V startup script during a make install], , ) +AM_CONDITIONAL([INSTALL_SYSTEMV], [test "x$with_systemv" = "xyes"]) + +# Check to see if we should include the systemd stuff to define it as a service + +AC_ARG_WITH([systemd], +[ --with-systemd = install a systemd service description file during a make install], , ) +AM_CONDITIONAL([INSTALL_SYSTEMD], [test "x$with_systemd" = "xyes"]) + +# Add the libconfig package +if test "x${with_pkg_config}" = xyes ; then + PKG_CHECK_MODULES( + [LIBCONFIG], [libconfig], + [LIBS="${LIBCONFIG_LIBS} ${LIBS}"]) +else + AC_CHECK_LIB([config],[config_init], , AC_MSG_ERROR([libconfig library needed])) +fi + +AC_ARG_WITH([configfiles], +[ --with-configfiles = install configuration files during a make install ], ,[with_configfiles=yes]) +AM_CONDITIONAL([INSTALL_CONFIG_FILES], [test "x$with_configfiles" = "xyes"]) + + # Look for piddir flag AC_ARG_WITH(piddir, [ --with-piddir= Specify a pathname to a directory in which to write the PID file.], [ AC_MSG_CHECKING(--with-piddir argument) @@ -57,11 +106,18 @@ AC_ARG_WITH(ssl, [ choose --with-ssl=openssl or --with-ssl=polarssl for encrypti AC_MSG_ERROR(choose either "openssl" or "polarssl" encryption) fi if test "x${with_ssl}" = xopenssl ; then - PKG_CHECK_MODULES( - [SSL], [libssl,libcrypto], - [LIBS="${SSL_LIBS} ${LIBS}" - AC_DEFINE([HAVE_LIBCRYPTO],[1],[Define to 1 if you have libcrypto]) - AC_DEFINE([HAVE_LIBSSL],[1],[Define to 1 if you have libssl])]) + if test "x${with_pkg_config}" = x1 ; then + PKG_CHECK_MODULES( + [SSL], [libssl,libcrypto], + [LIBS="${SSL_LIBS} ${LIBS}" + AC_DEFINE([HAVE_LIBCRYPTO],[1],[Define to 1 if you have libcrypto]) + AC_DEFINE([HAVE_LIBSSL],[1],[Define to 1 if you have libssl])]) + else + AC_CHECK_LIB([crypto], [main], , AC_MSG_ERROR(libcrypto selected but the library cannot be found!)) + AC_CHECK_LIB([ssl], [main], , AC_MSG_ERROR(libssl selected but the library cannot be found!)) + AC_DEFINE([HAVE_LIBCRYPTO],[1],[Define to 1 if you have libcrypto]) + AC_DEFINE([HAVE_LIBSSL],[1],[Define to 1 if you have libssl]) + fi elif test "x${with_ssl}" = xpolarssl ; then AC_CHECK_LIB([polarssl],[ssl_init], , AC_MSG_ERROR(PolarSSL selected but the library cannot be found!)) else @@ -74,6 +130,12 @@ AC_ARG_WITH(soxr, [ --with-soxr = choose libsoxr for high-quality interpolation AC_MSG_RESULT(>>Including support for soxr-based interpolation) AC_CHECK_LIB([soxr],[soxr_create], , AC_MSG_ERROR(soxr support requested but libsoxr not found!))], ) +# Look for metadata flag -- set flag for conditional compilation +AC_ARG_WITH(metadata, [ --with-metadata = include support for a metadata feed], [ + AC_MSG_RESULT(>>Including metadata support) + AC_DEFINE([CONFIG_METADATA], 1, [Needed by the compiler.])], ) +AM_CONDITIONAL([USE_METADATA], [test "x$with_metadata" = "xyes"]) + # What follows is a bit messy, because if the relevant library is requested, a compiler flag is defined, a file is included in the compilation # and the relevant link files are added. @@ -99,10 +161,15 @@ AC_ARG_WITH(alsa, [ --with-alsa = choose native ALSA API support (GNU/Linux onl HAS_ALSA=1 AM_CONDITIONAL([USE_ALSA], [test 0]) AC_DEFINE([CONFIG_ALSA], 1, [Needed by the compiler.]) - PKG_CHECK_MODULES( - [ALSA], [alsa], - [LIBS="${ALSA_LIBS} ${LIBS}" - AC_DEFINE([HAVE_LIBASOUND],[1],[Define to 1 if you have ALSA])])]) + if test "x${with_pkg_config}" = x1 ; then + PKG_CHECK_MODULES( + [ALSA], [alsa], + [LIBS="${ALSA_LIBS} ${LIBS}" + AC_DEFINE([HAVE_LIBASOUND],[1],[Define to 1 if you have ALSA])]) + else + AC_CHECK_LIB([asound], [snd_pcm_open], , AC_MSG_ERROR(ALSA support requires the asound library!)) + AC_DEFINE([HAVE_LIBASOUND],[1],[Define to 1 if you have ALSA]) + fi ]) AM_CONDITIONAL([USE_ALSA], [test "x$HAS_ALSA" = "x1"]) # Look for SNDIO flag @@ -122,8 +189,8 @@ AC_ARG_WITH(ao, [ --with-ao = choose native AO (Audio Output?) API support -- p AM_CONDITIONAL([USE_AO], [test "x$HAS_AO" = "x1"]) # Look for pulseaudio flag -AC_ARG_WITH(pulseaudio, [ --with-pulseaudio = choose native PulseAudio API support -- probably broken], [ - AC_MSG_RESULT(>>Including a PulseAudio back end --- N.B. this is probably broken!) +AC_ARG_WITH(pulseaudio, [ --with-pulseaudio = choose native PulseAudio API support -- N.B. no synchronisation -- so underflow or overflow is inevitable!], [ + AC_MSG_RESULT(>>Including a PulseAudio back end -- N.B. no synchronisation -- so underflow or overflow is inevitable!) HAS_PULSE=1 AC_DEFINE([CONFIG_PULSE], 1, [Needed by the compiler.]) AC_CHECK_LIB([pulse-simple], [pa_simple_new], , AC_MSG_ERROR(PulseAudio support requires the pulse-simple library!)) @@ -141,7 +208,7 @@ AM_CONDITIONAL([USE_DNS_SD], [test "x$HAS_DNS_SD" = "x1"]) # Checks for header files. AC_HEADER_STDC AC_CHECK_HEADERS([getopt_long.h]) -AC_CHECK_HEADERS([arpa/inet.h fcntl.h mach/mach.h memory.h netdb.h netinet/in.h stdint.h stdlib.h string.h sys/ioctl.h sys/socket.h sys/time.h syslog.h unistd.h]) +AC_CHECK_HEADERS([arpa/inet.h fcntl.h limits.h mach/mach.h memory.h netdb.h netinet/in.h stdint.h stdlib.h string.h sys/ioctl.h sys/socket.h sys/time.h syslog.h unistd.h]) # Checks for typedefs, structures, and compiler characteristics. AC_C_INLINE @@ -157,11 +224,12 @@ AC_TYPE_UINT64_T AC_TYPE_UINT8_T # Checks for library functions. +AC_FUNC_ALLOCA AC_FUNC_ERROR_AT_LINE AC_FUNC_FORK AC_FUNC_MALLOC AC_FUNC_REALLOC -AC_CHECK_FUNCS([clock_gettime gethostname gettimeofday inet_ntoa memchr memmove memset pow select socket stpcpy strcasecmp strchr strdup strerror strstr strtol strtoul]) +AC_CHECK_FUNCS([atexit clock_gettime gethostname inet_ntoa memchr memmove memset mkfifo pow select socket stpcpy strcasecmp strchr strdup strerror strstr strtol strtoul]) AC_CONFIG_FILES([Makefile man/Makefile]) AC_OUTPUT diff --git a/man/shairport-sync.7 b/man/shairport-sync.7 index 1d2e65c4..1449c43d 100644 --- a/man/shairport-sync.7 +++ b/man/shairport-sync.7 @@ -2,7 +2,7 @@ .SH NAME shairport-sync \- Synchronised Audio Player for iTunes / AirPlay .SH SYNOPSIS -\fBshairport-sync [-dvw]\fB [-a \fB\fIname\fB]\fB [-A \fB\fIlatency\fB]\fB [-B \fB\fIcommand\fB]\fB [-E \fB\fIcommand\fB]\fB [--forkedDaapdLatency=\fB\fIlatency\fB]\fB [-i \fB\fIlatency\fB]\fB [-L \fB\fIlatency\fB]\fB [-m \fB\fIbackend\fB]\fB [-o \fB\fIbackend\fB]\fB [--password=\fB\fIsecret\fB]\fB [-r \fB\fIthreshold\fB]\fB [--statistics]\fB [-S \fB\fImode\fB]\fB [-t \fB\fItimeout\fB]\fB [--tolerance=\fB\fIframes\fB]\fB [-- \fB\fIaudio_backend_options\fB]\fB +\fBshairport-sync [-dvw]\fB [-a \fB\fIname\fB]\fB [-A \fB\fIlatency\fB]\fB [-B \fB\fIcommand\fB]\fB [-c \fB\fIconfigurationfile\fB]\fB [-E \fB\fIcommand\fB]\fB [--forkedDaapdLatency=\fB\fIlatency\fB]\fB [--get-cover-art]\fB [-i \fB\fIlatency\fB]\fB [-L \fB\fIlatency\fB]\fB [-m \fB\fIbackend\fB]\fB [--meta-dir=\fB\fIdirectory\fB]\fB [-o \fB\fIbackend\fB]\fB [--password=\fB\fIsecret\fB]\fB [-r \fB\fIthreshold\fB]\fB [--statistics]\fB [-S \fB\fImode\fB]\fB [-t \fB\fItimeout\fB]\fB [--tolerance=\fB\fIframes\fB]\fB [-- \fB\fIaudio_backend_options\fB]\fB shairport-sync -D\fB @@ -17,10 +17,190 @@ shairport-sync -R\fB shairport-sync -V\fB \f1 .SH DESCRIPTION -shairport-sync plays audio streamed from iTunes or from an AirPlay device to an audio device connected via an audio back end. At present, the only fully-implemented back end is for ALSA. +shairport-sync plays audio streamed from iTunes or from an AirPlay device to an ALSA-compatible audio output device. A feature of shairport-sync is that the audio is played synchronously. This means that if many devices are playing the same stream at the same time, all the outputs will stay in step with one another. This allows multiple devices play the same source without getting out of phase with one another, enabling, for example, simultaneous multi-room operation. + +shairport-sync can additionally be compiled and configured to stream raw audio to a pipe or to stdout. + +Settings can be made using the configuration file (recommended for all new installations) or by using command-line options. +.SH CONFIGURATION FILE SETTINGS +You should use the configuration file for setting up shairport-sync. This file is normally \fI/etc/shairport-sync.conf\f1. You may need to have root privileges to modify it. + +Settings are organised into groups, for example, there is a "general" group of standard settings, and there is an "alsa" group with settings that pertain to the ALSA back end. Here is an example of a typical configuration file: + +\fBgeneral = {\f1 + +\fBname = "Mike's Boombox";\f1 + +\fBinterpolation = "soxr";\f1 + +\fBpassword = "secret";\f1 + +\fB};\f1 + +\fB\f1 + +\fBalsa = {\f1 + +\fBoutput_device = "hw:0";\f1 + +\fBmixer_control_name = "PCM";\f1 + +\fB};\f1 + +Most settings have sensible default values, so -- as in the example above -- users generally only need to set (1) the service name, (2) a password (if desired) and (3) the output device. If the output device has a mixer that can be used for volume control, then (4) the volume control's name should be specificed. It is highly desirable to use the output device's mixer for volume control, if available -- response time is reduced to zero and the processor load is reduced. In the example above, "soxr" interpolation was also enabled. + +A sample configuration file with all possible settings, but with all of them commented out, is installed at \fI/etc/shairport-sync.conf.sample\f1. + +To retain backwards compatability with previous versions of shairport-sync you can use still use command line options, but any new features, etc. will be available only via configuration file settings. + +The configuration file is processed using the \fIlibconfig\f1 library -- see \fBhttp://www.hyperrealm.com/libconfig/libconfig_manual.html\f1. +.TP +\fB"GENERAL" SETTINGS\f1 +These are the settings available within the \fBgeneral\f1 group: +.TP +\fBname=\f1\fI"service_name"\f1\fB;\f1 +Use this \fIservice_name\f1 to identify this player in iTunes, etc. The default name is "Shairport Sync on ". +.TP +\fBpassword=\f1\fI"password"\f1\fB;\f1 +Require the password \fIpassword\f1 to connect to the service. If you leave this setting commented out, no password is needed. +.TP +\fBinterpolation=\f1\fI"mode"\f1\fB;\f1 +Interpolate, or "stuff", the audio stream using the \fImode\f1. Interpolation here refers to the process of adding or removing frames of audio to or from the stream sent to the output device to keep it exactly in synchrony with the player. The default mode, "basic", is normally almost completely inaudible. The alternative mode, "soxr", is even less obtrusive but requires much more processing power. For this mode, support for libsoxr, the SoX Resampler Library, must be selected when shairport-sync is compiled. +.TP +\fBstatistics=\f1\fI"setting"\f1\fB;\f1 +Use this \fIsetting\f1 to enable ("yes") or disable ("no") the output of some statistical information on the console or in the log. The default is to disable statistics. +.TP +\fBmdns_backend=\f1\fI"backend"\f1\fB;\f1 +shairport-sync has a number of modules of code ("backends") for interacting with the mDNS service to be used to advertise itself. Normally, the first mDNS backend that works is selected. This setting forces the selection of the specific mDNS \fIbackend\f1. The default is "avahi". Perform the command \fBshairport-sync -h\f1 to get a list of available mDNS modules. +.TP +\fBoutput_backend=\f1\fI"backend"\f1\fB;\f1 +shairport-sync has a number of modules of code ("backends") through which audio is output. Normally, the first audio backend that works is selected. This setting forces the selection of the specific audio \fIbackend\f1. The default is "alsa". Perform the command \fBshairport-sync -h\f1 to get a list of available audio backends. Only the alsa backend supports synchronisation. +.TP +\fBport=\f1\fIportnumber\f1\fB;\f1 +Use this to specify the \fIportnumber\f1 shairport-sync uses to listen for service requests from iTunes, etc. The default is port 5000. +.TP +\fBudp_port_base=\f1\fIportnumber\f1\fB;\f1 +When shairport-sync starts to play audio, it establises three UDP connections to the audio source. Use this setting to specify the starting \fIportnumber\f1 for these three ports. It will pick the first three unused ports starting from \fIportnumber\f1. The default is port 6001. +.TP +\fBudp_port_range=\f1\fIrange\f1\fB;\f1 +Use this in conjunction with the prevous setting to specify the \fIrange\f1 of ports that can be checked for availability. Only three ports are needed. The default is 100, thus 100 ports will be checked from port 6001 upwards until three are found. +.TP +\fBdrift=\f1\fIframes\f1\fB;\f1 +Allow playback to drift up to \fIframes\f1 out of exact synchronization before attempting to correct it. The default is 88 frames, i.e. 2 ms. The smaller the tolerance, the more likely it is that overcorrection will occur. Overcorrection is when more corrections (insertions and deletions) are made than are strictly necessary to keep the stream in sync. Use the \fBstatistics\f1 setting to monitor correction levels. Corrections should not greatly exceed net corrections. +.TP +\fBresync_threshold=\f1\fIthreshold\f1\fB;\f1 +Resynchronise if timings differ by more than \fIthreshold\f1 frames. If the output timing differs from the source timing by more than the threshold, output will be muted and a full resynchronisation will occur. The default threshold is 2,205 frames, i.e. 50 milliseconds. Specify 0 to disable resynchronisation. +.TP +\fBlog_verbosity=\f1\fI0\f1\fB;\f1 +Use this to specify how much debugging information should be output or logged. "0" means no debug information, "3" means most debug information. The default is "0". +.TP +\fBignore_volume_control=\f1\fI"choice"\f1\fB;\f1 +Set this \fIchoice\f1 to "yes" if you want the volume to be at 100% no matter what the source's volume control is set to. This might be useful if you want to set the volume on the output device, independently of the setting at the source. The default is "no". +.TP +\fB"LATENCIES" SETTINGS\f1 +There are four default latency settings, chosen automatically. One latency matches the latency used by recent versions of iTunes when playing audio and another matches the latency used by so-called "AirPlay" devices, including iOS devices and iTunes and Quicktime Player when they are playing video. A third latency is used when the audio source is forked-daapd. The fourth latency is the default if no other latency is chosen and is used for older versions of iTunes. + +If you want to change latencies to compensate for a delay in the audio output device (which will have the same effect on all sources), instead of changing these individual latencies, consider using the \fBaudio_backend_latency_offset\f1 setting in the \fBalsa\f1 group (or the appropriate other group if you're not outputing through the alsa backend). +.TP +\fBitunes=\f1\fIlatency\f1\fB;\f1 +This is the \fIlatency\f1, in frames, used for iTunes 10 or later. Default is 99,400. +.TP +\fBairplay=\f1\fIlatency\f1\fB;\f1 +This is the \fIlatency\f1, in frames, used for AirPlay devices, including iOS devices and iTunes and Quicktime Player when they are playing video. Default is 88,200. +.TP +\fBforkedDaapd=\f1\fIlatency\f1\fB;\f1 +This is the \fIlatency\f1, in frames, used for forkedDaapd sources. Default is 99,400. +.TP +\fBdefault=\f1\fIlatency\f1\fB;\f1 +This is the \fIlatency\f1, in frames, used when the source is unrecognised. Default is 88,200. +.TP +\fB"METADATA" SETTINGS\f1 +shairport-sync can process metadata provided by the source, such as Track Number, Album Name, cover art, etc. and can provide additional metadata such as volume level, pause/resume, etc. It provides the metadata to a pipe, by default \fI/tmp/shairport-sync-metadata\f1. To process metadata, shairport-sync must have been compiled with metadata support included. You can check that this is so by running \fBshairport-sync -V\f1; the identification string will contain the word \fBmetadata\f1. + +The \fBmetadata\f1 group of settings allow you to enable metadata handling and to control certain aspects of it: +.TP +\fBenabled=\f1\fI"choice"\f1\fB;\f1 +Set the \fIchoice\f1 to "yes" to enable shairport-sync to look for metadata from the audio source and to forward it, along with metadata generated by shairport-sync itself, to the metadata pipe. The default is "no". +.TP +\fBinclude_cover_art=\f1\fI"choice"\f1\fB;\f1 +Set the \fIchoice\f1 to "yes" to enable shairport-sync to look for cover art from the audio source and to include it in the feed to the metadata pipe. You must also enable metadata (see above). One reason for not including cover art is that the images can sometimes be very large and may delay transmission of subsequent metadata through the pipe. The default is "no". +.TP +\fBpipe_name=\f1\fI"filepathname"\f1\fB;\f1 +Specify the absolute path name of the pipe through which metadata should be sent The default is \fI/tmp/shairport-sync-metadata\f1". +.TP +\fB"SESSIONCONTROL" SETTINGS\f1 +shairport-sync can run programs just before it starts to play an audio stream and just after it finishes. You specify them using the sessioncontrol group settings run_this_before_play_begins and run_this_after_play_ends. +.TP +\fBrun_this_before_play_begins=\f1\fI"/path/to/application and args"\f1\fB;\f1 +Here you can specify a program and its arguments that will be run just before a play session begins. Be careful to include the full path to the application. The application must be marked as executable and, if it is a script, its first line must begin with the standard \fI#!/bin/...\f1 as appropriate. +.TP +\fBrun_this_after_play_ends=\f1\fI"/path/to/application and args"\f1\fB;\f1 +Here you can specify a program and its arguments that will be run just after a play session ends. Be careful to include the full path to the application. The application must be marked as executable and, if it is a script, its first line must begin with the standard \fI#!/bin/...\f1 as appropriate. +.TP +\fBwait_for_completion=\f1\fI"choice"\f1\fB;\f1 +Set \fIchoice\f1 to "yes" to make shairport-sync wait until the programs specified in the \fBrun_this_before_play_begins\f1 and \fBrun_this_after_play_ends\f1 have completed execution before continuing. The default is "no". +.TP +\fBallow_session_interruption=\f1\fI"choice"\f1\fB;\f1 +If \fBchoice\f1 is set to "yes", then another source will be able to interrupt an existing play session and start a new one. When set to "no" (the default), other devices attempting to interrupt a session will fail, receiving a busy signal. +.TP +\fBsession_timeout=\f1\fIseconds\f1\fB;\f1 +If a play session has been established and the source disappears without warning (such as a device going out of range of a network) then wait for \fIseconds\f1 seconds before ending the session. Once the session has terminated, other devices can use it. The default is 120 seconds. +.TP +\fB"ALSA" SETTINGS\f1 +These settings are for the ALSA back end, used to communicate with audio output devices in the ALSA system. (By the way, you can use tools such as \fBalsamixer\f1 or \fBaplay\f1 to discover what devices are available.) Use these settings to select the output device and the mixer control to be used to control the output volume. You can additionally set the desired size of the output buffer and you can adjust overall latency. Here are the \fBalsa\f1 group settings: +.TP +\fBoutput_device=\f1\fI"output_device"\f1\fB;\f1 +Use the output device called \fIoutput_device\f1. The default is the device called "default". +.TP +.TP +\fBmixer_control_name=\f1\fI"name"\f1\fB;\f1 +Specify the \fIname\f1 of the mixer control to be used by shairport-sync to control the volume. The mixer control must be on the mixer device, which by default is the output device. If you do not specify a mixer control name, shairport-sync will adjust the volume in software. \fBmixer_type=\f1\fI"mixer_type"\f1\fB;\f1 +This setting is deprecated and will be removed soon. If you wish to use a mixer control to control the volume, then set \fImixer_type\f1 to "hardware". The default is "software". +.TP +\fBmixer_device=\f1\fI"mixer_device"\f1\fB;\f1 +By default, the mixer is assumed to be output_device. Use this setting to specify a device other than the output device. +.TP +\fBaudio_backend_latency_offset=\f1\fIoffset\f1\fB;\f1 +Set this \fIoffset\f1, in frames, to compensate for a fixed delay in the audio back end. For example, if the output device delays by 100 ms, set this to -4410. +.TP +\fBaudio_backend_buffer_desired_length=\f1\fIlength\f1\fB;\f1 +Use this to set the desired number frames to be in the output device's hardware output buffer. The default is 6,615 frames, or 0.15 seconds. If set too small, buffer underflow may occur on low-powered machines. If too large, the response times when using software volume control (i.e. when not using a mixer control to control volume) become annoying, or it may exceed the hardware buffer size. It may need to be larger on low-powered machines that are also performing other tasks, such as processing metadata. +.TP +\fB"PIPE" SETTINGS\f1 +These settings are for the PIPE backend, used to route audio to a named unix pipe. The audio is in raw CD audio format: PCM 16 bit little endian, 44,100 samples per second, stereo. + +Use the \fIname\f1 setting to set the name and location of the pipe. + +There are two further settings affecting timing that might be useful if the pipe reader is, for example, a program to play an audio stream such as \fBaplay\f1. The \fIaudio_backend_latency_offset\f1 affects precisely when the first audio packet is sent and the \fIaudio_backend_buffer_desired_length\f1 setting affects the nominal output buffer size. + +These are the settings available within the \fBpipe\f1 group: +.TP +\fBname=\f1\fI"/path/to/pipe"\f1\fB;\f1 +Use this to specify the name and location of the pipe. The pipe will be created and opened when shairport-sync starts up and will be closed upon shutdown. Frames of audio will be sent to the pipe in packets of 352 frames and will be discarded if the pipe has not have a reader attached. The sender will wait for up to five seconds for a packet to be written before discarding it. +.TP +\fBaudio_backend_latency_offset=\f1\fIoffset_in_frames\f1\fB;\f1 +Packets of audio frames are written to the pipe synchronously -- that is, they are written to at exactly the time they should be played. You can offset the time of initial audio output relative to its nominal time using this setting. For example to send an audio stream to the pipe 100 milliseconds before it is due to be played, set this to -4410. Default setting is 0. +.TP +\fBaudio_backend_buffer_desired_length=\f1\fIbuffer_length_in_frames\f1\fB;\f1 +Use this setting, in frames, to set the size of the output buffer. It works by determining how soon the second and subsequent packets of audio frames are sent to the pipe. For example, if you send the first packet of audio exactly when it is due and, using a \fIaudio_backend_buffer_desired_length\f1 setting of 44100, send subsequent packets of audio a second before they are due to be played, they will be buffered in the pipe reader's buffer, giving it a nominal buffer size of 44,100 frames. Note that if the pipe reader consumes audio packets faster or slower than they are supplied, the buffer will eventually empty or overflow -- shairport-sync performs no stuffing or interpolation when writing to a pipe. Default setting is 44,100 frames. +.TP +\fB"STDOUT" SETTINGS\f1 +These settings are for the STDOUT backend, used to route audio to standard output ("stdout"). The audio is in raw CD audio format: PCM 16 bit little endian, 44,100 samples per second, stereo. + +There are two settings affecting timing that might be useful if the stdout reader is, for example, a program to play an audio stream such as \fBaplay\f1. The \fIaudio_backend_latency_offset\f1 affects precisely when the first audio packet is sent and the \fIaudio_backend_buffer_desired_length\f1 setting affects the nominal output buffer size. + +These are the settings available within the \fBstdout\f1 group: +.TP +\fBaudio_backend_latency_offset=\f1\fIoffset_in_frames\f1\fB;\f1 +Packets of audio frames are written to stdout synchronously -- that is, they are written at exactly the time they should be played. You can offset the time of initial audio output relative to its nominal time using this setting. For example to send an audio stream to stdout 100 milliseconds before it is due to be played, set this to -4410. Default setting is 0. +.TP +\fBaudio_backend_buffer_desired_length=\f1\fIbuffer_length_in_frames\f1\fB;\f1 +Use this setting, in frames, to set the size of the output buffer. It works by determining how soon the second and subsequent packets of audio frames are sent to stdout. For example, if you send the first packet of audio exactly when it is due and, using a \fIaudio_backend_buffer_desired_length\f1 setting of 44100, send subsequent packets of audio a second before they are due to be played, they will be buffered in the stdout reader's buffer, giving it a nominal buffer size of 44,100 frames. Note that if the stdout reader consumes audio packets faster or slower than they are supplied, the buffer will eventually empty or overflow -- shairport-sync performs no stuffing or interpolation when writing to stdout. Default setting is 44,100 frames. .SH OPTIONS +Note: if you are setting up shairport-sync for the first time or are updating an existing installation, you are encouraged to use the configuration file settings described above. Most of the options described below simply replicate the configuration settings and are retained to provide backward compatability with older installations of shairport-sync. + Many of the options take sensible default values, so you can normally ignore most of them. See the EXAMPLES section for typical usages. The command line for shairport-sync can take two kinds of options: regular \fBprogram options\f1 and \fBaudio backend options\f1. Program options are always listed first, followed by any audio backend options, preceded by a \fB--\f1 symbol. @@ -38,6 +218,9 @@ Execute \fIprogram\f1 when playback is about to begin. Specify the full path to If you want shairport-sync to wait until the command has completed before starting to play, select the \fB-w\f1 option as well. .TP +\fB-c \f1\fIfilename\f1\fB | --configfile=\f1\fIfilename\f1 +Read configuration settings from \fIfilename\f1. The default is to read them from \fI/etc/shairport-sync.conf\f1. For information about configuration settings, see the "Configuration File Settings" section above. +.TP \fB-D | --disconnectFromOutput\f1 Disconnect the shairport-sync daemon from the output device and exit. (Requires that the daemon has written its PID to an agreed file -- see the \fB-d\f1 option). .TP @@ -52,6 +235,11 @@ If you want shairport-sync to wait until the command has completed before contin \fB--forkedDaapdLatency=\f1\fIlatency\f1 Use this \fIlatency\f1, in frames, for audio streamed from a forked-daapd based source. The default is 99,400 frames, where there are 44,100 frames to the second. .TP +\fB--get-coverart\f1 +This option requires the \fB--meta-dir\f1 option to be set, and enables shairport-sync to request cover art from the source and to transmit it through the metadata pipe. + +Please note that cover art data may be very large, and may place too great a burden on your network. Also, at the time of writing, there appears to be a serious bug in iTunes which stalls transmission of large cover art items, stalling and potentially crashing shairport-sync. +.TP \fB-h | --help\f1 Print brief help message and exit. .TP @@ -64,6 +252,9 @@ Kill the shairport-sync daemon and exit. (Requires that the daemon has written i \fB-L | --latency=\f1\fIlatency\f1 Use this to set the \fIdefault latency\f1, in frames, for audio coming from an unidentified source or from an iTunes Version 9 or earlier source. The standard value for the \fIdefault latency\f1 is 88,200 frames, where there are 44,100 frames to the second. .TP +\fB--meta-dir=\f1\fIdirectory\f1 +Listen for metadata coming from the source and send it, along with metadata from shairport-sync itself, to a pipe called \fIshairport-sync-metadata\f1 in the \fIdirectory\f1 you specify. If you add the \fB--get-cover-art\f1 then cover art will be sent through the pipe too. See \fBhttps://github.com/mikebrady/shairport-sync-metadata-reader\f1 for a sample metadata reader. +.TP \fB-m \f1\fImdnsbackend\f1\fB | --mdns=\f1\fImdnsbackend\f1 Force the use of the specified mDNS backend to advertise the player on the network. The default is to try all mDNS backends until one works. .TP @@ -119,17 +310,17 @@ Use the specified output \fIdevice\f1. You may specify a card, e.g. \fBhw:0\f1, Use the specified hardware \fImixer\f1 for volume control. Use this to specify where the mixer is to be found. For example, if the mixer is associated with a card, as is often the case, specify the card, e.g. \fBhw:0\f1. If (unusually) the mixer is associated with a specific device on a card, specify the device, e.g. \fBhw:0,1\f1. The default is the device named in the \fB-d\f1 option, if given, or the device named \fBdefault\f1. .TP \fB-t \f1\fIdevicetype\f1 -The type of the output device is \fIdevicetype\f1 which must be \fBhardware\f1 or \fBsoftware\f1. The default is \fBsoftware\f1. If you specify \fBhardware\f1 you must also specify the output device using the \fB-d\f1 option and the name of the volume control using the \fB-c\f1 option. You may also need to specify the mixer using the \fB-m\f1 option. +This option is deprecated and is ignored. For your information, its functionality has been automatically incorporated in the -c option -- if you specify a mixer name with the -c option, it is assumed that the mixer is implemented in hardware. .SH EXAMPLES Here is a slightly contrived but typical example: -shairport-sync \fB-d\f1 \fB-a "Joe's Stereo"\f1 \fB-S soxr\f1 \fB--\f1 \fB-d hw:1,0\f1 \fB-m hw:1\f1 \fB-t hardware\f1 \fB-c PCM\f1 +shairport-sync \fB-d\f1 \fB-a "Joe's Stereo"\f1 \fB-S soxr\f1 \fB--\f1 \fB-d hw:1,0\f1 \fB-m hw:1\f1 \fB-c PCM\f1 -The program will run in daemon mode ( \fB-d\f1 ), will be visible as "Joe's Stereo" ( \fB-a "Joe's Stereo"\f1 ) and will use the SoX Resampler Library-based stuffing ( \fB-S soxr\f1 ). The audio backend options following the \fB--\f1 separator specify that the audio will be output on output 0 of soundcard 1 ( \fB-d hw:1,0\f1 ) and will take advantage of the same sound card's hardware (\fB-t hardware\f1) mixer ( \fB-m hw:1\f1 ) using the level control named "PCM" ( \fB-c "PCM"\f1 ). +The program will run in daemon mode ( \fB-d\f1 ), will be visible as "Joe's Stereo" ( \fB-a "Joe's Stereo"\f1 ) and will use the SoX Resampler Library-based stuffing ( \fB-S soxr\f1 ). The audio backend options following the \fB--\f1 separator specify that the audio will be output on output 0 of soundcard 1 ( \fB-d hw:1,0\f1 ) and will take advantage of the same sound card's mixer ( \fB-m hw:1\f1 ) using the level control named "PCM" ( \fB-c "PCM"\f1 ). The example above is slightly contrived in order to show the use of the \fB-m\f1 option. Typically, output 0 is the default output of a card, so the output device could be written \fB-d hw:1\f1 and then the mixer option would be unnecessary, giving the following, simpler, command: -shairport-sync \fB-d\f1 \fB-a "Joe's Stereo"\f1 \fB-S soxr\f1 \fB--\f1 \fB-d hw:1\f1 \fB-t hardware\f1 \fB-c PCM\f1 +shairport-sync \fB-d\f1 \fB-a "Joe's Stereo"\f1 \fB-S soxr\f1 \fB--\f1 \fB-d hw:1\f1 \fB-c PCM\f1 .SH CREDITS Mike Brady developed shairport-sync from the original shairport by James Laird. diff --git a/man/shairport-sync.7.xml b/man/shairport-sync.7.xml index fcd01340..14c8be65 100644 --- a/man/shairport-sync.7.xml +++ b/man/shairport-sync.7.xml @@ -2,7 +2,7 @@ - + --> @@ -39,11 +39,14 @@ [-a name] [-A latency] [-B command] + [-c configurationfile] [-E command] [--forkedDaapdLatency=latency] + [--get-cover-art] [-i latency] [-L latency] [-m backend] + [--meta-dir=directory] [-o backend] [--password=secret] [-r threshold] @@ -63,8 +66,7 @@

shairport-sync plays audio streamed from iTunes or from an AirPlay - device to an audio device connected via an audio back end. At present, - the only fully-implemented back end is for ALSA.

+ device to an ALSA-compatible audio output device.

A feature of shairport-sync is that the audio is played synchronously. This means that if many devices are playing the same stream at the same @@ -72,11 +74,312 @@ This allows multiple devices play the same source without getting out of phase with one another, enabling, for example, simultaneous multi-room operation.

- + +

shairport-sync can additionally be compiled and configured to stream raw audio to a pipe or to stdout.

+ +

Settings can be made using the configuration file (recommended for all new installations) or by using command-line options.

+ + +
+

You should use the configuration file for setting up shairport-sync. + This file is normally /etc/shairport-sync.conf. + You may need to have root privileges to modify it.

+ +

Settings are organised into groups, for example, there is a "general" group of + standard settings, and there is an "alsa" group with settings that pertain to the ALSA + back end. Here is an example of a typical configuration file:

+ +

general = {

+

name = "Mike's Boombox";

+

interpolation = "soxr";

+

password = "secret";

+

};

+

+

alsa = {

+

output_device = "hw:0";

+

mixer_control_name = "PCM";

+

};

+ +

Most settings have sensible default values, so -- as in the example above -- users generally only need to set (1) the service name, (2) a password (if desired) and + (3) the output device. If the output device has a mixer that can be used for volume control, then (4) the volume control's name should be specificed. It is highly desirable to use the output device's mixer for volume control, if available -- response time is reduced to zero and the processor load is reduced. In the example above, "soxr" interpolation was also enabled.

+ +

A sample configuration file with all possible settings, but with all of them commented out, is installed at /etc/shairport-sync.conf.sample.

+ +

To retain backwards compatability with previous versions of shairport-sync + you can use still use command line options, but any new features, etc. will + be available only via configuration file settings.

+ +

The configuration file is processed using the libconfig library + -- see .

+ + +

These are the settings available within the general group:

+ + + + + + + + + + + + + + + + + + + +

There are four default latency settings, chosen automatically. One latency matches the latency used by recent versions of iTunes when playing audio and another matches the latency used by so-called "AirPlay" devices, including iOS devices and iTunes and Quicktime Player when they are playing video. A third latency is used when the audio source is forked-daapd. The fourth latency is the default if no other latency is chosen and is used for older versions of iTunes.

+

If you want to change latencies to compensate for a delay in the audio output device (which will have the same effect on all sources), instead of changing these individual latencies, consider using the audio_backend_latency_offset setting in the alsa group (or the appropriate other group if you're not outputing through the alsa backend).

+ + + + + + + + +

shairport-sync can process metadata provided by the source, such as Track Number, Album Name, cover art, etc. and can provide additional metadata such as volume level, + pause/resume, etc. It provides the metadata to a pipe, by default /tmp/shairport-sync-metadata. + To process metadata, shairport-sync must have been compiled with metadata support included. + You can check that this is so by running shairport-sync -V; the identification string will contain the word metadata.

+

The metadata group of settings allow you to enable metadata handling and to control certain aspects of it:

+ + + + + + + +

shairport-sync can run programs just before it starts to play an audio stream and just after it finishes. + You specify them using the sessioncontrol group settings run_this_before_play_begins and run_this_after_play_ends.

+ + + + + + + + + +

These settings are for the ALSA back end, used to communicate with audio output devices in the ALSA system. + (By the way, you can use tools such as alsamixer or aplay to discover what devices are available.) + Use these settings to select the output device and the mixer control to be used to control the output volume. + You can additionally set the desired size of the output buffer and you can adjust overall latency. Here are the alsa group settings:

+ + + +

mixer_type="mixer_type";

+ This setting is deprecated and will be removed soon. If you wish to use a mixer control to control the volume, then set mixer_type to "hardware". + The default is "software". + + + + + + +

These settings are for the PIPE backend, used to route audio to a named unix pipe. The audio is in raw CD audio format: PCM 16 bit little endian, 44,100 samples per second, + stereo.

+

Use the name setting to set the name and location of the pipe.

+

There are two further settings affecting timing that might be useful if the pipe reader is, for example, + a program to play an audio stream such as aplay. The audio_backend_latency_offset affects precisely when the first audio packet is sent + and the audio_backend_buffer_desired_length setting affects the nominal output buffer size.

+

These are the settings available within the pipe group:

+ + + + + + + + +

These settings are for the STDOUT backend, used to route audio to standard output ("stdout"). + The audio is in raw CD audio format: PCM 16 bit little endian, 44,100 samples per second, stereo.

+

There are two settings affecting timing that might be useful if the stdout reader is, for example, + a program to play an audio stream such as aplay. The audio_backend_latency_offset affects precisely when the first audio packet is sent + and the audio_backend_buffer_desired_length setting affects the nominal output buffer size.

+

These are the settings available within the stdout group:

+ + + + + +
+

Note: if you are setting up shairport-sync for the first time or are updating an existing installation, + you are encouraged to use the configuration file settings described above. Most of the options described below + simply replicate the configuration settings and are retained to provide backward compatability with older installations of shairport-sync.

+

Many of the options take sensible default values, so you can normally ignore most of them. See the EXAMPLES section for typical usages.

@@ -120,6 +423,13 @@

+ + + + + + @@ -389,7 +720,6 @@ -- -d hw:1,0 -m hw:1 - -t hardware -c PCM

The program will run in daemon mode ( -d ), will be visible as @@ -397,8 +727,7 @@ Library-based stuffing ( -S soxr ). The audio backend options following the -- separator specify that the audio will be output on output 0 of soundcard 1 ( - -d hw:1,0 ) and will take advantage of the same sound card's hardware - (-t hardware) mixer ( -m hw:1 ) + -d hw:1,0 ) and will take advantage of the same sound card's mixer ( -m hw:1 ) using the level control named "PCM" ( -c "PCM" ).

The example above is slightly contrived in order to show the use of the -m option. @@ -410,7 +739,6 @@ -S soxr -- -d hw:1 - -t hardware -c PCM diff --git a/man/shairport-sync.html b/man/shairport-sync.html index 3d76270f..c587b03e 100644 --- a/man/shairport-sync.html +++ b/man/shairport-sync.html @@ -10,11 +10,14 @@ [-a name] [-A latency] [-B command] + [-c configurationfile] [-E command] [--forkedDaapdLatency=latency] + [--get-cover-art] [-i latency] [-L latency] [-m backend] + [--meta-dir=directory] [-o backend] [--password=secret] [-r threshold] @@ -43,8 +46,7 @@

Description

shairport-sync plays audio streamed from iTunes or from an AirPlay - device to an audio device connected via an audio back end. At present, - the only fully-implemented back end is for ALSA.

+ device to an ALSA-compatible audio output device.

A feature of shairport-sync is that the audio is played synchronously. This means that if many devices are playing the same stream at the same @@ -52,13 +54,316 @@ This allows multiple devices play the same source without getting out of phase with one another, enabling, for example, simultaneous multi-room operation.

+ +

shairport-sync can additionally be compiled and configured to stream raw audio to a pipe or to stdout.

+ +

Settings can be made using the configuration file (recommended for all new installations) or by using command-line options.

+ + +

Configuration File Settings

+ +

You should use the configuration file for setting up shairport-sync. + This file is normally /etc/shairport-sync.conf. + You may need to have root privileges to modify it.

+ +

Settings are organised into groups, for example, there is a "general" group of + standard settings, and there is an "alsa" group with settings that pertain to the ALSA + back end. Here is an example of a typical configuration file:

+ +

general = {

+

name = "Mike's Boombox";

+

interpolation = "soxr";

+

password = "secret";

+

};

+

+

alsa = {

+

output_device = "hw:0";

+

mixer_control_name = "PCM";

+

};

+ +

Most settings have sensible default values, so -- as in the example above -- users generally only need to set (1) the service name, (2) a password (if desired) and + (3) the output device. If the output device has a mixer that can be used for volume control, then (4) the volume control's name should be specificed. It is highly desirable to use the output device's mixer for volume control, if available -- response time is reduced to zero and the processor load is reduced. In the example above, "soxr" interpolation was also enabled.

+ +

A sample configuration file with all possible settings, but with all of them commented out, is installed at /etc/shairport-sync.conf.sample.

+ +

To retain backwards compatability with previous versions of shairport-sync + you can use still use command line options, but any new features, etc. will + be available only via configuration file settings.

+ +

The configuration file is processed using the libconfig library + -- see http://www.hyperrealm.com/libconfig/libconfig_manual.html.

+ +

"GENERAL" SETTINGS

+

These are the settings available within the general group:

+ + +

name="service_name";

+ + Use this service_name to identify this player in iTunes, etc. + The default name is "Shairport Sync on <hostname>". + + + + +

password="password";

+ Require the password password to connect to the service. If you leave this setting commented out, no password is needed. + + +

interpolation="mode";

+ Interpolate, or "stuff", the audio stream using the mode. Interpolation here refers to the + process of adding or removing frames of audio to or from the + stream sent to the output device to keep it exactly in synchrony + with the player. + The default mode, "basic", is normally almost completely inaudible. + The alternative mode, "soxr", is even less obtrusive but + requires much more processing power. For this mode, support for + libsoxr, the SoX Resampler Library, must be selected when + shairport-sync is compiled. + + + + +

statistics="setting";

+ Use this setting to enable ("yes") or disable ("no") the output of some statistical information on the console or in the log. The default is to disable statistics. + + + +

mdns_backend="backend";

+ shairport-sync has a number of modules of code ("backends") for interacting with the mDNS service to be used to advertise itself. Normally, the first mDNS backend that works is selected. This setting forces the selection of the specific mDNS backend. The default is "avahi". Perform the command shairport-sync -h to get a list of available mDNS modules. + + +

output_backend="backend";

+ shairport-sync has a number of modules of code ("backends") through which audio is output. Normally, the first audio backend that works is selected. This setting forces the selection of the specific audio backend. The default is "alsa". Perform the command shairport-sync -h to get a list of available audio backends. Only the alsa backend supports synchronisation. + + +

port=portnumber;

+ Use this to specify the portnumber shairport-sync uses to listen for service requests from iTunes, etc. The default is port 5000. + + +

udp_port_base=portnumber;

+ When shairport-sync starts to play audio, it establises three UDP connections to the audio source. Use this setting to specify the starting portnumber for these three ports. It will pick the first three unused ports starting from portnumber. The default is port 6001. + + +

udp_port_range=range;

+ Use this in conjunction with the prevous setting to specify the range of ports that can be checked for availability. Only three ports are needed. The default is 100, thus 100 ports will be checked from port 6001 upwards until three are found. + + +

drift=frames;

+ Allow playback to drift up to frames out of exact synchronization before attempting to correct it. + The default is 88 frames, i.e. 2 ms. The smaller the tolerance, the more likely it is that overcorrection will occur. + Overcorrection is when more corrections (insertions and deletions) are made than are strictly necessary to keep the stream in sync. Use the statistics setting to + monitor correction levels. Corrections should not greatly exceed net corrections. + + + +

resync_threshold=threshold;

+ Resynchronise if timings differ by more than threshold frames. + If the output timing differs from the source timing by more than + the threshold, output will be muted and a full resynchronisation + will occur. The default threshold is 2,205 frames, i.e. 50 + milliseconds. Specify 0 to disable resynchronisation. + + +

log_verbosity=0;

+ Use this to specify how much debugging information should be output or logged. "0" means no debug information, "3" means most debug information. The default is "0". + + +

ignore_volume_control="choice";

+ Set this choice to "yes" if you want the volume to be at 100% no matter what the source's volume control is set to. This might be useful if you want to set the volume on the output device, independently of the setting at the source. The default is "no". + + +

"LATENCIES" SETTINGS

+

There are four default latency settings, chosen automatically. One latency matches the latency used by recent versions of iTunes when playing audio and another matches the latency used by so-called "AirPlay" devices, including iOS devices and iTunes and Quicktime Player when they are playing video. A third latency is used when the audio source is forked-daapd. The fourth latency is the default if no other latency is chosen and is used for older versions of iTunes.

+

If you want to change latencies to compensate for a delay in the audio output device (which will have the same effect on all sources), instead of changing these individual latencies, consider using the audio_backend_latency_offset setting in the alsa group (or the appropriate other group if you're not outputing through the alsa backend).

+ + + +

itunes=latency;

+ This is the latency, in frames, used for iTunes 10 or later. Default is 99,400. + + +

airplay=latency;

+ This is the latency, in frames, used for AirPlay devices, including iOS devices and iTunes and Quicktime Player when they are playing video. Default is 88,200. + + +

forkedDaapd=latency;

+ This is the latency, in frames, used for forkedDaapd sources. Default is 99,400. + + +

default=latency;

+ This is the latency, in frames, used when the source is unrecognised. Default is 88,200. + + +

"METADATA" SETTINGS

+

shairport-sync can process metadata provided by the source, such as Track Number, Album Name, cover art, etc. and can provide additional metadata such as volume level, + pause/resume, etc. It provides the metadata to a pipe, by default /tmp/shairport-sync-metadata. + To process metadata, shairport-sync must have been compiled with metadata support included. + You can check that this is so by running shairport-sync -V; the identification string will contain the word metadata.

+

The metadata group of settings allow you to enable metadata handling and to control certain aspects of it:

+ + + +

enabled="choice";

+ Set the choice to "yes" to enable shairport-sync to look for metadata from the audio source and to forward it, + along with metadata generated by shairport-sync itself, to the metadata pipe. The default is "no". + + +

include_cover_art="choice";

+ Set the choice to "yes" to enable shairport-sync to look for cover art from the audio source and to include it in the feed to the metadata pipe. + You must also enable metadata (see above). + One reason for not including cover art is that the images can sometimes be very large and may delay transmission of subsequent metadata through the pipe. + The default is "no". + + +

pipe_name="filepathname";

+ Specify the absolute path name of the pipe through which metadata should be sent The default is /tmp/shairport-sync-metadata". + + +

"SESSIONCONTROL" SETTINGS

+

shairport-sync can run programs just before it starts to play an audio stream and just after it finishes. + You specify them using the sessioncontrol group settings run_this_before_play_begins and run_this_after_play_ends.

+ + + +

run_this_before_play_begins="/path/to/application and args";

+ Here you can specify a program and its arguments that will be run just before a play session begins. Be careful to include the full path to the application. + The application must be marked as executable and, if it is a script, its first line must begin with the standard #!/bin/... as appropriate. + + +

run_this_after_play_ends="/path/to/application and args";

+ Here you can specify a program and its arguments that will be run just after a play session ends. Be careful to include the full path to the application. + The application must be marked as executable and, if it is a script, its first line must begin with the standard #!/bin/... as appropriate. + + +

wait_for_completion="choice";

+ Set choice to "yes" to make shairport-sync wait until the programs specified in the run_this_before_play_begins + and run_this_after_play_ends have completed execution before continuing. The default is "no". + + +

allow_session_interruption="choice";

+ If choice is set to "yes", then another source will be able to interrupt an existing play session and start a new one. + When set to "no" (the default), other devices attempting to interrupt a session will fail, receiving a busy signal. + + +

session_timeout=seconds;

+ If a play session has been established and the source disappears without warning (such as a device going out of range of a network) + then wait for seconds seconds before ending the session. Once the session has terminated, other devices can use it. + The default is 120 seconds. + + +

"ALSA" SETTINGS

+

These settings are for the ALSA back end, used to communicate with audio output devices in the ALSA system. + (By the way, you can use tools such as alsamixer or aplay to discover what devices are available.) + Use these settings to select the output device and the mixer control to be used to control the output volume. + You can additionally set the desired size of the output buffer and you can adjust overall latency. Here are the alsa group settings:

+ + +

output_device="output_device";

+ Use the output device called output_device. The default is the device called "default". + + + +

mixer_control_name="name";

+ Specify the name of the mixer control to be used by shairport-sync to control the volume. + The mixer control must be on the mixer device, which by default is the output device. + If you do not specify a mixer control name, shairport-sync will adjust the volume in software. + +

mixer_type="mixer_type";

+ This setting is deprecated and will be removed soon. If you wish to use a mixer control to control the volume, then set mixer_type to "hardware". + The default is "software". + + +

mixer_device="mixer_device";

+ By default, the mixer is assumed to be output_device. Use this setting to specify a device other than the output device. + + +

audio_backend_latency_offset=offset;

+ Set this offset, in frames, to compensate for a fixed delay in the audio back end. + For example, if the output device delays by 100 ms, set this to -4410. + + +

audio_backend_buffer_desired_length=length;

+ Use this to set the desired number frames to be in the output device's hardware output buffer. + The default is 6,615 frames, or 0.15 seconds. If set too small, buffer underflow may occur on low-powered machines. + If too large, the response times when using software volume control (i.e. when not using a mixer control to control volume) become annoying, + or it may exceed the hardware buffer size. + It may need to be larger on low-powered machines that are also performing other tasks, such as processing metadata. + + +

"PIPE" SETTINGS

+

These settings are for the PIPE backend, used to route audio to a named unix pipe. The audio is in raw CD audio format: PCM 16 bit little endian, 44,100 samples per second, + stereo.

+

Use the name setting to set the name and location of the pipe.

+

There are two further settings affecting timing that might be useful if the pipe reader is, for example, + a program to play an audio stream such as aplay. The audio_backend_latency_offset affects precisely when the first audio packet is sent + and the audio_backend_buffer_desired_length setting affects the nominal output buffer size.

+

These are the settings available within the pipe group:

+ + +

name="/path/to/pipe";

+ Use this to specify the name and location of the pipe. The pipe will be created and opened when shairport-sync starts up and will be closed upon shutdown. + Frames of audio will be sent to the pipe in packets of 352 frames and will be discarded if the pipe has not have a reader attached. + The sender will wait for up to five seconds for a packet to be written before discarding it. + + + +

audio_backend_latency_offset=offset_in_frames;

+ + Packets of audio frames are written to the pipe synchronously -- that is, they are written to at exactly the time they should be played. + You can offset the time of initial audio output relative to its nominal time using this setting. + For example to send an audio stream to the pipe 100 milliseconds before it is due to be played, set this to -4410. Default setting is 0. + + + +

audio_backend_buffer_desired_length=buffer_length_in_frames;

+ + Use this setting, in frames, to set the size of the output buffer. It works by determining how soon the second and subsequent packets of audio frames are sent to the pipe. + For example, if you send the first packet of audio exactly when it is due and, using a audio_backend_buffer_desired_length setting of 44100, + send subsequent packets of audio a second before they are due to be played, they will be buffered in the pipe reader's buffer, giving it a nominal buffer size of 44,100 frames. + Note that if the pipe reader consumes audio packets faster or slower than they are supplied, the buffer will eventually empty or overflow -- + shairport-sync performs no stuffing or interpolation when writing to a pipe. Default setting is 44,100 frames. + + + +

"STDOUT" SETTINGS

+

These settings are for the STDOUT backend, used to route audio to standard output ("stdout"). + The audio is in raw CD audio format: PCM 16 bit little endian, 44,100 samples per second, stereo.

+

There are two settings affecting timing that might be useful if the stdout reader is, for example, + a program to play an audio stream such as aplay. The audio_backend_latency_offset affects precisely when the first audio packet is sent + and the audio_backend_buffer_desired_length setting affects the nominal output buffer size.

+

These are the settings available within the stdout group:

+ + +

audio_backend_latency_offset=offset_in_frames;

+ + Packets of audio frames are written to stdout synchronously -- that is, they are written at exactly the time they should be played. + You can offset the time of initial audio output relative to its nominal time using this setting. + For example to send an audio stream to stdout 100 milliseconds before it is due to be played, set this to -4410. Default setting is 0. + + + +

audio_backend_buffer_desired_length=buffer_length_in_frames;

+ + Use this setting, in frames, to set the size of the output buffer. It works by determining how soon the second and subsequent packets of audio frames are sent to stdout. + For example, if you send the first packet of audio exactly when it is due and, using a audio_backend_buffer_desired_length setting of 44100, + send subsequent packets of audio a second before they are due to be played, they will be buffered in the stdout reader's buffer, giving it a nominal buffer size of 44,100 frames. + Note that if the stdout reader consumes audio packets faster or slower than they are supplied, the buffer will eventually empty or overflow -- + shairport-sync performs no stuffing or interpolation when writing to stdout. Default setting is 44,100 frames. + + + +

Options

+

Note: if you are setting up shairport-sync for the first time or are updating an existing installation, + you are encouraged to use the configuration file settings described above. Most of the options described below + simply replicate the configuration settings and are retained to provide backward compatability with older installations of shairport-sync.

+

Many of the options take sensible default values, so you can normally ignore most of them. See the EXAMPLES section for typical usages.

@@ -105,6 +410,13 @@ +

-c filename | --configfile=filename

+

+ Read configuration settings from filename. The default is to read them from /etc/shairport-sync.conf. For information about configuration settings, see the "Configuration File Settings" section above. +

+ + +

-D | --disconnectFromOutput

Disconnect the shairport-sync daemon from the output device and @@ -144,6 +456,19 @@ frames to the second.

+ + +

--get-coverart

+

+ This option requires the --meta-dir option to be set, and enables + shairport-sync to request cover art from the source and to transmit it through + the metadata pipe.

+

Please note that cover art data may be very large, and may place too great a + burden on your network. Also, at the time of writing, there appears to be a + serious bug in iTunes which stalls transmission of large cover art items, + stalling and potentially crashing shairport-sync. +

+

-h | --help

@@ -178,6 +503,17 @@ +

--meta-dir=directory

+

+ Listen for metadata coming from the source and send it, along with metadata from + shairport-sync itself, to a pipe called shairport-sync-metadata + in the directory you specify. If you add the --get-cover-art then + cover art will be sent through the pipe too. See https://github.com/mikebrady/shairport-sync-metadata-reader + for a sample metadata reader. +

+ + +

-m mdnsbackend | --mdns=mdnsbackend

Force the use of the specified mDNS backend to advertise the @@ -358,11 +694,8 @@

-t devicetype

- The type of the output device is devicetype which must be - hardware or software. The default is software. - If you specify hardware you must also specify the output device using the - -d option and the name of the volume control using the -c option. - You may also need to specify the mixer using the -m option. + This option is deprecated and is ignored. For your information, its functionality has been automatically incorporated in the -c option + -- if you specify a mixer name with the -c option, it is assumed that the mixer is implemented in hardware.

@@ -376,7 +709,6 @@ -- -d hw:1,0 -m hw:1 - -t hardware -c PCM
@@ -385,8 +717,7 @@ Library-based stuffing ( -S soxr ). The audio backend options following the -- separator specify that the audio will be output on output 0 of soundcard 1 ( - -d hw:1,0 ) and will take advantage of the same sound card's hardware - (-t hardware) mixer ( -m hw:1 ) + -d hw:1,0 ) and will take advantage of the same sound card's mixer ( -m hw:1 ) using the level control named "PCM" ( -c "PCM" ).

The example above is slightly contrived in order to show the use of the -m option. @@ -398,7 +729,6 @@ -S soxr -- -d hw:1 - -t hardware -c PCM
diff --git a/mdns.c b/mdns.c index 53b5df76..f4491fd1 100644 --- a/mdns.c +++ b/mdns.c @@ -24,7 +24,6 @@ * OTHER DEALINGS IN THE SOFTWARE. */ - #include #include #include @@ -47,78 +46,66 @@ extern mdns_backend mdns_tinysvcmdns; static mdns_backend *mdns_backends[] = { #ifdef CONFIG_AVAHI - &mdns_avahi, - &mdns_external_avahi, + &mdns_avahi, &mdns_external_avahi, #endif #ifdef CONFIG_HAVE_DNS_SD_H - &mdns_dns_sd, - &mdns_external_dns_sd, + &mdns_dns_sd, &mdns_external_dns_sd, #endif #ifdef CONFIG_TINYSVCMDNS &mdns_tinysvcmdns, #endif - NULL -}; + NULL}; void mdns_register(void) { - char *mdns_apname = alloca(strlen(config.apname) + 14); - char *p = mdns_apname; - int i; - for (i=0; i<6; i++) { - sprintf(p, "%02X", config.hw_addr[i]); - p += 2; - } - *p++ = '@'; - strcpy(p, config.apname); + char *mdns_apname = alloca(strlen(config.apname) + 14); + char *p = mdns_apname; + int i; + for (i = 0; i < 6; i++) { + sprintf(p, "%02X", config.hw_addr[i]); + p += 2; + } + *p++ = '@'; + strcpy(p, config.apname); - mdns_backend **b = NULL; - - if (config.mdns_name != NULL) - { - for (b = mdns_backends; *b; b++) - { - if (strcmp((*b)->name, config.mdns_name) != 0) // Not the one we are looking for - continue; - int error = (*b)->mdns_register(mdns_apname, config.port); - if (error >= 0) - { - config.mdns = *b; - } - break; - } + mdns_backend **b = NULL; - if (*b == NULL) - warn("%s mDNS backend not found"); - } - else - { - for (b = mdns_backends; *b; b++) - { - int error = (*b)->mdns_register(mdns_apname, config.port); - if (error >= 0) - { - config.mdns = *b; - break; - } - } + if (config.mdns_name != NULL) { + for (b = mdns_backends; *b; b++) { + if (strcmp((*b)->name, config.mdns_name) != 0) // Not the one we are looking for + continue; + int error = (*b)->mdns_register(mdns_apname, config.port); + if (error >= 0) { + config.mdns = *b; + } + break; } - if (config.mdns == NULL) - die("Could not establish mDNS advertisement!"); + if (*b == NULL) + warn("%s mDNS backend not found"); + } else { + for (b = mdns_backends; *b; b++) { + int error = (*b)->mdns_register(mdns_apname, config.port); + if (error >= 0) { + config.mdns = *b; + break; + } + } + } + + if (config.mdns == NULL) + die("Could not establish mDNS advertisement!"); } void mdns_unregister(void) { - if (config.mdns) { - config.mdns->mdns_unregister(); - } + if (config.mdns) { + config.mdns->mdns_unregister(); + } } void mdns_ls_backends(void) { - mdns_backend **b = NULL; - printf("Available mDNS backends: \n"); - for (b = mdns_backends; *b; b++) - { - printf(" %s\n", (*b)->name); - } + mdns_backend **b = NULL; + printf("Available mDNS backends: \n"); + for (b = mdns_backends; *b; b++) { + printf(" %s\n", (*b)->name); + } } - diff --git a/mdns.h b/mdns.h index 5fa30d19..a15b77fa 100644 --- a/mdns.h +++ b/mdns.h @@ -1,6 +1,8 @@ #ifndef _MDNS_H #define _MDNS_H +#include "config.h" + extern int mdns_pid; void mdns_unregister(void); @@ -8,13 +10,24 @@ void mdns_register(void); void mdns_ls_backends(void); typedef struct { - char *name; - int (*mdns_register)(char *apname, int port); - void (*mdns_unregister)(void); + char *name; + int (*mdns_register)(char *apname, int port); + void (*mdns_unregister)(void); } mdns_backend; -#define MDNS_RECORD_WITHOUT_METADATA "tp=UDP", "sm=false", "ek=1", "et=0,1", "cn=0,1", "ch=2", \ - "ss=16", "sr=44100", "vn=3", "txtvers=1", \ - config.password ? "pw=true" : "pw=false" +// text and progress only -- picture feed really buggy from iTunes + +#ifdef CONFIG_METADATA + +#define METADATA_EXPRESSION config.get_coverart ? "md=0,1,2" : "md=0,2" + +#define MDNS_RECORD_WITH_METADATA \ + "tp=UDP", "sm=false", "ek=1", "et=0,1", "cn=0,1", "ch=2", METADATA_EXPRESSION, "ss=16", \ + "sr=44100", "vn=3", "txtvers=1", config.password ? "pw=true" : "pw=false" +#endif + +#define MDNS_RECORD_WITHOUT_METADATA \ + "tp=UDP", "sm=false", "ek=1", "et=0,1", "cn=0,1", "ch=2", "ss=16", "sr=44100", "vn=3", \ + "txtvers=1", config.password ? "pw=true" : "pw=false" #endif // _MDNS_H diff --git a/mdns_avahi.c b/mdns_avahi.c index b142dd55..31800b8b 100644 --- a/mdns_avahi.c +++ b/mdns_avahi.c @@ -42,111 +42,104 @@ static AvahiThreadedPoll *tpoll = NULL; static char *name = NULL; static int port = 0; -static void egroup_callback(AvahiEntryGroup *g, - AvahiEntryGroupState state, +static void egroup_callback(AvahiEntryGroup *g, AvahiEntryGroupState state, AVAHI_GCC_UNUSED void *userdata) { - if (state==AVAHI_ENTRY_GROUP_COLLISION) - die("service name already exists on network!"); - if (state==AVAHI_ENTRY_GROUP_FAILURE) - die("avahi entry group failure!"); + if (state == AVAHI_ENTRY_GROUP_COLLISION) + die("service name already exists on network!"); + if (state == AVAHI_ENTRY_GROUP_FAILURE) + die("avahi entry group failure!"); } static void register_service(AvahiClient *c) { - debug(1, "avahi: register_service."); - if (!group) - group = avahi_entry_group_new(c, egroup_callback, NULL); - if (!group) - die("avahi_entry_group_new failed"); + debug(1, "avahi: register_service."); + if (!group) + group = avahi_entry_group_new(c, egroup_callback, NULL); + if (!group) + die("avahi_entry_group_new failed"); - if (!avahi_entry_group_is_empty(group)) - return; + if (!avahi_entry_group_is_empty(group)) + return; - int ret; - debug(1,"Avahi without metadata"); - ret = avahi_entry_group_add_service(group, - AVAHI_IF_UNSPEC, - AVAHI_PROTO_UNSPEC, - 0, - name, - "_raop._tcp", - NULL, - NULL, - port, - MDNS_RECORD_WITHOUT_METADATA, + int ret; +#ifdef CONFIG_METADATA + if (config.metadata_enabled) { + debug(1, "Avahi with metadata"); + ret = avahi_entry_group_add_service(group, AVAHI_IF_UNSPEC, AVAHI_PROTO_UNSPEC, 0, name, + "_raop._tcp", NULL, NULL, port, MDNS_RECORD_WITH_METADATA, NULL); - - if (ret < 0) - die("avahi_entry_group_add_service failed"); + } else { +#endif + debug(1, "Avahi without metadata"); + ret = avahi_entry_group_add_service(group, AVAHI_IF_UNSPEC, AVAHI_PROTO_UNSPEC, 0, name, + "_raop._tcp", NULL, NULL, port, + MDNS_RECORD_WITHOUT_METADATA, NULL); +#ifdef CONFIG_METADATA + } +#endif - ret = avahi_entry_group_commit(group); - if (ret < 0) - die("avahi_entry_group_commit failed"); + if (ret < 0) + die("avahi_entry_group_add_service failed"); + + ret = avahi_entry_group_commit(group); + if (ret < 0) + die("avahi_entry_group_commit failed"); } -static void client_callback(AvahiClient *c, - AvahiClientState state, - AVAHI_GCC_UNUSED void * userdata) { - switch (state) { - case AVAHI_CLIENT_S_REGISTERING: - if (group) - avahi_entry_group_reset(group); - break; +static void client_callback(AvahiClient *c, AvahiClientState state, + AVAHI_GCC_UNUSED void *userdata) { + switch (state) { + case AVAHI_CLIENT_S_REGISTERING: + if (group) + avahi_entry_group_reset(group); + break; - case AVAHI_CLIENT_S_RUNNING: - register_service(c); - break; + case AVAHI_CLIENT_S_RUNNING: + register_service(c); + break; - case AVAHI_CLIENT_FAILURE: - case AVAHI_CLIENT_S_COLLISION: - die("avahi client failure"); + case AVAHI_CLIENT_FAILURE: + case AVAHI_CLIENT_S_COLLISION: + die("avahi client failure"); - case AVAHI_CLIENT_CONNECTING: - break; - } + case AVAHI_CLIENT_CONNECTING: + break; + } } static int avahi_register(char *srvname, int srvport) { - debug(1, "avahi: avahi_register."); - name = strdup(srvname); - port = srvport; + debug(1, "avahi: avahi_register."); + name = strdup(srvname); + port = srvport; - int err; - if (!(tpoll = avahi_threaded_poll_new())) { - warn("couldn't create avahi threaded tpoll!"); - return -1; - } - if (!(client = avahi_client_new(avahi_threaded_poll_get(tpoll), - 0, - client_callback, - NULL, - &err))) { - warn("couldn't create avahi client: %s!", avahi_strerror(err)); - return -1; - } + int err; + if (!(tpoll = avahi_threaded_poll_new())) { + warn("couldn't create avahi threaded tpoll!"); + return -1; + } + if (!(client = + avahi_client_new(avahi_threaded_poll_get(tpoll), 0, client_callback, NULL, &err))) { + warn("couldn't create avahi client: %s!", avahi_strerror(err)); + return -1; + } - if (avahi_threaded_poll_start(tpoll) < 0) { - warn("couldn't start avahi tpoll thread"); - return -1; - } + if (avahi_threaded_poll_start(tpoll) < 0) { + warn("couldn't start avahi tpoll thread"); + return -1; + } - return 0; + return 0; } static void avahi_unregister(void) { - debug(1, "avahi: avahi_unregister."); - if (tpoll) - avahi_threaded_poll_stop(tpoll); - tpoll = NULL; + debug(1, "avahi: avahi_unregister."); + if (tpoll) + avahi_threaded_poll_stop(tpoll); + tpoll = NULL; - if (name) - free(name); - name = NULL; + if (name) + free(name); + name = NULL; } -mdns_backend mdns_avahi = -{ - .name = "avahi", - .mdns_register = avahi_register, - .mdns_unregister = avahi_unregister -}; - +mdns_backend mdns_avahi = { + .name = "avahi", .mdns_register = avahi_register, .mdns_unregister = avahi_unregister}; diff --git a/mdns_dns_sd.c b/mdns_dns_sd.c index bbd42ca2..b9e11b45 100644 --- a/mdns_dns_sd.c +++ b/mdns_dns_sd.c @@ -33,70 +33,62 @@ static DNSServiceRef service; static int mdns_dns_sd_register(char *apname, int port) { - const char *recordwithoutmetadata[] = { MDNS_RECORD_WITHOUT_METADATA, NULL }; - const char **record = recordwithoutmetadata; - - uint16_t length = 0; - const char **field; + const char *recordwithoutmetadata[] = {MDNS_RECORD_WITHOUT_METADATA, NULL}; +#ifdef CONFIG_METADATA + const char *recordwithmetadata[] = {MDNS_RECORD_WITH_METADATA, NULL}; +#endif + char **record; +#ifdef CONFIG_METADATA + if (config.meta_dir) + record = recordwithmetadata; + else +#endif + record = recordwithoutmetadata; - // Concatenate string contained i record into buf. + uint16_t length = 0; + const char **field; - for (field = record; *field; field ++) - { - length += strlen(*field) + 1; // One byte for length each time - } + // Concatenate string contained i record into buf. - char *buf = malloc(length * sizeof(char)); - if (buf == NULL) - { - warn("dns_sd: buffer record allocation failed"); - return -1; - } + for (field = record; *field; field++) { + length += strlen(*field) + 1; // One byte for length each time + } - char *p = buf; + char *buf = malloc(length * sizeof(char)); + if (buf == NULL) { + warn("dns_sd: buffer record allocation failed"); + return -1; + } - for (field = record; *field; field ++) - { - char * newp = stpcpy(p + 1, *field); - *p = newp - p - 1; - p = newp; - } + char *p = buf; - DNSServiceErrorType error; - error = DNSServiceRegister(&service, - 0, - kDNSServiceInterfaceIndexAny, - apname, - "_raop._tcp", - "", - NULL, - htons((uint16_t)port), - length, - buf, - NULL, - NULL); + for (field = record; *field; field++) { + char *newp = stpcpy(p + 1, *field); + *p = newp - p - 1; + p = newp; + } - free(buf); + DNSServiceErrorType error; + error = DNSServiceRegister(&service, 0, kDNSServiceInterfaceIndexAny, apname, "_raop._tcp", "", + NULL, htons((uint16_t)port), length, buf, NULL, NULL); - if (error == kDNSServiceErr_NoError) - return 0; - else - { - warn("dns-sd: DNSServiceRegister error %d", error); - return -1; - } + free(buf); + + if (error == kDNSServiceErr_NoError) + return 0; + else { + warn("dns-sd: DNSServiceRegister error %d", error); + return -1; + } } static void mdns_dns_sd_unregister(void) { - if (service) - { - DNSServiceRefDeallocate(service); - service = NULL; - } + if (service) { + DNSServiceRefDeallocate(service); + service = NULL; + } } -mdns_backend mdns_dns_sd = { - .name = "dns-sd", - .mdns_register = mdns_dns_sd_register, - .mdns_unregister = mdns_dns_sd_unregister -}; +mdns_backend mdns_dns_sd = {.name = "dns-sd", + .mdns_register = mdns_dns_sd_register, + .mdns_unregister = mdns_dns_sd_unregister}; diff --git a/mdns_external.c b/mdns_external.c index e93ea37b..f9ba20c6 100644 --- a/mdns_external.c +++ b/mdns_external.c @@ -41,120 +41,128 @@ int mdns_pid = 0; * Check errno for error details. */ static int fork_execvp(const char *file, char *const argv[]) { - int execpipe[2]; - int pid = 0; - if (pipe(execpipe) < 0) { - return -1; - } + int execpipe[2]; + int pid = 0; + if (pipe(execpipe) < 0) { + return -1; + } - if (fcntl(execpipe[1], F_SETFD, fcntl(execpipe[1], F_GETFD) | FD_CLOEXEC) < 0) { - close(execpipe[0]); - close(execpipe[1]); - return -1; - } + if (fcntl(execpipe[1], F_SETFD, fcntl(execpipe[1], F_GETFD) | FD_CLOEXEC) < 0) { + close(execpipe[0]); + close(execpipe[1]); + return -1; + } - pid = fork(); - if (pid < 0) { - close(execpipe[0]); - close(execpipe[1]); - return -1; - } - else if(pid == 0) { // Child - close(execpipe[0]); // Close the read end - execvp(file, argv); + pid = fork(); + if (pid < 0) { + close(execpipe[0]); + close(execpipe[1]); + return -1; + } else if (pid == 0) { // Child + close(execpipe[0]); // Close the read end + execvp(file, argv); - // If we reach this point then execve has failed. - // Write erno's value into the pipe and exit. - int ignore = write(execpipe[1], &errno, sizeof(errno)); + // If we reach this point then execve has failed. + // Write erno's value into the pipe and exit. + int ignore = write(execpipe[1], &errno, sizeof(errno)); - _exit(-1); - return 0; // Just to make the compiler happy. - } - else { // Parent - close(execpipe[1]); // Close the write end + _exit(-1); + return 0; // Just to make the compiler happy. + } else { // Parent + close(execpipe[1]); // Close the write end - int childErrno; - // Block until child closes the pipe or sends errno. - if(read(execpipe[0], &childErrno, sizeof(childErrno)) == sizeof(childErrno)) { // We received errno - errno = childErrno; - return -1; - } - else { // Child closed the pipe. execvp was successful. - return pid; - } + int childErrno; + // Block until child closes the pipe or sends errno. + if (read(execpipe[0], &childErrno, sizeof(childErrno)) == + sizeof(childErrno)) { // We received errno + errno = childErrno; + return -1; + } else { // Child closed the pipe. execvp was successful. + return pid; } + } } static int mdns_external_avahi_register(char *apname, int port) { - char mdns_port[6]; - sprintf(mdns_port, "%d", config.port); + char mdns_port[6]; + sprintf(mdns_port, "%d", config.port); - char *argvwithoutmetadata[] = { - NULL, apname, "_raop._tcp", mdns_port, MDNS_RECORD_WITHOUT_METADATA, NULL - }; - char **argv = argvwithoutmetadata; - - argv[0] = "avahi-publish-service"; - int pid = fork_execvp(argv[0], argv); - if (pid >= 0) - { - mdns_pid = pid; - return 0; - } - else - warn("Calling %s failed !", argv[0]); + char *argvwithoutmetadata[] = {NULL, apname, "_raop._tcp", mdns_port, + MDNS_RECORD_WITHOUT_METADATA, NULL}; +#ifdef CONFIG_METADATA + char *argvwithmetadata[] = {NULL, apname, "_raop._tcp", mdns_port, MDNS_RECORD_WITH_METADATA, + NULL}; +#endif + char **argv; - argv[0] = "mDNSPublish"; - pid = fork_execvp(argv[0], argv); - if (pid >= 0) - { - mdns_pid = pid; - return 0; - } - else - warn("Calling %s failed !", argv[0]); - - // If we reach here, both execvp calls failed. - return -1; +#ifdef CONFIG_METADATA + if (config.metadata_enabled) + argv = argvwithmetadata; + else +#endif + argv = argvwithoutmetadata; + + argv[0] = "avahi-publish-service"; + int pid = fork_execvp(argv[0], argv); + if (pid >= 0) { + mdns_pid = pid; + return 0; + } else + warn("Calling %s failed !", argv[0]); + + argv[0] = "mDNSPublish"; + pid = fork_execvp(argv[0], argv); + if (pid >= 0) { + mdns_pid = pid; + return 0; + } else + warn("Calling %s failed !", argv[0]); + + // If we reach here, both execvp calls failed. + return -1; } static int mdns_external_dns_sd_register(char *apname, int port) { - char mdns_port[6]; - sprintf(mdns_port, "%d", config.port); + char mdns_port[6]; + sprintf(mdns_port, "%d", config.port); - char *argvwithoutmetadata[] = { - NULL, apname, "_raop._tcp", mdns_port, MDNS_RECORD_WITHOUT_METADATA, NULL - }; + char *argvwithoutmetadata[] = {NULL, apname, "_raop._tcp", mdns_port, + MDNS_RECORD_WITHOUT_METADATA, NULL}; - char **argv=argvwithoutmetadata; +#ifdef CONFIG_METADATA + char *argvwithmetadata[] = {NULL, apname, "_raop._tcp", mdns_port, MDNS_RECORD_WITH_METADATA, + NULL}; +#endif - int pid = fork_execvp(argv[0], argv); - if (pid >= 0) - { - mdns_pid = pid; - return 0; - } - else - warn("Calling %s failed !", argv[0]); + char **argv; +#ifdef CONFIG_METADATA + if (config.metadata_enabled) + argv = argvwithmetadata; + else +#endif - return -1; + argv = argvwithoutmetadata; + + int pid = fork_execvp(argv[0], argv); + if (pid >= 0) { + mdns_pid = pid; + return 0; + } else + warn("Calling %s failed !", argv[0]); + + return -1; } static void kill_mdns_child(void) { - if (mdns_pid) - kill(mdns_pid, SIGTERM); - mdns_pid = 0; + if (mdns_pid) + kill(mdns_pid, SIGTERM); + mdns_pid = 0; } -mdns_backend mdns_external_avahi = { - .name = "external-avahi", - .mdns_register = mdns_external_avahi_register, - .mdns_unregister = kill_mdns_child -}; - -mdns_backend mdns_external_dns_sd = { - .name = "external-dns-sd", - .mdns_register = mdns_external_dns_sd_register, - .mdns_unregister = kill_mdns_child -}; +mdns_backend mdns_external_avahi = {.name = "external-avahi", + .mdns_register = mdns_external_avahi_register, + .mdns_unregister = kill_mdns_child}; +mdns_backend mdns_external_dns_sd = {.name = "external-dns-sd", + .mdns_register = mdns_external_dns_sd_register, + .mdns_unregister = kill_mdns_child}; diff --git a/mdns_tinysvcmdns.c b/mdns_tinysvcmdns.c index 48361133..a3c475a2 100644 --- a/mdns_tinysvcmdns.c +++ b/mdns_tinysvcmdns.c @@ -27,7 +27,7 @@ #include #include #include -#include +#include #include #include #include @@ -39,117 +39,108 @@ static struct mdnsd *svr = NULL; static int mdns_tinysvcmdns_register(char *apname, int port) { - struct ifaddrs *ifalist; - struct ifaddrs *ifa; + struct ifaddrs *ifalist; + struct ifaddrs *ifa; - svr = mdnsd_start(); - if (svr == NULL) { - warn("tinysvcmdns: mdnsd_start() failed"); - return -1; + svr = mdnsd_start(); + if (svr == NULL) { + warn("tinysvcmdns: mdnsd_start() failed"); + return -1; + } + + // Thanks to Paul Lietar for this + // room for name + .local + NULL + char hostname[100 + 6]; + gethostname(hostname, 99); + // according to POSIX, this may be truncated without a final NULL ! + hostname[99] = 0; + + // will not work if the hostname doesn't end in .local + char *hostend = hostname + strlen(hostname); + if ((strlen(hostname) < strlen(".local")) || (strcmp(hostend - 6, ".local") != 0)) { + strcat(hostname, ".local"); + } + + if (getifaddrs(&ifalist) < 0) { + warn("tinysvcmdns: getifaddrs() failed"); + return -1; + } + + ifa = ifalist; + + // Look for an ipv4/ipv6 non-loopback interface to use as the main one. + for (ifa = ifalist; ifa != NULL; ifa = ifa->ifa_next) { + if (!(ifa->ifa_flags & IFF_LOOPBACK) && ifa->ifa_addr && ifa->ifa_addr->sa_family == AF_INET) { + uint32_t main_ip = ((struct sockaddr_in *)ifa->ifa_addr)->sin_addr.s_addr; + + mdnsd_set_hostname(svr, hostname, main_ip); // TTL should be 120 seconds + break; + } else if (!(ifa->ifa_flags & IFF_LOOPBACK) && ifa->ifa_addr && + ifa->ifa_addr->sa_family == AF_INET6) { + struct in6_addr *addr = &((struct sockaddr_in6 *)ifa->ifa_addr)->sin6_addr; + + mdnsd_set_hostname_v6(svr, hostname, addr); // TTL should be 120 seconds + break; } + } - // Thanks to Paul Lietar for this - // room for name + .local + NULL - char hostname[100 + 6]; - gethostname(hostname, 99); - // according to POSIX, this may be truncated without a final NULL ! - hostname[99] = 0; + if (ifa == NULL) { + warn("tinysvcmdns: no non-loopback ipv4 or ipv6 interface found"); + return -1; + } - // will not work if the hostname doesn't end in .local - char *hostend = hostname + strlen(hostname); - if ((strlen(hostname) < strlen(".local")) || (strcmp(hostend - 6, ".local")!=0)) { - strcat(hostname, ".local"); - } - - if (getifaddrs(&ifalist) < 0) - { - warn("tinysvcmdns: getifaddrs() failed"); - return -1; + // Skip the first one, it was already added by set_hostname + for (ifa = ifa->ifa_next; ifa != NULL; ifa = ifa->ifa_next) { + if (ifa->ifa_flags & IFF_LOOPBACK) // Skip loop-back interfaces + continue; + + switch (ifa->ifa_addr->sa_family) { + case AF_INET: { // ipv4 + uint32_t ip = ((struct sockaddr_in *)ifa->ifa_addr)->sin_addr.s_addr; + struct rr_entry *a_e = rr_create_a(create_nlabel(hostname), ip); // TTL should be 120 seconds + mdnsd_add_rr(svr, a_e); + } break; + case AF_INET6: { // ipv6 + struct in6_addr *addr = &((struct sockaddr_in6 *)ifa->ifa_addr)->sin6_addr; + struct rr_entry *aaaa_e = + rr_create_aaaa(create_nlabel(hostname), addr); // TTL should be 120 seconds + mdnsd_add_rr(svr, aaaa_e); + } break; } + } - ifa = ifalist; + freeifaddrs(ifa); - // Look for an ipv4/ipv6 non-loopback interface to use as the main one. - for (ifa = ifalist; ifa != NULL; ifa = ifa->ifa_next) - { - if (!(ifa->ifa_flags & IFF_LOOPBACK) && ifa->ifa_addr && - ifa->ifa_addr->sa_family == AF_INET) - { - uint32_t main_ip = ((struct sockaddr_in *)ifa->ifa_addr)->sin_addr.s_addr; + char *txtwithoutmetadata[] = {MDNS_RECORD_WITHOUT_METADATA, NULL}; +#ifdef CONFIG_METADATA + char *txtwithmetadata[] = {MDNS_RECORD_WITH_METADATA, NULL}; +#endif + char **txt; - mdnsd_set_hostname(svr, hostname, main_ip); // TTL should be 120 seconds - break; - } - else if (!(ifa->ifa_flags & IFF_LOOPBACK) && ifa->ifa_addr && - ifa->ifa_addr->sa_family == AF_INET6) - { - struct in6_addr *addr = &((struct sockaddr_in6 *)ifa->ifa_addr)->sin6_addr; +#ifdef CONFIG_METADATA + if (config.meta_dir) + txt = txtwithmetadata; + else +#endif - mdnsd_set_hostname_v6(svr, hostname, addr); // TTL should be 120 seconds - break; - } - } + txt = txtwithoutmetadata; - if (ifa == NULL) - { - warn("tinysvcmdns: no non-loopback ipv4 or ipv6 interface found"); - return -1; - } + struct mdns_service *svc = + mdnsd_register_svc(svr, apname, "_raop._tcp.local", port, NULL, + (const char **)txt); // TTL should be 75 minutes, i.e. 4500 seconds + mdns_service_destroy(svc); - // Skip the first one, it was already added by set_hostname - for (ifa = ifa->ifa_next; ifa != NULL; ifa = ifa->ifa_next) - { - if (ifa->ifa_flags & IFF_LOOPBACK) // Skip loop-back interfaces - continue; - - switch (ifa->ifa_addr->sa_family) - { - case AF_INET: { // ipv4 - uint32_t ip = ((struct sockaddr_in *)ifa->ifa_addr)->sin_addr.s_addr; - struct rr_entry *a_e = rr_create_a(create_nlabel(hostname), ip); // TTL should be 120 seconds - mdnsd_add_rr(svr, a_e); - } - break; - case AF_INET6: { // ipv6 - struct in6_addr *addr = &((struct sockaddr_in6 *)ifa->ifa_addr)->sin6_addr; - struct rr_entry *aaaa_e = rr_create_aaaa(create_nlabel(hostname), addr); // TTL should be 120 seconds - mdnsd_add_rr(svr, aaaa_e); - } - break; - } - } - - freeifaddrs(ifa); - - char *txtwithoutmetadata[] = { MDNS_RECORD_WITHOUT_METADATA, NULL }; - char **txt = txtwithoutmetadata; - - - - struct mdns_service *svc = mdnsd_register_svc(svr, - apname, - "_raop._tcp.local", - port, - NULL, - (const char **)txt); // TTL should be 75 minutes, i.e. 4500 seconds - - mdns_service_destroy(svc); - - return 0; + return 0; } static void mdns_tinysvcmdns_unregister(void) { - if (svr) - { - mdnsd_stop(svr); - svr = NULL; - } + if (svr) { + mdnsd_stop(svr); + svr = NULL; + } } -mdns_backend mdns_tinysvcmdns = { - .name = "tinysvcmdns", - .mdns_register = mdns_tinysvcmdns_register, - .mdns_unregister = mdns_tinysvcmdns_unregister -}; - +mdns_backend mdns_tinysvcmdns = {.name = "tinysvcmdns", + .mdns_register = mdns_tinysvcmdns_register, + .mdns_unregister = mdns_tinysvcmdns_unregister}; diff --git a/player.c b/player.c index 8ddc32d4..425bc879 100644 --- a/player.c +++ b/player.c @@ -73,16 +73,17 @@ static AES_KEY aes; #endif static int sampling_rate, frame_size; -#define FRAME_BYTES(frame_size) (4*frame_size) +#define FRAME_BYTES(frame_size) (4 * frame_size) // maximal resampling shift - conservative -#define OUTFRAME_BYTES(frame_size) (4*(frame_size+3)) +#define OUTFRAME_BYTES(frame_size) (4 * (frame_size + 3)) -#ifdef HAVE_LIBPOLARSSL +#ifdef HAVE_LIBPOLARSSL static aes_context dctx; #endif static pthread_t player_thread; static int please_stop; +static int encrypted; // Normally the audio is encrypted, but it may not be static int connection_state_to_output; // if true, then play incoming stuff; if false drop everything @@ -93,8 +94,6 @@ static int late_packet_message_sent; static uint64_t packet_count = 0; static int32_t last_seqno_read; - - // interthread variables static double software_mixer_volume = 1.0; static int fix_volume = 0x10000; @@ -102,14 +101,13 @@ static pthread_mutex_t vol_mutex = PTHREAD_MUTEX_INITIALIZER; // default buffer size // needs to be a power of 2 because of the way BUFIDX(seqno) works -#define BUFFER_FRAMES 512 -#define MAX_PACKET 2048 +#define BUFFER_FRAMES 512 +#define MAX_PACKET 2048 // DAC buffer occupancy stuff -#define DAC_BUFFER_QUEUE_DESIRED_LENGTH 6615 #define DAC_BUFFER_QUEUE_MINIMUM_LENGTH 5000 -typedef struct audio_buffer_entry { // decoded audio packets +typedef struct audio_buffer_entry { // decoded audio packets int ready; uint32_t timestamp; seq_t sequence_number; @@ -132,16 +130,16 @@ static pthread_mutex_t ab_mutex = PTHREAD_MUTEX_INITIALIZER; static pthread_mutex_t flush_mutex = PTHREAD_MUTEX_INITIALIZER; static pthread_cond_t flowcontrol; -static int64_t first_packet_time_to_play; // nanoseconds +static int64_t first_packet_time_to_play, time_since_play_started; // nanoseconds static audio_parameters audio_information; // stats -static uint64_t missing_packets,late_packets,too_late_packets,resend_requests; +static uint64_t missing_packets, late_packets, too_late_packets, resend_requests; static void ab_resync(void) { int i; - for (i=0; i 0; + int32_t d = ORDINATE(b) - ORDINATE(a); + return d > 0; } static inline seq_t seq_sum(seq_t a, seq_t b) { uint32_t p = a & 0xffff; uint32_t q = b & 0x0ffff; - uint32_t r = (a+b) & 0xffff; + uint32_t r = (a + b) & 0xffff; return r; } - // now for 32-bit wrapping in timestamps // this returns true if the second arg is strictly after the first // on the assumption that the gap between them is never greater than (2^31)-1 // Represent a and b in 64 bits static inline int seq32_order(uint32_t a, uint32_t b) { - if (a==b) - return 0; - int64_t A = a & 0xffffffff; - int64_t B = b & 0xffffffff; - int64_t C = B-A; - // if bit 31 is set, it means either b is before (i.e. less than) a or - // b is (2^31)-1 ahead of a. - - // If we assume the gap between b and a should never reach 2 billion, then - // bit 31 == 0 means b is strictly after a + if (a == b) + return 0; + int64_t A = a & 0xffffffff; + int64_t B = b & 0xffffffff; + int64_t C = B - A; + // if bit 31 is set, it means either b is before (i.e. less than) a or + // b is (2^31)-1 ahead of a. + + // If we assume the gap between b and a should never reach 2 billion, then + // bit 31 == 0 means b is strictly after a return (C & 0x80000000) == 0; } - - static void alac_decode(short *dest, uint8_t *buf, int len) { unsigned char packet[MAX_PACKET]; unsigned char packetp[MAX_PACKET]; - assert(len<=MAX_PACKET); - - unsigned char iv[16]; - int aeslen = len & ~0xf; - memcpy(iv, aesiv, sizeof(iv)); - -#ifdef HAVE_LIBPOLARSSL - aes_crypt_cbc(&dctx,AES_DECRYPT,aeslen,iv,buf, packet); -#endif -#ifdef HAVE_LIBSSL - AES_cbc_encrypt(buf, packet, aeslen, &aes, iv, AES_DECRYPT); -#endif - - memcpy(packet+aeslen, buf+aeslen, len-aeslen); - + assert(len <= MAX_PACKET); int outsize; - alac_decode_frame(decoder_info, packet, dest, &outsize); + if (encrypted) { + unsigned char iv[16]; + int aeslen = len & ~0xf; + memcpy(iv, aesiv, sizeof(iv)); +#ifdef HAVE_LIBPOLARSSL + aes_crypt_cbc(&dctx, AES_DECRYPT, aeslen, iv, buf, packet); +#endif +#ifdef HAVE_LIBSSL + AES_cbc_encrypt(buf, packet, aeslen, &aes, iv, AES_DECRYPT); +#endif + memcpy(packet + aeslen, buf + aeslen, len - aeslen); + alac_decode_frame(decoder_info, packet, dest, &outsize); + } else { + alac_decode_frame(decoder_info, buf, dest, &outsize); + } assert(outsize == FRAME_BYTES(frame_size)); } @@ -268,112 +263,114 @@ static int init_decoder(int32_t fmtp[12]) { decoder_info = alac; alac->setinfo_max_samples_per_frame = frame_size; - alac->setinfo_7a = fmtp[2]; + alac->setinfo_7a = fmtp[2]; alac->setinfo_sample_size = sample_size; alac->setinfo_rice_historymult = fmtp[4]; alac->setinfo_rice_initialhistory = fmtp[5]; alac->setinfo_rice_kmodifier = fmtp[6]; - alac->setinfo_7f = fmtp[7]; - alac->setinfo_80 = fmtp[8]; - alac->setinfo_82 = fmtp[9]; - alac->setinfo_86 = fmtp[10]; + alac->setinfo_7f = fmtp[7]; + alac->setinfo_80 = fmtp[8]; + alac->setinfo_82 = fmtp[9]; + alac->setinfo_86 = fmtp[10]; alac->setinfo_8a_rate = fmtp[11]; alac_allocate_buffers(alac); return 0; } -static void free_decoder(void) { - alac_free(decoder_info); -} +static void free_decoder(void) { alac_free(decoder_info); } static void init_buffer(void) { int i; - for (i=0; i(BUFFER_FRAMES*7)/8) - // debug(1,"An interval of %u frames has opened, with ab_read: %u, ab_write: %u and seqno: %u.",seq_diff(ab_read,seqno),ab_read,ab_write,seqno); - int32_t gap = seq_diff(ab_write,PREDECESSOR(seqno))+1; - if (gap<=0) - debug(1,"Unexpected gap size: %d.",gap); - int i; - for (i=0;iready = 0; // to be sure, to be sure - abuf->timestamp = 0; - abuf->sequence_number = 0; - } - // debug(1,"N %d s %u.",seq_diff(ab_write,PREDECESSOR(seqno))+1,ab_write); - abuf = audio_buffer + BUFIDX(seqno); - rtp_request_resend(ab_write,gap); - resend_requests++; - ab_write = SUCCESSOR(seqno); - } else if (seq_order(ab_read, seqno)) { // late but not yet played - late_packets++; - abuf = audio_buffer + BUFIDX(seqno); - } else { // too late. - too_late_packets++; - /* - if (!late_packet_message_sent) { - debug(1, "too-late packet received: %u; ab_read: %u; ab_write: %u.", seqno, ab_read, ab_write); - late_packet_message_sent=1; - } - */ - } - // pthread_mutex_unlock(&ab_mutex); + abuf_t *abuf = 0; - if (abuf) { - alac_decode(abuf->data, data, len); - abuf->ready = 1; - abuf->timestamp = timestamp; - abuf->sequence_number = seqno; - } - - // pthread_mutex_lock(&ab_mutex); - + if (!ab_synced) { + debug(2, "syncing to seqno %u.", seqno); + ab_write = seqno; + ab_read = seqno; + ab_synced = 1; + } + if (ab_write == seqno) { // expected packet + abuf = audio_buffer + BUFIDX(seqno); + ab_write = SUCCESSOR(seqno); + } else if (seq_order(ab_write, seqno)) { // newer than expected + // if (ORDINATE(seqno)>(BUFFER_FRAMES*7)/8) + // debug(1,"An interval of %u frames has opened, with ab_read: %u, ab_write: %u and seqno: + // %u.",seq_diff(ab_read,seqno),ab_read,ab_write,seqno); + int32_t gap = seq_diff(ab_write, PREDECESSOR(seqno)) + 1; + if (gap <= 0) + debug(1, "Unexpected gap size: %d.", gap); + int i; + for (i = 0; i < gap; i++) { + abuf = audio_buffer + BUFIDX(seq_sum(ab_write, i)); + abuf->ready = 0; // to be sure, to be sure + abuf->timestamp = 0; + abuf->sequence_number = 0; + } + // debug(1,"N %d s %u.",seq_diff(ab_write,PREDECESSOR(seqno))+1,ab_write); + abuf = audio_buffer + BUFIDX(seqno); + rtp_request_resend(ab_write, gap); + resend_requests++; + ab_write = SUCCESSOR(seqno); + } else if (seq_order(ab_read, seqno)) { // late but not yet played + late_packets++; + abuf = audio_buffer + BUFIDX(seqno); + } else { // too late. + too_late_packets++; + /* + if (!late_packet_message_sent) { + debug(1, "too-late packet received: %u; ab_read: %u; ab_write: %u.", seqno, ab_read, + ab_write); + late_packet_message_sent=1; + } + */ + } + // pthread_mutex_unlock(&ab_mutex); + + if (abuf) { + alac_decode(abuf->data, data, len); + abuf->ready = 1; + abuf->timestamp = timestamp; + abuf->sequence_number = seqno; + } + + // pthread_mutex_lock(&ab_mutex); } int rc = pthread_cond_signal(&flowcontrol); if (rc) - debug(1,"Error signalling flowcontrol."); - } + debug(1, "Error signalling flowcontrol."); + } pthread_mutex_unlock(&ab_mutex); } - static inline short lcg_rand(void) { static unsigned long lcg_prev = 12345; lcg_prev = lcg_prev * 69069 + 3; @@ -391,7 +388,7 @@ static inline short dithered_vol(short sample) { out += rand_a; out -= rand_b; } - return out>>16; + return out >> 16; } // get the next frame, when available. return 0 if underrun/stream reset. @@ -402,205 +399,294 @@ static abuf_t *buffer_get_frame(void) { abuf_t *abuf = 0; int i; abuf_t *curframe; - + pthread_mutex_lock(&ab_mutex); int wait; int32_t dac_delay = 0; do { // get the time - local_time_now = get_absolute_time_in_fp(); + local_time_now = get_absolute_time_in_fp(); - // if config.timeout (default 120) seconds have elapsed since the last audio packet was received, then we should stop. - // config.timeout of zero means don't check..., but iTunes may be confused by a long gap followed by a resumption... - if ((time_of_last_audio_packet!=0) && (shutdown_requested==0) && (config.timeout!=0)) { - uint64_t ct = config.timeout; // go from int to 64-bit int - if (local_time_now-time_of_last_audio_packet>=ct<<32) { - debug(1,"As Yeats almost said, \"Too long a silence / can make a stone of the heart\""); + // if config.timeout (default 120) seconds have elapsed since the last audio packet was + // received, then we should stop. + // config.timeout of zero means don't check..., but iTunes may be confused by a long gap + // followed by a resumption... + + if ((time_of_last_audio_packet != 0) && (shutdown_requested == 0) && + (config.dont_check_timeout == 0)) { + uint64_t ct = config.timeout; // go from int to 64-bit int + if ((local_time_now > time_of_last_audio_packet) && + (local_time_now - time_of_last_audio_packet >= ct << 32)) { + debug(1, "As Yeats almost said, \"Too long a silence / can make a stone of the heart\""); rtsp_request_shutdown_stream(); - shutdown_requested=1; + shutdown_requested = 1; } } int rco = get_requested_connection_state_to_output(); - + if (connection_state_to_output != rco) { - connection_state_to_output=rco; + connection_state_to_output = rco; // change happening - if (connection_state_to_output==0) { //going off - pthread_mutex_lock(&flush_mutex); - flush_requested=1; - pthread_mutex_unlock(&flush_mutex); + if (connection_state_to_output == 0) { // going off + pthread_mutex_lock(&flush_mutex); + flush_requested = 1; + pthread_mutex_unlock(&flush_mutex); } } - + pthread_mutex_lock(&flush_mutex); - if (flush_requested==1) { + if (flush_requested == 1) { if (config.output->flush) config.output->flush(); ab_resync(); first_packet_timestamp = 0; first_packet_time_to_play = 0; - flush_requested=0; + time_since_play_started = 0; + flush_requested = 0; } pthread_mutex_unlock(&flush_mutex); uint32_t flush_limit = 0; if (ab_synced) { - do { - curframe = audio_buffer + BUFIDX(ab_read); - if (curframe->ready) { - - if (curframe->sequence_number!=ab_read) { - // some kind of sync problem has occurred. - if (BUFIDX(curframe->sequence_number)==BUFIDX(ab_read)) { - // it looks like some kind of aliasing has happened - if (seq_order(ab_read,curframe->sequence_number)) { - ab_read=curframe->sequence_number; - debug(1,"Aliasing of buffer index -- reset."); - } - } else { - debug(1,"Inconsistent sequence numbers detected"); - } - } - - if ((flush_rtp_timestamp!=0) && ((curframe->timestamp==flush_rtp_timestamp) || seq32_order(curframe->timestamp,flush_rtp_timestamp))) { - debug(1,"Dropping flushed packet seqno %u, timestamp %u",curframe->sequence_number,curframe->timestamp); - curframe->ready=0; - flush_limit++; - ab_read=SUCCESSOR(ab_read); - } - if ((flush_rtp_timestamp!=0) && (!seq32_order(curframe->timestamp,flush_rtp_timestamp))) // if we have gone past the flush boundary time - flush_rtp_timestamp=0; - } - } while ((flush_rtp_timestamp!=0) && (flush_limit<=8820) && (curframe->ready==0)); - - if (flush_limit==8820) { - debug(1,"Flush hit the 8820 frame limit!"); - flush_limit=0; - } - - dac_delay = 0; - curframe = audio_buffer + BUFIDX(ab_read); - if (config.output->delay) { - dac_delay = config.output->delay(); - if (dac_delay==-1) { - debug(1,"Error getting dac_delay at start of loop."); - dac_delay=0; + do { + curframe = audio_buffer + BUFIDX(ab_read); + if (curframe->ready) { + + if (curframe->sequence_number != ab_read) { + // some kind of sync problem has occurred. + if (BUFIDX(curframe->sequence_number) == BUFIDX(ab_read)) { + // it looks like some kind of aliasing has happened + if (seq_order(ab_read, curframe->sequence_number)) { + ab_read = curframe->sequence_number; + debug(1, "Aliasing of buffer index -- reset."); + } + } else { + debug(1, "Inconsistent sequence numbers detected"); + } + } + + if ((flush_rtp_timestamp != 0) && + ((curframe->timestamp == flush_rtp_timestamp) || + seq32_order(curframe->timestamp, flush_rtp_timestamp))) { + debug(1, "Dropping flushed packet seqno %u, timestamp %u", curframe->sequence_number, + curframe->timestamp); + curframe->ready = 0; + flush_limit++; + ab_read = SUCCESSOR(ab_read); + } + if ((flush_rtp_timestamp != 0) && + (!seq32_order(curframe->timestamp, + flush_rtp_timestamp))) // if we have gone past the flush boundary time + flush_rtp_timestamp = 0; } + } while ((flush_rtp_timestamp != 0) && (flush_limit <= 8820) && (curframe->ready == 0)); + + if (flush_limit == 8820) { + debug(1, "Flush hit the 8820 frame limit!"); + flush_limit = 0; } + curframe = audio_buffer + BUFIDX(ab_read); + if (curframe->ready) { -/* - // This is broken -- it causes infinite loops. It's in the wrong place, and anyway isn't used much. - if ((flush_rtp_timestamp) && (flush_rtp_timestamp>=curframe->timestamp)) { - debug(1,"Dropping flushed packet seqno %u, timestamp %u",curframe->sequence_number,curframe->timestamp); - curframe->ready=0; - if (flush_rtp_timestamp==curframe->timestamp) // if we are finished... - flush_rtp_timestamp=0; - ab_read==SUCCESSOR(ab_read); - } else -*/ - if (ab_buffering) { // if we are getting packets but not yet forwarding them to the player - if (first_packet_timestamp==0) { // if this is the very first packet - // debug(1,"First frame seen, time %u, with %d frames...",curframe->timestamp,seq_diff(ab_read, ab_write)); - uint32_t reference_timestamp; - uint64_t reference_timestamp_time; - get_reference_timestamp_stuff(&reference_timestamp,&reference_timestamp_time); + if (ab_buffering) { // if we are getting packets but not yet forwarding them to the player + if (first_packet_timestamp == 0) { // if this is the very first packet + // debug(1,"First frame seen, time %u, with %d + // frames...",curframe->timestamp,seq_diff(ab_read, ab_write)); + uint32_t reference_timestamp; + uint64_t reference_timestamp_time,remote_reference_timestamp_time; + get_reference_timestamp_stuff(&reference_timestamp, &reference_timestamp_time, &remote_reference_timestamp_time); if (reference_timestamp) { // if we have a reference time // debug(1,"First frame seen with timestamp..."); - first_packet_timestamp=curframe->timestamp; // we will keep buffering until we are supposed to start playing this - - // here, see if we should start playing. We need to know when to allow the packets to be sent to the player - // we will get a fix every second or so, which will be stored as a pair consisting of - // the time when the packet with a particular timestamp should be played. - // it might not be the timestamp of our first packet, however, so we might have to do some calculations. - - int64_t delta = ((int64_t)first_packet_timestamp-(int64_t)reference_timestamp); + first_packet_timestamp = curframe->timestamp; // we will keep buffering until we are + // supposed to start playing this - first_packet_time_to_play = reference_timestamp_time+((delta+(int64_t)config.latency)<<32)/44100; // using the latency requested... - if (local_time_now>=first_packet_time_to_play) { - debug(2,"First packet is late! It should have played before now. Flushing 0.1 seconds"); - player_flush(first_packet_timestamp+4410); + // Here, calculate when we should start playing. We need to know when to allow the + // packets to be sent to the player. + // We will send packets of silence from now until that time and then we will send the + // first packet, which will be followed by the subsequent packets. + + // we will get a fix every second or so, which will be stored as a pair consisting of + // the time when the packet with a particular timestamp should be played, neglecting + // latencies, etc. + + // It probably won't be the timestamp of our first packet, however, so we might have + // to do some calculations. + + // To calculate when the first packet will be played, we figure out the exact time the + // packet should be played according to its timestamp and the reference time. + // We then need to add the desired latency, typically 88200 frames. + + // Then we need to offset this by the backend latency offset. For example, if we knew + // that the audio back end has a latency of 100 ms, we would + // ask for the first packet to be emitted 100 ms earlier than it should, i.e. -4410 + // frames, so that when it got through the audio back end, + // if would be in sync. To do this, we would give it a latency offset of -100 ms, i.e. + // -4410 frames. + + int64_t delta = ((int64_t)first_packet_timestamp - (int64_t)reference_timestamp); + + first_packet_time_to_play = + reference_timestamp_time + + ((delta + (int64_t)config.latency + (int64_t)config.audio_backend_latency_offset) + << 32) / + 44100; + + if (local_time_now >= first_packet_time_to_play) { + debug( + 1, + "First packet is late! It should have played before now. Flushing 0.1 seconds"); + player_flush(first_packet_timestamp + 4410); } } - } + } - if (first_packet_time_to_play!=0) { + if (first_packet_time_to_play != 0) { uint32_t filler_size = frame_size; - uint32_t max_dac_delay = DAC_BUFFER_QUEUE_DESIRED_LENGTH; - // if (dac_delay==0) // i.e. if this is the first fill - filler_size = 4410; // 0.1 second -- the maximum we'll add to the DAC + uint32_t max_dac_delay = 4410; + filler_size = 4410; // 0.1 second -- the maximum we'll add to the DAC - if (local_time_now>=first_packet_time_to_play) { + if (local_time_now >= first_packet_time_to_play) { // we've gone past the time... - // debug(1,"Run past the exact start time by %llu frames, with time now of %llx, fpttp of %llx and dac_delay of %d and %d packets; flush.",(((tn-first_packet_time_to_play)*44100)>>32)+dac_delay,tn,first_packet_time_to_play,dac_delay,seq_diff(ab_read, ab_write)); - + // debug(1,"Run past the exact start time by %llu frames, with time now of %llx, fpttp + // of %llx and dac_delay of %d and %d packets; + // flush.",(((tn-first_packet_time_to_play)*44100)>>32)+dac_delay,tn,first_packet_time_to_play,dac_delay,seq_diff(ab_read, + // ab_write)); + if (config.output->flush) config.output->flush(); ab_resync(); first_packet_timestamp = 0; first_packet_time_to_play = 0; + time_since_play_started = 0; } else { - uint64_t gross_frame_gap = ((first_packet_time_to_play-local_time_now)*44100)>>32; - int64_t exact_frame_gap = gross_frame_gap-dac_delay; - if (exact_frame_gap<=0) { + if (config.output->delay) { + dac_delay = config.output->delay(); + if (dac_delay == -1) { + debug(1, "Error getting dac_delay in buffer_get_frame."); + dac_delay = 0; + } + } else + dac_delay = 0; + uint64_t gross_frame_gap = + ((first_packet_time_to_play - local_time_now) * 44100) >> 32; + int64_t exact_frame_gap = gross_frame_gap - dac_delay; + if (exact_frame_gap <= 0) { // we've gone past the time... - // debug(1,"Run a bit past the exact start time by %lld frames, with time now of %llx, fpttp of %llx and dac_delay of %d and %d packets; flush.",-exact_frame_gap,tn,first_packet_time_to_play,dac_delay,seq_diff(ab_read, ab_write)); + // debug(1,"Run a bit past the exact start time by %lld frames, with time now of + // %llx, fpttp of %llx and dac_delay of %d and %d packets; + // flush.",-exact_frame_gap,tn,first_packet_time_to_play,dac_delay,seq_diff(ab_read, + // ab_write)); if (config.output->flush) config.output->flush(); ab_resync(); first_packet_timestamp = 0; first_packet_time_to_play = 0; } else { - uint32_t fs=filler_size; - if (fs>(max_dac_delay-dac_delay)) - fs=max_dac_delay-dac_delay; - if ((exact_frame_gap<=fs) || (exact_frame_gap<=frame_size*2)) { - fs=exact_frame_gap; - // debug(1,"Exact frame gap is %llu; play %d frames of silence. Dac_delay is %d, with %d packets, ab_read is %04x, ab_write is %04x.",exact_frame_gap,fs,dac_delay,seq_diff(ab_read, ab_write),ab_read,ab_write); + uint32_t fs = filler_size; + if (fs > (max_dac_delay - dac_delay)) + fs = max_dac_delay - dac_delay; + if ((exact_frame_gap <= fs) || (exact_frame_gap <= frame_size * 2)) { + fs = exact_frame_gap; + // debug(1,"Exact frame gap is %llu; play %d frames of silence. Dac_delay is %d, + // with %d packets, ab_read is %04x, ab_write is + // %04x.",exact_frame_gap,fs,dac_delay,seq_diff(ab_read, + // ab_write),ab_read,ab_write); ab_buffering = 0; } signed short *silence; silence = malloc(FRAME_BYTES(fs)); memset(silence, 0, FRAME_BYTES(fs)); - // debug(1,"Exact frame gap is %llu; play %d frames of silence. Dac_delay is %d, with %d packets.",exact_frame_gap,fs,dac_delay,seq_diff(ab_read, ab_write)); + // debug(1,"Exact frame gap is %llu; play %d frames of silence. Dac_delay is %d, + // with %d packets.",exact_frame_gap,fs,dac_delay,seq_diff(ab_read, ab_write)); config.output->play(silence, fs); free(silence); + if (ab_buffering == 0) { + uint64_t reference_timestamp_time; // don't need this... + get_reference_timestamp_stuff(&play_segment_reference_frame, &reference_timestamp_time, &play_segment_reference_frame_remote_time); +#ifdef CONFIG_METADATA + send_ssnc_metadata('prsm', NULL, 0, 0); // "resume", but don't wait if the queue is locked +#endif + } } } } } } } - wait = (ab_buffering || (dac_delay>=DAC_BUFFER_QUEUE_DESIRED_LENGTH) || (!ab_synced)) && (!please_stop); -// wait = (ab_buffering || (seq_diff(ab_read, ab_write) < (config.latency-22000)/(352)) || (!ab_synced)) && (!please_stop); + + // Here, we work out whether to release a packet or wait + // We release a buffer when the time is right. + + // To work out when the time is right, we need to take account of (1) the actual time the packet + // should be released, + // (2) the latency requested, (3) the audio backend latency offset and (4) the desired length of + // the audio backend's buffer + + // The time is right if the current time is later or the same as + // The packet time + (latency + latency offset - backend_buffer_length). + // Note: the last three items are expressed in frames and must be converted to time. + + int do_wait = 1; + if ((ab_synced) && (curframe) && (curframe->ready) && (curframe->timestamp)) { + uint32_t reference_timestamp; + uint64_t reference_timestamp_time,remote_reference_timestamp_time; + get_reference_timestamp_stuff(&reference_timestamp, &reference_timestamp_time, &remote_reference_timestamp_time); + if (reference_timestamp) { // if we have a reference time + uint32_t packet_timestamp = curframe->timestamp; + int64_t delta = ((int64_t)packet_timestamp - (int64_t)reference_timestamp); + int64_t offset = (int64_t)config.latency + config.audio_backend_latency_offset - + (int64_t)config.audio_backend_buffer_desired_length; + int64_t net_offset = delta + offset; + int64_t time_to_play = reference_timestamp_time; + int64_t net_offset_fp_sec; + if (net_offset >= 0) { + net_offset_fp_sec = (net_offset << 32) / 44100; + time_to_play += net_offset_fp_sec; // using the latency requested... + // debug(2,"Net Offset: %lld, adjusted: %lld.",net_offset,net_offset_fp_sec); + } else { + net_offset_fp_sec = ((-net_offset) << 32) / 44100; + time_to_play -= net_offset_fp_sec; + // debug(2,"Net Offset: %lld, adjusted: -%lld.",net_offset,net_offset_fp_sec); + } + + if (local_time_now >= time_to_play) { + do_wait = 0; + } + } + } + wait = (ab_buffering || (do_wait != 0) || (!ab_synced)) && (!please_stop); + if (wait) { - uint64_t time_to_wait_for_wakeup_fp = ((uint64_t)1<<32)/44100; // this is time period of one frame - time_to_wait_for_wakeup_fp *= 4*352; // four full 352-frame packets - time_to_wait_for_wakeup_fp /= 3; //four thirds of a packet time - -#ifdef COMPILE_FOR_LINUX - uint64_t time_of_wakeup_fp = local_time_now+time_to_wait_for_wakeup_fp; - uint64_t sec = time_of_wakeup_fp>>32; - uint64_t nsec = ((time_of_wakeup_fp&0xffffffff)*1000000000)>>32; - + uint64_t time_to_wait_for_wakeup_fp = + ((uint64_t)1 << 32) / 44100; // this is time period of one frame + time_to_wait_for_wakeup_fp *= 4 * 352; // four full 352-frame packets + time_to_wait_for_wakeup_fp /= 3; // four thirds of a packet time + +#ifdef COMPILE_FOR_LINUX_AND_FREEBSD + uint64_t time_of_wakeup_fp = local_time_now + time_to_wait_for_wakeup_fp; + uint64_t sec = time_of_wakeup_fp >> 32; + uint64_t nsec = ((time_of_wakeup_fp & 0xffffffff) * 1000000000) >> 32; + struct timespec time_of_wakeup; time_of_wakeup.tv_sec = sec; time_of_wakeup.tv_nsec = nsec; - - pthread_cond_timedwait(&flowcontrol,&ab_mutex,&time_of_wakeup); - // int rc = pthread_cond_timedwait(&flowcontrol,&ab_mutex,&time_of_wakeup); - // if (rc!=0) - // debug(1,"pthread_cond_timedwait returned error code %d.",rc); + + pthread_cond_timedwait(&flowcontrol, &ab_mutex, &time_of_wakeup); +// int rc = pthread_cond_timedwait(&flowcontrol,&ab_mutex,&time_of_wakeup); +// if (rc!=0) +// debug(1,"pthread_cond_timedwait returned error code %d.",rc); #endif #ifdef COMPILE_FOR_OSX - uint64_t sec = time_to_wait_for_wakeup_fp>>32;; - uint64_t nsec = ((time_to_wait_for_wakeup_fp&0xffffffff)*1000000000)>>32; + uint64_t sec = time_to_wait_for_wakeup_fp >> 32; + ; + uint64_t nsec = ((time_to_wait_for_wakeup_fp & 0xffffffff) * 1000000000) >> 32; struct timespec time_to_wait; time_to_wait.tv_sec = sec; time_to_wait.tv_nsec = nsec; - pthread_cond_timedwait_relative_np(&flowcontrol,&ab_mutex,&time_to_wait); + pthread_cond_timedwait_relative_np(&flowcontrol, &ab_mutex, &time_to_wait); #endif - } + } } while (wait); if (please_stop) { @@ -612,11 +698,10 @@ static abuf_t *buffer_get_frame(void) { // check if t+8, t+16, t+32, t+64, t+128, ... (buffer_start_fill / 2) // packets have arrived... last-chance resend - - + if (!ab_buffering) { - for (i = 8; i < (seq_diff(ab_read,ab_write) / 2); i = (i * 2)) { - seq_t next = seq_sum(ab_read,i); + for (i = 8; i < (seq_diff(ab_read, ab_write) / 2); i = (i * 2)) { + seq_t next = seq_sum(ab_read, i); abuf = audio_buffer + BUFIDX(next); if (!abuf->ready) { rtp_request_resend(next, 1); @@ -625,15 +710,15 @@ static abuf_t *buffer_get_frame(void) { } } } - + if (!curframe->ready) { // debug(1, " %d. Supplying a silent frame.", read); missing_packets++; memset(curframe->data, 0, FRAME_BYTES(frame_size)); - curframe->timestamp=0; + curframe->timestamp = 0; } curframe->ready = 0; - ab_read=SUCCESSOR(ab_read); + ab_read = SUCCESSOR(ab_read); pthread_mutex_unlock(&ab_mutex); return curframe; } @@ -641,65 +726,66 @@ static abuf_t *buffer_get_frame(void) { static inline short shortmean(short a, short b) { long al = (long)a; long bl = (long)b; - long longmean = (al+bl)/2; + long longmean = (al + bl) / 2; short r = (short)longmean; - if (r!=longmean) - debug(1,"Error calculating average of two shorts"); + if (r != longmean) + debug(1, "Error calculating average of two shorts"); return r; } // stuff: 1 means add 1; 0 means do nothing; -1 means remove 1 static int stuff_buffer_basic(short *inptr, short *outptr, int stuff) { - if ((stuff>1) || (stuff<-1)) { - debug(1,"Stuff argument to stuff_buffer must be from -1 to +1."); - return frame_size; - } - int i; - int stuffsamp = frame_size; - if (stuff) -// stuffsamp = rand() % (frame_size - 1); - stuffsamp = (rand() % (frame_size-2))+1; // ensure there's always a sample before and after the item + if ((stuff > 1) || (stuff < -1)) { + debug(1, "Stuff argument to stuff_buffer must be from -1 to +1."); + return frame_size; + } + int i; + int stuffsamp = frame_size; + if (stuff) + // stuffsamp = rand() % (frame_size - 1); + stuffsamp = + (rand() % (frame_size - 2)) + 1; // ensure there's always a sample before and after the item - pthread_mutex_lock(&vol_mutex); - for (i=0; i> 1); - //*outptr++ = dithered_vol(((long)inptr[-1] + (long)inptr[1]) >> 1); - *outptr++ = dithered_vol(shortmean(inptr[-2],inptr[0])); - *outptr++ = dithered_vol(shortmean(inptr[-1],inptr[1])); - } else if (stuff==-1) { - debug(3, "---------"); - inptr++; - inptr++; - } - for (i=stuffsamp; i> 1); + //*outptr++ = dithered_vol(((long)inptr[-1] + (long)inptr[1]) >> 1); + *outptr++ = dithered_vol(shortmean(inptr[-2], inptr[0])); + *outptr++ = dithered_vol(shortmean(inptr[-1], inptr[1])); + } else if (stuff == -1) { + debug(3, "---------"); + inptr++; + inptr++; } - pthread_mutex_unlock(&vol_mutex); + for (i = stuffsamp; i < frame_size + stuff; i++) { + *outptr++ = dithered_vol(*inptr++); + *outptr++ = dithered_vol(*inptr++); + } + } + pthread_mutex_unlock(&vol_mutex); - return frame_size + stuff; + return frame_size + stuff; } #ifdef HAVE_LIBSOXR // stuff: 1 means add 1; 0 means do nothing; -1 means remove 1 static int stuff_buffer_soxr(short *inptr, short *outptr, int stuff) { - if ((stuff>1) || (stuff<-1)) { - debug(1,"Stuff argument to sox_stuff_buffer must be from -1 to +1."); + if ((stuff > 1) || (stuff < -1)) { + debug(1, "Stuff argument to sox_stuff_buffer must be from -1 to +1."); return frame_size; } int i; - short *ip,*op; - ip=inptr; - op=outptr; - + short *ip, *op; + ip = inptr; + op = outptr; + if (stuff) { // debug(1,"Stuff %d.",stuff); soxr_io_spec_t io_spec; @@ -712,10 +798,10 @@ static int stuff_buffer_soxr(short *inptr, short *outptr, int stuff) { size_t odone; soxr_error_t error = soxr_oneshot(frame_size, frame_size + stuff, 2, /* Rates and # of chans. */ - inptr, frame_size, NULL, /* Input. */ - outptr, frame_size + stuff, &odone, /* Output. */ - &io_spec, /* Input, output and transfer spec. */ - NULL, NULL); /* Default configuration.*/ + inptr, frame_size, NULL, /* Input. */ + outptr, frame_size + stuff, &odone, /* Output. */ + &io_spec, /* Input, output and transfer spec. */ + NULL, NULL); /* Default configuration.*/ if (error) die("soxr error: %s\n", "error: %s\n", soxr_strerror(error)); @@ -724,26 +810,26 @@ static int stuff_buffer_soxr(short *inptr, short *outptr, int stuff) { die("odone = %d!\n", odone); const int gpm = 5; - + // keep the first (dpm) samples, to mitigate the Gibbs phenomenon - for (i=0;i BUFFER_FRAMES) + die("Not enough buffers available for a total latency of %d frames. A maximum of %d 352-frame packets may be accommodated.",maximum_latency,BUFFER_FRAMES); connection_state_to_output = get_requested_connection_state_to_output(); -//this is about half a minute +// this is about half a minute #define trend_interval 3758 stats_t statistics[trend_interval]; - int number_of_statistics,oldest_statistic,newest_statistic; - int64_t tsum_of_sync_errors,tsum_of_corrections,tsum_of_insertions_and_deletions,tsum_of_drifts; - int64_t previous_sync_error,previous_correction; + int number_of_statistics, oldest_statistic, newest_statistic; + int at_least_one_frame_seen = 0; + int64_t tsum_of_sync_errors, tsum_of_corrections, tsum_of_insertions_and_deletions, + tsum_of_drifts; + int64_t previous_sync_error, previous_correction; int64_t minimum_dac_queue_size = 1000000; int32_t minimum_buffer_occupancy = BUFFER_FRAMES; int32_t maximum_buffer_occupancy = 0; - audio_information.valid=0; - + + audio_information.valid = 0; + buffer_occupancy = 0; + int play_samples; int64_t current_delay; int play_number = 0; @@ -791,20 +888,23 @@ static void *player_thread_func(void *arg) { tsum_of_sync_errors = tsum_of_corrections = tsum_of_insertions_and_deletions = tsum_of_drifts = 0; const int print_interval = trend_interval; // don't ask... - // I think it's useful to keep this prime to prevent it from falling into a pattern with some other process. - - char rnstate[256]; - initstate(time(NULL),rnstate,256); - + // I think it's useful to keep this prime to prevent it from falling into a pattern with some + // other process. + + char rnstate[256]; + initstate(time(NULL), rnstate, 256); + signed short *inbuf, *outbuf, *silence; outbuf = malloc(OUTFRAME_BYTES(frame_size)); silence = malloc(OUTFRAME_BYTES(frame_size)); memset(silence, 0, OUTFRAME_BYTES(frame_size)); - - late_packet_message_sent=0; - missing_packets=late_packets=too_late_packets=resend_requests=0; - flush_rtp_timestamp=0; // it seems this number has a special significance -- it seems to be used as a null operand, so we'll use it like that too + late_packet_message_sent = 0; + missing_packets = late_packets = too_late_packets = resend_requests = 0; + flush_rtp_timestamp = 0; // it seems this number has a special significance -- it seems to be used + // as a null operand, so we'll use it like that too int sync_error_out_of_bounds = 0; // number of times in a row that there's been a serious sync error + + uint64_t tens_of_seconds = 0; while (!please_stop) { abuf_t *inframe = buffer_get_frame(); if (inframe) { @@ -812,206 +912,270 @@ static void *player_thread_func(void *arg) { if (inbuf) { play_number++; // if it's a supplied silent frame, let us know... - if (inframe->timestamp==0) { + if (inframe->timestamp == 0) { // debug(1,"Player has a supplied silent frame."); - last_seqno_read = (SUCCESSOR(last_seqno_read)&0xffff); //manage the packet out of sequence minder + last_seqno_read = + (SUCCESSOR(last_seqno_read) & 0xffff); // manage the packet out of sequence minder config.output->play(inbuf, frame_size); } else { - // We have a frame of data. We need to see if we want to add or remove a frame from it to keep in sync. + // We have a frame of data. We need to see if we want to add or remove a frame from it to + // keep in sync. // So we calculate the timing error for the first frame in the DAC. // If it's ahead of time, we add one audio frame to this frame to delay a subsequent frame - // If it's late, we remove an audio frame from this frame to bring a subsequent frame forward in time - - uint32_t reference_timestamp; - uint64_t reference_timestamp_time; - get_reference_timestamp_stuff(&reference_timestamp,&reference_timestamp_time); + // If it's late, we remove an audio frame from this frame to bring a subsequent frame + // forward in time - int64_t rt,nt; + at_least_one_frame_seen = 1; + + uint32_t reference_timestamp; + uint64_t reference_timestamp_time,remote_reference_timestamp_time; + get_reference_timestamp_stuff(&reference_timestamp, &reference_timestamp_time, &remote_reference_timestamp_time); + + int64_t rt, nt; rt = reference_timestamp; nt = inframe->timestamp; - + uint64_t local_time_now = get_absolute_time_in_fp(); - //struct timespec tn; - //clock_gettime(CLOCK_MONOTONIC,&tn); - //uint64_t local_time_now=((uint64_t)tn.tv_sec<<32)+((uint64_t)tn.tv_nsec<<32)/1000000000; + // struct timespec tn; + // clock_gettime(CLOCK_MONOTONIC,&tn); + // uint64_t + // local_time_now=((uint64_t)tn.tv_sec<<32)+((uint64_t)tn.tv_nsec<<32)/1000000000; int64_t td_in_frames; - int64_t td = local_time_now-reference_timestamp_time; - // debug(1,"td is %lld.",td); - if (td>=0) { - td_in_frames = (td*44100)>>32; + int64_t td = local_time_now - reference_timestamp_time; + // debug(1,"td is %lld.",td); + if (td >= 0) { + td_in_frames = (td * 44100) >> 32; } else { - td_in_frames = -((-td*44100)>>32); + td_in_frames = -((-td * 44100) >> 32); } + // This is the timing error for the next audio frame in the DAC, if applicable + int64_t sync_error = 0; + + int amount_to_stuff = 0; + + // check sequencing + if (last_seqno_read == -1) + last_seqno_read = inframe->sequence_number; + else { + last_seqno_read = (SUCCESSOR(last_seqno_read) & 0xffff); + if (inframe->sequence_number != last_seqno_read) { + debug( + 1, + "Player: packets out of sequence: expected: %d, got: %d, sync error: %d frames.", + last_seqno_read, inframe->sequence_number, sync_error); + last_seqno_read = inframe->sequence_number; // reset warning... + } + } + + buffer_occupancy = seq_diff(ab_read, ab_write); + + if (buffer_occupancy < minimum_buffer_occupancy) + minimum_buffer_occupancy = buffer_occupancy; + + if (buffer_occupancy > maximum_buffer_occupancy) + maximum_buffer_occupancy = buffer_occupancy; + if (config.output->delay) { current_delay = config.output->delay(); - if (current_delay==-1) { - debug(1,"Delay error when checking running latency."); - current_delay=0; + if (current_delay == -1) { + debug(1, "Delay error when checking running latency."); + current_delay = 0; } - } else { - current_delay = 0; - } - if (current_delaymaximum_buffer_occupancy) - maximum_buffer_occupancy=bo; + // this is the actual delay, including the latency we actually want, which will + // fluctuate a good bit about a potentially rising or falling trend. + int64_t delay = td_in_frames + rt - (nt - current_delay); - // this is the actual delay, including the latency we actually want, which will fluctuate a good bit about a potentially rising or falling trend. - int64_t delay = td_in_frames+rt-(nt-current_delay); - - // This is the timing error for the next audio frame in the DAC. - int64_t sync_error = delay-config.latency; - - // before we finally commit to this frame, check its sequencing and timing - // check sequencing - if (last_seqno_read==-1) - last_seqno_read=inframe->sequence_number; - else { - last_seqno_read = (SUCCESSOR(last_seqno_read) & 0xffff); - if (inframe->sequence_number!=last_seqno_read) - debug(1,"Player: packets out of sequence: expected: %d, got: %d, sync error: %d frames.",last_seqno_read,inframe->sequence_number,sync_error); - last_seqno_read=inframe->sequence_number; // reset warning... - } - - int amount_to_stuff = 0; - // require a certain error before bothering to fix it... - if (sync_error>config.tolerance) { - amount_to_stuff = -1; - } - if (sync_error<-config.tolerance) { - amount_to_stuff = 1; - } - - // only allow stuffing if there is enough time to do it -- check DAC buffer... - if (current_delay352) - amount_to_stuff=0; - } + // before we finally commit to this frame, check its sequencing and timing + + // require a certain error before bothering to fix it... + if (sync_error > config.tolerance) { + amount_to_stuff = -1; + } + if (sync_error < -config.tolerance) { + amount_to_stuff = 1; + } + + // only allow stuffing if there is enough time to do it -- check DAC buffer... + if (current_delay < DAC_BUFFER_QUEUE_MINIMUM_LENGTH) { + // debug(1,"DAC buffer too short to allow stuffing."); + amount_to_stuff = 0; + } + + // try to keep the corrections definitely below 1 in 1000 audio frames - if ((amount_to_stuff==0) && (fix_volume==0x10000)) { - // if no stuffing needed and no volume adjustment, then - // don't send to stuff_buffer_* and don't copy to outbuf; just send directly to the output device... - config.output->play(inbuf, frame_size); - } else { + // calculate the time elapsed since the play session started. + + if (amount_to_stuff) { + if ((local_time_now) && (first_packet_time_to_play) && (local_time_now >= first_packet_time_to_play)) { + + int64_t tp = (local_time_now - first_packet_time_to_play)>>32; // seconds + + if (tp<5) + amount_to_stuff = 0; // wait at least five seconds + else if (tp<30) { + if ((random() % 1000) > 352) // keep it to about 1:1000 for the first thirty seconds + amount_to_stuff = 0; + } + } + } + + if ((amount_to_stuff == 0) && (fix_volume == 0x10000)) { + // if no stuffing needed and no volume adjustment, then + // don't send to stuff_buffer_* and don't copy to outbuf; just send directly to the + // output device... + config.output->play(inbuf, frame_size); + } else { #ifdef HAVE_LIBSOXR - switch (config.packet_stuffing) { + switch (config.packet_stuffing) { case ST_basic: -// if (amount_to_stuff) debug(1,"Basic stuff..."); - play_samples = stuff_buffer_basic(inbuf, outbuf,amount_to_stuff); + // if (amount_to_stuff) debug(1,"Basic stuff..."); + play_samples = stuff_buffer_basic(inbuf, outbuf, amount_to_stuff); break; case ST_soxr: -// if (amount_to_stuff) debug(1,"Soxr stuff..."); - play_samples = stuff_buffer_soxr(inbuf, outbuf,amount_to_stuff); + // if (amount_to_stuff) debug(1,"Soxr stuff..."); + play_samples = stuff_buffer_soxr(inbuf, outbuf, amount_to_stuff); break; - } + } #else -// if (amount_to_stuff) debug(1,"Standard stuff..."); - play_samples = stuff_buffer_basic(inbuf, outbuf,amount_to_stuff); + // if (amount_to_stuff) debug(1,"Standard stuff..."); + play_samples = stuff_buffer_basic(inbuf, outbuf, amount_to_stuff); #endif - /* - { - int co; - int is_silent=1; - short *p = outbuf; - for (co=0;coplay(outbuf, play_samples); - } - - // check for loss of sync - // timestamp of zero means an inserted silent frame in place of a missing frame - if ((inframe->timestamp!=0) && (!please_stop) && (config.resyncthreshold!=0) && (abs(sync_error)>config.resyncthreshold)) { - sync_error_out_of_bounds++; - // debug(1,"Sync error out of bounds: Error: %lld; previous error: %lld; DAC: %lld; timestamp: %llx, time now %llx",sync_error,previous_sync_error,current_delay,inframe->timestamp,local_time_now); - if (sync_error_out_of_bounds>3) { - debug(1,"Lost sync with source for %d consecutive packets -- flushing and resyncing. Error: %lld.",sync_error_out_of_bounds,sync_error); + config.output->play(outbuf, play_samples); + } + + // check for loss of sync + // timestamp of zero means an inserted silent frame in place of a missing frame + if ((inframe->timestamp != 0) && (!please_stop) && (config.resyncthreshold != 0) && + (abs(sync_error) > config.resyncthreshold)) { + sync_error_out_of_bounds++; + // debug(1,"Sync error out of bounds: Error: %lld; previous error: %lld; DAC: %lld; + // timestamp: %llx, time now + // %llx",sync_error,previous_sync_error,current_delay,inframe->timestamp,local_time_now); + if (sync_error_out_of_bounds > 3) { + debug(1, "Lost sync with source for %d consecutive packets -- flushing and " + "resyncing. Error: %lld.", + sync_error_out_of_bounds, sync_error); + sync_error_out_of_bounds = 0; + player_flush(nt); + } + } else { sync_error_out_of_bounds = 0; - player_flush(nt); - } + } } else { - sync_error_out_of_bounds = 0; + // if there is no delay procedure, there can be no synchronising + if (fix_volume == 0x10000) + config.output->play(inbuf, frame_size); + else { + play_samples = stuff_buffer_basic(inbuf, outbuf, 0); + config.output->play(outbuf, frame_size); + } } - // mark the frame as finished - inframe->timestamp=0; + // mark the frame as finished + inframe->timestamp = 0; inframe->sequence_number = 0; - + // debug(1,"Sync error %lld frames. Amount to stuff %d." ,sync_error,amount_to_stuff); - - // new stats calculation. We want a running average of sync error, drift, adjustment, number of additions+subtractions - - if (number_of_statistics==trend_interval) { + + // new stats calculation. We want a running average of sync error, drift, adjustment, + // number of additions+subtractions + + if (number_of_statistics == trend_interval) { // here we remove the oldest statistical data and take it from the summaries as well - tsum_of_sync_errors-=statistics[oldest_statistic].sync_error; + tsum_of_sync_errors -= statistics[oldest_statistic].sync_error; tsum_of_drifts -= statistics[oldest_statistic].drift; - if (statistics[oldest_statistic].correction>0) - tsum_of_insertions_and_deletions-=statistics[oldest_statistic].correction; + if (statistics[oldest_statistic].correction > 0) + tsum_of_insertions_and_deletions -= statistics[oldest_statistic].correction; else - tsum_of_insertions_and_deletions+=statistics[oldest_statistic].correction; - tsum_of_corrections-=statistics[oldest_statistic].correction; - oldest_statistic=(oldest_statistic+1)%trend_interval; - number_of_statistics--; + tsum_of_insertions_and_deletions += statistics[oldest_statistic].correction; + tsum_of_corrections -= statistics[oldest_statistic].correction; + oldest_statistic = (oldest_statistic + 1) % trend_interval; + number_of_statistics--; } - + statistics[newest_statistic].sync_error = sync_error; statistics[newest_statistic].correction = amount_to_stuff; - if (number_of_statistics==0) + if (number_of_statistics == 0) statistics[newest_statistic].drift = 0; else - statistics[newest_statistic].drift = sync_error-previous_sync_error-previous_correction; + statistics[newest_statistic].drift = + sync_error - previous_sync_error - previous_correction; previous_sync_error = sync_error; previous_correction = amount_to_stuff; tsum_of_sync_errors += sync_error; tsum_of_drifts += statistics[newest_statistic].drift; - if (amount_to_stuff>0) - tsum_of_insertions_and_deletions+=amount_to_stuff; - else - tsum_of_insertions_and_deletions-=amount_to_stuff; - tsum_of_corrections+=amount_to_stuff; - - newest_statistic=(newest_statistic+1)%trend_interval; + if (amount_to_stuff > 0) { + tsum_of_insertions_and_deletions += amount_to_stuff; + } else { + tsum_of_insertions_and_deletions -= amount_to_stuff; + } + tsum_of_corrections += amount_to_stuff; + session_corrections += amount_to_stuff; + + + newest_statistic = (newest_statistic + 1) % trend_interval; number_of_statistics++; - } - if (play_number%print_interval==0) { - // we can now calculate running averages for sync error (frames), corrections (ppm), insertions plus deletions (ppm), drift (ppm) - double moving_average_sync_error = (1.0*tsum_of_sync_errors)/number_of_statistics; - double moving_average_correction = (1.0*tsum_of_corrections)/number_of_statistics; - double moving_average_insertions_plus_deletions = (1.0*tsum_of_insertions_and_deletions)/number_of_statistics; - double moving_average_drift = (1.0*tsum_of_drifts)/number_of_statistics; + if (play_number % print_interval == 0) { + // we can now calculate running averages for sync error (frames), corrections (ppm), + // insertions plus deletions (ppm), drift (ppm) + double moving_average_sync_error = (1.0 * tsum_of_sync_errors) / number_of_statistics; + double moving_average_correction = (1.0 * tsum_of_corrections) / number_of_statistics; + double moving_average_insertions_plus_deletions = + (1.0 * tsum_of_insertions_and_deletions) / number_of_statistics; + double moving_average_drift = (1.0 * tsum_of_drifts) / number_of_statistics; // if ((play_number/print_interval)%20==0) if (config.statistics_requested) - inform("Sync error: %.1f (frames); net correction: %.1f (ppm); corrections: %.1f (ppm); missing packets %llu; late packets %llu; too late packets %llu; resend requests %llu; min DAC queue size %lli, min and max buffer occupancy %u and %u.", moving_average_sync_error, moving_average_correction*1000000/352, moving_average_insertions_plus_deletions*1000000/352,missing_packets,late_packets,too_late_packets,resend_requests,minimum_dac_queue_size,minimum_buffer_occupancy,maximum_buffer_occupancy); - minimum_dac_queue_size=1000000; // hack reset - maximum_buffer_occupancy = 0; // can't be less than this + if (at_least_one_frame_seen) + if (config.output->delay) + inform("Sync error: %.1f (frames); net correction: %.1f (ppm); corrections: %.1f " + "(ppm); missing packets %llu; late packets %llu; too late packets %llu; " + "resend requests %llu; min DAC queue size %lli, min and max buffer occupancy " + "%u and %u.", + moving_average_sync_error, moving_average_correction * 1000000 / 352, + moving_average_insertions_plus_deletions * 1000000 / 352, missing_packets, + late_packets, too_late_packets, resend_requests, minimum_dac_queue_size, + minimum_buffer_occupancy, maximum_buffer_occupancy); + else + inform("Synchronisation disabled. Missing packets %llu; late packets %llu; too late packets %llu; " + "resend requests %llu; min and max buffer occupancy " + "%u and %u.", + missing_packets, + late_packets, too_late_packets, resend_requests, + minimum_buffer_occupancy, maximum_buffer_occupancy); + else + inform("No frames received in the last sampling interval."); + minimum_dac_queue_size = 1000000; // hack reset + maximum_buffer_occupancy = 0; // can't be less than this minimum_buffer_occupancy = BUFFER_FRAMES; // can't be more than this + at_least_one_frame_seen = 0; } } } @@ -1021,29 +1185,33 @@ static void *player_thread_func(void *arg) { return 0; } - // takes the volume as specified by the airplay protocol void player_volume(double f) { -// The volume ranges -144.0 (mute) or -30 -- 0. See http://git.zx2c4.com/Airtunes2/about/#setting-volume -// By examination, the -30 -- 0 range is linear on the slider; i.e. the slider is calibrated in 30 equal increments -// So, we will pass this on without any weighting if we have a hardware mixer, as we expect the mixer to be calibrated in dB. + // The volume ranges -144.0 (mute) or -30 -- 0. See + // http://git.zx2c4.com/Airtunes2/about/#setting-volume + // By examination, the -30 -- 0 range is linear on the slider; i.e. the slider is calibrated in 30 + // equal increments + // So, we will pass this on without any weighting if we have a hardware mixer, as we expect the + // mixer to be calibrated in dB. -// Here, we ask for an attenuation we will apply in software. The dB range of a value from 1 to 65536 is about 48.1 dB (log10 of 65536 is 4.8164). -// Thus, we ask our vol2attn function for an appropriate dB between -48.1 and 0 dB and translate it back to a number. - - double scaled_volume = vol2attn(f,0,-4810); - double linear_volume = pow(10,scaled_volume/1000); - - if(f == -144.0) + // Here, we ask for an attenuation we will apply in software. The dB range of a value from 1 to + // 65536 is about 48.1 dB (log10 of 65536 is 4.8164). + // Thus, we ask our vol2attn function for an appropriate dB between -48.1 and 0 dB and translate + // it back to a number. + + double scaled_volume = vol2attn(f, 0, -4810); + double linear_volume = pow(10, scaled_volume / 1000); + + if (f == -144.0) linear_volume = 0.0; - + if (config.output->volume) { - config.output->volume(f); - linear_volume=1.0; // no attenuation needed -- this value is used as a flag to avoid calculations + config.output->volume(f); // volume will be sent as metadata by the config.output device + linear_volume = + 1.0; // no attenuation needed -- this value is used as a flag to avoid calculations } - - + if (config.output->parameters) config.output->parameters(&audio_information); else { @@ -1054,58 +1222,76 @@ void player_volume(double f) { audio_information.has_true_mute = 0; audio_information.is_muted = 0; } - audio_information.valid=1; + audio_information.valid = 1; pthread_mutex_lock(&vol_mutex); - software_mixer_volume = linear_volume; - fix_volume = 65536.0 * software_mixer_volume; - pthread_mutex_unlock(&vol_mutex); + software_mixer_volume = linear_volume; + fix_volume = 65536.0 * software_mixer_volume; + pthread_mutex_unlock(&vol_mutex); +#ifdef CONFIG_METADATA + char *dv = malloc(64); // will be freed in the metadata thread + if (dv) { + memset(dv, 0, 64); + snprintf(dv, 63, "%.2f,%.2f,%.2f,%.2f", audio_information.airplay_volume, + audio_information.current_volume_dB / 100.0, + audio_information.minimum_volume_dB / 100.0, + audio_information.maximum_volume_dB / 100.0); + send_ssnc_metadata('pvol', dv, strlen(dv), 1); + } +#endif } void player_flush(uint32_t timestamp) { - // debug(1,"Flush requested up to %u. It seems as if 0 is special.",timestamp); + // debug(1,"Flush requested up to %u. It seems as if 0 is special.",timestamp); pthread_mutex_lock(&flush_mutex); - flush_requested=1; - //if (timestamp!=0) - flush_rtp_timestamp=timestamp; // flush all packets up to (and including?) this + flush_requested = 1; + // if (timestamp!=0) + flush_rtp_timestamp = timestamp; // flush all packets up to (and including?) this pthread_mutex_unlock(&flush_mutex); + play_segment_reference_frame = 0; +#ifdef CONFIG_METADATA + send_ssnc_metadata('pfls', NULL, 0, 1); +#endif } int player_play(stream_cfg *stream) { packet_count = 0; + encrypted = stream->encrypted; if (config.buffer_start_fill > BUFFER_FRAMES) - die("specified buffer starting fill %d > buffer size %d", - config.buffer_start_fill, BUFFER_FRAMES); - + die("specified buffer starting fill %d > buffer size %d", config.buffer_start_fill, + BUFFER_FRAMES); + if (encrypted) { #ifdef HAVE_LIBPOLARSSL - memset(&dctx,0,sizeof(aes_context)); - aes_setkey_dec(&dctx, stream->aeskey, 128); + memset(&dctx, 0, sizeof(aes_context)); + aes_setkey_dec(&dctx, stream->aeskey, 128); #endif #ifdef HAVE_LIBSSL - AES_set_decrypt_key(stream->aeskey, 128, &aes); + AES_set_decrypt_key(stream->aeskey, 128, &aes); #endif - - aesiv = stream->aesiv; + aesiv = stream->aesiv; + } init_decoder(stream->fmtp); // must be after decoder init init_buffer(); please_stop = 0; command_start(); - - // set the flowcontrol condition variable to wait on a monotonic clock -#ifdef COMPILE_FOR_LINUX +#ifdef CONFIG_METADATA + send_ssnc_metadata('pbeg', NULL, 0, 1); +#endif + +// set the flowcontrol condition variable to wait on a monotonic clock +#ifdef COMPILE_FOR_LINUX_AND_FREEBSD pthread_condattr_t attr; pthread_condattr_init(&attr); - pthread_condattr_setclock( &attr, CLOCK_MONOTONIC); // can't do this in OS X, and don't need it. - int rc = pthread_cond_init(&flowcontrol,&attr); + pthread_condattr_setclock(&attr, CLOCK_MONOTONIC); // can't do this in OS X, and don't need it. + int rc = pthread_cond_init(&flowcontrol, &attr); #endif #ifdef COMPILE_FOR_OSX - int rc = pthread_cond_init(&flowcontrol,NULL); + int rc = pthread_cond_init(&flowcontrol, NULL); #endif if (rc) - debug(1,"Error initialising condition variable."); + debug(1, "Error initialising condition variable."); config.output->start(sampling_rate); - size_t size = (PTHREAD_STACK_MIN + 256 * 1024); pthread_attr_t tattr; pthread_attr_init(&tattr); @@ -1114,7 +1300,6 @@ int player_play(stream_cfg *stream) { debug(1, "Error setting stack size for player_thread: %s", strerror(errno)); pthread_create(&player_thread, &tattr, player_thread_func, NULL); pthread_attr_destroy(&tattr); - return 0; } @@ -1122,11 +1307,14 @@ void player_stop(void) { please_stop = 1; pthread_cond_signal(&flowcontrol); // tell it to give up pthread_join(player_thread, NULL); +#ifdef CONFIG_METADATA + send_ssnc_metadata('pend', NULL, 0, 1); +#endif config.output->stop(); command_stop(); free_buffer(); free_decoder(); int rc = pthread_cond_destroy(&flowcontrol); if (rc) - debug(1,"Error destroying condition variable."); + debug(1, "Error destroying condition variable."); } diff --git a/player.h b/player.h index 0a9f82a5..b9cc900a 100644 --- a/player.h +++ b/player.h @@ -4,8 +4,9 @@ #include "audio.h" typedef struct { - uint8_t aesiv[16], aeskey[16]; - int32_t fmtp[12]; + int encrypted; + uint8_t aesiv[16], aeskey[16]; + int32_t fmtp[12]; } stream_cfg; typedef uint16_t seq_t; @@ -19,6 +20,6 @@ void player_stop(void); void player_volume(double f); void player_flush(uint32_t timestamp); -void player_put_packet(seq_t seqno,uint32_t timestamp, uint8_t *data, int len); +void player_put_packet(seq_t seqno, uint32_t timestamp, uint8_t *data, int len); #endif //_PLAYER_H diff --git a/rtp.c b/rtp.c index dea71c3c..e00a7536 100644 --- a/rtp.c +++ b/rtp.c @@ -41,15 +41,16 @@ #include "player.h" #include "rtp.h" - typedef struct { - uint32_t seconds; - uint32_t fraction; + uint32_t seconds; + uint32_t fraction; } ntp_timestamp; typedef struct time_ping_record { uint64_t local_to_remote_difference; uint64_t dispersion; + uint64_t local_time; + uint64_t remote_time; } time_ping_record; // only one RTP session can be active at a time. @@ -57,184 +58,211 @@ static int running = 0; static int please_shutdown; static char client_ip_string[INET6_ADDRSTRLEN]; // the ip string pointing to the client -static short client_ip_family; // AF_INET / AF_INET6 -static uint32_t client_active_remote; // used when you want to control the client... +static short client_ip_family; // AF_INET / AF_INET6 +static uint32_t client_active_remote; // used when you want to control the client... static SOCKADDR rtp_client_control_socket; // a socket pointing to the control port of the client -static SOCKADDR rtp_client_timing_socket; // a socket pointing to the timing port of the client -static int audio_socket; // our local [server] audio socket -static int control_socket; // our local [server] control socket -static int timing_socket; // local timing socket +static SOCKADDR rtp_client_timing_socket; // a socket pointing to the timing port of the client +static int audio_socket; // our local [server] audio socket +static int control_socket; // our local [server] control socket +static int timing_socket; // local timing socket static pthread_t rtp_audio_thread, rtp_control_thread, rtp_timing_thread; static uint32_t reference_timestamp; static uint64_t reference_timestamp_time; +static uint64_t remote_reference_timestamp_time; // debug variables static int request_sent; -#define time_ping_history 8 -#define time_ping_fudge_factor 100000 +#define time_ping_history 8 +#define time_ping_fudge_factor 100000 static uint8_t time_ping_count; struct time_ping_record time_pings[time_ping_history]; -//static struct timespec dtt; // dangerous -- this assumes that there will never be two timing request in flight at the same time -static uint64_t departure_time; // dangerous -- this assumes that there will never be two timing request in flight at the same time +// static struct timespec dtt; // dangerous -- this assumes that there will never be two timing +// request in flight at the same time +static uint64_t departure_time; // dangerous -- this assumes that there will never be two timing + // request in flight at the same time static pthread_mutex_t reference_time_mutex = PTHREAD_MUTEX_INITIALIZER; uint64_t static local_to_remote_time_difference; // used to switch between local and remote clocks static void *rtp_audio_receiver(void *arg) { - // we inherit the signal mask (SIGUSR1) - - int32_t last_seqno = -1; - uint8_t packet[2048], *pktp; + // we inherit the signal mask (SIGUSR1) - ssize_t nread; - while (1) { - if (please_shutdown) - break; - nread = recv(audio_socket, packet, sizeof(packet), 0); - if (nread < 0) - break; + int32_t last_seqno = -1; + uint8_t packet[2048], *pktp; - ssize_t plen = nread; - uint8_t type = packet[1] & ~0x80; - if (type == 0x60 || type == 0x56) { // audio data / resend - pktp = packet; - if (type==0x56) { - pktp += 4; - plen -= 4; - } - seq_t seqno = ntohs(*(unsigned short *)(pktp+2)); - // increment last_seqno and see if it's the same as the incoming seqno + ssize_t nread; + while (1) { + if (please_shutdown) + break; + nread = recv(audio_socket, packet, sizeof(packet), 0); + if (nread < 0) + break; - if (last_seqno==-1) - last_seqno=seqno; - else { - last_seqno = (last_seqno+1)&0xffff; - if (seqno!=last_seqno) - debug(2,"RTP: Packets out of sequence: expected: %d, got %d.",last_seqno,seqno); - last_seqno=seqno; // reset warning... - } - uint32_t timestamp = ntohl(*(unsigned long *)(pktp+4)); - - //if (packet[1]&0x10) - // debug(1,"Audio packet Extension bit set."); + ssize_t plen = nread; + uint8_t type = packet[1] & ~0x80; + if (type == 0x60 || type == 0x56) { // audio data / resend + pktp = packet; + if (type == 0x56) { + pktp += 4; + plen -= 4; + } + seq_t seqno = ntohs(*(unsigned short *)(pktp + 2)); + // increment last_seqno and see if it's the same as the incoming seqno - pktp += 12; - plen -= 12; + if (last_seqno == -1) + last_seqno = seqno; + else { + last_seqno = (last_seqno + 1) & 0xffff; + if (seqno != last_seqno) + debug(2, "RTP: Packets out of sequence: expected: %d, got %d.", last_seqno, seqno); + last_seqno = seqno; // reset warning... + } + uint32_t timestamp = ntohl(*(unsigned long *)(pktp + 4)); - // check if packet contains enough content to be reasonable - if (plen >= 16) { - player_put_packet(seqno,timestamp, pktp, plen); - continue; - } - if (type == 0x56 && seqno == 0) { - debug(2, "resend-related request packet received, ignoring."); - continue; - } - debug(1, "Audio receiver -- Unknown RTP packet of type 0x%02X length %d seqno %d", type, nread, seqno); - } - warn("Audio receiver -- Unknown RTP packet of type 0x%02X length %d.", type, nread); + // if (packet[1]&0x10) + // debug(1,"Audio packet Extension bit set."); + + pktp += 12; + plen -= 12; + + // check if packet contains enough content to be reasonable + if (plen >= 16) { + player_put_packet(seqno, timestamp, pktp, plen); + continue; + } + if (type == 0x56 && seqno == 0) { + debug(2, "resend-related request packet received, ignoring."); + continue; + } + debug(1, "Audio receiver -- Unknown RTP packet of type 0x%02X length %d seqno %d", type, + nread, seqno); } + warn("Audio receiver -- Unknown RTP packet of type 0x%02X length %d.", type, nread); + } - debug(1, "Audio receiver -- Server RTP thread interrupted. terminating."); - close(audio_socket); + debug(1, "Audio receiver -- Server RTP thread interrupted. terminating."); + close(audio_socket); - return NULL; + return NULL; } static void *rtp_control_receiver(void *arg) { - // we inherit the signal mask (SIGUSR1) - reference_timestamp=0; // nothing valid received yet - uint8_t packet[2048]; - struct timespec tn; - uint64_t remote_time_of_sync,local_time_now, remote_time_now; - uint32_t sync_rtp_timestamp,rtp_timestamp_less_latency; - ssize_t nread; - while (1) { - if (please_shutdown) - break; - nread = recv(control_socket, packet, sizeof(packet), 0); - local_time_now=get_absolute_time_in_fp(); -// clock_gettime(CLOCK_MONOTONIC,&tn); -// local_time_now=((uint64_t)tn.tv_sec<<32)+((uint64_t)tn.tv_nsec<<32)/1000000000; - - if (nread < 0) - break; + // we inherit the signal mask (SIGUSR1) + reference_timestamp = 0; // nothing valid received yet + uint8_t packet[2048], *pktp; + struct timespec tn; + uint64_t remote_time_of_sync, local_time_now, remote_time_now; + uint32_t sync_rtp_timestamp, rtp_timestamp_less_latency; + ssize_t nread; + while (1) { + if (please_shutdown) + break; + nread = recv(control_socket, packet, sizeof(packet), 0); + local_time_now = get_absolute_time_in_fp(); + // clock_gettime(CLOCK_MONOTONIC,&tn); + // local_time_now=((uint64_t)tn.tv_sec<<32)+((uint64_t)tn.tv_nsec<<32)/1000000000; - ssize_t plen = nread; - if (packet[1] == 0xd4) { // sync data - /* - char obf[4096]; - char *obfp = obf; - int obfc; - for (obfc=0;obfc>32); - //debug(1,"Sync Time is %lld us late (local times).",((local_time_now-reference_timestamp_time)*1000000)>>32); - } else { - debug(1,"Sync packet received before we got a timing packet back."); - } - } else - debug(1,"Control Port -- Unknown RTP packet of type 0x%02X length %d.", packet[1], nread); - } + if (nread < 0) + break; - debug(1, "Control RTP thread interrupted. terminating."); - close(control_socket); + ssize_t plen = nread; + if (packet[1] == 0xd4) { // sync data + /* + char obf[4096]; + char *obfp = obf; + int obfc; + for (obfc=0;obfc>32); + // debug(1,"Sync Time is %lld us late (local + // times).",((local_time_now-reference_timestamp_time)*1000000)>>32); + } else { + debug(1, "Sync packet received before we got a timing packet back."); + } + } else if (packet[1] == 0xd6) { // resent audio data in the control path -- whaale only? + // debug(1, "Control Port -- Retransmitted Audio Data Packet received."); + pktp = packet+4; + plen -= 4; + seq_t seqno = ntohs(*(unsigned short *)(pktp + 2)); + + uint32_t timestamp = ntohl(*(unsigned long *)(pktp + 4)); + + pktp += 12; + plen -= 12; + + // check if packet contains enough content to be reasonable + if (plen >= 16) { + player_put_packet(seqno, timestamp, pktp, plen); + continue; + } else { + debug(1, "Too-short retransmitted audio packet received in control port, ignored."); + } + } else + debug(1, "Control Port -- Unknown RTP packet of type 0x%02X length %d.", packet[1], nread); + } + + debug(1, "Control RTP thread interrupted. terminating."); + close(control_socket); + + return NULL; } static void *rtp_timing_sender(void *arg) { - struct timing_request { + struct timing_request { char leader; char type; uint16_t seqno; uint32_t filler; - uint64_t origin,receive,transmit; + uint64_t origin, receive, transmit; }; - - uint64_t request_number=0; - - struct timing_request req; // *not* a standard RTCP NACK - + + uint64_t request_number = 0; + + struct timing_request req; // *not* a standard RTCP NACK + req.leader = 0x80; - req.type = 0xd2; // Timing request + req.type = 0xd2; // Timing request req.filler = 0; - req.seqno=htons(7); - + req.seqno = htons(7); + time_ping_count = 0; // we inherit the signal mask (SIGUSR1) @@ -245,392 +273,492 @@ static void *rtp_timing_sender(void *arg) { if (!running) die("rtp_timing_sender called without active stream!"); - //debug(1, "Requesting ntp timestamp exchange."); + // debug(1, "Requesting ntp timestamp exchange."); req.filler = 0; - req.origin = req.receive = req.transmit=0; + req.origin = req.receive = req.transmit = 0; -// clock_gettime(CLOCK_MONOTONIC,&dtt); - departure_time=get_absolute_time_in_fp(); - socklen_t msgsize = sizeof(struct sockaddr_in); + // clock_gettime(CLOCK_MONOTONIC,&dtt); + departure_time = get_absolute_time_in_fp(); + socklen_t msgsize = sizeof(struct sockaddr_in); #ifdef AF_INET6 - if (rtp_client_timing_socket.SAFAMILY==AF_INET6) { - msgsize = sizeof(struct sockaddr_in6); - } + if (rtp_client_timing_socket.SAFAMILY == AF_INET6) { + msgsize = sizeof(struct sockaddr_in6); + } #endif - if (sendto(timing_socket, &req, sizeof(req), 0, (struct sockaddr*)&rtp_client_timing_socket, msgsize)==-1) { - perror("Error sendto-ing to timing socket"); - } + if (sendto(timing_socket, &req, sizeof(req), 0, (struct sockaddr *)&rtp_client_timing_socket, + msgsize) == -1) { + perror("Error sendto-ing to timing socket"); + } request_number++; - if (request_number<=4) + if (request_number <= 4) usleep(500000); else sleep(3); } - debug(1, "rtp_timing_sender thread interrupted. terminating."); + debug(1, "rtp_timing_sender thread interrupted. terminating."); return NULL; } static void *rtp_timing_receiver(void *arg) { - // we inherit the signal mask (SIGUSR1) - uint8_t packet[2048], *pktp; - ssize_t nread; - pthread_t timer_requester; - pthread_create(&timer_requester, NULL, &rtp_timing_sender, NULL); -// struct timespec att; - uint64_t distant_receive_time,distant_transmit_time,arrival_time,return_time,transit_time,processing_time; - local_to_remote_time_jitters = 0; - local_to_remote_time_jitters_count = 0; - uint64_t l2rtd=0; - while (1) { - if (please_shutdown) - break; - nread = recv(timing_socket, packet, sizeof(packet), 0); - arrival_time=get_absolute_time_in_fp(); -// clock_gettime(CLOCK_MONOTONIC,&att); - - if (nread < 0) - break; + // we inherit the signal mask (SIGUSR1) + uint8_t packet[2048], *pktp; + ssize_t nread; + pthread_t timer_requester; + pthread_create(&timer_requester, NULL, &rtp_timing_sender, NULL); + // struct timespec att; + uint64_t distant_receive_time, distant_transmit_time, arrival_time, return_time, transit_time, + processing_time; + local_to_remote_time_jitters = 0; + local_to_remote_time_jitters_count = 0; + uint64_t first_remote_time = 0; + uint64_t first_local_time = 0; + + uint64_t first_local_to_remote_time_difference = 0; + uint64_t first_local_to_remote_time_difference_time; + uint64_t l2rtd = 0; + while (1) { + if (please_shutdown) + break; + nread = recv(timing_socket, packet, sizeof(packet), 0); + arrival_time = get_absolute_time_in_fp(); + // clock_gettime(CLOCK_MONOTONIC,&att); - ssize_t plen = nread; - //debug(1,"Packet Received on Timing Port."); - if (packet[1] == 0xd3) { // timing reply - /* - char obf[4096]; - char *obfp = obf; - int obfc; - for (obfc=0;obfc>32; debug(1,"Time ping turnaround time: %lld us.",rtus); - - //distant_receive_time = ((uint64_t)ntohl(*((uint32_t*)&packet[16])))<<32+ntohl(*((uint32_t*)&packet[20])); - - distant_receive_time = (uint64_t)ntohl(*((uint32_t*)&packet[16]))<<32; - distant_receive_time += ntohl(*((uint32_t*)&packet[20])); - - //distant_transmit_time = ((uint64_t)ntohl(*((uint32_t*)&packet[24])))<<32+ntohl(*((uint32_t*)&packet[28])); - - distant_transmit_time = (uint64_t)ntohl(*((uint32_t*)&packet[24]))<<32; - distant_transmit_time += ntohl(*((uint32_t*)&packet[28])); - - processing_time = distant_transmit_time-distant_receive_time; - - // debug(1,"Return trip time: %lluuS, remote processing time: %lluuS.",(return_time*1000000)>>32,(processing_time*1000000)>>32); + ssize_t plen = nread; + // debug(1,"Packet Received on Timing Port."); + if (packet[1] == 0xd3) { // timing reply + /* + char obf[4096]; + char *obfp = obf; + int obfc; + for (obfc=0;obfc0;cc--) { - time_pings[cc]=time_pings[cc-1]; - time_pings[cc].dispersion = (time_pings[cc].dispersion*133)/100; // make the dispersions 'age' by this rational factor - } - time_pings[0].local_to_remote_difference = local_time_by_remote_clock-arrival_time; - time_pings[0].dispersion = return_time; - if (time_ping_count1) { - if (l2rtd>local_to_remote_time_difference) { - local_to_remote_time_jitters=local_to_remote_time_jitters+l2rtd-local_to_remote_time_difference; - ji = l2rtd-local_to_remote_time_difference; - } else { - local_to_remote_time_jitters=local_to_remote_time_jitters+local_to_remote_time_difference-l2rtd; - ji = - (local_to_remote_time_difference-l2rtd); - } - local_to_remote_time_jitters_count+=1; - } - // uncomment below to print jitter between client's clock and oour clock - // int64_t rtus = (tld*1000000)>>32; ji = (ji*1000000)>>32; debug(1,"Choosing time difference with dispersion of %lld us with delta of %lld us",rtus,ji); + return_time = arrival_time - departure_time; - local_to_remote_time_difference=l2rtd; - } else { - debug(1, "Timing port -- Unknown RTP packet of type 0x%02X length %d.", packet[1], nread); + // uint64_t rtus = (return_time*1000000)>>32; debug(1,"Time ping turnaround time: %lld + // us.",rtus); + + // distant_receive_time = + // ((uint64_t)ntohl(*((uint32_t*)&packet[16])))<<32+ntohl(*((uint32_t*)&packet[20])); + + distant_receive_time = (uint64_t)ntohl(*((uint32_t *)&packet[16])) << 32; + distant_receive_time += ntohl(*((uint32_t *)&packet[20])); + + // distant_transmit_time = + // ((uint64_t)ntohl(*((uint32_t*)&packet[24])))<<32+ntohl(*((uint32_t*)&packet[28])); + + distant_transmit_time = (uint64_t)ntohl(*((uint32_t *)&packet[24])) << 32; + distant_transmit_time += ntohl(*((uint32_t *)&packet[28])); + + processing_time = distant_transmit_time - distant_receive_time; + + // debug(1,"Return trip time: %lluuS, remote processing time: + // %lluuS.",(return_time*1000000)>>32,(processing_time*1000000)>>32); + + uint64_t local_time_by_remote_clock = distant_transmit_time + return_time / 2; + + unsigned int cc; + for (cc = time_ping_history - 1; cc > 0; cc--) { + time_pings[cc] = time_pings[cc - 1]; + time_pings[cc].dispersion = (time_pings[cc].dispersion * 133) / + 100; // make the dispersions 'age' by this rational factor } + // these are for diagnostics only -- not used + time_pings[0].local_time = arrival_time; + time_pings[0].remote_time = distant_transmit_time; + + time_pings[0].local_to_remote_difference = local_time_by_remote_clock - arrival_time; + time_pings[0].dispersion = return_time; + if (time_ping_count < time_ping_history) + time_ping_count++; + + uint64_t local_time_chosen = arrival_time;; + uint64_t remote_time_chosen = distant_transmit_time; + // now pick the timestamp with the lowest dispersion + uint64_t l2rtd = time_pings[0].local_to_remote_difference; + uint64_t tld = time_pings[0].dispersion; + for (cc = 1; cc < time_ping_count; cc++) + if (time_pings[cc].dispersion < tld) { + l2rtd = time_pings[cc].local_to_remote_difference; + tld = time_pings[cc].dispersion; + local_time_chosen = time_pings[cc].local_time; + remote_time_chosen = time_pings[cc].remote_time; + } + int64_t ji; + + if (time_ping_count > 1) { + if (l2rtd > local_to_remote_time_difference) { + local_to_remote_time_jitters = + local_to_remote_time_jitters + l2rtd - local_to_remote_time_difference; + ji = l2rtd - local_to_remote_time_difference; + } else { + local_to_remote_time_jitters = + local_to_remote_time_jitters + local_to_remote_time_difference - l2rtd; + ji = -(local_to_remote_time_difference - l2rtd); + } + local_to_remote_time_jitters_count += 1; + } + // uncomment below to print jitter between client's clock and oour clock + // int64_t rtus = (tld*1000000)>>32; ji = (ji*1000000)>>32; debug(1,"Choosing time difference + // with dispersion of %lld us with delta of %lld us",rtus,ji); + + local_to_remote_time_difference = l2rtd; + if (first_local_to_remote_time_difference==0) { + first_local_to_remote_time_difference = local_to_remote_time_difference; + first_local_to_remote_time_difference_time = get_absolute_time_in_fp(); + } + + int64_t clock_drift, clock_drift_in_usec; + if (first_local_time==0) { + first_local_time = local_time_chosen; + first_remote_time = remote_time_chosen; + uint64_t clock_drift = 0; + } else { + uint64_t local_time_change = local_time_chosen - first_local_time; + uint64_t remote_time_change = remote_time_chosen - first_remote_time; + + + if (remote_time_change >= local_time_change) + clock_drift = remote_time_change - local_time_change; + else + clock_drift = -(local_time_change - remote_time_change); + } + if (clock_drift>=0) + clock_drift_in_usec = (clock_drift * 1000000)>>32; + else + clock_drift_in_usec = -(((-clock_drift) * 1000000)>>32); + + + + int64_t source_drift_usec; + if (play_segment_reference_frame!=0) { + uint32_t reference_timestamp; + uint64_t reference_timestamp_time,remote_reference_timestamp_time; + get_reference_timestamp_stuff(&reference_timestamp, &reference_timestamp_time, &remote_reference_timestamp_time); + uint64_t frame_difference = 0; + if (reference_timestamp>=play_segment_reference_frame) + frame_difference = (uint64_t)reference_timestamp-(uint64_t)play_segment_reference_frame; + else // rollover + frame_difference = (uint64_t)reference_timestamp+0x100000000-(uint64_t)play_segment_reference_frame; + uint64_t frame_time_difference_calculated = (((uint64_t)frame_difference<<32)/44100); + uint64_t frame_time_difference_actual = remote_reference_timestamp_time-play_segment_reference_frame_remote_time; // this is all done by reference to the sources' system clock + // debug(1,"%llu frames since play started, %llu usec calculated, %llu usec actual",frame_difference, (frame_time_difference_calculated*1000000)>>32, (frame_time_difference_actual*1000000)>>32); + if (frame_time_difference_calculated>=frame_time_difference_actual) // i.e. if the time it should have taken to send the packets is greater than the actual time difference measured on the source clock + // then the source DAC's clock is running fast relative to the source system clock + source_drift_usec = frame_time_difference_calculated-frame_time_difference_actual; + else + // otherwise the source DAC's clock is running slow relative to the source system clock + source_drift_usec = -(frame_time_difference_actual-frame_time_difference_calculated); + } else + source_drift_usec = 0; + source_drift_usec = (source_drift_usec*1000000)>>32; // turn it to microseconds + + int64_t current_delay = 0; + if (config.output->delay) { + current_delay = config.output->delay(); + } + // Useful for troubleshooting: + // clock_drift between source and local clock -- +ve means source is faster + // session_corrections -- the amount of correction done, in microseconds. +ve means frames added + // current_delay = delay in DAC buffer in frames + // source_drift_usec = how much faster (+ve) or slower the source DAC is running relative to the source clock + // buffer_occupancy = the number of buffers occupied. Crude, but should show no long term trend if source and device are in sync. + // return_time = the time from soliciting a timing packet to getting it back. It should be short ( < 5 ms) and pretty consistent. + // debug(1, "%lld\t%lld\t%lld\t%lld\t%u\t%llu", clock_drift_in_usec,(session_corrections*1000000)/44100,current_delay,source_drift_usec,buffer_occupancy,(return_time*1000000)>>32); + + } else { + debug(1, "Timing port -- Unknown RTP packet of type 0x%02X length %d.", packet[1], nread); } + } - debug(1, "Timing RTP thread interrupted. terminating."); - void *retval; - pthread_kill(timer_requester, SIGUSR1); - pthread_join(timer_requester, &retval); - debug(1,"Closed and terminated timer requester thread."); - debug(1, "Timing RTP thread terminated."); - close(timing_socket); + debug(1, "Timing RTP thread interrupted. terminating."); + void *retval; + pthread_kill(timer_requester, SIGUSR1); + pthread_join(timer_requester, &retval); + debug(1, "Closed and terminated timer requester thread."); + debug(1, "Timing RTP thread terminated."); + close(timing_socket); - return NULL; + return NULL; } -static int bind_port(SOCKADDR *remote,int *sock, int desired_port ) { - struct - addrinfo hints, *info; +static int bind_port(SOCKADDR *remote, int *sock) { + struct addrinfo hints, *info; - memset(&hints, 0, sizeof(hints)); - hints.ai_family = remote->SAFAMILY; - hints.ai_socktype = SOCK_DGRAM; - hints.ai_flags = AI_PASSIVE; - - char buffer[10]; + memset(&hints, 0, sizeof(hints)); + hints.ai_family = remote->SAFAMILY; + hints.ai_socktype = SOCK_DGRAM; + hints.ai_flags = AI_PASSIVE; + + char buffer[10]; + + // look for a port in the range, if any was specified. + int desired_port = config.udp_port_base; + int ret; + do { snprintf(buffer, 10, "%d", desired_port); - - int ret = getaddrinfo(NULL,buffer, &hints, &info); + + ret = getaddrinfo(NULL, buffer, &hints, &info); if (ret < 0) - die("failed to get usable addrinfo?! %s.", gai_strerror(ret)); + die("failed to get usable addrinfo?! %s.", gai_strerror(ret)); *sock = socket(remote->SAFAMILY, SOCK_DGRAM, IPPROTO_UDP); ret = bind(*sock, info->ai_addr, info->ai_addrlen); freeaddrinfo(info); - if (ret < 0) - die("could not bind a UDP port!"); - - int sport; - SOCKADDR local; - socklen_t local_len = sizeof(local); - getsockname(*sock, (struct sockaddr*)&local, &local_len); + } while ((ret<0) && (errno==EADDRINUSE) && (desired_port!=0) && (desired_port++ < config.udp_port_base+config.udp_port_range)); + + // debug(1,"UDP port chosen: %d.",desired_port); + + if (ret < 0) { + die("error: could not bind a UDP port!"); + } + + + int sport; + SOCKADDR local; + socklen_t local_len = sizeof(local); + getsockname(*sock, (struct sockaddr *)&local, &local_len); #ifdef AF_INET6 - if (local.SAFAMILY == AF_INET6) { - struct sockaddr_in6 *sa6 = (struct sockaddr_in6*)&local; - sport = ntohs(sa6->sin6_port); - } else + if (local.SAFAMILY == AF_INET6) { + struct sockaddr_in6 *sa6 = (struct sockaddr_in6 *)&local; + sport = ntohs(sa6->sin6_port); + } else #endif - { - struct sockaddr_in *sa = (struct sockaddr_in*)&local; - sport = ntohs(sa->sin_port); - } + { + struct sockaddr_in *sa = (struct sockaddr_in *)&local; + sport = ntohs(sa->sin_port); + } - return sport; + return sport; } +void rtp_setup(SOCKADDR *remote, int cport, int tport, uint32_t active_remote, int *lsport, + int *lcport, int *ltport) { + if (running) + die("rtp_setup called with active stream!"); -void rtp_setup(SOCKADDR *remote, int cport, int tport, uint32_t active_remote, int *lsport, int *lcport, int *ltport) { - if (running) - die("rtp_setup called with active stream!"); + debug(2, "rtp_setup: cport=%d tport=%d.", cport, tport); - debug(2, "rtp_setup: cport=%d tport=%d.", cport, tport); - - client_active_remote = active_remote; - - // print out what we know about the client - void *addr; - char *ipver; - int port; - char portstr[20]; - client_ip_family = remote->SAFAMILY; // keep information about the kind of ip of the client + client_active_remote = active_remote; + + // print out what we know about the client + void *addr; + char *ipver; + int port; + char portstr[20]; + client_ip_family = remote->SAFAMILY; // keep information about the kind of ip of the client #ifdef AF_INET6 - if (remote->SAFAMILY == AF_INET6) { - struct sockaddr_in6 *sa6 = (struct sockaddr_in6*)remote; - addr = &(sa6->sin6_addr); - port = ntohs(sa6->sin6_port); - ipver = "IPv6"; - } + if (remote->SAFAMILY == AF_INET6) { + struct sockaddr_in6 *sa6 = (struct sockaddr_in6 *)remote; + addr = &(sa6->sin6_addr); + port = ntohs(sa6->sin6_port); + ipver = "IPv6"; + } #endif - if (remote->SAFAMILY == AF_INET) { - struct sockaddr_in *sa4 = (struct sockaddr_in*)remote; - addr = &(sa4->sin_addr); - port = ntohs(sa4->sin_port); - ipver = "IPv4"; - } - inet_ntop(remote->SAFAMILY,addr,client_ip_string,sizeof(client_ip_string)); // keep the client's ip number - debug(1,"Connection from %s: %s:%d",ipver,client_ip_string,port); - - + if (remote->SAFAMILY == AF_INET) { + struct sockaddr_in *sa4 = (struct sockaddr_in *)remote; + addr = &(sa4->sin_addr); + port = ntohs(sa4->sin_port); + ipver = "IPv4"; + } + inet_ntop(remote->SAFAMILY, addr, client_ip_string, + sizeof(client_ip_string)); // keep the client's ip number + debug(1, "Connection from %s: %s:%d", ipver, client_ip_string, port); - // set up a the record of the remote's control socket - struct addrinfo hints; - struct addrinfo *servinfo; + // set up a the record of the remote's control socket + struct addrinfo hints; + struct addrinfo *servinfo; - memset(&rtp_client_control_socket,0,sizeof(rtp_client_control_socket)); - memset(&hints,0, sizeof hints); - hints.ai_family = remote->SAFAMILY; - hints.ai_socktype = SOCK_DGRAM; - snprintf(portstr, 20, "%d", cport); - if (getaddrinfo(client_ip_string,portstr,&hints,&servinfo)!=0) - die("Can't get address of client's control port"); + memset(&rtp_client_control_socket, 0, sizeof(rtp_client_control_socket)); + memset(&hints, 0, sizeof hints); + hints.ai_family = remote->SAFAMILY; + hints.ai_socktype = SOCK_DGRAM; + snprintf(portstr, 20, "%d", cport); + if (getaddrinfo(client_ip_string, portstr, &hints, &servinfo) != 0) + die("Can't get address of client's control port"); #ifdef AF_INET6 - if (servinfo->ai_family == AF_INET6) - memcpy(&rtp_client_control_socket,servinfo->ai_addr,sizeof(struct sockaddr_in6)); - else + if (servinfo->ai_family == AF_INET6) + memcpy(&rtp_client_control_socket, servinfo->ai_addr, sizeof(struct sockaddr_in6)); + else #endif - memcpy(&rtp_client_control_socket,servinfo->ai_addr,sizeof(struct sockaddr_in)); - freeaddrinfo(servinfo); - - // set up a the record of the remote's timing socket - memset(&rtp_client_timing_socket,0,sizeof(rtp_client_timing_socket)); - memset(&hints,0, sizeof hints); - hints.ai_family = remote->SAFAMILY; - hints.ai_socktype = SOCK_DGRAM; - snprintf(portstr, 20, "%d", tport); - if (getaddrinfo(client_ip_string,portstr,&hints,&servinfo)!=0) - die("Can't get address of client's timing port"); + memcpy(&rtp_client_control_socket, servinfo->ai_addr, sizeof(struct sockaddr_in)); + freeaddrinfo(servinfo); + + // set up a the record of the remote's timing socket + memset(&rtp_client_timing_socket, 0, sizeof(rtp_client_timing_socket)); + memset(&hints, 0, sizeof hints); + hints.ai_family = remote->SAFAMILY; + hints.ai_socktype = SOCK_DGRAM; + snprintf(portstr, 20, "%d", tport); + if (getaddrinfo(client_ip_string, portstr, &hints, &servinfo) != 0) + die("Can't get address of client's timing port"); #ifdef AF_INET6 - if (servinfo->ai_family == AF_INET6) - memcpy(&rtp_client_timing_socket,servinfo->ai_addr,sizeof(struct sockaddr_in6)); - else + if (servinfo->ai_family == AF_INET6) + memcpy(&rtp_client_timing_socket, servinfo->ai_addr, sizeof(struct sockaddr_in6)); + else #endif - memcpy(&rtp_client_timing_socket,servinfo->ai_addr,sizeof(struct sockaddr_in)); - freeaddrinfo(servinfo); - - // now, we open three sockets -- one for the audio stream, one for the timing and one for the control - - *lsport = bind_port(remote,&audio_socket,0); - *lcport = bind_port(remote,&control_socket,0); - *ltport = bind_port(remote,&timing_socket,0); + memcpy(&rtp_client_timing_socket, servinfo->ai_addr, sizeof(struct sockaddr_in)); + freeaddrinfo(servinfo); - debug(2, "listening for audio, control and timing on ports %d, %d, %d.", *lsport, *lcport, *ltport); + // now, we open three sockets -- one for the audio stream, one for the timing and one for the + // control - please_shutdown = 0; - reference_timestamp=0; - pthread_create(&rtp_audio_thread, NULL, &rtp_audio_receiver, NULL); - pthread_create(&rtp_control_thread, NULL, &rtp_control_receiver, NULL); - pthread_create(&rtp_timing_thread, NULL, &rtp_timing_receiver, NULL); + *lsport = bind_port(remote, &audio_socket); + *lcport = bind_port(remote, &control_socket); + *ltport = bind_port(remote, &timing_socket); - running = 1; - request_sent=0; + debug(2, "listening for audio, control and timing on ports %d, %d, %d.", *lsport, *lcport, + *ltport); + + please_shutdown = 0; + reference_timestamp = 0; + pthread_create(&rtp_audio_thread, NULL, &rtp_audio_receiver, NULL); + pthread_create(&rtp_control_thread, NULL, &rtp_control_receiver, NULL); + pthread_create(&rtp_timing_thread, NULL, &rtp_timing_receiver, NULL); + + running = 1; + request_sent = 0; } -void get_reference_timestamp_stuff(uint32_t *timestamp,uint64_t *timestamp_time) { +void get_reference_timestamp_stuff(uint32_t *timestamp, uint64_t *timestamp_time, uint64_t *remote_timestamp_time) { pthread_mutex_lock(&reference_time_mutex); - *timestamp=reference_timestamp; - *timestamp_time = reference_timestamp_time; + *timestamp = reference_timestamp; + *timestamp_time = reference_timestamp_time; + *remote_timestamp_time = remote_reference_timestamp_time; pthread_mutex_unlock(&reference_time_mutex); } void clear_reference_timestamp(void) { pthread_mutex_lock(&reference_time_mutex); - reference_timestamp=0; - reference_timestamp_time=0; + reference_timestamp = 0; + reference_timestamp_time = 0; pthread_mutex_unlock(&reference_time_mutex); } void rtp_shutdown(void) { - if (!running) - die("rtp_shutdown called without active stream!"); + if (!running) + die("rtp_shutdown called without active stream!"); - debug(2, "shutting down RTP thread"); - please_shutdown = 1; - void *retval; - reference_timestamp=0; - pthread_kill(rtp_audio_thread, SIGUSR1); - pthread_join(rtp_audio_thread, &retval); - pthread_kill(rtp_control_thread, SIGUSR1); - pthread_join(rtp_control_thread, &retval); - pthread_kill(rtp_timing_thread, SIGUSR1); - pthread_join(rtp_timing_thread, &retval); - running = 0; + debug(2, "shutting down RTP thread"); + please_shutdown = 1; + void *retval; + reference_timestamp = 0; + pthread_kill(rtp_audio_thread, SIGUSR1); + pthread_join(rtp_audio_thread, &retval); + pthread_kill(rtp_control_thread, SIGUSR1); + pthread_join(rtp_control_thread, &retval); + pthread_kill(rtp_timing_thread, SIGUSR1); + pthread_join(rtp_timing_thread, &retval); + running = 0; } void rtp_request_resend(seq_t first, uint32_t count) { - if (running) { - if (!request_sent) { - debug(2, "requesting resend on %d packets starting at %u.",count,first); - request_sent=1; - } - - char req[8]; // *not* a standard RTCP NACK - req[0] = 0x80; - req[1] = 0x55|0x80; // Apple 'resend' - *(unsigned short *)(req+2) = htons(1); // our seqnum - *(unsigned short *)(req+4) = htons(first); // missed seqnum - *(unsigned short *)(req+6) = htons(count); // count - socklen_t msgsize = sizeof(struct sockaddr_in); -#ifdef AF_INET6 - if (rtp_client_timing_socket.SAFAMILY==AF_INET6) { - msgsize = sizeof(struct sockaddr_in6); - } -#endif - if (sendto(audio_socket, req, sizeof(req), 0, (struct sockaddr*)&rtp_client_control_socket,msgsize)==-1) { - perror("Error sendto-ing to audio socket"); - } - } else { - if (!request_sent) { - debug(2,"rtp_request_resend called without active stream!"); - request_sent=1; - } + if (running) { + if (!request_sent) { + debug(2, "requesting resend on %d packets starting at %u.", count, first); + request_sent = 1; } + + char req[8]; // *not* a standard RTCP NACK + req[0] = 0x80; + req[1] = 0x55 | 0x80; // Apple 'resend' + *(unsigned short *)(req + 2) = htons(1); // our seqnum + *(unsigned short *)(req + 4) = htons(first); // missed seqnum + *(unsigned short *)(req + 6) = htons(count); // count + socklen_t msgsize = sizeof(struct sockaddr_in); +#ifdef AF_INET6 + if (rtp_client_timing_socket.SAFAMILY == AF_INET6) { + msgsize = sizeof(struct sockaddr_in6); + } +#endif + if (sendto(audio_socket, req, sizeof(req), 0, (struct sockaddr *)&rtp_client_control_socket, + msgsize) == -1) { + perror("Error sendto-ing to audio socket"); + } + } else { + if (!request_sent) { + debug(2, "rtp_request_resend called without active stream!"); + request_sent = 1; + } + } } void rtp_request_client_pause() { if (running) { - if (client_active_remote==0) { - debug(1,"Can't request a client pause: no valid active remote."); - } else { - // debug(1,"Send a client pause request to %s:3689 with active remote %u.",client_ip_string,client_active_remote); - - - struct addrinfo hints, *res; - int sockfd; - - char message[1000] , server_reply[2000]; + if (client_active_remote == 0) { + debug(1, "Can't request a client pause: no valid active remote."); + } else { + // debug(1,"Send a client pause request to %s:3689 with active remote + // %u.",client_ip_string,client_active_remote); - // first, load up address structs with getaddrinfo(): + struct addrinfo hints, *res; + int sockfd; - memset(&hints, 0, sizeof hints); - hints.ai_family = AF_UNSPEC; - hints.ai_socktype = SOCK_STREAM; + char message[1000], server_reply[2000]; - getaddrinfo(client_ip_string, "3689", &hints, &res); + // first, load up address structs with getaddrinfo(): - // make a socket: + memset(&hints, 0, sizeof hints); + hints.ai_family = AF_UNSPEC; + hints.ai_socktype = SOCK_STREAM; - sockfd = socket(res->ai_family, res->ai_socktype, res->ai_protocol); + getaddrinfo(client_ip_string, "3689", &hints, &res); - if (sockfd == -1) { - die("Could not create socket"); - } - // debug(1,"Socket created"); - - // connect! + // make a socket: - if (connect(sockfd, res->ai_addr, res->ai_addrlen) < 0) { - die("connect failed. Error"); - } - // debug(1,"Connect successful"); - - sprintf(message,"GET /ctrl-int/1/pause HTTP/1.1\r\nHost: %s:3689\r\nActive-Remote: %u\r\n\r\n",client_ip_string,client_active_remote); - // debug(1,"Sending this message: \"%s\".",message); - - //Send some data - if( send(sockfd , message , strlen(message) , 0) < 0) { - debug(1,"Send failed"); - } - - //Receive a reply from the server - if( recv(sockfd , server_reply , 2000 , 0) < 0) { - debug(1,"recv failed"); - } - - // debug(1,"Server replied: \"%s\".",server_reply); - - if (strstr(server_reply,"HTTP/1.1 204 No Content")!=server_reply) - debug(1,"Client pause request failed."); - // debug(1,"Client pause request failed: \"%s\".",server_reply); - close(sockfd); + sockfd = socket(res->ai_family, res->ai_socktype, res->ai_protocol); + + if (sockfd == -1) { + die("Could not create socket"); + } + // debug(1,"Socket created"); + + // connect! + + if (connect(sockfd, res->ai_addr, res->ai_addrlen) < 0) { + die("connect failed. Error"); + } + // debug(1,"Connect successful"); + + sprintf(message, + "GET /ctrl-int/1/pause HTTP/1.1\r\nHost: %s:3689\r\nActive-Remote: %u\r\n\r\n", + client_ip_string, client_active_remote); + // debug(1,"Sending this message: \"%s\".",message); + + // Send some data + if (send(sockfd, message, strlen(message), 0) < 0) { + debug(1, "Send failed"); + } + + // Receive a reply from the server + if (recv(sockfd, server_reply, 2000, 0) < 0) { + debug(1, "recv failed"); + } + + // debug(1,"Server replied: \"%s\".",server_reply); + + if (strstr(server_reply, "HTTP/1.1 204 No Content") != server_reply) + debug(1, "Client pause request failed."); + // debug(1,"Client pause request failed: \"%s\".",server_reply); + close(sockfd); } } else { - debug(1,"Request to pause non-existent play stream -- ignored."); + debug(1, "Request to pause non-existent play stream -- ignored."); } } diff --git a/rtp.h b/rtp.h index 5b679a60..8045d2bb 100644 --- a/rtp.h +++ b/rtp.h @@ -5,13 +5,14 @@ #include "player.h" -void rtp_setup(SOCKADDR *remote, int controlport, int timingport, uint32_t active_remote, int *local_server_port, int *local_control_port, int *local_timing_port); +void rtp_setup(SOCKADDR *remote, int controlport, int timingport, uint32_t active_remote, + int *local_server_port, int *local_control_port, int *local_timing_port); void rtp_shutdown(void); void rtp_request_resend(seq_t first, uint32_t count); void rtp_request_client_pause(void); // ask the client to pause -void get_reference_timestamp_stuff(uint32_t *timestamp,uint64_t *timestamp_time); -void clear_reference_timestamp(void); +void get_reference_timestamp_stuff(uint32_t *timestamp, uint64_t *timestamp_time, uint64_t *remote_timestamp_time); +void clear_reference_timestamp(void); uint64_t static local_to_remote_time_jitters; uint64_t static local_to_remote_time_jitters_count; diff --git a/rtsp.c b/rtsp.c index 379662a2..9340dbfb 100644 --- a/rtsp.c +++ b/rtsp.c @@ -40,6 +40,8 @@ #include #include #include +#include + #include "config.h" @@ -76,7 +78,6 @@ static pthread_mutex_t play_lock = PTHREAD_MUTEX_INITIALIZER; // if a reference count is read as zero, it means the it's being deallocated. static pthread_mutex_t reference_counter_lock = PTHREAD_MUTEX_INITIALIZER; - // only one thread is allowed to use the player at once. // it monitors the request variable (at least when interrupted) static pthread_mutex_t playing_mutex = PTHREAD_MUTEX_INITIALIZER; @@ -84,11 +85,11 @@ static int please_shutdown = 0; static pthread_t playing_thread = 0; typedef struct { - int fd; - stream_cfg stream; - SOCKADDR remote; - int running; - pthread_t thread; + int fd; + stream_cfg stream; + SOCKADDR remote; + int running; + pthread_t thread; } rtsp_conn_info; #ifdef CONFIG_METADATA @@ -96,17 +97,17 @@ typedef struct { pthread_mutex_t pc_queue_lock; pthread_cond_t pc_queue_item_added_signal; pthread_cond_t pc_queue_item_removed_signal; - size_t item_size; // number of bytes in each item - uint32_t count; // number of items in the queue + size_t item_size; // number of bytes in each item + uint32_t count; // number of items in the queue uint32_t capacity; // maximum number of items - uint32_t toq; // first item to take - uint32_t eoq; // free space at end of queue - void *items; // a pointer to where the items are actually stored - } pc_queue; // producer-consumer queue + uint32_t toq; // first item to take + uint32_t eoq; // free space at end of queue + void *items; // a pointer to where the items are actually stored +} pc_queue; // producer-consumer queue #endif typedef struct { - uint32_t referenceCount; // we might start using this... + uint32_t referenceCount; // we might start using this... int nheaders; char *name[16]; char *value[16]; @@ -127,10 +128,10 @@ typedef struct { uint32_t code; char *data; uint32_t length; - rtsp_message* carrier; + rtsp_message *carrier; } metadata_package; -void pc_queue_init(pc_queue* the_queue, char* items, size_t item_size, uint32_t number_of_items) { +void pc_queue_init(pc_queue *the_queue, char *items, size_t item_size, uint32_t number_of_items) { the_queue->item_size = item_size; the_queue->items = items; the_queue->count = 0; @@ -139,85 +140,86 @@ void pc_queue_init(pc_queue* the_queue, char* items, size_t item_size, uint32_t the_queue->eoq = 0; } -int send_metadata(uint32_t type,uint32_t code,char *data,uint32_t length,rtsp_message* carrier,int block); +int send_metadata(uint32_t type, uint32_t code, char *data, uint32_t length, rtsp_message *carrier, + int block); -int send_ssnc_metadata(uint32_t code,char *data,uint32_t length,int block) { - return send_metadata('ssnc',code,data,length,NULL,block); +int send_ssnc_metadata(uint32_t code, char *data, uint32_t length, int block) { + return send_metadata('ssnc', code, data, length, NULL, block); } -int pc_queue_add_item(pc_queue* the_queue,const void* the_stuff, int block) { +int pc_queue_add_item(pc_queue *the_queue, const void *the_stuff, int block) { int rc; if (the_queue) { - if (block==0) { + if (block == 0) { rc = pthread_mutex_trylock(&the_queue->pc_queue_lock); - if (rc==EBUSY) + if (rc == EBUSY) return EBUSY; } else rc = pthread_mutex_lock(&the_queue->pc_queue_lock); if (rc) - debug(1,"Error locking for pc_queue_add_item"); - while(the_queue->count==the_queue->capacity) { - rc = pthread_cond_wait(&the_queue->pc_queue_item_removed_signal,&the_queue->pc_queue_lock); + debug(1, "Error locking for pc_queue_add_item"); + while (the_queue->count == the_queue->capacity) { + rc = pthread_cond_wait(&the_queue->pc_queue_item_removed_signal, &the_queue->pc_queue_lock); if (rc) - debug(1,"Error waiting for item to be removed"); + debug(1, "Error waiting for item to be removed"); } uint32_t i = the_queue->eoq; - void * p = the_queue->items + the_queue->item_size*i; -// void * p = &the_queue->qbase + the_queue->item_size*the_queue->eoq; - memcpy(p,the_stuff,the_queue->item_size); - + void *p = the_queue->items + the_queue->item_size * i; + // void * p = &the_queue->qbase + the_queue->item_size*the_queue->eoq; + memcpy(p, the_stuff, the_queue->item_size); + // update the pointer i++; - if (i==the_queue->capacity) + if (i == the_queue->capacity) // fold pointer if necessary - i=0; + i = 0; the_queue->eoq = i; the_queue->count++; - if (the_queue->count==the_queue->capacity) - debug(1,"pc_queue is full!"); + if (the_queue->count == the_queue->capacity) + debug(1, "pc_queue is full!"); rc = pthread_cond_signal(&the_queue->pc_queue_item_added_signal); if (rc) - debug(1,"Error signalling after pc_queue_add_item"); + debug(1, "Error signalling after pc_queue_add_item"); rc = pthread_mutex_unlock(&the_queue->pc_queue_lock); if (rc) - debug(1,"Error unlocking for pc_queue_add_item"); + debug(1, "Error unlocking for pc_queue_add_item"); } else { - debug(1,"Adding an item to a NULL queue"); + debug(1, "Adding an item to a NULL queue"); } return 0; } -int pc_queue_get_item(pc_queue* the_queue,void* the_stuff) { +int pc_queue_get_item(pc_queue *the_queue, void *the_stuff) { int rc; if (the_queue) { rc = pthread_mutex_lock(&the_queue->pc_queue_lock); if (rc) - debug(1,"Error locking for pc_queue_get_item"); - while(the_queue->count==0) { - rc = pthread_cond_wait(&the_queue->pc_queue_item_added_signal,&the_queue->pc_queue_lock); + debug(1, "Error locking for pc_queue_get_item"); + while (the_queue->count == 0) { + rc = pthread_cond_wait(&the_queue->pc_queue_item_added_signal, &the_queue->pc_queue_lock); if (rc) - debug(1,"Error waiting for item to be added"); + debug(1, "Error waiting for item to be added"); } uint32_t i = the_queue->toq; -// void * p = &the_queue->qbase + the_queue->item_size*the_queue->toq; - void * p = the_queue->items + the_queue->item_size*i; - memcpy(the_stuff,p,the_queue->item_size); - + // void * p = &the_queue->qbase + the_queue->item_size*the_queue->toq; + void *p = the_queue->items + the_queue->item_size * i; + memcpy(the_stuff, p, the_queue->item_size); + // update the pointer i++; - if (i==the_queue->capacity) + if (i == the_queue->capacity) // fold pointer if necessary - i=0; + i = 0; the_queue->toq = i; the_queue->count--; rc = pthread_cond_signal(&the_queue->pc_queue_item_removed_signal); if (rc) - debug(1,"Error signalling after pc_queue_removed_item"); + debug(1, "Error signalling after pc_queue_removed_item"); rc = pthread_mutex_unlock(&the_queue->pc_queue_lock); if (rc) - debug(1,"Error unlocking for pc_queue_get_item"); + debug(1, "Error unlocking for pc_queue_get_item"); } else { - debug(1,"Removing an item from a NULL queue"); + debug(1, "Removing an item from a NULL queue"); } return 0; } @@ -226,12 +228,12 @@ int pc_queue_get_item(pc_queue* the_queue,void* the_stuff) { // determine if we are the currently playing thread static inline int rtsp_playing(void) { - if (pthread_mutex_trylock(&playing_mutex)) { - return pthread_equal(playing_thread, pthread_self()); - } else { - pthread_mutex_unlock(&playing_mutex); - return 0; - } + if (pthread_mutex_trylock(&playing_mutex)) { + return pthread_equal(playing_thread, pthread_self()); + } else { + pthread_mutex_unlock(&playing_mutex); + return 0; + } } void rtsp_request_shutdown_stream(void) { @@ -239,561 +241,589 @@ void rtsp_request_shutdown_stream(void) { pthread_kill(playing_thread, SIGUSR1); } - static void rtsp_take_player(void) { - if (rtsp_playing()) - return; + if (rtsp_playing()) + return; - if (pthread_mutex_trylock(&playing_mutex)) { - debug(1, "shutting down playing thread."); - // XXX minor race condition between please_shutdown and signal delivery - please_shutdown = 1; - pthread_kill(playing_thread, SIGUSR1); - pthread_mutex_lock(&playing_mutex); - } - playing_thread = pthread_self(); // make us the currently-playing thread (why?) + if (pthread_mutex_trylock(&playing_mutex)) { + debug(1, "shutting down playing thread."); + // XXX minor race condition between please_shutdown and signal delivery + please_shutdown = 1; + pthread_kill(playing_thread, SIGUSR1); + pthread_mutex_lock(&playing_mutex); + } + playing_thread = pthread_self(); // make us the currently-playing thread (why?) } void rtsp_shutdown_stream(void) { - rtsp_take_player(); - pthread_mutex_unlock(&playing_mutex); + rtsp_take_player(); + pthread_mutex_unlock(&playing_mutex); } // keep track of the threads we have spawned so we can join() them static rtsp_conn_info **conns = NULL; static int nconns = 0; static void track_thread(rtsp_conn_info *conn) { - conns = realloc(conns, sizeof(rtsp_conn_info*) * (nconns + 1)); - conns[nconns] = conn; - nconns++; + conns = realloc(conns, sizeof(rtsp_conn_info *) * (nconns + 1)); + conns[nconns] = conn; + nconns++; } static void cleanup_threads(void) { - void *retval; - int i; - debug(2, "culling threads."); - for (i=0; irunning == 0) { - pthread_join(conns[i]->thread, &retval); - free(conns[i]); - debug(2, "one joined..."); - nconns--; - if (nconns) - conns[i] = conns[nconns]; - } else { - i++; - } + void *retval; + int i; + debug(2, "culling threads."); + for (i = 0; i < nconns;) { + if (conns[i]->running == 0) { + pthread_join(conns[i]->thread, &retval); + free(conns[i]); + debug(2, "one joined..."); + nconns--; + if (nconns) + conns[i] = conns[nconns]; + } else { + i++; } + } } // park a null at the line ending, and return the next line pointer // accept \r, \n, or \r\n static char *nextline(char *in, int inbuf) { - char *out = NULL; - while (inbuf) { - if (*in == '\r') { - *in++ = 0; - out = in; - } - if (*in == '\n') { - *in++ = 0; - out = in; - } - - if (out) - break; - - in++; - inbuf--; + char *out = NULL; + while (inbuf) { + if (*in == '\r') { + *in++ = 0; + out = in; } - return out; + if (*in == '\n') { + *in++ = 0; + out = in; + } + + if (out) + break; + + in++; + inbuf--; + } + return out; } -static void msg_retain(rtsp_message * msg) { +static void msg_retain(rtsp_message *msg) { if (msg) { - int rc = pthread_mutex_lock(&reference_counter_lock); - if (rc) - debug(1,"Error %d locking reference counter lock"); - msg->referenceCount++; - rc = pthread_mutex_unlock(&reference_counter_lock); - if (rc) - debug(1,"Error %d unlocking reference counter lock"); + int rc = pthread_mutex_lock(&reference_counter_lock); + if (rc) + debug(1, "Error %d locking reference counter lock"); + msg->referenceCount++; + rc = pthread_mutex_unlock(&reference_counter_lock); + if (rc) + debug(1, "Error %d unlocking reference counter lock"); } else { - debug(1,"null rtsp_message pointer passed to retain"); + debug(1, "null rtsp_message pointer passed to retain"); } } -static rtsp_message * msg_init(void) { - rtsp_message *msg = malloc(sizeof(rtsp_message)); - memset(msg, 0, sizeof(rtsp_message)); - msg->referenceCount = 1; // from now on, any access to this must be protected with the lock - return msg; +static rtsp_message *msg_init(void) { + rtsp_message *msg = malloc(sizeof(rtsp_message)); + memset(msg, 0, sizeof(rtsp_message)); + msg->referenceCount = 1; // from now on, any access to this must be protected with the lock + return msg; } static int msg_add_header(rtsp_message *msg, char *name, char *value) { - if (msg->nheaders >= sizeof(msg->name)/sizeof(char*)) { - warn("too many headers?!"); - return 1; - } + if (msg->nheaders >= sizeof(msg->name) / sizeof(char *)) { + warn("too many headers?!"); + return 1; + } - msg->name[msg->nheaders] = strdup(name); - msg->value[msg->nheaders] = strdup(value); - msg->nheaders++; + msg->name[msg->nheaders] = strdup(name); + msg->value[msg->nheaders] = strdup(value); + msg->nheaders++; - return 0; + return 0; } static char *msg_get_header(rtsp_message *msg, char *name) { - int i; - for (i=0; inheaders; i++) - if (!strcasecmp(msg->name[i], name)) - return msg->value[i]; - return NULL; + int i; + for (i = 0; i < msg->nheaders; i++) + if (!strcasecmp(msg->name[i], name)) + return msg->value[i]; + return NULL; } static void msg_print_debug_headers(rtsp_message *msg) { int i; - for (i=0; inheaders; i++) { - debug(1," Type: \"%s\", content: \"%s\"",msg->name[i],msg->value[i]); + for (i = 0; i < msg->nheaders; i++) { + debug(1, " Type: \"%s\", content: \"%s\"", msg->name[i], msg->value[i]); } } static void msg_free(rtsp_message *msg) { - + if (msg) { int rc = pthread_mutex_lock(&reference_counter_lock); if (rc) - debug(1,"Error %d locking reference counter lock during msg_free()",rc); + debug(1, "Error %d locking reference counter lock during msg_free()", rc); msg->referenceCount--; rc = pthread_mutex_unlock(&reference_counter_lock); if (rc) - debug(1,"Error %d unlocking reference counter lock during msg_free()",rc); - if (msg->referenceCount==0) { + debug(1, "Error %d unlocking reference counter lock during msg_free()", rc); + if (msg->referenceCount == 0) { int i; - for (i=0; inheaders; i++) { - free(msg->name[i]); - free(msg->value[i]); + for (i = 0; i < msg->nheaders; i++) { + free(msg->name[i]); + free(msg->value[i]); } if (msg->content) free(msg->content); free(msg); } // else { // debug(1,"rtsp_message reference count non-zero: %d!",msg->referenceCount); - //} + //} } else { - debug(1,"null rtsp_message pointer passed to msg_free()"); - } + debug(1, "null rtsp_message pointer passed to msg_free()"); + } } - static int msg_handle_line(rtsp_message **pmsg, char *line) { - rtsp_message *msg = *pmsg; + rtsp_message *msg = *pmsg; - if (!msg) { - msg = msg_init(); - *pmsg = msg; - char *sp, *p; + if (!msg) { + msg = msg_init(); + *pmsg = msg; + char *sp, *p; - // debug(1, "received request: %s", line); + // debug(1, "received request: %s", line); - p = strtok_r(line, " ", &sp); - if (!p) - goto fail; - strncpy(msg->method, p, sizeof(msg->method)-1); + p = strtok_r(line, " ", &sp); + if (!p) + goto fail; + strncpy(msg->method, p, sizeof(msg->method) - 1); - p = strtok_r(NULL, " ", &sp); - if (!p) - goto fail; + p = strtok_r(NULL, " ", &sp); + if (!p) + goto fail; - p = strtok_r(NULL, " ", &sp); - if (!p) - goto fail; - if (strcmp(p, "RTSP/1.0")) - goto fail; + p = strtok_r(NULL, " ", &sp); + if (!p) + goto fail; + if (strcmp(p, "RTSP/1.0")) + goto fail; - return -1; - } + return -1; + } - if (strlen(line)) { - char *p; - p = strstr(line, ": "); - if (!p) { - warn("bad header: >>%s<<", line); - goto fail; - } - *p = 0; - p += 2; - msg_add_header(msg, line, p); - debug(2, " %s: %s.", line, p); - return -1; - } else { - char *cl = msg_get_header(msg, "Content-Length"); - if (cl) - return atoi(cl); - else - return 0; + if (strlen(line)) { + char *p; + p = strstr(line, ": "); + if (!p) { + warn("bad header: >>%s<<", line); + goto fail; } + *p = 0; + p += 2; + msg_add_header(msg, line, p); + debug(2, " %s: %s.", line, p); + return -1; + } else { + char *cl = msg_get_header(msg, "Content-Length"); + if (cl) + return atoi(cl); + else + return 0; + } fail: - *pmsg = NULL; - msg_free(msg); - return 0; + *pmsg = NULL; + msg_free(msg); + return 0; } -static enum rtsp_read_request_response rtsp_read_request(int fd, rtsp_message** the_packet) { - enum rtsp_read_request_response reply=rtsp_read_request_response_ok; - ssize_t buflen = 512; - char *buf = malloc(buflen+1); +static enum rtsp_read_request_response rtsp_read_request(int fd, rtsp_message **the_packet) { + enum rtsp_read_request_response reply = rtsp_read_request_response_ok; + ssize_t buflen = 512; + char *buf = malloc(buflen + 1); - rtsp_message *msg = NULL; + rtsp_message *msg = NULL; - ssize_t nread; - ssize_t inbuf = 0; - int msg_size = -1; - - while (msg_size < 0) { - if (please_shutdown) { - debug(1, "RTSP shutdown requested."); - reply = rtsp_read_request_response_shutdown_requested; - goto shutdown; - } - nread = read(fd, buf+inbuf, buflen - inbuf); - if (!nread) { - debug(1, "RTSP connection closed."); - reply = rtsp_read_request_response_shutdown_requested; - goto shutdown; - } - if (nread < 0) { - if (errno==EINTR) - continue; - perror("read failure"); - reply = rtsp_read_request_response_error; - goto shutdown; - } - inbuf += nread; + ssize_t nread; + ssize_t inbuf = 0; + int msg_size = -1; - char *next; - while (msg_size < 0 && (next = nextline(buf, inbuf))) { - msg_size = msg_handle_line(&msg, buf); - - if (!msg) { - warn("no RTSP header received"); - reply = rtsp_read_request_response_bad_packet; - goto shutdown; - } - - inbuf -= next-buf; - if (inbuf) - memmove(buf, next, inbuf); - } + while (msg_size < 0) { + if (please_shutdown) { + debug(1, "RTSP shutdown requested."); + reply = rtsp_read_request_response_shutdown_requested; + goto shutdown; } - - if (msg_size > buflen) { - buf = realloc(buf, msg_size); - if (!buf) { - warn("too much content"); - reply = rtsp_read_request_response_error; - goto shutdown; - } - buflen = msg_size; + nread = read(fd, buf + inbuf, buflen - inbuf); + if (!nread) { + debug(1, "RTSP connection closed."); + reply = rtsp_read_request_response_shutdown_requested; + goto shutdown; } - - uint64_t threshold_time = get_absolute_time_in_fp() + ((uint64_t)5<<32); // i.e. five seconds from now - int warning_message_sent = 0; - - const size_t max_read_chunk = 50000; - while (inbuf < msg_size) { - - // we are going to read the stream in chunks and time how long it takes to do so. - // If it's taking too long, (and we find out about it), we will send an error message as metadata - - if (warning_message_sent==0) { - uint64_t time_now = get_absolute_time_in_fp(); - if (time_now>threshold_time) { // it's taking too long - debug(1,"Error receiving metadata from source -- transmission seems to be stalled."); -#ifdef CONFIG_METADATA - send_ssnc_metadata('stal',NULL,0,1); -#endif - warning_message_sent = 1; - } - } - ssize_t read_chunk = msg_size-inbuf; - if (read_chunk > max_read_chunk) - read_chunk = max_read_chunk; - nread = read(fd, buf+inbuf, read_chunk); - if (!nread) { - reply = rtsp_read_request_response_error; - goto shutdown; - } - if (nread==EINTR) + if (nread < 0) { + if (errno == EINTR) continue; - if (nread < 0) { - perror("read failure"); - reply = rtsp_read_request_response_error; + perror("read failure"); + reply = rtsp_read_request_response_error; + goto shutdown; + } + inbuf += nread; + + char *next; + while (msg_size < 0 && (next = nextline(buf, inbuf))) { + msg_size = msg_handle_line(&msg, buf); + + if (!msg) { + warn("no RTSP header received"); + reply = rtsp_read_request_response_bad_packet; goto shutdown; } - inbuf += nread; - } - msg->contentlength = inbuf; - msg->content = buf; - *the_packet = msg; - return reply; + inbuf -= next - buf; + if (inbuf) + memmove(buf, next, inbuf); + } + } + + if (msg_size > buflen) { + buf = realloc(buf, msg_size); + if (!buf) { + warn("too much content"); + reply = rtsp_read_request_response_error; + goto shutdown; + } + buflen = msg_size; + } + + uint64_t threshold_time = + get_absolute_time_in_fp() + ((uint64_t)5 << 32); // i.e. five seconds from now + int warning_message_sent = 0; + + const size_t max_read_chunk = 50000; + while (inbuf < msg_size) { + + // we are going to read the stream in chunks and time how long it takes to do so. + // If it's taking too long, (and we find out about it), we will send an error message as + // metadata + + if (warning_message_sent == 0) { + uint64_t time_now = get_absolute_time_in_fp(); + if (time_now > threshold_time) { // it's taking too long + debug(1, "Error receiving metadata from source -- transmission seems to be stalled."); +#ifdef CONFIG_METADATA + send_ssnc_metadata('stal', NULL, 0, 1); +#endif + warning_message_sent = 1; + } + } + ssize_t read_chunk = msg_size - inbuf; + if (read_chunk > max_read_chunk) + read_chunk = max_read_chunk; + nread = read(fd, buf + inbuf, read_chunk); + if (!nread) { + reply = rtsp_read_request_response_error; + goto shutdown; + } + if (nread == EINTR) + continue; + if (nread < 0) { + perror("read failure"); + reply = rtsp_read_request_response_error; + goto shutdown; + } + inbuf += nread; + } + + msg->contentlength = inbuf; + msg->content = buf; + *the_packet = msg; + return reply; shutdown: - if (msg) { - msg_free(msg); // which will free the content and everything else - } - // in case the message wasn't formed or wasn't fully initialised - if ((msg) && (msg-> content == NULL) || (!msg)) - free(buf); - *the_packet = NULL; - return reply; + if (msg) { + msg_free(msg); // which will free the content and everything else + } + // in case the message wasn't formed or wasn't fully initialised + if ((msg) && (msg->content == NULL) || (!msg)) + free(buf); + *the_packet = NULL; + return reply; } static void msg_write_response(int fd, rtsp_message *resp) { - char pkt[1024]; - int pktfree = sizeof(pkt); - char *p = pkt; - int i, n; + char pkt[1024]; + int pktfree = sizeof(pkt); + char *p = pkt; + int i, n; - n = snprintf(p, pktfree, - "RTSP/1.0 %d %s\r\n", resp->respcode, - resp->respcode==200 ? "OK" : "Error"); - // debug(1, "sending response: %s", pkt); + n = snprintf(p, pktfree, "RTSP/1.0 %d %s\r\n", resp->respcode, + resp->respcode == 200 ? "OK" : "Error"); + // debug(1, "sending response: %s", pkt); + pktfree -= n; + p += n; + + for (i = 0; i < resp->nheaders; i++) { + debug(2, " %s: %s.", resp->name[i], resp->value[i]); + n = snprintf(p, pktfree, "%s: %s\r\n", resp->name[i], resp->value[i]); pktfree -= n; p += n; + if (pktfree <= 0) + die("Attempted to write overlong RTSP packet"); + } - for (i=0; inheaders; i++) { - debug(2, " %s: %s.", resp->name[i], resp->value[i]); - n = snprintf(p, pktfree, "%s: %s\r\n", resp->name[i], resp->value[i]); - pktfree -= n; - p += n; - if (pktfree <= 0) - die("Attempted to write overlong RTSP packet"); - } + if (pktfree < 3) + die("Attempted to write overlong RTSP packet"); - if (pktfree < 3) - die("Attempted to write overlong RTSP packet"); - - strcpy(p, "\r\n"); - int ignore = write(fd, pkt, p-pkt+2); + strcpy(p, "\r\n"); + int ignore = write(fd, pkt, p - pkt + 2); } -static void handle_record(rtsp_conn_info *conn, - rtsp_message *req, rtsp_message *resp) { +static void handle_record(rtsp_conn_info *conn, rtsp_message *req, rtsp_message *resp) { resp->respcode = 200; - msg_add_header(resp, "Audio-Latency","88200"); + msg_add_header(resp, "Audio-Latency", "88200"); } -static void handle_options(rtsp_conn_info *conn, - rtsp_message *req, rtsp_message *resp) { - resp->respcode = 200; - msg_add_header(resp, "Public", - "ANNOUNCE, SETUP, RECORD, " - "PAUSE, FLUSH, TEARDOWN, " - "OPTIONS, GET_PARAMETER, SET_PARAMETER"); +static void handle_options(rtsp_conn_info *conn, rtsp_message *req, rtsp_message *resp) { + resp->respcode = 200; + msg_add_header(resp, "Public", "ANNOUNCE, SETUP, RECORD, " + "PAUSE, FLUSH, TEARDOWN, " + "OPTIONS, GET_PARAMETER, SET_PARAMETER"); } -static void handle_teardown(rtsp_conn_info *conn, - rtsp_message *req, rtsp_message *resp) { - if (!rtsp_playing()) - return; - resp->respcode = 200; - msg_add_header(resp, "Connection", "close"); - please_shutdown = 1; +static void handle_teardown(rtsp_conn_info *conn, rtsp_message *req, rtsp_message *resp) { + if (!rtsp_playing()) + return; + resp->respcode = 200; + msg_add_header(resp, "Connection", "close"); + please_shutdown = 1; } -static void handle_flush(rtsp_conn_info *conn, - rtsp_message *req, rtsp_message *resp) { - if (!rtsp_playing()) - return; - char *p; - uint32_t rtptime=0; - char * hdr = msg_get_header(req,"RTP-Info"); - - if (hdr) { - // debug(1,"FLUSH message received: \"%s\".",hdr); - // get the rtp timestamp - p = strstr(hdr, "rtptime="); - if (p) { - p = strchr(p, '=') + 1; - if (p) - rtptime = uatoi(p); // unsigned integer -- up to 2^32-1 - } - } - // debug(1,"RTSP Flush Requested."); - player_flush(rtptime); - resp->respcode = 200; -} +static void handle_flush(rtsp_conn_info *conn, rtsp_message *req, rtsp_message *resp) { + if (!rtsp_playing()) + return; + char *p; + uint32_t rtptime = 0; + char *hdr = msg_get_header(req, "RTP-Info"); -static void handle_setup(rtsp_conn_info *conn, - rtsp_message *req, rtsp_message *resp) { - int cport, tport; - int lsport,lcport,ltport; - uint32_t active_remote=0; - - char * ar = msg_get_header(req,"Active-Remote"); - if (ar) { - // debug(1,"Active-Remote string seen: \"%s\".",ar); - // get the active remote - char *p; - active_remote = strtoul(ar,&p,10); - // debug(1,"Active Remote is %u.",active_remote); - } - - // select latency - // if iTunes V10 or later is detected, use the iTunes latency setting - // if AirPlay is detected, use the AirPlay latency setting - // for everything else, use the general latency setting, if given, or - // else use the default latency setting - - config.latency=88200; - - if (config.userSuppliedLatency) - config.latency=config.userSuppliedLatency; - - char * ua = msg_get_header(req,"User-Agent"); - if (ua==0) { - debug(1,"No User-Agent string found in the SETUP message. Using latency of %d frames.",config.latency); - } else { - if (strstr(ua,"iTunes")==ua) { - int iTunesVersion=0; - // now check it's version 10 or later - char *pp = strchr(ua,'/') + 1; - if (pp) - iTunesVersion=atoi(pp); - else - debug(2,"iTunes Version Number not found."); - if (iTunesVersion>=10) { - debug(2,"User-Agent is iTunes 10 or better, (actual version is %d); selecting the iTunes latency of %d frames.",iTunesVersion,config.iTunesLatency); - config.latency=config.iTunesLatency; - } - } else if (strstr(ua,"AirPlay")==ua) { - debug(2,"User-Agent is AirPlay; selecting the AirPlay latency of %d frames.",config.AirPlayLatency); - config.latency=config.AirPlayLatency; - } else if (strstr(ua,"forked-daapd")==ua) { - debug(2,"User-Agent is forked-daapd; selecting the forked-daapd latency of %d frames.",config.ForkedDaapdLatency); - config.latency=config.ForkedDaapdLatency; - } else { - debug(2,"Unrecognised User-Agent. Using latency of %d frames.",config.latency); - } - } - char *hdr = msg_get_header(req, "Transport"); - if (!hdr) - return; - - char *p; - p = strstr(hdr, "control_port="); - if (!p) - return; - p = strchr(p, '=') + 1; - cport = atoi(p); - - p = strstr(hdr, "timing_port="); - if (!p) - return; - p = strchr(p, '=') + 1; - tport = atoi(p); - - rtsp_take_player(); - rtp_setup(&conn->remote, cport, tport, active_remote, &lsport,&lcport,<port); - if (!lsport) - return; - char *q; - p = strstr(hdr,"control_port="); + if (hdr) { + // debug(1,"FLUSH message received: \"%s\".",hdr); + // get the rtp timestamp + p = strstr(hdr, "rtptime="); if (p) { - q = strchr(p,';'); // get past the control port entry - *p++=0; - if (q++) - strcat(hdr,q); // should unsplice the control port entry + p = strchr(p, '=') + 1; + if (p) + rtptime = uatoi(p); // unsigned integer -- up to 2^32-1 } - p = strstr(hdr,"timing_port="); - if (p) { - q = strchr(p,';'); // get past the timing port entry - *p++=0; - if (q++) - strcat(hdr,q); // should unsplice the timing port entry - } - - player_play(&conn->stream); - - char *resphdr = alloca(200); - *resphdr=0; - sprintf(resphdr, "RTP/AVP/UDP;unicast;interleaved=0-1;mode=record;control_port=%d;timing_port=%d;server_port=%d", lcport, ltport, lsport); - - msg_add_header(resp, "Transport", resphdr); - - msg_add_header(resp, "Session", "1"); - - resp->respcode = 200; + } + // debug(1,"RTSP Flush Requested: %u.",rtptime); + player_flush(rtptime); + resp->respcode = 200; } -static void handle_ignore(rtsp_conn_info *conn, - rtsp_message *req, rtsp_message *resp) { - resp->respcode = 200; -} +static void handle_setup(rtsp_conn_info *conn, rtsp_message *req, rtsp_message *resp) { + int cport, tport; + int lsport, lcport, ltport; + uint32_t active_remote = 0; -static void handle_set_parameter_parameter(rtsp_conn_info *conn, - rtsp_message *req, rtsp_message *resp) { - char *cp = req->content; - int cp_left = req->contentlength; - char *next; - while (cp_left && cp) { - next = nextline(cp, cp_left); - cp_left -= next-cp; + char *ar = msg_get_header(req, "Active-Remote"); + if (ar) { + debug(1,"Active-Remote string seen: \"%s\".",ar); + // get the active remote + char *p; + active_remote = strtoul(ar, &p, 10); +#ifdef CONFIG_METADATA + send_metadata('ssnc','acre',ar,strlen(ar),req,1); +#endif + } - if (!strncmp(cp, "volume: ", 8)) { - float volume = atof(cp + 8); - debug(2, "volume: %f\n", volume); - player_volume(volume); - } else -#ifdef CONFIG_METADATA - if(!strncmp(cp, "progress: ", 10)) { - char *progress = cp + 10; - debug(2, "progress: \"%s\"\n", progress); // rtpstampstart/rtpstampnow/rtpstampend 44100 per second - send_ssnc_metadata('prgr',strdup(progress),strlen(progress),1); - } else -#endif - { - debug(1, "unrecognised parameter: \"%s\" (%d)\n", cp, strlen(cp)); - } - cp = next; +#ifdef CONFIG_METADATA + ar = msg_get_header(req, "DACP-ID"); + if (ar) { + debug(1,"DACP-ID string seen: \"%s\".",ar); + send_metadata('ssnc','daid',ar,strlen(ar),req,1); + } +#endif + + // select latency + // if iTunes V10 or later is detected, use the iTunes latency setting + // if AirPlay is detected, use the AirPlay latency setting + // for everything else, use the general latency setting, if given, or + // else use the default latency setting + + config.latency = 88200; + + if (config.userSuppliedLatency) + config.latency = config.userSuppliedLatency; + + char *ua = msg_get_header(req, "User-Agent"); + if (ua == 0) { + debug(1, "No User-Agent string found in the SETUP message. Using latency of %d frames.", + config.latency); + } else { + if (strstr(ua, "iTunes") == ua) { + int iTunesVersion = 0; + // now check it's version 10 or later + char *pp = strchr(ua, '/') + 1; + if (pp) + iTunesVersion = atoi(pp); + else + debug(2, "iTunes Version Number not found."); + if (iTunesVersion >= 10) { + debug(2, "User-Agent is iTunes 10 or better, (actual version is %d); selecting the iTunes " + "latency of %d frames.", + iTunesVersion, config.iTunesLatency); + config.latency = config.iTunesLatency; + } + } else if (strstr(ua, "AirPlay") == ua) { + debug(2, "User-Agent is AirPlay; selecting the AirPlay latency of %d frames.", + config.AirPlayLatency); + config.latency = config.AirPlayLatency; + } else if (strstr(ua, "forked-daapd") == ua) { + debug(2, "User-Agent is forked-daapd; selecting the forked-daapd latency of %d frames.", + config.ForkedDaapdLatency); + config.latency = config.ForkedDaapdLatency; + } else { + debug(2, "Unrecognised User-Agent. Using latency of %d frames.", config.latency); } + } + char *hdr = msg_get_header(req, "Transport"); + if (!hdr) + goto error; + + char *p; + p = strstr(hdr, "control_port="); + if (!p) + goto error; + p = strchr(p, '=') + 1; + cport = atoi(p); + + p = strstr(hdr, "timing_port="); + if (!p) + goto error; + p = strchr(p, '=') + 1; + tport = atoi(p); + + rtsp_take_player(); + rtp_setup(&conn->remote, cport, tport, active_remote, &lsport, &lcport, <port); + if (!lsport) + goto error; + char *q; + p = strstr(hdr, "control_port="); + if (p) { + q = strchr(p, ';'); // get past the control port entry + *p++ = 0; + if (q++) + strcat(hdr, q); // should unsplice the control port entry + } + p = strstr(hdr, "timing_port="); + if (p) { + q = strchr(p, ';'); // get past the timing port entry + *p++ = 0; + if (q++) + strcat(hdr, q); // should unsplice the timing port entry + } + + player_play(&conn->stream); + + char *resphdr = alloca(200); + *resphdr = 0; + sprintf(resphdr, "RTP/AVP/" + "UDP;unicast;interleaved=0-1;mode=record;control_port=%d;timing_port=%d;server_" + "port=%d", + lcport, ltport, lsport); + + msg_add_header(resp, "Transport", resphdr); + + msg_add_header(resp, "Session", "1"); + + resp->respcode = 200; + return; + +error: + warn("Error in setup request."); + pthread_mutex_unlock(&play_lock); + resp->respcode = 451; // invalid arguments } +static void handle_ignore(rtsp_conn_info *conn, rtsp_message *req, rtsp_message *resp) { + resp->respcode = 200; +} + +static void handle_set_parameter_parameter(rtsp_conn_info *conn, rtsp_message *req, + rtsp_message *resp) { + char *cp = req->content; + int cp_left = req->contentlength; + char *next; + while (cp_left && cp) { + next = nextline(cp, cp_left); + cp_left -= next - cp; + + if (!strncmp(cp, "volume: ", 8)) { + if (config.ignore_volume_control == 0) { + float volume = atof(cp + 8); + debug(2, "volume: %f\n", volume); + player_volume(volume); + } + } else +#ifdef CONFIG_METADATA + if (!strncmp(cp, "progress: ", 10)) { + char *progress = cp + 10; + debug(2, "progress: \"%s\"\n", + progress); // rtpstampstart/rtpstampnow/rtpstampend 44100 per second + send_ssnc_metadata('prgr', strdup(progress), strlen(progress), 1); + } else +#endif + { + debug(1, "unrecognised parameter: \"%s\" (%d)\n", cp, strlen(cp)); + } + cp = next; + } +} #ifdef CONFIG_METADATA // Metadata is not used by shairport-sync. -// Instead we send all metadata to a fifo pipe, so that other apps can listen to the pipe and use the metadata. +// Instead we send all metadata to a fifo pipe, so that other apps can listen to the pipe and use +// the metadata. -// We use two 4-character codes to identify each piece of data and we send the data itself, if any, in base64 form. +// We use two 4-character codes to identify each piece of data and we send the data itself, if any, +// in base64 form. // The first 4-character code, called the "type", is either: -// 'core' for all the regular metadadata coming from iTunes, etc., or -// 'ssnc' (for 'shairport-sync') for all metadata coming from Shairport Sync itself, such as start/end delimiters, etc. +// 'core' for all the regular metadadata coming from iTunes, etc., or +// 'ssnc' (for 'shairport-sync') for all metadata coming from Shairport Sync itself, such as +// start/end delimiters, etc. -// For 'core' metadata, the second 4-character code is the 4-character metadata code coming from iTunes etc. +// For 'core' metadata, the second 4-character code is the 4-character metadata code coming from +// iTunes etc. // For 'ssnc' metadata, the second 4-character code is used to distinguish the messages. -// Cover art is not tagged in the same way as other metadata, it seems, so is sent as an 'ssnc' type metadata message with the code 'PICT' +// Cover art is not tagged in the same way as other metadata, it seems, so is sent as an 'ssnc' type +// metadata message with the code 'PICT' // Here are the 'ssnc' codes defined so far: -// 'PICT' -- the payload is a picture, either a JPEG or a PNG. Check the first few bytes to see which. +// 'PICT' -- the payload is a picture, either a JPEG or a PNG. Check the first few bytes to see +// which. // 'pbeg' -- play stream begin. No arguments // 'pend' -- play stream end. No arguments // 'pfls' -- play stream flush. No arguments // 'prsm' -- play stream resume. No arguments -// 'pvol' -- play volume. The volume is sent as a string -- "airplay_volume,volume,lowest_volume,highest_volume,has_true_mute,is_muted" +// 'pvol' -- play volume. The volume is sent as a string -- +// "airplay_volume,volume,lowest_volume,highest_volume,has_true_mute,is_muted" // volume, lowest_volume and highest_volume are given in dB -// is_muted is 1 if [true] mute is enabled, 0 otherwise. -// The "airplay_volume" is what's sent to the player, and is from 0.00 down to -30.00, with -144.00 meaning mute. +// is_muted is 1 if [true] mute is enabled, 0 otherwise. +// The "airplay_volume" is what's sent to the player, and is from 0.00 down to -30.00, +// with -144.00 meaning mute. // This is linear on the volume control slider of iTunes or iOS AirPLay -// 'prgr' -- progress -- this is metadata from AirPlay consisting of RTP timestamps for the start of the current play sequence, the current play point and the end of the play sequence. +// 'prgr' -- progress -- this is metadata from AirPlay consisting of RTP timestamps for the start +// of the current play sequence, the current play point and the end of the play sequence. // I guess the timestamps wrap at 2^32. // 'mdst' -- a sequence of metadata is about to start // 'mden' -- a sequence of metadata has ended -// 'snam' -- the name of the originator -- e.g. "Joe's iPhone" or "iTunes...". -// +// 'snam' -- A device -- e.g. "Joe's iPhone" -- has opened a play session. Specifically, it's the "X-Apple-Client-Name" string +// 'snua' -- A "user agent" -- e.g. "iTunes/12..." -- has opened a play session. Specifically, it's the "User-Agent" string +// // including a simple base64 encoder to minimise malloc/free activity // From Stack Overflow, with thanks: @@ -804,49 +834,45 @@ static void handle_set_parameter_parameter(rtsp_conn_info *conn, // add _so to end of name to avoid confusion with polarssl's implementation -static char encoding_table[] = {'A', 'B', 'C', 'D', 'E', 'F', 'G', 'H', - 'I', 'J', 'K', 'L', 'M', 'N', 'O', 'P', - 'Q', 'R', 'S', 'T', 'U', 'V', 'W', 'X', - 'Y', 'Z', 'a', 'b', 'c', 'd', 'e', 'f', - 'g', 'h', 'i', 'j', 'k', 'l', 'm', 'n', - 'o', 'p', 'q', 'r', 's', 't', 'u', 'v', - 'w', 'x', 'y', 'z', '0', '1', '2', '3', - '4', '5', '6', '7', '8', '9', '+', '/'}; +static char encoding_table[] = {'A', 'B', 'C', 'D', 'E', 'F', 'G', 'H', 'I', 'J', 'K', 'L', 'M', + 'N', 'O', 'P', 'Q', 'R', 'S', 'T', 'U', 'V', 'W', 'X', 'Y', 'Z', + 'a', 'b', 'c', 'd', 'e', 'f', 'g', 'h', 'i', 'j', 'k', 'l', 'm', + 'n', 'o', 'p', 'q', 'r', 's', 't', 'u', 'v', 'w', 'x', 'y', 'z', + '0', '1', '2', '3', '4', '5', '6', '7', '8', '9', '+', '/'}; static int mod_table[] = {0, 2, 1}; -// pass in a pointer to the data, its length, a pointer to the output buffer and a pointer to an int containing its maximum length +// pass in a pointer to the data, its length, a pointer to the output buffer and a pointer to an int +// containing its maximum length // the actual length will be returned. -char *base64_encode_so(const unsigned char *data, - size_t input_length, - char *encoded_data, - size_t *output_length) { +char *base64_encode_so(const unsigned char *data, size_t input_length, char *encoded_data, + size_t *output_length) { - size_t calculated_output_length = 4 * ((input_length + 2) / 3); - if (calculated_output_length> *output_length) - return(NULL); - *output_length = calculated_output_length; - - int i,j; - for (i = 0, j = 0; i < input_length;) { + size_t calculated_output_length = 4 * ((input_length + 2) / 3); + if (calculated_output_length > *output_length) + return (NULL); + *output_length = calculated_output_length; - uint32_t octet_a = i < input_length ? (unsigned char)data[i++] : 0; - uint32_t octet_b = i < input_length ? (unsigned char)data[i++] : 0; - uint32_t octet_c = i < input_length ? (unsigned char)data[i++] : 0; + int i, j; + for (i = 0, j = 0; i < input_length;) { - uint32_t triple = (octet_a << 0x10) + (octet_b << 0x08) + octet_c; + uint32_t octet_a = i < input_length ? (unsigned char)data[i++] : 0; + uint32_t octet_b = i < input_length ? (unsigned char)data[i++] : 0; + uint32_t octet_c = i < input_length ? (unsigned char)data[i++] : 0; - encoded_data[j++] = encoding_table[(triple >> 3 * 6) & 0x3F]; - encoded_data[j++] = encoding_table[(triple >> 2 * 6) & 0x3F]; - encoded_data[j++] = encoding_table[(triple >> 1 * 6) & 0x3F]; - encoded_data[j++] = encoding_table[(triple >> 0 * 6) & 0x3F]; - } + uint32_t triple = (octet_a << 0x10) + (octet_b << 0x08) + octet_c; - for (i = 0; i < mod_table[input_length % 3]; i++) - encoded_data[*output_length - 1 - i] = '='; + encoded_data[j++] = encoding_table[(triple >> 3 * 6) & 0x3F]; + encoded_data[j++] = encoding_table[(triple >> 2 * 6) & 0x3F]; + encoded_data[j++] = encoding_table[(triple >> 1 * 6) & 0x3F]; + encoded_data[j++] = encoding_table[(triple >> 0 * 6) & 0x3F]; + } - return encoded_data; + for (i = 0; i < mod_table[input_length % 3]; i++) + encoded_data[*output_length - 1 - i] = '='; + + return encoded_data; } // with thanks! @@ -861,70 +887,43 @@ metadata_package metadata_queue_items[metadata_queue_size]; static pthread_t metadata_thread; void metadata_create(void) { - if (!config.meta_dir) - return; + if (config.metadata_enabled == 0) + return; - const char fn[] = "shairport-sync-metadata"; - size_t pl = strlen(config.meta_dir) + 1 + strlen(fn); + size_t pl = strlen(config.metadata_pipename) + 1; - char* path = malloc(pl+1); - snprintf(path, pl+1, "%s/%s", config.meta_dir, fn); + char *path = malloc(pl + 1); + snprintf(path, pl + 1, "%s", config.metadata_pipename); - if (mkfifo(path, 0644) && errno != EEXIST) - die("Could not create metadata FIFO %s", path); + if (mkfifo(path, 0644) && errno != EEXIST) + die("Could not create metadata FIFO %s", path); - free(path); + free(path); } void metadata_open(void) { - if (!config.meta_dir) - return; + if (config.metadata_enabled == 0) + return; - const char fn[] = "shairport-sync-metadata"; - size_t pl = strlen(config.meta_dir) + 1 + strlen(fn); + size_t pl = strlen(config.metadata_pipename) + 1; - char* path = malloc(pl+1); - snprintf(path, pl+1, "%s/%s", config.meta_dir, fn); + char *path = malloc(pl + 1); + snprintf(path, pl + 1, "%s", config.metadata_pipename); - fd = open(path, O_WRONLY | O_NONBLOCK); - //if (fd < 0) - // debug(1, "Could not open metadata FIFO %s. Will try again later.", path); + fd = open(path, O_WRONLY | O_NONBLOCK); + // if (fd < 0) + // debug(1, "Could not open metadata FIFO %s. Will try again later.", path); - free(path); + free(path); } static void metadata_close(void) { - close(fd); - fd = -1; + close(fd); + fd = -1; } -ssize_t non_blocking_write(int fd, const void *buf, size_t count) { - // debug(1,"writing %u to pipe...",count); - // we are assuming that the count is always smaller than the FIFO's buffer - struct pollfd ufds[1]; - ssize_t reply; - do { - ufds[0].fd=fd; - ufds[0].events = POLLOUT; - int rv = poll(ufds,1,5000); - if (rv==-1) - debug(1,"error waiting for pipe to unblock..."); - if (rv==0) - debug(1,"timeout waiting for pipe to unblock"); - reply=write(fd,buf,count); - if ((reply==-1) && ((errno == EAGAIN) || (errno == EWOULDBLOCK))) - debug(1,"writing to pipe will block..."); -// else -// debug(1,"writing %u to pipe done...",reply); - } while ((reply==-1) && ((errno == EAGAIN) || (errno == EWOULDBLOCK))); - return reply; - - -// return write(fd,buf,count); -} - -void metadata_process(uint32_t type,uint32_t code,char *data,uint32_t length) { - debug(2,"Process metadata with type %x, code %x and length %u.",type,code,length); +void metadata_process(uint32_t type, uint32_t code, char *data, uint32_t length) { + debug(2, "Process metadata with type %x, code %x and length %u.", type, code, length); int ret; // readers may go away and come back if (fd < 0) @@ -932,14 +931,15 @@ void metadata_process(uint32_t type,uint32_t code,char *data,uint32_t length) { if (fd < 0) return; char thestring[1024]; - snprintf(thestring,1024,"%x%x%u\n",type,code,length); + snprintf(thestring, 1024, "%x%x%u", type, code, + length); ret = non_blocking_write(fd, thestring, strlen(thestring)); if (ret < 1) return; - if ((data!=NULL) && (length>0)) { - snprintf(thestring,1024,"\n"); + if ((data != NULL) && (length > 0)) { + snprintf(thestring, 1024, "\n\n"); ret = non_blocking_write(fd, thestring, strlen(thestring)); - if (ret < 1) // no reader + if (ret < 1) // no reader return; // here, we write the data in base64 form using our nice base64 encoder // but, we break it into lines of 76 output characters, except for the last one. @@ -948,37 +948,38 @@ void metadata_process(uint32_t type,uint32_t code,char *data,uint32_t length) { char *remaining_data = data; size_t towrite_count; char outbuf[76]; - while ((remaining_count) && (ret>=0)) { - size_t towrite_count = remaining_count; - if (towrite_count>57) - towrite_count = 57; - size_t outbuf_size = 76; // size of output buffer on entry, length of result on exit - if (base64_encode_so(remaining_data, towrite_count, outbuf, &outbuf_size)==NULL) - debug(1,"Error encoding base64 data."); - //debug(1,"Remaining count: %d ret: %d, outbuf_size: %d.",remaining_count,ret,outbuf_size); - ret = non_blocking_write(fd,outbuf,outbuf_size); - if (ret<0) - return; - remaining_data+=towrite_count; - remaining_count-=towrite_count; - // ret = write(fd,"\r\n",2); - // if (ret<0) - // return; + while ((remaining_count) && (ret >= 0)) { + size_t towrite_count = remaining_count; + if (towrite_count > 57) + towrite_count = 57; + size_t outbuf_size = 76; // size of output buffer on entry, length of result on exit + if (base64_encode_so(remaining_data, towrite_count, outbuf, &outbuf_size) == NULL) + debug(1, "Error encoding base64 data."); + // debug(1,"Remaining count: %d ret: %d, outbuf_size: %d.",remaining_count,ret,outbuf_size); + ret = non_blocking_write(fd, outbuf, outbuf_size); + if (ret < 0) + return; + remaining_data += towrite_count; + remaining_count -= towrite_count; } - snprintf(thestring,1024,"\n"); + snprintf(thestring, 1024, ""); ret = non_blocking_write(fd, thestring, strlen(thestring)); - if (ret < 1) // no reader + if (ret < 1) // no reader return; } + snprintf(thestring, 1024, "\n"); + ret = non_blocking_write(fd, thestring, strlen(thestring)); + if (ret < 1) // no reader + return; } -void* metadata_thread_function(void *ignore) { +void *metadata_thread_function(void *ignore) { metadata_create(); metadata_package pack; while (1) { pc_queue_get_item(&metadata_queue, &pack); - if (config.meta_dir) - metadata_process(pack.type,pack.code,pack.data,pack.length); + if (config.metadata_enabled) + metadata_process(pack.type, pack.code, pack.data, pack.length); if (pack.carrier) msg_free(pack.carrier); // release the message else if (pack.data) @@ -989,13 +990,33 @@ void* metadata_thread_function(void *ignore) { void metadata_init(void) { // create a pc_queue for passing information to a threaded metadata handler - pc_queue_init(&metadata_queue,(char *)&metadata_queue_items,sizeof(metadata_package),metadata_queue_size); + pc_queue_init(&metadata_queue, (char *)&metadata_queue_items, sizeof(metadata_package), + metadata_queue_size); int ret = pthread_create(&metadata_thread, NULL, metadata_thread_function, NULL); if (ret) - debug(1,"Failed to create metadata thread!"); + debug(1, "Failed to create metadata thread!"); } -int send_metadata(uint32_t type,uint32_t code,char *data,uint32_t length, rtsp_message* carrier, int block) { +int send_metadata(uint32_t type, uint32_t code, char *data, uint32_t length, rtsp_message *carrier, + int block) { + + // parameters: type, code, pointer to data or NULL, length of data or NULL, the rtsp_message or + // NULL + // the rtsp_message is sent for 'core' messages, because it contains the data and must not be + // freed until the data has been read. So, it is passed to send_metadata to be retained, + // sent to the thread where metadata is processed and released (and probably freed). + + // The rtsp_message is also sent for certain non-'core' messages. + + // The reading of the parameters is a bit complex + // If the rtsp_message field is non-null, then it represents an rtsp_message which should be freed + // in the thread handler when the parameter pointed to by the pointer and specified by the length + // is finished with + // If the rtsp_message is NULL, then if the pointer is non-null, it points to a malloc'ed block + // and should be freed when the thread is finished with it. The length of the data in the block is + // given in length + // If the rtsp_message is NULL and the pointer is also NULL, nothing further is done. + metadata_package pack; pack.type = type; pack.code = code; @@ -1004,102 +1025,87 @@ int send_metadata(uint32_t type,uint32_t code,char *data,uint32_t length, rtsp_m if (carrier) msg_retain(carrier); pack.carrier = carrier; - int rc = pc_queue_add_item(&metadata_queue,&pack,block); - if ((rc==EBUSY) && (carrier)) + int rc = pc_queue_add_item(&metadata_queue, &pack, block); + if ((rc == EBUSY) && (carrier)) msg_free(carrier); - if (rc==EBUSY) - warn("Metadata queue is busy, dropping message of type 0x%08X, code 0x%08X.",type,code); + if (rc == EBUSY) + warn("Metadata queue is busy, dropping message of type 0x%08X, code 0x%08X.", type, code); return rc; } -static void handle_set_parameter_metadata(rtsp_conn_info *conn, - rtsp_message *req, - rtsp_message *resp) { - char *cp = req->content; - int cl = req->contentlength; +static void handle_set_parameter_metadata(rtsp_conn_info *conn, rtsp_message *req, + rtsp_message *resp) { + char *cp = req->content; + int cl = req->contentlength; - unsigned int off = 8; - - // inform the listener that a set of metadata is starting - // this doesn't include the cover art though... - - // parameters: type, code, pointer to data or NULL, length of data or NULL, the rtsp_message or NULL - // the rtsp_message is sent for 'core' messages, because it contains the data and must not be - // freed until the data has been read. So, it is passed to send_metadata to be retained, - // sent to the thread where metadata is processed and released (and probably freed). - - // The reading of the parameters is a bit complex - // If the rtsp_message field is non-null, then it represents an rtsp_message which should be freed in the thread handler when the parameter pointed to by the pointer and specified by the length is finished with - // If the rtsp_message is NULL, then if the pointer is non-null, it points to a malloc'ed block and should be freed when the thread is finished with it. The length of the data in the block is given in length - // If the rtsp_message is NULL and the pointer is also NULL, nothing further is done. - - send_metadata('ssnc','mdst',NULL,0,NULL,1); + unsigned int off = 8; - while (off < cl) { - // pick up the metadata tag as an unsigned longint - uint32_t itag = ntohl(*(uint32_t *)(cp+off)); - off += sizeof(uint32_t); + // inform the listener that a set of metadata is starting + // this doesn't include the cover art though... - // pick up the length of the data - uint32_t vl = ntohl(*(uint32_t *)(cp+off)); - off += sizeof(uint32_t); - - // pass the data over - if (vl==0) - send_metadata('core',itag,NULL,0,NULL,1); - else - send_metadata('core',itag,(char *)(cp+off),vl,req,1); - - // move on to the next item - off += vl; - } - - // inform the listener that a set of metadata is ending - send_metadata('ssnc','mden',NULL,0,NULL,1); - // send the user some shairport-originated metadata - // send the name of the player, e.g. "Joe's iPhone" or "iTunes" - send_metadata('ssnc','sndr',strdup(sender_name),strlen(sender_name),NULL,1); + send_metadata('ssnc', 'mdst', NULL, 0, NULL, 1); + + while (off < cl) { + // pick up the metadata tag as an unsigned longint + uint32_t itag = ntohl(*(uint32_t *)(cp + off)); + off += sizeof(uint32_t); + + // pick up the length of the data + uint32_t vl = ntohl(*(uint32_t *)(cp + off)); + off += sizeof(uint32_t); + + // pass the data over + if (vl == 0) + send_metadata('core', itag, NULL, 0, NULL, 1); + else + send_metadata('core', itag, (char *)(cp + off), vl, req, 1); + + // move on to the next item + off += vl; + } + + // inform the listener that a set of metadata is ending + send_metadata('ssnc', 'mden', NULL, 0, NULL, 1); } #endif -static void handle_set_parameter(rtsp_conn_info *conn, - rtsp_message *req, rtsp_message *resp) { - //if (!req->contentlength) - // debug(1, "received empty SET_PARAMETER request."); +static void handle_set_parameter(rtsp_conn_info *conn, rtsp_message *req, rtsp_message *resp) { + // if (!req->contentlength) + // debug(1, "received empty SET_PARAMETER request."); - char *ct = msg_get_header(req, "Content-Type"); + char *ct = msg_get_header(req, "Content-Type"); - if (ct) { - debug(2, "SET_PARAMETER Content-Type:\"%s\".", ct); + if (ct) { + debug(2, "SET_PARAMETER Content-Type:\"%s\".", ct); #ifdef CONFIG_METADATA - if (!strncmp(ct, "application/x-dmap-tagged", 25)) { - debug(2, "received metadata tags in SET_PARAMETER request."); - handle_set_parameter_metadata(conn, req, resp); - } else if (!strncmp(ct, "image", 5)) { - // debug(1, "received image in SET_PARAMETER request."); - // note: the image/type tag isn't reliable, so it's not being sent - // -- best look at the first few bytes of the image - send_metadata('ssnc','PICT',req->content,req->contentlength,req,1); - } else -#endif - if (!strncmp(ct, "text/parameters", 15)) { - debug(2, "received parameters in SET_PARAMETER request."); - handle_set_parameter_parameter(conn, req, resp); - } else { - debug(1, "received unknown Content-Type \"%s\" in SET_PARAMETER request.", ct); - } + if (!strncmp(ct, "application/x-dmap-tagged", 25)) { + debug(2, "received metadata tags in SET_PARAMETER request."); + handle_set_parameter_metadata(conn, req, resp); + } else if (!strncmp(ct, "image", 5)) { + // debug(1, "received image in SET_PARAMETER request."); + // note: the image/type tag isn't reliable, so it's not being sent + // -- best look at the first few bytes of the image + send_metadata('ssnc', 'PICT', req->content, req->contentlength, req, 1); + } else +#endif + if (!strncmp(ct, "text/parameters", 15)) { + debug(2, "received parameters in SET_PARAMETER request."); + handle_set_parameter_parameter(conn, req, resp); } else { - debug(1, "missing Content-Type header in SET_PARAMETER request."); + debug(1, "received unknown Content-Type \"%s\" in SET_PARAMETER request.", ct); } + } else { + debug(1, "missing Content-Type header in SET_PARAMETER request."); + } - resp->respcode = 200; + resp->respcode = 200; } -static void handle_announce(rtsp_conn_info *conn, - rtsp_message *req, rtsp_message *resp) { - // allow a session to be interrupted if the timeout is set to zero - if ((config.timeout==0) || (pthread_mutex_trylock(&play_lock) == 0)) { +static void handle_announce(rtsp_conn_info *conn, rtsp_message *req, rtsp_message *resp) { + // interrupt session if permitted + if ((config.allow_session_interruption == 1) || (pthread_mutex_trylock(&play_lock) == 0)) { + resp->respcode = 456; // 456 - Header Field Not Valid for Resource char *paesiv = NULL; char *prsaaeskey = NULL; char *pfmtp = NULL; @@ -1108,485 +1114,505 @@ static void handle_announce(rtsp_conn_info *conn, char *next; while (cp_left && cp) { next = nextline(cp, cp_left); - cp_left -= next-cp; + cp_left -= next - cp; if (!strncmp(cp, "a=fmtp:", 7)) - pfmtp = cp+7; + pfmtp = cp + 7; if (!strncmp(cp, "a=aesiv:", 8)) - paesiv = cp+8; + paesiv = cp + 8; if (!strncmp(cp, "a=rsaaeskey:", 12)) - prsaaeskey = cp+12; + prsaaeskey = cp + 12; cp = next; } - - if (!paesiv || !prsaaeskey || !pfmtp) { - warn("required params missing from announce"); - return; - } - - int len, keylen; - uint8_t *aesiv = base64_dec(paesiv, &len); - if (len != 16) { - warn("client announced aeskey of %d bytes, wanted 16", len); - free(aesiv); - return; - } - memcpy(conn->stream.aesiv, aesiv, 16); - free(aesiv); - - uint8_t *rsaaeskey = base64_dec(prsaaeskey, &len); - uint8_t *aeskey = rsa_apply(rsaaeskey, len, &keylen, RSA_MODE_KEY); - free(rsaaeskey); - if (keylen != 16) { - warn("client announced rsaaeskey of %d bytes, wanted 16", keylen); - free(aeskey); - return; - } - memcpy(conn->stream.aeskey, aeskey, 16); - free(aeskey); - - int i; - for (i=0; istream.fmtp)/sizeof(conn->stream.fmtp[0]); i++) - conn->stream.fmtp[i] = atoi(strsep(&pfmtp, " \t")); + if ((paesiv==NULL) && (prsaaeskey==NULL)) { + //debug(1,"Unencrypted session requested?"); + conn->stream.encrypted = 0; + } else { + conn->stream.encrypted = 1; + //debug(1,"Encrypted session requested"); + } + + if (!pfmtp) { + warn("FMTP params missing from the following ANNOUNCE message:"); + // print each line of the request content + // the problem is that nextline has replace all returns, newlines, etc. by NULLs + char *cp = req->content; + int cp_left = req->contentlength; + while (cp_left>1) { + if (strlen(cp)!=0) + warn(" %s",cp); + cp+=strlen(cp)+1; + cp_left-=strlen(cp)+1; + } + goto out; + } + + if (conn->stream.encrypted) { + int len, keylen; + uint8_t *aesiv = base64_dec(paesiv, &len); + if (len != 16) { + warn("client announced aeskey of %d bytes, wanted 16", len); + free(aesiv); + goto out; + } + memcpy(conn->stream.aesiv, aesiv, 16); + free(aesiv); + + uint8_t *rsaaeskey = base64_dec(prsaaeskey, &len); + uint8_t *aeskey = rsa_apply(rsaaeskey, len, &keylen, RSA_MODE_KEY); + free(rsaaeskey); + if (keylen != 16) { + warn("client announced rsaaeskey of %d bytes, wanted 16", keylen); + free(aeskey); + goto out; + } + memcpy(conn->stream.aeskey, aeskey, 16); + free(aeskey); + } + int i; + for (i = 0; i < sizeof(conn->stream.fmtp) / sizeof(conn->stream.fmtp[0]); i++) + conn->stream.fmtp[i] = atoi(strsep(&pfmtp, " \t")); + char *hdr = msg_get_header(req, "X-Apple-Client-Name"); if (hdr) { - strncpy(sender_name,hdr,1024); - debug(1,"Play connection from \"%s\".",hdr); - } else { - hdr = msg_get_header(req, "User-Agent"); - if (hdr) { - debug(1,"Play connection from \"%s\".",hdr); - strncpy(sender_name,hdr,1024); - } else - sender_name[0]=0; - } + debug(1, "Play connection from device named \"%s\".", hdr); + #ifdef CONFIG_METADATA + send_metadata('ssnc','snam',hdr,strlen(hdr),req,1); + #endif + } + hdr = msg_get_header(req, "User-Agent"); + if (hdr) { + debug(1, "Play connection from user agent \"%s\".", hdr); + #ifdef CONFIG_METADATA + send_metadata('ssnc','snua',hdr,strlen(hdr),req,1); + #endif + } resp->respcode = 200; } else { resp->respcode = 453; - debug(1,"Already playing."); + debug(1, "Already playing."); + } + +out: + if (resp->respcode != 200 && resp->respcode != 453) { + pthread_mutex_unlock(&play_lock); } } - static struct method_handler { - char *method; - void (*handler)(rtsp_conn_info *conn, rtsp_message *req, - rtsp_message *resp); -} method_handlers[] = { - {"OPTIONS", handle_options}, - {"ANNOUNCE", handle_announce}, - {"FLUSH", handle_flush}, - {"TEARDOWN", handle_teardown}, - {"SETUP", handle_setup}, - {"GET_PARAMETER", handle_ignore}, - {"SET_PARAMETER", handle_set_parameter}, - {"RECORD", handle_record}, - {NULL, NULL} -}; + char *method; + void (*handler)(rtsp_conn_info *conn, rtsp_message *req, rtsp_message *resp); +} method_handlers[] = {{"OPTIONS", handle_options}, + {"ANNOUNCE", handle_announce}, + {"FLUSH", handle_flush}, + {"TEARDOWN", handle_teardown}, + {"SETUP", handle_setup}, + {"GET_PARAMETER", handle_ignore}, + {"SET_PARAMETER", handle_set_parameter}, + {"RECORD", handle_record}, + {NULL, NULL}}; static void apple_challenge(int fd, rtsp_message *req, rtsp_message *resp) { - char *hdr = msg_get_header(req, "Apple-Challenge"); - if (!hdr) - return; + char *hdr = msg_get_header(req, "Apple-Challenge"); + if (!hdr) + return; - SOCKADDR fdsa; - socklen_t sa_len = sizeof(fdsa); - getsockname(fd, (struct sockaddr*)&fdsa, &sa_len); + SOCKADDR fdsa; + socklen_t sa_len = sizeof(fdsa); + getsockname(fd, (struct sockaddr *)&fdsa, &sa_len); - int chall_len; - uint8_t *chall = base64_dec(hdr, &chall_len); - uint8_t buf[48], *bp = buf; - int i; - memset(buf, 0, sizeof(buf)); + int chall_len; + uint8_t *chall = base64_dec(hdr, &chall_len); + uint8_t buf[48], *bp = buf; + int i; + memset(buf, 0, sizeof(buf)); - if (chall_len > 16) { - warn("oversized Apple-Challenge!"); - free(chall); - return; - } - memcpy(bp, chall, chall_len); + if (chall_len > 16) { + warn("oversized Apple-Challenge!"); free(chall); - bp += chall_len; + return; + } + memcpy(bp, chall, chall_len); + free(chall); + bp += chall_len; #ifdef AF_INET6 - if (fdsa.SAFAMILY == AF_INET6) { - struct sockaddr_in6 *sa6 = (struct sockaddr_in6*)(&fdsa); - memcpy(bp, sa6->sin6_addr.s6_addr, 16); - bp += 16; - } else + if (fdsa.SAFAMILY == AF_INET6) { + struct sockaddr_in6 *sa6 = (struct sockaddr_in6 *)(&fdsa); + memcpy(bp, sa6->sin6_addr.s6_addr, 16); + bp += 16; + } else #endif - { - struct sockaddr_in *sa = (struct sockaddr_in*)(&fdsa); - memcpy(bp, &sa->sin_addr.s_addr, 4); - bp += 4; - } + { + struct sockaddr_in *sa = (struct sockaddr_in *)(&fdsa); + memcpy(bp, &sa->sin_addr.s_addr, 4); + bp += 4; + } - for (i=0; i<6; i++) - *bp++ = config.hw_addr[i]; + for (i = 0; i < 6; i++) + *bp++ = config.hw_addr[i]; - int buflen, resplen; - buflen = bp-buf; - if (buflen < 0x20) - buflen = 0x20; + int buflen, resplen; + buflen = bp - buf; + if (buflen < 0x20) + buflen = 0x20; - uint8_t *challresp = rsa_apply(buf, buflen, &resplen, RSA_MODE_AUTH); - char *encoded = base64_enc(challresp, resplen); + uint8_t *challresp = rsa_apply(buf, buflen, &resplen, RSA_MODE_AUTH); + char *encoded = base64_enc(challresp, resplen); - // strip the padding. - char *padding = strchr(encoded, '='); - if (padding) - *padding = 0; + // strip the padding. + char *padding = strchr(encoded, '='); + if (padding) + *padding = 0; - msg_add_header(resp, "Apple-Response", encoded); - free(challresp); - free(encoded); + msg_add_header(resp, "Apple-Response", encoded); + free(challresp); + free(encoded); } static char *make_nonce(void) { - uint8_t random[8]; - int fd = open("/dev/random", O_RDONLY); - if (fd < 0) - die("could not open /dev/random!"); - int ignore = read(fd, random, sizeof(random)); - close(fd); - return base64_enc(random, 8); + uint8_t random[8]; + int fd = open("/dev/random", O_RDONLY); + if (fd < 0) + die("could not open /dev/random!"); + int ignore = read(fd, random, sizeof(random)); + close(fd); + return base64_enc(random, 8); } static int rtsp_auth(char **nonce, rtsp_message *req, rtsp_message *resp) { - if (!config.password) - return 0; - if (!*nonce) { - *nonce = make_nonce(); - goto authenticate; - } + if (!config.password) + return 0; + if (!*nonce) { + *nonce = make_nonce(); + goto authenticate; + } - char *hdr = msg_get_header(req, "Authorization"); - if (!hdr || strncmp(hdr, "Digest ", 7)) - goto authenticate; + char *hdr = msg_get_header(req, "Authorization"); + if (!hdr || strncmp(hdr, "Digest ", 7)) + goto authenticate; - char *realm = strstr(hdr, "realm=\""); - char *username = strstr(hdr, "username=\""); - char *response = strstr(hdr, "response=\""); - char *uri = strstr(hdr, "uri=\""); + char *realm = strstr(hdr, "realm=\""); + char *username = strstr(hdr, "username=\""); + char *response = strstr(hdr, "response=\""); + char *uri = strstr(hdr, "uri=\""); - if (!realm || !username || !response || !uri) - goto authenticate; + if (!realm || !username || !response || !uri) + goto authenticate; - char *quote; - realm = strchr(realm, '"') + 1; - if (!(quote = strchr(realm, '"'))) - goto authenticate; - *quote = 0; - username = strchr(username, '"') + 1; - if (!(quote = strchr(username, '"'))) - goto authenticate; - *quote = 0; - response = strchr(response, '"') + 1; - if (!(quote = strchr(response, '"'))) - goto authenticate; - *quote = 0; - uri = strchr(uri, '"') + 1; - if (!(quote = strchr(uri, '"'))) - goto authenticate; - *quote = 0; + char *quote; + realm = strchr(realm, '"') + 1; + if (!(quote = strchr(realm, '"'))) + goto authenticate; + *quote = 0; + username = strchr(username, '"') + 1; + if (!(quote = strchr(username, '"'))) + goto authenticate; + *quote = 0; + response = strchr(response, '"') + 1; + if (!(quote = strchr(response, '"'))) + goto authenticate; + *quote = 0; + uri = strchr(uri, '"') + 1; + if (!(quote = strchr(uri, '"'))) + goto authenticate; + *quote = 0; + + uint8_t digest_urp[16], digest_mu[16], digest_total[16]; - uint8_t digest_urp[16], digest_mu[16], digest_total[16]; - #ifdef HAVE_LIBSSL - MD5_CTX ctx; - - MD5_Init(&ctx); - MD5_Update(&ctx, username, strlen(username)); - MD5_Update(&ctx, ":", 1); - MD5_Update(&ctx, realm, strlen(realm)); - MD5_Update(&ctx, ":", 1); - MD5_Update(&ctx, config.password, strlen(config.password)); - MD5_Final(digest_urp, &ctx); - MD5_Init(&ctx); - MD5_Update(&ctx, req->method, strlen(req->method)); - MD5_Update(&ctx, ":", 1); - MD5_Update(&ctx, uri, strlen(uri)); - MD5_Final(digest_mu, &ctx); + MD5_CTX ctx; + + MD5_Init(&ctx); + MD5_Update(&ctx, username, strlen(username)); + MD5_Update(&ctx, ":", 1); + MD5_Update(&ctx, realm, strlen(realm)); + MD5_Update(&ctx, ":", 1); + MD5_Update(&ctx, config.password, strlen(config.password)); + MD5_Final(digest_urp, &ctx); + MD5_Init(&ctx); + MD5_Update(&ctx, req->method, strlen(req->method)); + MD5_Update(&ctx, ":", 1); + MD5_Update(&ctx, uri, strlen(uri)); + MD5_Final(digest_mu, &ctx); #endif - #ifdef HAVE_LIBPOLARSSL - md5_context tctx; - md5_starts(&tctx); - md5_update(&tctx, (const unsigned char *)username, strlen(username)); - md5_update(&tctx, (unsigned char *) ":", 1); - md5_update(&tctx, (const unsigned char *)realm, strlen(realm)); - md5_update(&tctx, (unsigned char *) ":", 1); - md5_update(&tctx, (const unsigned char *)config.password, strlen(config.password)); - md5_finish(&tctx,digest_urp); - md5_starts(&tctx); - md5_update(&tctx, (const unsigned char *)req->method, strlen(req->method)); - md5_update(&tctx, (unsigned char *) ":", 1); - md5_update(&tctx, (const unsigned char *)uri, strlen(uri)); - md5_finish(&tctx,digest_mu); + md5_context tctx; + md5_starts(&tctx); + md5_update(&tctx, (const unsigned char *)username, strlen(username)); + md5_update(&tctx, (unsigned char *)":", 1); + md5_update(&tctx, (const unsigned char *)realm, strlen(realm)); + md5_update(&tctx, (unsigned char *)":", 1); + md5_update(&tctx, (const unsigned char *)config.password, strlen(config.password)); + md5_finish(&tctx, digest_urp); + md5_starts(&tctx); + md5_update(&tctx, (const unsigned char *)req->method, strlen(req->method)); + md5_update(&tctx, (unsigned char *)":", 1); + md5_update(&tctx, (const unsigned char *)uri, strlen(uri)); + md5_finish(&tctx, digest_mu); #endif - - int i; - unsigned char buf[33]; - for (i=0; i<16; i++) - sprintf((char *)buf + 2*i, "%02X", digest_urp[i]); - + + int i; + unsigned char buf[33]; + for (i = 0; i < 16; i++) + sprintf((char *)buf + 2 * i, "%02X", digest_urp[i]); + #ifdef HAVE_LIBSSL - MD5_Init(&ctx); - MD5_Update(&ctx, buf, 32); - MD5_Update(&ctx, ":", 1); - MD5_Update(&ctx, *nonce, strlen(*nonce)); - MD5_Update(&ctx, ":", 1); - for (i=0; i<16; i++) - sprintf(buf + 2*i, "%02X", digest_mu[i]); - MD5_Update(&ctx, buf, 32); - MD5_Final(digest_total, &ctx); - #endif - - -#ifdef HAVE_LIBPOLARSSL - md5_starts(&tctx); - md5_update(&tctx, buf, 32); - md5_update(&tctx, (unsigned char *) ":", 1); - md5_update(&tctx, (const unsigned char *)*nonce, strlen(*nonce)); - md5_update(&tctx, (unsigned char *) ":", 1); - for (i=0; i<16; i++) - sprintf((char *)buf + 2*i,"%02X", digest_mu[i]); - md5_update(&tctx, buf, 32); - md5_finish(&tctx,digest_total); + MD5_Init(&ctx); + MD5_Update(&ctx, buf, 32); + MD5_Update(&ctx, ":", 1); + MD5_Update(&ctx, *nonce, strlen(*nonce)); + MD5_Update(&ctx, ":", 1); + for (i = 0; i < 16; i++) + sprintf(buf + 2 * i, "%02X", digest_mu[i]); + MD5_Update(&ctx, buf, 32); + MD5_Final(digest_total, &ctx); #endif - for (i=0; i<16; i++) - sprintf((char *)buf + 2*i,"%02X", digest_total[i]); +#ifdef HAVE_LIBPOLARSSL + md5_starts(&tctx); + md5_update(&tctx, buf, 32); + md5_update(&tctx, (unsigned char *)":", 1); + md5_update(&tctx, (const unsigned char *)*nonce, strlen(*nonce)); + md5_update(&tctx, (unsigned char *)":", 1); + for (i = 0; i < 16; i++) + sprintf((char *)buf + 2 * i, "%02X", digest_mu[i]); + md5_update(&tctx, buf, 32); + md5_finish(&tctx, digest_total); +#endif - if (!strcmp(response, (const char *)buf)) - return 0; - warn("auth failed"); + for (i = 0; i < 16; i++) + sprintf((char *)buf + 2 * i, "%02X", digest_total[i]); + + if (!strcmp(response, (const char *)buf)) + return 0; + warn("auth failed"); authenticate: - resp->respcode = 401; - int hdrlen = strlen(*nonce) + 40; - char *authhdr = malloc(hdrlen); - snprintf(authhdr, hdrlen, "Digest realm=\"taco\", nonce=\"%s\"", *nonce); - msg_add_header(resp, "WWW-Authenticate", authhdr); - free(authhdr); - return 1; + resp->respcode = 401; + int hdrlen = strlen(*nonce) + 40; + char *authhdr = malloc(hdrlen); + snprintf(authhdr, hdrlen, "Digest realm=\"taco\", nonce=\"%s\"", *nonce); + msg_add_header(resp, "WWW-Authenticate", authhdr); + free(authhdr); + return 1; } static void *rtsp_conversation_thread_func(void *pconn) { - // SIGUSR1 is used to interrupt this thread if blocked for read - sigset_t set; - sigemptyset(&set); - sigaddset(&set, SIGUSR1); - pthread_sigmask(SIG_UNBLOCK, &set, NULL); + // SIGUSR1 is used to interrupt this thread if blocked for read + sigset_t set; + sigemptyset(&set); + sigaddset(&set, SIGUSR1); + pthread_sigmask(SIG_UNBLOCK, &set, NULL); - rtsp_conn_info *conn = pconn; + rtsp_conn_info *conn = pconn; - rtsp_message *req, *resp; - char *hdr, *auth_nonce = NULL; - - enum rtsp_read_request_response reply; - - do { - reply=rtsp_read_request(conn->fd,&req); - if (reply==rtsp_read_request_response_ok) { - resp = msg_init(); - resp->respcode = 400; + rtsp_message *req, *resp; + char *hdr, *auth_nonce = NULL; - apple_challenge(conn->fd, req, resp); - hdr = msg_get_header(req, "CSeq"); - if (hdr) - msg_add_header(resp, "CSeq", hdr); - msg_add_header(resp, "Audio-Jack-Status", "connected; type=analog"); + enum rtsp_read_request_response reply; - if (rtsp_auth(&auth_nonce, req, resp)) - goto respond; + do { + reply = rtsp_read_request(conn->fd, &req); + if (reply == rtsp_read_request_response_ok) { + resp = msg_init(); + resp->respcode = 400; - struct method_handler *mh; - for (mh=method_handlers; mh->method; mh++) { - if (!strcmp(mh->method, req->method)) { - // debug(1,"RTSP Packet received of type \"%s\":",mh->method), - // msg_print_debug_headers(req); - mh->handler(conn, req, resp); - // debug(1,"RTSP Response:"); - // msg_print_debug_headers(resp); - break; - } + apple_challenge(conn->fd, req, resp); + hdr = msg_get_header(req, "CSeq"); + if (hdr) + msg_add_header(resp, "CSeq", hdr); + msg_add_header(resp, "Audio-Jack-Status", "connected; type=analog"); + + if (rtsp_auth(&auth_nonce, req, resp)) + goto respond; + + struct method_handler *mh; + for (mh = method_handlers; mh->method; mh++) { + if (!strcmp(mh->method, req->method)) { + //debug(1,"RTSP Packet received of type \"%s\":",mh->method), + //msg_print_debug_headers(req); + mh->handler(conn, req, resp); + //debug(1,"RTSP Response:"); + //msg_print_debug_headers(resp); + break; } - -respond: - msg_write_response(conn->fd, resp); - msg_free(req); - msg_free(resp); - } else { - if (reply!=rtsp_read_request_response_shutdown_requested) - debug(1,"rtsp_read_request error %d, packet ignored.",(int)reply); } - } while (reply!=rtsp_read_request_response_shutdown_requested); - debug(1, "closing RTSP connection."); - if (conn->fd > 0) - close(conn->fd); - if (rtsp_playing()) { - rtp_shutdown(); - player_stop(); - pthread_mutex_unlock(&play_lock); - please_shutdown = 0; - pthread_mutex_unlock(&playing_mutex); + respond: + msg_write_response(conn->fd, resp); + msg_free(req); + msg_free(resp); + } else { + if (reply != rtsp_read_request_response_shutdown_requested) + debug(1, "rtsp_read_request error %d, packet ignored.", (int)reply); } - if (auth_nonce) - free(auth_nonce); - conn->running = 0; - debug(2, "terminating RTSP thread."); - return NULL; + } while (reply != rtsp_read_request_response_shutdown_requested); + + debug(1, "closing RTSP connection."); + if (conn->fd > 0) + close(conn->fd); + if (rtsp_playing()) { + rtp_shutdown(); + player_stop(); + pthread_mutex_unlock(&play_lock); + please_shutdown = 0; + pthread_mutex_unlock(&playing_mutex); + } + if (auth_nonce) + free(auth_nonce); + conn->running = 0; + debug(2, "terminating RTSP thread."); + return NULL; } // this function is not thread safe. -static const char* format_address(struct sockaddr *fsa) { - static char string[INETx_ADDRSTRLEN]; - void *addr; +static const char *format_address(struct sockaddr *fsa) { + static char string[INETx_ADDRSTRLEN]; + void *addr; #ifdef AF_INET6 - if (fsa->sa_family == AF_INET6) { - struct sockaddr_in6 *sa6 = (struct sockaddr_in6*)(fsa); - addr = &(sa6->sin6_addr); - } else + if (fsa->sa_family == AF_INET6) { + struct sockaddr_in6 *sa6 = (struct sockaddr_in6 *)(fsa); + addr = &(sa6->sin6_addr); + } else #endif - { - struct sockaddr_in *sa = (struct sockaddr_in*)(fsa); - addr = &(sa->sin_addr); - } - return inet_ntop(fsa->sa_family, addr, string, sizeof(string)); + { + struct sockaddr_in *sa = (struct sockaddr_in *)(fsa); + addr = &(sa->sin_addr); + } + return inet_ntop(fsa->sa_family, addr, string, sizeof(string)); } void rtsp_listen_loop(void) { - struct addrinfo hints, *info, *p; - char portstr[6]; - int *sockfd = NULL; - int nsock = 0; - int i, ret; + struct addrinfo hints, *info, *p; + char portstr[6]; + int *sockfd = NULL; + int nsock = 0; + int i, ret; - memset(&hints, 0, sizeof(hints)); - hints.ai_family = AF_UNSPEC; - hints.ai_socktype = SOCK_STREAM; - hints.ai_flags = AI_PASSIVE; + memset(&hints, 0, sizeof(hints)); + hints.ai_family = AF_UNSPEC; + hints.ai_socktype = SOCK_STREAM; + hints.ai_flags = AI_PASSIVE; - snprintf(portstr, 6, "%d", config.port); - - // debug(1,"listen socket port request is \"%s\".",portstr); + snprintf(portstr, 6, "%d", config.port); - ret = getaddrinfo(NULL, portstr, &hints, &info); - if (ret) { - die("getaddrinfo failed: %s", gai_strerror(ret)); + // debug(1,"listen socket port request is \"%s\".",portstr); + + ret = getaddrinfo(NULL, portstr, &hints, &info); + if (ret) { + die("getaddrinfo failed: %s", gai_strerror(ret)); + } + + for (p = info; p; p = p->ai_next) { + int fd = socket(p->ai_family, p->ai_socktype, IPPROTO_TCP); + int yes = 1; + + // Handle socket open failures if protocol unavailable (or IPV6 not handled) + if (fd == -1) { + // debug(1, "Failed to get socket: fam=%d, %s\n", p->ai_family, strerror(errno)); + continue; } - for (p=info; p; p=p->ai_next) { - int fd = socket(p->ai_family, p->ai_socktype, IPPROTO_TCP); - int yes = 1; - - // Handle socket open failures if protocol unavailable (or IPV6 not handled) - if (fd == -1) { - // debug(1, "Failed to get socket: fam=%d, %s\n", p->ai_family, strerror(errno)); - continue; - } - - ret = setsockopt(fd, SOL_SOCKET, SO_REUSEADDR, &yes, sizeof(yes)); + ret = setsockopt(fd, SOL_SOCKET, SO_REUSEADDR, &yes, sizeof(yes)); #ifdef IPV6_V6ONLY - // some systems don't support v4 access on v6 sockets, but some do. - // since we need to account for two sockets we might as well - // always. - if (p->ai_family == AF_INET6) { - ret |= setsockopt(fd, IPPROTO_IPV6, IPV6_V6ONLY, &yes, sizeof(yes)); - } + // some systems don't support v4 access on v6 sockets, but some do. + // since we need to account for two sockets we might as well + // always. + if (p->ai_family == AF_INET6) { + ret |= setsockopt(fd, IPPROTO_IPV6, IPV6_V6ONLY, &yes, sizeof(yes)); + } #endif - if (!ret) - ret = bind(fd, p->ai_addr, p->ai_addrlen); + if (!ret) + ret = bind(fd, p->ai_addr, p->ai_addrlen); - // one of the address families will fail on some systems that - // report its availability. do not complain. - if (ret) { - debug(1, "Failed to bind to address %s.", format_address(p->ai_addr)); - continue; - } - - listen(fd, 5); - nsock++; - sockfd = realloc(sockfd, nsock*sizeof(int)); - sockfd[nsock-1] = fd; + // one of the address families will fail on some systems that + // report its availability. do not complain. + if (ret) { + debug(1, "Failed to bind to address %s.", format_address(p->ai_addr)); + continue; } - freeaddrinfo(info); + listen(fd, 5); + nsock++; + sockfd = realloc(sockfd, nsock * sizeof(int)); + sockfd[nsock - 1] = fd; + } - if (!nsock) - die("could not bind any listen sockets!"); + freeaddrinfo(info); - int maxfd = -1; - fd_set fds; - FD_ZERO(&fds); - for (i=0; i maxfd) - maxfd = sockfd[i]; + if (!nsock) + die("could not bind any listen sockets!"); + + int maxfd = -1; + fd_set fds; + FD_ZERO(&fds); + for (i = 0; i < nsock; i++) { + if (sockfd[i] > maxfd) + maxfd = sockfd[i]; + } + + mdns_register(); + + // printf("Listening for connections."); + // shairport_startup_complete(); + + int acceptfd; + struct timeval tv; + while (1) { + tv.tv_sec = 300; + tv.tv_usec = 0; + + for (i = 0; i < nsock; i++) + FD_SET(sockfd[i], &fds); + + ret = select(maxfd + 1, &fds, 0, 0, &tv); + if (ret < 0) { + if (errno == EINTR) + continue; + break; } - mdns_register(); + cleanup_threads(); - // printf("Listening for connections."); - // shairport_startup_complete(); - - int acceptfd; - struct timeval tv; - while (1) { - tv.tv_sec = 300; - tv.tv_usec = 0; - - for (i=0; iremote); - - debug(1, "new RTSP connection."); - conn->fd = accept(acceptfd, (struct sockaddr *)&conn->remote, &slen); - if (conn->fd < 0) { - perror("failed to accept connection"); - free(conn); - } else { - pthread_t rtsp_conversation_thread; - ret = pthread_create(&rtsp_conversation_thread, NULL, rtsp_conversation_thread_func, conn); - if (ret) - die("Failed to create RTSP receiver thread!"); - - conn->thread = rtsp_conversation_thread; - conn->running = 1; - track_thread(conn); - } + acceptfd = -1; + for (i = 0; i < nsock; i++) { + if (FD_ISSET(sockfd[i], &fds)) { + acceptfd = sockfd[i]; + break; + } } - perror("select"); - die("fell out of the RTSP select loop"); + if (acceptfd < 0) // timeout + continue; + + rtsp_conn_info *conn = malloc(sizeof(rtsp_conn_info)); + memset(conn, 0, sizeof(rtsp_conn_info)); + socklen_t slen = sizeof(conn->remote); + + debug(1, "new RTSP connection."); + conn->fd = accept(acceptfd, (struct sockaddr *)&conn->remote, &slen); + if (conn->fd < 0) { + perror("failed to accept connection"); + free(conn); + } else { + pthread_t rtsp_conversation_thread; + ret = pthread_create(&rtsp_conversation_thread, NULL, rtsp_conversation_thread_func, conn); + if (ret) + die("Failed to create RTSP receiver thread!"); + + conn->thread = rtsp_conversation_thread; + conn->running = 1; + track_thread(conn); + } + } + perror("select"); + die("fell out of the RTSP select loop"); } diff --git a/rtsp.h b/rtsp.h index 74fdd52e..49650055 100644 --- a/rtsp.h +++ b/rtsp.h @@ -5,12 +5,17 @@ void rtsp_listen_loop(void); void rtsp_shutdown_stream(void); void rtsp_request_shutdown_stream(void); + +// initialise the metadata stuff + +void metadata_init(void); + // sends metadata out to the metadata pipe, if enabled. // It is sent with the type 'ssnc' the given code, data and length // The handler at the other end must kknow what to do with the data // e.g. it it's malloced, to free it, etc. // nothing is done automatically -int send_ssnc_metadata(uint32_t code,char *data,uint32_t length, int block); +int send_ssnc_metadata(uint32_t code, char *data, uint32_t length, int block); #endif // _RTSP_H diff --git a/scripts/shairport-sync.conf b/scripts/shairport-sync.conf new file mode 100644 index 00000000..b2ac51a5 --- /dev/null +++ b/scripts/shairport-sync.conf @@ -0,0 +1,77 @@ +// Sample Configuration File for Shairport Sync +// Commented out settings are generally the defaults, except where noted. + +// General Settings +general = +{ +// name = "Shairport Sync Player"; // This is the name the service will advertise to iTunes. The default is "Shairport Sync on " +// password = "secret"; // comment out this line if you want to have no password +// interpolation = "basic"; // aka "stuffing". Default is "basic", alternative is "soxr". Use "soxr" only if you have a reasonably fast processor. +// output_backend = "alsa"; // Run "shairport-sync -h" to get a list of all output_backends, e.g. "alsa", "pipe", "stdout". The default is the first one. +// mdns_backend = "avahi"; // Run "shairport-sync -h" to get a list of all mdns_backends. The default is the first one. +// port = 5000; // Listen for service requests on this port +// udp_port_base = 6001; // start allocating UDP ports from this port number when needed +// udp_port_range = 100; // look for free ports in this number of places, starting at the UDP port base (only three are needed). +// statistics = "no"; // set to "yes" to print statistics in the log +// drift = 88; // allow this number of frames of drift away from exact synchronisation before attempting to correct it +// resync_threshold = 2205; // a synchronisation error greater than this will cause resynchronisation; 0 disables it +// log_verbosity = 0; // "0" means no debug verbosity, "3" is most verbose. +// ignore_volume_control = "no"; // set this to "yes" if you want the volume to be at 100% no matter what the source's volume control is set to. +}; + +// Latencies for different sources. These have been estimated from listening tests. +// It's probably better to compensate for a delay in the output device using the alsa "audio_backend_latency_offset" setting -- see below. +latencies = +{ +// default = 88200; // used for unrecognised sources and for iTunes up to and including iTunes 9.X. +// itunes = 99400; // used for iTunes 10 or later +// airplay = 88200; +// forkedDaapd = 99400; +}; + +// How to deal with metadata, including artwork +metadata = +{ +// enabled = "no"; // et to yes to get Shairport Sync to solicit metadata from the source and to pass it on via a pipe +// include_cover_art = "no"; // set to "yes" to get Shairport Sync to solicit cover art from the source and pass it via the pipe. You must also set "enabled" to "yes". +// pipe_name = "/tmp/shairport-sync-metadata"; +}; + +// Advanced parameters for controlling how a Shairport Sync runs +sessioncontrol = +{ +// run_this_before_play_begins = "/full/path/to/application and args"; // make sure the application has executable permission. It it's a script, include the #!... stuff on the first line +// run_this_after_play_ends = "/full/path/to/application and args"; // make sure the application has executable permission. It it's a script, include the #!... stuff on the first line +// wait_for_completion = "no"; // set to "yes" to get Shairport Sync to wait until the "run_this..." applications have terminated before continuing +// allow_session_interruption = "no"; // set to "yes" to allow another device to interrupt Shairport Sync while it's playing from an existing audio source +// session_timeout = 120; // wait for this number of seconds after a source disappears before terminating the session and becoming available again. +}; + +// +// Back End Settings +// + +// These are parameters for the "alsa" audio back end, the only back end that supports synchronised audio. +alsa = +{ +// output_device = "default"; // the name of the alsa output device. Use "alsamixer" or "aplay" to find out the names of devices, mixers, etc. +// mixer_control_name = "PCM"; // the name of the mixer to use to adjust output volume. If not specified, volume in adjusted in software. +// mixer_device = "default"; // the mixer_device default is whatever the output_device is. Normally you wouldn't have to use this. +// audio_backend_latency_offset = 0; // Set this offset to compensate for a fixed delay in the audio back end. E.g. if the output device delays by 100 ms, set this to -4410. +// audio_backend_buffer_desired_length = 6615; // If set too small, buffer underflow occurs on low-powered machines. Too long and the response times with software mixer become annoying. +}; + +// These are parameters for the "pipe" audio back end, a back end that directs raw CD-style audio output to a pipe. No interpolation is done. +pipe = +{ +// name = "/path/to/pipe"; // there is no default pipe name for the output +// audio_backend_latency_offset = 0; // Set this offset to compensate for a fixed delay in the audio back end. E.g. if the output device delays by 100 ms, set this to -4410. +// audio_backend_buffer_desired_length = 44100; // Having started to send audio at the right time, send all subsequent audio this many frames ahead of time, creating a buffer this size. +}; + +// These are parameters for the "stdout" audio back end, a back end that directs raw CD-style audio output to stdout. No interpolation is done. +stdout = +{ +// audio_backend_latency_offset = 0; // Set this offset to compensate for a fixed delay in the audio back end. E.g. if the output device delays by 100 ms, set this to -4410. +// audio_backend_buffer_desired_length = 44100; // Having started to send audio at the right time, send all subsequent audio this many frames ahead of time, creating a buffer this size. +}; diff --git a/scripts/shairport-sync.service b/scripts/shairport-sync.service new file mode 100644 index 00000000..2cd9bd48 --- /dev/null +++ b/scripts/shairport-sync.service @@ -0,0 +1,13 @@ +[Unit] +Description=ShairportSync AirTunes receiver +After=sound.target +Requires=avahi-daemon.service +After=avahi-daemon.service + +[Service] +ExecStart=/usr/bin/shairport-sync +User=shairport-sync +Group=shairport-sync + +[Install] +WantedBy=multi-user.target diff --git a/shairport-sync.spec b/shairport-sync.spec new file mode 100644 index 00000000..5c1308e8 --- /dev/null +++ b/shairport-sync.spec @@ -0,0 +1,65 @@ +Name: shairport-sync +Version: 2.4 +Release: 1%{?dist} +Summary: AirTunes emulator. Shairport Sync adds multi-room capability with Audio Synchronisation + +Group: Applications/Multimedia +License: GPL +URL: https://github.com/mikebrady/shairport-sync +Source0: https://github.com/mikebrady/%{name}/archive/%{version}.tar.gz#/%{name}-%{version}.tar.gz + +BuildRequires: autoconf +BuildRequires: automake +BuildRequires: libconfig-devel +BuildRequires: popt-devel +BuildRequires: openssl-devel +BuildRequires: libdaemon-devel +BuildRequires: avahi-devel +BuildRequires: alsa-lib-devel +BuildRequires: systemd-units +BuildRequires: soxr-devel +Requires: popt +Requires: openssl +Requires: avahi +Requires: libdaemon +Requires: alsa-lib +Requires: soxr + +%description +Shairport Sync emulates an AirPort Express for the purpose of streaming audio from iTunes, iPods, iPhones, iPads and AppleTVs. Audio played by a Shairport Sync-powered device stays synchronised with the source and hence with similar devices playing the same source. Thus, for example, synchronised multi-room audio is possible without difficulty. (Hence the name Shairport Sync, BTW.) + +Shairport Sync does not support AirPlay video or photo streaming. + +%prep +%setup -q + +%build +autoreconf -i -f +%configure --with-avahi --with-alsa --with-ssl=openssl --with-soxr --with-systemd +make %{?_smp_mflags} + + +%install +make install DESTDIR=%{buildroot} +rm %{buildroot}/etc/shairport-sync.conf.sample + +%pre +getent group %{name} &>/dev/null || groupadd --system %{name} >/dev/null +getent passwd %{name} &> /dev/null || useradd --system -c "%{name} User" \ + -d %{_localstatedir}/%{name} -m -g %{name} -s /sbin/nologin \ + -G audio %{name} >/dev/null + +%files +%config /etc/shairport-sync.conf +/usr/bin/shairport-sync +/usr/share/man/man7/shairport-sync.7.gz +%{_unitdir}/%{name}.service +%doc AUTHORS LICENSES README.md + +%changelog +* Thu Aug 27 2015 Mike Brady 2.4 +- Prepare for stable release 2.4 +* Wed Aug 26 2015 Mike Brady 2.3.13.1-1 +- Harmonise release numbers +* Fri Jul 24 2015 Bill Peck 2.3.7-1 +- Initial spec file diff --git a/shairport.c b/shairport.c index e16b670c..e6a9ea0a 100644 --- a/shairport.c +++ b/shairport.c @@ -36,6 +36,8 @@ #include #include #include +#include +#include #include "config.h" @@ -58,485 +60,827 @@ #include #include - static int shutting_down = 0; +static char *appName = NULL; +char configuration_file_path[4096+1]; +char actual_configuration_file_path[4096+1]; void shairport_shutdown() { - if (shutting_down) - return; - shutting_down = 1; - mdns_unregister(); - rtsp_shutdown_stream(); - if (config.output) - config.output->deinit(); + if (shutting_down) + return; + shutting_down = 1; + mdns_unregister(); + rtsp_shutdown_stream(); + if (config.output) + config.output->deinit(); } -static void sig_ignore(int foo, siginfo_t *bar, void *baz) { -} +static void sig_ignore(int foo, siginfo_t *bar, void *baz) {} static void sig_shutdown(int foo, siginfo_t *bar, void *baz) { - debug(1, "shutdown requested..."); - shairport_shutdown(); - daemon_log(LOG_NOTICE, "exit..."); - daemon_retval_send(255); - daemon_pid_file_remove(); - exit(0); + debug(1, "shutdown requested..."); + shairport_shutdown(); + daemon_log(LOG_NOTICE, "exit..."); + daemon_retval_send(255); + daemon_pid_file_remove(); + exit(0); } static void sig_child(int foo, siginfo_t *bar, void *baz) { - pid_t pid; - while ((pid = waitpid((pid_t)-1, 0, WNOHANG)) > 0) { - if (pid == mdns_pid && !shutting_down) { - die("MDNS child process died unexpectedly!"); - } - } + pid_t pid; + while ((pid = waitpid((pid_t)-1, 0, WNOHANG)) > 0) { + if (pid == mdns_pid && !shutting_down) { + die("MDNS child process died unexpectedly!"); + } + } } static void sig_disconnect_audio_output(int foo, siginfo_t *bar, void *baz) { - debug(1,"disconnect audio output requested."); - set_requested_connection_state_to_output(0); + debug(1, "disconnect audio output requested."); + set_requested_connection_state_to_output(0); } static void sig_connect_audio_output(int foo, siginfo_t *bar, void *baz) { - debug(1,"connect audio output requested."); - set_requested_connection_state_to_output(1); + debug(1, "connect audio output requested."); + set_requested_connection_state_to_output(1); } void print_version(void) { char version_string[200]; - strcpy(version_string,PACKAGE_VERSION); + strcpy(version_string, PACKAGE_VERSION); #ifdef HAVE_LIBPOLARSSL - strcat(version_string,"-polarssl"); + strcat(version_string, "-polarssl"); #endif #ifdef HAVE_LIBSSL - strcat(version_string,"-openssl"); + strcat(version_string, "-openssl"); #endif #ifdef CONFIG_TINYSVCMDNS - strcat(version_string,"-tinysvcmdns"); + strcat(version_string, "-tinysvcmdns"); #endif #ifdef CONFIG_AVAHI - strcat(version_string,"-Avahi"); + strcat(version_string, "-Avahi"); #endif #ifdef CONFIG_DNS_SD - strcat(version_string,"-dns_sd"); + strcat(version_string, "-dns_sd"); #endif #ifdef CONFIG_ALSA - strcat(version_string,"-ALSA"); + strcat(version_string, "-ALSA"); #endif #ifdef CONFIG_SNDIO - strcat(version_string,"-sndio"); + strcat(version_string, "-sndio"); #endif #ifdef CONFIG_AO - strcat(version_string,"-ao"); + strcat(version_string, "-ao"); #endif #ifdef CONFIG_PULSE - strcat(version_string,"-pulse"); + strcat(version_string, "-pulse"); +#endif +#ifdef CONFIG_DUMMY + strcat(version_string, "-dummy"); +#endif +#ifdef CONFIG_STDOUT + strcat(version_string, "-stdout"); +#endif +#ifdef CONFIG_PIPE + strcat(version_string, "-pipe"); #endif #ifdef HAVE_LIBSOXR - strcat(version_string,"-soxr"); + strcat(version_string, "-soxr"); #endif - printf("%s\n",version_string); +#ifdef CONFIG_METADATA + strcat(version_string, "-metadata"); +#endif + printf("%s\n", version_string); } void usage(char *progname) { - printf("Usage: %s [options...]\n", progname); - printf(" or: %s [options...] -- [audio output-specific options]\n", progname); + printf("Usage: %s [options...]\n", progname); + printf(" or: %s [options...] -- [audio output-specific options]\n", progname); + printf("\n"); + printf("Options:\n"); + printf(" -h, --help show this help\n"); + printf(" -d, --daemon daemonise.\n"); + printf(" -V, --version show version information\n"); + printf(" -k, --kill kill the existing shairport daemon.\n"); + printf(" -D, --disconnectFromOutput disconnect immediately from the output device.\n"); + printf(" -R, --reconnectToOutput reconnect to the output device.\n"); + printf(" -c, --configfile=FILE read configuration settings from FILE. Default is " + "/etc/shairport-sync.conf.\n"); - printf("\n"); - printf("Mandatory arguments to long options are mandatory for short options too.\n"); - - printf("\n"); - printf("Options:\n"); - printf(" -h, --help show this help\n"); - printf(" -V, --version show version information\n"); - printf(" -v, --verbose -v print debug information; -vv more; -vvv lots\n"); - printf(" -p, --port=PORT set RTSP listening port\n"); - printf(" -a, --name=NAME set advertised name\n"); - printf(" -A, --AirPlayLatency=FRAMES set the latency for audio sent from an AirPlay device.\n"); - printf(" The default value is %u frames.\n", config.AirPlayLatency); - printf(" -i, --iTunesLatency=FRAMES set the latency for audio sent from iTunes 10 or later.\n"); - printf(" The default value is %u frames.\n", config.iTunesLatency); - printf(" -L, --latency=FRAMES set the latency for audio sent from an unknown device\n"); - printf(" or from an old version of iTunes. Default is %d frames.\n",config.latency); - printf(" --forkedDaapdLatency=FRAMES set the latency for audio sent from forked-daapd.\n"); - printf(" -S, --stuffing=MODE set how to adjust current latency to match desired latency \n"); - printf(" \"basic\" (default) inserts or deletes audio frames from packet frames with low processor overhead.\n"); - printf(" \"soxr\" uses libsoxr to minimally resample packet frames -- moderate processor overhead.\n"); - printf(" \"soxr\" option only available if built with soxr support.\n"); - printf(" -d, --daemon daemonise.\n"); - printf(" -k, --kill kill the existing shairport daemon.\n"); - printf(" -D, --disconnectFromOutput disconnect immediately from the output device.\n"); - printf(" -R, --reconnectToOutput reconnect to the output device.\n"); - printf(" -B, --on-start=PROGRAM run PROGRAM when playback is about to begin.\n"); - printf(" -E, --on-stop=PROGRAM run PROGRAM when playback has ended.\n"); - printf(" For -B and -E options, specify the full path to the program, e.g. /usr/bin/logger.\n"); - printf(" Executable scripts work, but must have #!/bin/sh (or whatever) in the headline.\n"); - printf(" -w, --wait-cmd wait until the -B or -E programs finish before continuing\n"); - printf(" -o, --output=BACKEND select audio output method\n"); - printf(" -m, --mdns=BACKEND force the use of BACKEND to advertize the service\n"); - printf(" if no mdns provider is specified,\n"); - printf(" shairport tries them all until one works.\n"); - printf(" -r, --resync=THRESHOLD resync if error exceeds this number of frames. Set to 0 to stop resyncing.\n"); - printf(" -t, --timeout=SECONDS go back to idle mode from play mode after a break in communications of this many seconds (default 120). Set to 0 never to exit play mode.\n"); - printf(" --statistics print some interesting statistics -- output to the logfile if running as a daemon.\n"); - printf(" --tolerance=TOLERANCE allow a synchronization error of TOLERANCE frames (default 88) before trying to correct it.\n"); - printf(" --password=PASSWORD require PASSWORD to connect. Default is not to require a password.\n"); - printf("\n"); - mdns_ls_backends(); - printf("\n"); - audio_ls_outputs(); + printf(" -v, --verbose -v print debug information; -vv more; -vvv lots\n"); + printf(" -p, --port=PORT set RTSP listening port\n"); + printf(" -a, --name=NAME set advertised name\n"); + printf( + " -A, --AirPlayLatency=FRAMES set the latency for audio sent from an AirPlay device.\n"); + printf(" The default value is %u frames.\n", config.AirPlayLatency); + printf( + " -i, --iTunesLatency=FRAMES set the latency for audio sent from iTunes 10 or later.\n"); + printf(" The default value is %u frames.\n", config.iTunesLatency); + printf(" -L, --latency=FRAMES set the latency for audio sent from an unknown device\n"); + printf(" or from an old version of iTunes. Default is %d frames.\n", + config.latency); + printf(" --forkedDaapdLatency=FRAMES set the latency for audio sent from forked-daapd.\n"); + printf(" -S, --stuffing=MODE set how to adjust current latency to match desired latency \n"); + printf(" \"basic\" (default) inserts or deletes audio frames from " + "packet frames with low processor overhead.\n"); + printf(" \"soxr\" uses libsoxr to minimally resample packet frames -- " + "moderate processor overhead.\n"); + printf( + " \"soxr\" option only available if built with soxr support.\n"); + printf(" -B, --on-start=PROGRAM run PROGRAM when playback is about to begin.\n"); + printf(" -E, --on-stop=PROGRAM run PROGRAM when playback has ended.\n"); + printf(" For -B and -E options, specify the full path to the program, " + "e.g. /usr/bin/logger.\n"); + printf(" Executable scripts work, but must have #!/bin/sh (or " + "whatever) in the headline.\n"); + printf(" -w, --wait-cmd wait until the -B or -E programs finish before continuing\n"); + printf(" -o, --output=BACKEND select audio output method\n"); + printf(" -m, --mdns=BACKEND force the use of BACKEND to advertize the service\n"); + printf(" if no mdns provider is specified,\n"); + printf(" shairport tries them all until one works.\n"); + printf(" -r, --resync=THRESHOLD resync if error exceeds this number of frames. Set to 0 to " + "stop resyncing.\n"); + printf(" -t, --timeout=SECONDS go back to idle mode from play mode after a break in " + "communications of this many seconds (default 120). Set to 0 never to exit play mode.\n"); + printf(" --statistics print some interesting statistics -- output to the logfile " + "if running as a daemon.\n"); + printf(" --tolerance=TOLERANCE allow a synchronization error of TOLERANCE frames (default " + "88) before trying to correct it.\n"); + printf(" --password=PASSWORD require PASSWORD to connect. Default is not to require a " + "password.\n"); +#ifdef CONFIG_METADATA + printf(" --metadata-pipename=PIPE send metadata to PIPE, e.g. " + "--metadata-pipename=/tmp/shairport-sync-metadata.\n"); + printf(" --get-coverart send cover art through the metadata pipe.\n"); +#endif + printf("\nGeneral options can be configured in /etc/%s.conf.\n", appName); + printf("\n"); + mdns_ls_backends(); + printf("\n"); + audio_ls_outputs(); } int parse_options(int argc, char **argv) { - signed char c; /* used for argument parsing */ - int i = 0; /* used for tracking options */ - char *stuffing = NULL; /* used for picking up the stuffing option */ - poptContext optCon; /* context for parsing command-line options */ + char *stuffing = NULL; /* used for picking up the stuffing option */ + signed char c; /* used for argument parsing */ + int i = 0; /* used for tracking options */ + poptContext optCon; /* context for parsing command-line options */ struct poptOption optionsTable[] = { - { "statistics", 0, POPT_ARG_NONE, &config.statistics_requested, 0, NULL}, - { "version", 'V', POPT_ARG_NONE, NULL, 0, NULL}, - { "verbose", 'v', POPT_ARG_NONE, NULL, 'v', NULL }, - { "daemon", 'd', POPT_ARG_NONE, &config.daemonise, 0, NULL }, - { "disconnectFromOutput", 'D', POPT_ARG_NONE, NULL, 0, NULL }, - { "reconnectToOutput", 'R', POPT_ARG_NONE, NULL, 0, NULL }, - { "kill", 'k', POPT_ARG_NONE, NULL, 0, NULL }, - { "port", 'p', POPT_ARG_INT, &config.port, 0, NULL } , - { "name", 'a', POPT_ARG_STRING, &config.apname, 0, NULL } , - { "output", 'o', POPT_ARG_STRING, &config.output_name, 0, NULL } , - { "on-start", 'B', POPT_ARG_STRING, &config.cmd_start, 0, NULL } , - { "on-stop", 'E', POPT_ARG_STRING, &config.cmd_stop, 0, NULL } , - { "wait-cmd", 'w', POPT_ARG_NONE, &config.cmd_blocking, 0, NULL } , - { "mdns", 'm', POPT_ARG_STRING, &config.mdns_name, 0, NULL } , - { "latency", 'L', POPT_ARG_INT, &config.userSuppliedLatency, 0, NULL } , - { "AirPlayLatency", 'A', POPT_ARG_INT, &config.AirPlayLatency, 0, NULL } , - { "iTunesLatency", 'i', POPT_ARG_INT, &config.iTunesLatency, 0, NULL } , - { "forkedDaapdLatency", 0, POPT_ARG_INT, &config.ForkedDaapdLatency, 0, NULL } , - { "stuffing", 'S', POPT_ARG_STRING, &stuffing, 'S', NULL } , - { "resync", 'r', POPT_ARG_INT, &config.resyncthreshold, 0, NULL } , - { "timeout", 't', POPT_ARG_INT, &config.timeout, 0, NULL } , - { "password", 0, POPT_ARG_STRING, &config.password, 0, NULL } , - { "tolerance", 0, POPT_ARG_INT, &config.tolerance, 0, NULL } , - POPT_AUTOHELP - { NULL, 0, 0, NULL, 0 } - }; - - int optind=argc; - int j; - for (j=0;j= 0) { switch (c) { - case 'v': - debuglev++; - break; - case 'S': - if (strcmp(stuffing,"basic")==0) - config.packet_stuffing = ST_basic; - else if (strcmp(stuffing,"soxr")==0) -#ifdef HAVE_LIBSOXR - config.packet_stuffing = ST_soxr; -#else - die("soxr option not available -- this version of shairport-sync was built without libsoxr support"); -#endif - else - die("Illegal stuffing option \"%s\" -- must be \"basic\" or \"soxr\"",stuffing); - break; + case 'v': + debuglev++; + break; } } if (c < -1) { - die("%s: %s",poptBadOption(optCon, POPT_BADOPTION_NOALIAS),poptStrerror(c)); + die("%s: %s", poptBadOption(optCon, POPT_BADOPTION_NOALIAS), poptStrerror(c)); } - /* Print out options */ + + config_setting_t *setting; + const char *str; + int value; + + debug(1,"Looking for the configuration file \"%s\".",config.configfile); - debug(2,"statistics_requester status is %d.",config.statistics_requested); - debug(2,"daemon status is %d.",config.daemonise); - debug(2,"rtsp listening port is %d.",config.port); - debug(2,"Shairport Sync player name is \"%s\".",config.apname); - debug(2,"Audio Output name is \"%s\".",config.output_name); - debug(2,"on-start action is \"%s\".",config.cmd_start); - debug(2,"on-stop action is \"%s\".",config.cmd_stop); - debug(2,"wait-cmd status is %d.",config.cmd_blocking); - debug(2,"mdns backend \"%s\".",config.mdns_name); - debug(2,"latency is %d.",config.userSuppliedLatency); - debug(2,"AirPlayLatency is %d.",config.AirPlayLatency); - debug(2,"iTunesLatency is %d.",config.iTunesLatency); - debug(2,"forkedDaapdLatency is %d.",config.ForkedDaapdLatency); - debug(2,"stuffing option is \"%s\".",stuffing); - debug(2,"resync time is %d.",config.resyncthreshold); - debug(2,"busy timeout time is %d.",config.timeout); - debug(2,"tolerance is %d frames.",config.tolerance); - debug(2,"password is \"%s\".",config.password); - return optind+1; + config_init(&config_file_stuff); + + char *config_file_real_path = realpath(config.configfile, NULL); + if (config_file_real_path==NULL) { + debug(2,"Can't resolve the configuration file \"%s\".",config.configfile); + } else { + debug(2, "Looking for configuration file at full path \"%s\"", config_file_real_path); + /* Read the file. If there is an error, report it and exit. */ + if (config_read_file(&config_file_stuff, config_file_real_path)) { + // make config.cfg point to it + config.cfg = &config_file_stuff; + /* Get the Service Name. */ + if (config_lookup_string(config.cfg, "general.name", &str)) + config.apname = (char *)str; + + /* Get the Daemonize setting. */ + if (config_lookup_string(config.cfg, "general.daemonize", &str)) { + if (strcasecmp(str, "no") == 0) + config.daemonise = 0; + else if (strcasecmp(str, "yes") == 0) + config.daemonise = 1; + else + die("Invalid daemonize option choice \"%s\". It should be \"yes\" or \"no\""); + } + + /* Get the mdns_backend setting. */ + if (config_lookup_string(config.cfg, "general.mdns_backend", &str)) + config.mdns_name = (char *)str; + + /* Get the output_backend setting. */ + if (config_lookup_string(config.cfg, "general.output_backend", &str)) + config.output_name = (char *)str; + + /* Get the port setting. */ + if (config_lookup_int(config.cfg, "general.port", &value)) { + if ((value < 0) || (value > 65535)) + die("Invalid port number \"%sd\". It should be between 0 and 65535, default is 5000", + value); + else + config.port = value; + } + + /* Get the udp port base setting. */ + if (config_lookup_int(config.cfg, "general.udp_port_base", &value)) { + if ((value < 0) || (value > 65535)) + die("Invalid port number \"%sd\". It should be between 0 and 65535, default is 6001", + value); + else + config.udp_port_base = value; + } + + /* Get the udp port range setting. This is number of ports that will be tried for free ports , starting at the port base. Only three ports are needed. */ + if (config_lookup_int(config.cfg, "general.udp_port_range", &value)) { + if ((value < 0) || (value > 65535)) + die("Invalid port range \"%sd\". It should be between 0 and 65535, default is 100", + value); + else + config.udp_port_range = value; + } + + /* Get the password setting. */ + if (config_lookup_string(config.cfg, "general.password", &str)) + config.password = (char *)str; + + if (config_lookup_string(config.cfg, "general.interpolation", &str)) { + if (strcasecmp(str, "basic") == 0) + config.packet_stuffing = ST_basic; + else if (strcasecmp(str, "soxr") == 0) + config.packet_stuffing = ST_soxr; + else + die("Invalid interpolation option choice \"%s\". It should be \"basic\" or \"soxr\""); + } + + /* Get the statistics setting. */ + if (config_lookup_string(config.cfg, "general.statistics", &str)) { + if (strcasecmp(str, "no") == 0) + config.statistics_requested = 0; + else if (strcasecmp(str, "yes") == 0) + config.statistics_requested = 1; + else + die("Invalid statistics option choice \"%s\". It should be \"yes\" or \"no\""); + } + + /* Get the drift tolerance setting. */ + if (config_lookup_int(config.cfg, "general.drift", &value)) + config.tolerance = value; + + /* Get the resync setting. */ + if (config_lookup_int(config.cfg, "general.resync_threshold", &value)) + config.resyncthreshold = value; + + /* Get the verbosity setting. */ + if (config_lookup_int(config.cfg, "general.log_verbosity", &value)) + if ((value >= 0) && (value <= 3)) + debuglev = value; + else + die("Invalid log verbosity setting option choice \"%d\". It should be between 0 and 3, " + "inclusive.", + value); + + /* Get the ignore_volume_control setting. */ + if (config_lookup_string(config.cfg, "general.ignore_volume_control", &str)) { + if (strcasecmp(str, "no") == 0) + config.ignore_volume_control = 0; + else if (strcasecmp(str, "yes") == 0) + config.ignore_volume_control = 1; + else + die("Invalid ignore_volume_control option choice \"%s\". It should be \"yes\" or \"no\""); + } + + /* Get the default latency. */ + if (config_lookup_int(config.cfg, "latencies.default", &value)) + config.latency = value; + + /* Get the itunes latency. */ + if (config_lookup_int(config.cfg, "latencies.itunes", &value)) + config.iTunesLatency = value; + + /* Get the AirPlay latency. */ + if (config_lookup_int(config.cfg, "latencies.airplay", &value)) + config.AirPlayLatency = value; + + /* Get the forkedDaapd latency. */ + if (config_lookup_int(config.cfg, "latencies.forkedDaapd", &value)) + config.ForkedDaapdLatency = value; + + #ifdef CONFIG_METADATA + /* Get the metadata setting. */ + if (config_lookup_string(config.cfg, "metadata.enabled", &str)) { + if (strcasecmp(str, "no") == 0) + config.metadata_enabled = 0; + else if (strcasecmp(str, "yes") == 0) + config.metadata_enabled = 1; + else + die("Invalid metadata enabled option choice \"%s\". It should be \"yes\" or \"no\""); + } + + if (config_lookup_string(config.cfg, "metadata.include_cover_art", &str)) { + if (strcasecmp(str, "no") == 0) + config.get_coverart = 0; + else if (strcasecmp(str, "yes") == 0) + config.get_coverart = 1; + else + die("Invalid metadata include_cover_art option choice \"%s\". It should be \"yes\" or " + "\"no\""); + } + + if (config_lookup_string(config.cfg, "metadata.pipe_name", &str)) { + config.metadata_pipename = (char *)str; + } + #endif + + if (config_lookup_string(config.cfg, "sessioncontrol.run_this_before_play_begins", &str)) { + config.cmd_start = (char *)str; + } + + if (config_lookup_string(config.cfg, "sessioncontrol.run_this_after_play_ends", &str)) { + config.cmd_stop = (char *)str; + } + + if (config_lookup_string(config.cfg, "sessioncontrol.wait_for_completion", &str)) { + if (strcasecmp(str, "no") == 0) + config.cmd_blocking = 0; + else if (strcasecmp(str, "yes") == 0) + config.cmd_blocking = 1; + else + die("Invalid session control wait_for_completion option choice \"%s\". It should be " + "\"yes\" or \"no\""); + } + + if (config_lookup_string(config.cfg, "sessioncontrol.allow_session_interruption", &str)) { + config.dont_check_timeout = 0; // this is for legacy -- only set by -t 0 + if (strcasecmp(str, "no") == 0) + config.allow_session_interruption = 0; + else if (strcasecmp(str, "yes") == 0) + config.allow_session_interruption = 1; + else + die("Invalid session control allow_interruption option choice \"%s\". It should be \"yes\" " + "or \"no\""); + } + + if (config_lookup_int(config.cfg, "sessioncontrol.session_timeout", &value)) { + config.timeout = value; + config.dont_check_timeout = 0; // this is for legacy -- only set by -t 0 + } + + } else { + if (config_error_type(&config_file_stuff) == CONFIG_ERR_FILE_IO) + debug(1, "Error reading configuration file \"%s\": \"%s\".", + config_error_file(&config_file_stuff), config_error_text(&config_file_stuff)); + else { + die("Line %d of the configuration file \"%s\":\n%s", config_error_line(&config_file_stuff), + config_error_file(&config_file_stuff), config_error_text(&config_file_stuff)); + } + } + free(config_file_real_path); + } + +// now, do the command line options again, but this time do them fully -- it's a unix convention that command line +// arguments have precedence over configuration file settings. + + optind = argc; + for (j = 0; j < argc; j++) + if (strcmp(argv[j], "--") == 0) + optind = j; + + optCon = poptGetContext(NULL, optind, (const char **)argv, optionsTable, 0); + poptSetOtherOptionHelp(optCon, "[OPTIONS]* "); + + /* Now do options processing, get portname */ + int tdebuglev = 0; + while ((c = poptGetNextOpt(optCon)) >= 0) { + switch (c) { + case 'v': + tdebuglev++; + break; + case 't': + if (config.timeout == 0) { + config.dont_check_timeout = 1; + config.allow_session_interruption = 1; + } else { + config.dont_check_timeout = 0; + config.allow_session_interruption = 0; + } + break; +#ifdef CONFIG_METADATA + case 'M': + config.metadata_enabled = 1; + break; + case 'g': + if (config.metadata_enabled == 0) + die("If you want to get cover art, you must also select the --metadata-pipename option."); + break; +#endif + case 'S': + if (strcmp(stuffing, "basic") == 0) + config.packet_stuffing = ST_basic; + else if (strcmp(stuffing, "soxr") == 0) +#ifdef HAVE_LIBSOXR + config.packet_stuffing = ST_soxr; +#else + die("soxr option not available -- this version of shairport-sync was built without libsoxr " + "support"); +#endif + else + die("Illegal stuffing option \"%s\" -- must be \"basic\" or \"soxr\"", stuffing); + break; + } + } + if (c < -1) { + die("%s: %s", poptBadOption(optCon, POPT_BADOPTION_NOALIAS), poptStrerror(c)); + } + if (tdebuglev!=0) + debuglev = tdebuglev; + return optind + 1; } void signal_setup(void) { - // mask off all signals before creating threads. - // this way we control which thread gets which signals. - // for now, we don't care which thread gets the following. - sigset_t set; - sigfillset(&set); - sigdelset(&set, SIGINT); - sigdelset(&set, SIGTERM); - sigdelset(&set, SIGHUP); - sigdelset(&set, SIGSTOP); - sigdelset(&set, SIGCHLD); - sigdelset(&set, SIGUSR2); - pthread_sigmask(SIG_BLOCK, &set, NULL); + // mask off all signals before creating threads. + // this way we control which thread gets which signals. + // for now, we don't care which thread gets the following. + sigset_t set; + sigfillset(&set); + sigdelset(&set, SIGINT); + sigdelset(&set, SIGTERM); + sigdelset(&set, SIGHUP); + sigdelset(&set, SIGSTOP); + sigdelset(&set, SIGCHLD); + sigdelset(&set, SIGUSR2); + pthread_sigmask(SIG_BLOCK, &set, NULL); - // setting this to SIG_IGN would prevent signalling any threads. - struct sigaction sa; - memset(&sa, 0, sizeof(sa)); - sa.sa_flags = SA_SIGINFO; - sa.sa_sigaction = &sig_ignore; - sigaction(SIGUSR1, &sa, NULL); + // setting this to SIG_IGN would prevent signalling any threads. + struct sigaction sa; + memset(&sa, 0, sizeof(sa)); + sa.sa_flags = SA_SIGINFO; + sa.sa_sigaction = &sig_ignore; + sigaction(SIGUSR1, &sa, NULL); - sa.sa_flags = SA_SIGINFO | SA_RESTART; - sa.sa_sigaction = &sig_shutdown; - sigaction(SIGINT, &sa, NULL); - sigaction(SIGTERM, &sa, NULL); + sa.sa_flags = SA_SIGINFO | SA_RESTART; + sa.sa_sigaction = &sig_shutdown; + sigaction(SIGINT, &sa, NULL); + sigaction(SIGTERM, &sa, NULL); - sa.sa_sigaction = &sig_disconnect_audio_output; - sigaction(SIGUSR2, &sa, NULL); + sa.sa_sigaction = &sig_disconnect_audio_output; + sigaction(SIGUSR2, &sa, NULL); - sa.sa_sigaction = &sig_connect_audio_output; - sigaction(SIGHUP, &sa, NULL); + sa.sa_sigaction = &sig_connect_audio_output; + sigaction(SIGHUP, &sa, NULL); - sa.sa_sigaction = &sig_child; - sigaction(SIGCHLD, &sa, NULL); + sa.sa_sigaction = &sig_child; + sigaction(SIGCHLD, &sa, NULL); } // forked daemon lets the spawner know it's up and running OK // should be called only once! void shairport_startup_complete(void) { - if (config.daemonise) { -// daemon_ready(); - } + if (config.daemonise) { + // daemon_ready(); + } } #ifdef USE_CUSTOM_PID_DIR const char *pid_file_proc(void) { #ifdef HAVE_ASPRINTF - static char *fn = NULL; - free(fn); - asprintf(&fn, "%s/%s.pid", PIDDIR, daemon_pid_file_ident ? daemon_pid_file_ident : "unknown"); + static char *fn = NULL; + free(fn); + asprintf(&fn, "%s/%s.pid", PIDDIR, daemon_pid_file_ident ? daemon_pid_file_ident : "unknown"); #else - static char fn[8192]; - snprintf(fn, sizeof(fn), "%s/%s.pid", PIDDIR, daemon_pid_file_ident ? daemon_pid_file_ident : "unknown"); + static char fn[8192]; + snprintf(fn, sizeof(fn), "%s/%s.pid", PIDDIR, + daemon_pid_file_ident ? daemon_pid_file_ident : "unknown"); #endif - return fn; + return fn; } #endif +void exit_function() { + if (config.cfg) + config_destroy(config.cfg); +} + int main(int argc, char **argv) { - daemon_set_verbosity(LOG_DEBUG); + daemon_set_verbosity(LOG_DEBUG); + memset(&config, 0, sizeof(config)); // also clears all strings, BTW + atexit(exit_function); - memset(&config, 0, sizeof(config)); // also clears all strings, BTW + // this is a bit weird, but apparently necessary + char *basec = strdup(argv[0]); + char *bname = basename(basec); + appName = strdup(bname); + free(basec); - // set defaults - config.statistics_requested - 0; // don't print stats in the log - config.latency = 88200; // AirPlay. Is also reset in rtsp.c when play is about to start - config.userSuppliedLatency = 0; // zero means none supplied - config.iTunesLatency = 99400; // this seems to work pretty well for iTunes from Version 10 (?) upwards-- two left-ear headphones, one from the iMac jack, one from an NSLU2 running a cheap "3D Sound" USB Soundcard - config.AirPlayLatency = 88200; // this seems to work pretty well for AirPlay -- Syncs sound and vision on AppleTV, but also used for iPhone/iPod/iPad sources - config.ForkedDaapdLatency = 99400; // Seems to be right - config.resyncthreshold = 441*5; // this number of frames is 50 ms - config.timeout = 120; // this number of seconds to wait for [more] audio before switching to idle. - config.tolerance = 88; // this number of frames of error before attempting to correct it. - config.buffer_start_fill = 220; - config.port = 5000; - config.packet_stuffing = ST_basic; // simple interpolation or deletion - char hostname[100]; - gethostname(hostname, 100); - config.apname = malloc(20 + 100); - snprintf(config.apname, 20 + 100, "Shairport Sync on %s", hostname); - set_requested_connection_state_to_output(1); // we expect to be able to connect to the output device - - /* Check if we are called with -V or --version parameter */ - if (argc >= 2 && ((strcmp(argv[1], "-V")==0) || (strcmp(argv[1], "--version")==0))) { - print_version(); - exit(1); - } + // set defaults + strcpy(configuration_file_path, "/etc/"); + strcat(configuration_file_path, appName); + strcat(configuration_file_path, ".conf"); + config.configfile = configuration_file_path; - /* Check if we are called with -h or --help parameter */ - if (argc >= 2 && ((strcmp(argv[1], "-h")==0) || (strcmp(argv[1], "--help")==0))) { - usage(argv[0]); - exit(1); - } + config.statistics_requested - 0; // don't print stats in the log + config.latency = 88200; // AirPlay. Is also reset in rtsp.c when play is about to start + config.userSuppliedLatency = 0; // zero means none supplied + config.iTunesLatency = 99400; // this seems to work pretty well for iTunes from Version 10 (?) + // upwards-- two left-ear headphones, one from the iMac jack, one + // from an NSLU2 running a cheap "3D Sound" USB Soundcard + config.AirPlayLatency = 88200; // this seems to work pretty well for AirPlay -- Syncs sound and + // vision on AppleTV, but also used for iPhone/iPod/iPad sources + config.ForkedDaapdLatency = 99400; // Seems to be right + config.resyncthreshold = 441 * 5; // this number of frames is 50 ms + config.timeout = 120; // this number of seconds to wait for [more] audio before switching to idle. + config.tolerance = 88; // this number of frames of error before attempting to correct it. + config.buffer_start_fill = 220; + config.port = 5000; + config.packet_stuffing = ST_basic; // simple interpolation or deletion + char hostname[100]; + gethostname(hostname, 100); + config.apname = malloc(20 + 100); + snprintf(config.apname, 20 + 100, "Shairport Sync on %s", hostname); + set_requested_connection_state_to_output( + 1); // we expect to be able to connect to the output device + config.audio_backend_buffer_desired_length = 6615; // 0.15 seconds. + config.udp_port_base = 6001; + config.udp_port_range = 100; - pid_t pid; + /* Check if we are called with -V or --version parameter */ + if (argc >= 2 && ((strcmp(argv[1], "-V") == 0) || (strcmp(argv[1], "--version") == 0))) { + print_version(); + exit(1); + } - /* Reset signal handlers */ - if (daemon_reset_sigs(-1) < 0) { - daemon_log(LOG_ERR, "Failed to reset all signal handlers: %s", strerror(errno)); - return 1; - } + /* Check if we are called with -h or --help parameter */ + if (argc >= 2 && ((strcmp(argv[1], "-h") == 0) || (strcmp(argv[1], "--help") == 0))) { + usage(argv[0]); + exit(1); + } + + pid_t pid; + + /* Reset signal handlers */ + if (daemon_reset_sigs(-1) < 0) { + daemon_log(LOG_ERR, "Failed to reset all signal handlers: %s", strerror(errno)); + return 1; + } + + /* Unblock signals */ + if (daemon_unblock_sigs(-1) < 0) { + daemon_log(LOG_ERR, "Failed to unblock all signals: %s", strerror(errno)); + return 1; + } - /* Unblock signals */ - if (daemon_unblock_sigs(-1) < 0) { - daemon_log(LOG_ERR, "Failed to unblock all signals: %s", strerror(errno)); - return 1; - } - - #if USE_CUSTOM_LOCAL_STATE_DIR - debug(1,"Locating localstatedir at \"%s\"",LOCALSTATEDIR); - /* Point to a function to help locate where the PID file will go */ - daemon_pid_file_proc = pid_file_proc; + debug(1, "Locating localstatedir at \"%s\"", LOCALSTATEDIR); + /* Point to a function to help locate where the PID file will go */ + daemon_pid_file_proc = pid_file_proc; #endif - /* Set indentification string for the daemon for both syslog and PID file */ - daemon_pid_file_ident = daemon_log_ident = daemon_ident_from_argv0(argv[0]); - - /* Check if we are called with -D or --disconnectFromOutput parameter */ - if (argc >= 2 && ((strcmp(argv[1], "-D")==0) || (strcmp(argv[1], "--disconnectFromOutput")==0))) { - if ((pid = daemon_pid_file_is_running()) >= 0) { - if (kill(pid,SIGUSR2)!=0) { // try to send the signal - daemon_log(LOG_WARNING, "Failed trying to send disconnectFromOutput command to daemon pid: %d: %s",pid, strerror(errno)); - } - } else { - daemon_log(LOG_WARNING, "Can't send a disconnectFromOutput request -- Failed to find daemon: %s", strerror(errno)); + /* Set indentification string for the daemon for both syslog and PID file */ + daemon_pid_file_ident = daemon_log_ident = daemon_ident_from_argv0(argv[0]); + + /* Check if we are called with -D or --disconnectFromOutput parameter */ + if (argc >= 2 && + ((strcmp(argv[1], "-D") == 0) || (strcmp(argv[1], "--disconnectFromOutput") == 0))) { + if ((pid = daemon_pid_file_is_running()) >= 0) { + if (kill(pid, SIGUSR2) != 0) { // try to send the signal + daemon_log(LOG_WARNING, + "Failed trying to send disconnectFromOutput command to daemon pid: %d: %s", pid, + strerror(errno)); } - exit(1); + } else { + daemon_log(LOG_WARNING, + "Can't send a disconnectFromOutput request -- Failed to find daemon: %s", + strerror(errno)); } - - /* Check if we are called with -R or --reconnectToOutput parameter */ - if (argc >= 2 && ((strcmp(argv[1], "-R")==0) || (strcmp(argv[1], "--reconnectToOutput")==0))) { - if ((pid = daemon_pid_file_is_running()) >= 0) { - if (kill(pid,SIGHUP)!=0) { // try to send the signal - daemon_log(LOG_WARNING, "Failed trying to send reconnectToOutput command to daemon pid: %d: %s",pid, strerror(errno)); - } - } else { - daemon_log(LOG_WARNING, "Can't send a reconnectToOutput request -- Failed to find daemon: %s", strerror(errno)); + exit(1); + } + + /* Check if we are called with -R or --reconnectToOutput parameter */ + if (argc >= 2 && + ((strcmp(argv[1], "-R") == 0) || (strcmp(argv[1], "--reconnectToOutput") == 0))) { + if ((pid = daemon_pid_file_is_running()) >= 0) { + if (kill(pid, SIGHUP) != 0) { // try to send the signal + daemon_log(LOG_WARNING, + "Failed trying to send reconnectToOutput command to daemon pid: %d: %s", pid, + strerror(errno)); } - exit(1); + } else { + daemon_log(LOG_WARNING, "Can't send a reconnectToOutput request -- Failed to find daemon: %s", + strerror(errno)); } - - /* Check if we are called with -k or --kill parameter */ - if (argc >= 2 && ((strcmp(argv[1], "-k")==0) || (strcmp(argv[1], "--kill")==0))) { + exit(1); + } + + /* Check if we are called with -k or --kill parameter */ + if (argc >= 2 && ((strcmp(argv[1], "-k") == 0) || (strcmp(argv[1], "--kill") == 0))) { + int ret; + + /* Kill daemon with SIGTERM */ + /* Check if the new function daemon_pid_file_kill_wait() is available, if it is, use it. */ + if ((ret = daemon_pid_file_kill_wait(SIGTERM, 5)) < 0) + daemon_log(LOG_WARNING, "Failed to kill daemon: %s", strerror(errno)); + else + daemon_pid_file_remove(); + return ret < 0 ? 1 : 0; + } + + /* Check that the daemon is not running twice at the same time */ + if ((pid = daemon_pid_file_is_running()) >= 0) { + daemon_log(LOG_ERR, "Daemon already running on PID file %u", pid); + return 1; + } + + // parse arguments into config + int audio_arg = parse_options(argc, argv); + + // mDNS supports maximum of 63-character names (we append 13). + if (strlen(config.apname) > 50) + die("Supplied name too long (max 50 characters)"); + + /* here, daemonise with libdaemon */ + + if (config.daemonise) { + /* Prepare for return value passing from the initialization procedure of the daemon process */ + if (daemon_retval_init() < 0) { + daemon_log(LOG_ERR, "Failed to create pipe."); + return 1; + } + + /* Do the fork */ + if ((pid = daemon_fork()) < 0) { + + /* Exit on error */ + daemon_retval_done(); + return 1; + + } else if (pid) { /* The parent */ int ret; - /* Kill daemon with SIGTERM */ - /* Check if the new function daemon_pid_file_kill_wait() is available, if it is, use it. */ - if ((ret = daemon_pid_file_kill_wait(SIGTERM, 5)) < 0) - daemon_log(LOG_WARNING, "Failed to kill daemon: %s", strerror(errno)); - else - daemon_pid_file_remove(); - return ret < 0 ? 1 : 0; - } - - /* Check that the daemon is not running twice at the same time */ - if ((pid = daemon_pid_file_is_running()) >= 0) { - daemon_log(LOG_ERR, "Daemon already running on PID file %u", pid); - return 1; - } - - // parse arguments into config - int audio_arg = parse_options(argc, argv); - - // mDNS supports maximum of 63-character names (we append 13). - if (strlen(config.apname) > 50) - die("Supplied name too long (max 50 characters)"); - - /* here, daemonise with libdaemon */ - - if (config.daemonise) { - /* Prepare for return value passing from the initialization procedure of the daemon process */ - if (daemon_retval_init() < 0) { - daemon_log(LOG_ERR, "Failed to create pipe."); - return 1; + /* Wait for 20 seconds for the return value passed from the daemon process */ + if ((ret = daemon_retval_wait(20)) < 0) { + daemon_log(LOG_ERR, "Could not receive return value from daemon process: %s", + strerror(errno)); + return 255; } - /* Do the fork */ - if ((pid = daemon_fork()) < 0) { + if (ret != 0) + daemon_log(ret != 0 ? LOG_ERR : LOG_INFO, "Daemon returned %i as return value.", ret); + return ret; - /* Exit on error */ - daemon_retval_done(); - return 1; + } else { /* The daemon */ - } else if (pid) { /* The parent */ - int ret; + /* Close FDs */ + if (daemon_close_all(-1) < 0) { + daemon_log(LOG_ERR, "Failed to close all file descriptors: %s", strerror(errno)); - /* Wait for 20 seconds for the return value passed from the daemon process */ - if ((ret = daemon_retval_wait(20)) < 0) { - daemon_log(LOG_ERR, "Could not receive return value from daemon process: %s", strerror(errno)); - return 255; - } - - if (ret != 0) - daemon_log(ret != 0 ? LOG_ERR : LOG_INFO, "Daemon returned %i as return value.", ret); - return ret; - - } else { /* The daemon */ - - /* Close FDs */ - if (daemon_close_all(-1) < 0) { - daemon_log(LOG_ERR, "Failed to close all file descriptors: %s", strerror(errno)); - - /* Send the error condition to the parent process */ - daemon_retval_send(1); - goto finish; - } - - /* Create the PID file */ - if (daemon_pid_file_create() < 0) { - daemon_log(LOG_ERR, "Could not create PID file (%s).", strerror(errno)); - daemon_retval_send(2); - goto finish; - } - - /* Send OK to parent process */ - daemon_retval_send(0); + /* Send the error condition to the parent process */ + daemon_retval_send(1); + goto finish; } - /* end libdaemon stuff */ + + /* Create the PID file */ + if (daemon_pid_file_create() < 0) { + daemon_log(LOG_ERR, "Could not create PID file (%s).", strerror(errno)); + daemon_retval_send(2); + goto finish; + } + + /* Send OK to parent process */ + daemon_retval_send(0); } + /* end libdaemon stuff */ + } - signal_setup(); - - // make sure the program can create files that group and world can read - umask(S_IWGRP | S_IWOTH); + signal_setup(); - config.output = audio_get_output(config.output_name); - if (!config.output) { - audio_ls_outputs(); - die("Invalid audio output specified!"); - } - config.output->init(argc-audio_arg, argv+audio_arg); + // make sure the program can create files that group and world can read + umask(S_IWGRP | S_IWOTH); - daemon_log(LOG_NOTICE, "startup"); + config.output = audio_get_output(config.output_name); + if (!config.output) { + audio_ls_outputs(); + die("Invalid audio output specified!"); + } + config.output->init(argc - audio_arg, argv + audio_arg); - uint8_t ap_md5[16]; + daemon_log(LOG_NOTICE, "startup"); + + /* Print out options */ + + debug(2, "statistics_requester status is %d.", config.statistics_requested); + debug(2, "daemon status is %d.", config.daemonise); + debug(2, "rtsp listening port is %d.", config.port); + debug(2, "udp base port is %d.", config.udp_port_base); + debug(2, "udp port range is %d.", config.udp_port_range); + debug(2, "Shairport Sync player name is \"%s\".", config.apname); + debug(2, "Audio Output name is \"%s\".", config.output_name); + debug(2, "on-start action is \"%s\".", config.cmd_start); + debug(2, "on-stop action is \"%s\".", config.cmd_stop); + debug(2, "wait-cmd status is %d.", config.cmd_blocking); + debug(2, "mdns backend \"%s\".", config.mdns_name); + debug(2, "userSuppliedLatency is %d.", config.userSuppliedLatency); + debug(2, "AirPlayLatency is %d.", config.AirPlayLatency); + debug(2, "iTunesLatency is %d.", config.iTunesLatency); + debug(2, "forkedDaapdLatency is %d.", config.ForkedDaapdLatency); + debug(2, "stuffing option is \"%d\".", config.packet_stuffing); + debug(2, "resync time is %d.", config.resyncthreshold); + debug(2, "allow a session to be interrupted: %d.", config.allow_session_interruption); + debug(2, "busy timeout time is %d.", config.timeout); + debug(2, "tolerance is %d frames.", config.tolerance); + debug(2, "password is \"%s\".", config.password); + debug(2, "ignore_volume_control is %d.", config.ignore_volume_control); + debug(2, "audio backend desired buffer length is %d.", + config.audio_backend_buffer_desired_length); + debug(2, "audio backend latency offset is %d.", config.audio_backend_latency_offset); + + char *realConfigPath = realpath(config.configfile,NULL); + if (realConfigPath) { + debug(2, "configuration file name \"%s\" resolves to \"%s\".", config.configfile,realConfigPath); + free(realConfigPath); + } else { + debug(2, "configuration file name \"%s\" can not be resolved.", config.configfile); + } +#ifdef CONFIG_METADATA + debug(2, "metdata enabled is %d.", config.metadata_enabled); + debug(2, "metadata pipename is \"%s\".", config.metadata_pipename); + debug(2, "get-coverart is %d.", config.get_coverart); +#endif + + uint8_t ap_md5[16]; #ifdef HAVE_LIBSSL - MD5_CTX ctx; - MD5_Init(&ctx); - MD5_Update(&ctx, config.apname, strlen(config.apname)); - MD5_Final(ap_md5, &ctx); + MD5_CTX ctx; + MD5_Init(&ctx); + MD5_Update(&ctx, config.apname, strlen(config.apname)); + MD5_Final(ap_md5, &ctx); #endif #ifdef HAVE_LIBPOLARSSL - md5_context tctx; - md5_starts(&tctx); - md5_update(&tctx, (unsigned char *)config.apname, strlen(config.apname)); - md5_finish(&tctx, ap_md5); + md5_context tctx; + md5_starts(&tctx); + md5_update(&tctx, (unsigned char *)config.apname, strlen(config.apname)); + md5_finish(&tctx, ap_md5); #endif - memcpy(config.hw_addr, ap_md5, sizeof(config.hw_addr)); + memcpy(config.hw_addr, ap_md5, sizeof(config.hw_addr)); +#ifdef CONFIG_METADATA + metadata_init(); // create the metadata pipe if necessary +#endif + rtsp_listen_loop(); - rtsp_listen_loop(); - - // should not reach this... - shairport_shutdown(); + // should not reach this... + shairport_shutdown(); finish: - daemon_log(LOG_NOTICE, "Unexpected exit..."); - daemon_retval_send(255); - daemon_pid_file_remove(); - return 1; - + daemon_log(LOG_NOTICE, "Unexpected exit..."); + daemon_retval_send(255); + daemon_pid_file_remove(); + return 1; } diff --git a/tinysvcmdns.c b/tinysvcmdns.c index 8c03b38e..bdb6c090 100644 --- a/tinysvcmdns.c +++ b/tinysvcmdns.c @@ -34,18 +34,16 @@ #include "common.h" #define DEBUG_PRINTF(...) debug(1, __VA_ARGS__) -#define log_message(level, ...) \ - do { \ - switch(level) \ - { \ - case LOG_ERR: \ - warn(__VA_ARGS__); \ - break; \ - default: \ - debug(1, __VA_ARGS__); \ - } \ - } while (0) - +#define log_message(level, ...) \ + do { \ + switch (level) { \ + case LOG_ERR: \ + warn(__VA_ARGS__); \ + break; \ + default: \ + debug(1, __VA_ARGS__); \ + } \ + } while (0) //******************************************************// // mdns.c // @@ -63,311 +61,320 @@ #include #endif -// See RFC 6762 Section 10 for an account of two TTLs -- 120 seconds for rrs with a host name as the record's name +// See RFC 6762 Section 10 for an account of two TTLs -- 120 seconds for rrs with a host name as the +// record's name // or a host name in the record's rdata // 75 minutes for everything else. // https://tools.ietf.org/html/rfc6762 -#define DEFAULT_TTL_FOR_RECORD_WITH_HOSTNAME 120 -#define DEFAULT_TTL 4500 +#define DEFAULT_TTL_FOR_RECORD_WITH_HOSTNAME 120 +#define DEFAULT_TTL 4500 struct name_comp { - uint8_t *label; // label - size_t pos; // position in msg + uint8_t *label; // label + size_t pos; // position in msg - struct name_comp *next; + struct name_comp *next; }; // ----- label functions ----- // duplicates a name inline uint8_t *dup_nlabel(const uint8_t *n) { - assert(n[0] <= 63); // prevent mis-use - return (uint8_t *) strdup((char *) n); + assert(n[0] <= 63); // prevent mis-use + return (uint8_t *)strdup((char *)n); } // duplicates a label uint8_t *dup_label(const uint8_t *label) { - int len = *label + 1; - if (len > 63) - return NULL; - uint8_t *newlabel = malloc(len + 1); - strncpy((char *) newlabel, (char *) label, len); - newlabel[len] = '\0'; - return newlabel; + int len = *label + 1; + if (len > 63) + return NULL; + uint8_t *newlabel = malloc(len + 1); + strncpy((char *)newlabel, (char *)label, len); + newlabel[len] = '\0'; + return newlabel; } uint8_t *join_nlabel(const uint8_t *n1, const uint8_t *n2) { - int len1, len2; - uint8_t *s; + int len1, len2; + uint8_t *s; - assert(n1[0] <= 63 && n2[0] <= 63); // detect misuse + assert(n1[0] <= 63 && n2[0] <= 63); // detect misuse - len1 = strlen((char *) n1); - len2 = strlen((char *) n2); + len1 = strlen((char *)n1); + len2 = strlen((char *)n2); - s = malloc(len1 + len2 + 1); - strncpy((char *) s, (char *) n1, len1); - strncpy((char *) s+len1, (char *) n2, len2); - s[len1 + len2] = '\0'; - return s; + s = malloc(len1 + len2 + 1); + strncpy((char *)s, (char *)n1, len1); + strncpy((char *)s + len1, (char *)n2, len2); + s[len1 + len2] = '\0'; + return s; } // returns a human-readable name label in dotted form char *nlabel_to_str(const uint8_t *name) { - char *label, *labelp; - const uint8_t *p; + char *label, *labelp; + const uint8_t *p; - assert(name != NULL); + assert(name != NULL); - label = labelp = malloc(256); + label = labelp = malloc(256); - for (p = name; *p; p++) { - strncpy(labelp, (char *) p + 1, *p); - labelp += *p; - *labelp = '.'; - labelp++; + for (p = name; *p; p++) { + strncpy(labelp, (char *)p + 1, *p); + labelp += *p; + *labelp = '.'; + labelp++; - p += *p; - } + p += *p; + } - *labelp = '\0'; + *labelp = '\0'; - return label; + return label; } // returns the length of a label field // does NOT uncompress the field, so it could be as small as 2 bytes // or 1 for the root static size_t label_len(uint8_t *pkt_buf, size_t pkt_len, size_t off) { - uint8_t *p; - uint8_t *e = pkt_buf + pkt_len; - size_t len = 0; + uint8_t *p; + uint8_t *e = pkt_buf + pkt_len; + size_t len = 0; - for (p = pkt_buf + off; p < e; p++) { - if (*p == 0) { - return len + 1; - } else if ((*p & 0xC0) == 0xC0) { - return len + 2; - } else { - len += *p + 1; - p += *p; - } - } + for (p = pkt_buf + off; p < e; p++) { + if (*p == 0) { + return len + 1; + } else if ((*p & 0xC0) == 0xC0) { + return len + 2; + } else { + len += *p + 1; + p += *p; + } + } - return len; + return len; } // creates a label // free() after use uint8_t *create_label(const char *txt) { - int len; - uint8_t *s; + int len; + uint8_t *s; - assert(txt != NULL); - len = strlen(txt); - if (len > 63) - return NULL; + assert(txt != NULL); + len = strlen(txt); + if (len > 63) + return NULL; - s = malloc(len + 2); - s[0] = len; - strncpy((char *) s + 1, txt, len); - s[len + 1] = '\0'; + s = malloc(len + 2); + s[0] = len; + strncpy((char *)s + 1, txt, len); + s[len + 1] = '\0'; - return s; + return s; } // creates a uncompressed name label given a DNS name like "apple.b.com" // free() after use uint8_t *create_nlabel(const char *name) { - char *label; - char *p, *e, *lenpos; - int len = 0; + char *label; + char *p, *e, *lenpos; + int len = 0; - assert(name != NULL); + assert(name != NULL); - len = strlen(name); - label = malloc(len + 1 + 1); - if (label == NULL) - return NULL; + len = strlen(name); + label = malloc(len + 1 + 1); + if (label == NULL) + return NULL; - strncpy((char *) label + 1, name, len); - label[len + 1] = '\0'; + strncpy((char *)label + 1, name, len); + label[len + 1] = '\0'; - p = label; - e = p + len; - lenpos = p; + p = label; + e = p + len; + lenpos = p; - while (p < e) { - *lenpos = 0; - char *dot = memchr(p + 1, '.', e - p - 1); - if (dot == NULL) - dot = e + 1; - *lenpos = dot - p - 1; + while (p < e) { + *lenpos = 0; + char *dot = memchr(p + 1, '.', e - p - 1); + if (dot == NULL) + dot = e + 1; + *lenpos = dot - p - 1; - p = dot; - lenpos = dot; - } + p = dot; + lenpos = dot; + } - return (uint8_t *) label; + return (uint8_t *)label; } // copies a label from the buffer into a newly-allocated string // free() after use static uint8_t *copy_label(uint8_t *pkt_buf, size_t pkt_len, size_t off) { - int len; + int len; - if (off > pkt_len) - return NULL; + if (off > pkt_len) + return NULL; - len = pkt_buf[off] + 1; - if (off + len > pkt_len) { - DEBUG_PRINTF("label length exceeds packet buffer\n"); - return NULL; - } + len = pkt_buf[off] + 1; + if (off + len > pkt_len) { + DEBUG_PRINTF("label length exceeds packet buffer\n"); + return NULL; + } - return dup_label(pkt_buf + off); + return dup_label(pkt_buf + off); } // uncompresses a name // free() after use static uint8_t *uncompress_nlabel(uint8_t *pkt_buf, size_t pkt_len, size_t off) { - uint8_t *p; - uint8_t *e = pkt_buf + pkt_len; - size_t len = 0; - char *str, *sp; - if (off >= pkt_len) - return NULL; + uint8_t *p; + uint8_t *e = pkt_buf + pkt_len; + size_t len = 0; + char *str, *sp; + if (off >= pkt_len) + return NULL; - // calculate length of uncompressed label - for (p = pkt_buf + off; *p && p < e; p++) { - size_t llen = 0; - if ((*p & 0xC0) == 0xC0) { - uint8_t *p2 = pkt_buf + (((p[0] & ~0xC0) << 8) | p[1]); - llen = *p2 + 1; - p = p2 + llen - 1; - } else { - llen = *p + 1; - p += llen - 1; - } - len += llen; - } + // calculate length of uncompressed label + for (p = pkt_buf + off; *p && p < e; p++) { + size_t llen = 0; + if ((*p & 0xC0) == 0xC0) { + uint8_t *p2 = pkt_buf + (((p[0] & ~0xC0) << 8) | p[1]); + llen = *p2 + 1; + p = p2 + llen - 1; + } else { + llen = *p + 1; + p += llen - 1; + } + len += llen; + } - str = sp = malloc(len + 1); - if (str == NULL) - return NULL; + str = sp = malloc(len + 1); + if (str == NULL) + return NULL; - // FIXME: must merge this with above code - for (p = pkt_buf + off; *p && p < e; p++) { - size_t llen = 0; - if ((*p & 0xC0) == 0xC0) { - uint8_t *p2 = pkt_buf + (((p[0] & ~0xC0) << 8) | p[1]); - llen = *p2 + 1; - strncpy(sp, (char *) p2, llen); - p = p2 + llen - 1; - } else { - llen = *p + 1; - strncpy(sp, (char *) p, llen); - p += llen - 1; - } - sp += llen; - } - *sp = '\0'; + // FIXME: must merge this with above code + for (p = pkt_buf + off; *p && p < e; p++) { + size_t llen = 0; + if ((*p & 0xC0) == 0xC0) { + uint8_t *p2 = pkt_buf + (((p[0] & ~0xC0) << 8) | p[1]); + llen = *p2 + 1; + strncpy(sp, (char *)p2, llen); + p = p2 + llen - 1; + } else { + llen = *p + 1; + strncpy(sp, (char *)p, llen); + p += llen - 1; + } + sp += llen; + } + *sp = '\0'; - return (uint8_t *) str; + return (uint8_t *)str; } // ----- RR list & group functions ----- const char *rr_get_type_name(enum rr_type type) { - switch (type) { - case RR_A: return "A"; - case RR_PTR: return "PTR"; - case RR_TXT: return "TXT"; - case RR_AAAA: return "AAAA"; - case RR_SRV: return "SRV"; - case RR_NSEC: return "NSEC"; - case RR_ANY: return "ANY"; - } - return NULL; + switch (type) { + case RR_A: + return "A"; + case RR_PTR: + return "PTR"; + case RR_TXT: + return "TXT"; + case RR_AAAA: + return "AAAA"; + case RR_SRV: + return "SRV"; + case RR_NSEC: + return "NSEC"; + case RR_ANY: + return "ANY"; + } + return NULL; } void rr_entry_destroy(struct rr_entry *rr) { - struct rr_data_txt *txt_rec; - assert(rr); + struct rr_data_txt *txt_rec; + assert(rr); - // check rr_type and free data elements - switch (rr->type) { - case RR_PTR: - if (rr->data.PTR.name) - free(rr->data.PTR.name); - // don't free entry - break; + // check rr_type and free data elements + switch (rr->type) { + case RR_PTR: + if (rr->data.PTR.name) + free(rr->data.PTR.name); + // don't free entry + break; - case RR_TXT: - txt_rec = &rr->data.TXT; - while (txt_rec) { - struct rr_data_txt *next = txt_rec->next; - if (txt_rec->txt) - free(txt_rec->txt); + case RR_TXT: + txt_rec = &rr->data.TXT; + while (txt_rec) { + struct rr_data_txt *next = txt_rec->next; + if (txt_rec->txt) + free(txt_rec->txt); - // only free() if it wasn't part of the struct - if (txt_rec != &rr->data.TXT) - free(txt_rec); + // only free() if it wasn't part of the struct + if (txt_rec != &rr->data.TXT) + free(txt_rec); - txt_rec = next; - } - break; + txt_rec = next; + } + break; - case RR_SRV: - if (rr->data.SRV.target) - free(rr->data.SRV.target); - break; + case RR_SRV: + if (rr->data.SRV.target) + free(rr->data.SRV.target); + break; - default: - // nothing to free - break; - } + default: + // nothing to free + break; + } - free(rr->name); - free(rr); + free(rr->name); + free(rr); } // destroys an RR list (and optionally, items) void rr_list_destroy(struct rr_list *rr, char destroy_items) { - struct rr_list *rr_next; + struct rr_list *rr_next; - for (; rr; rr = rr_next) { - rr_next = rr->next; - if (destroy_items) - rr_entry_destroy(rr->e); - free(rr); - } + for (; rr; rr = rr_next) { + rr_next = rr->next; + if (destroy_items) + rr_entry_destroy(rr->e); + free(rr); + } } int rr_list_count(struct rr_list *rr) { - int i = 0; - for (; rr; i++, rr = rr->next); - return i; + int i = 0; + for (; rr; i++, rr = rr->next) + ; + return i; } struct rr_entry *rr_list_remove(struct rr_list **rr_head, struct rr_entry *rr) { - struct rr_list *le = *rr_head, *pe = NULL; - for (; le; le = le->next) { - if (le->e == rr) { - if (pe == NULL) { - *rr_head = le->next; - free(le); - return rr; - } else { - pe->next = le->next; - free(le); - return rr; - } - } - pe = le; - } - return NULL; + struct rr_list *le = *rr_head, *pe = NULL; + for (; le; le = le->next) { + if (le->e == rr) { + if (pe == NULL) { + *rr_head = le->next; + free(le); + return rr; + } else { + pe->next = le->next; + free(le); + return rr; + } + } + pe = le; + } + return NULL; } // appends an rr_entry to an RR list @@ -375,650 +382,642 @@ struct rr_entry *rr_list_remove(struct rr_list **rr_head, struct rr_entry *rr) { // RRs are compared by memory location - not its contents // return value of 0 means item not added int rr_list_append(struct rr_list **rr_head, struct rr_entry *rr) { - struct rr_list *node = malloc(sizeof(struct rr_list)); - node->e = rr; - node->next = NULL; + struct rr_list *node = malloc(sizeof(struct rr_list)); + node->e = rr; + node->next = NULL; - if (*rr_head == NULL) { - *rr_head = node; - } else { - struct rr_list *e = *rr_head, *taile; - for (; e; e = e->next) { - // already in list - don't add - if (e->e == rr) { - free(node); - return 0; - } - if (e->next == NULL) - taile = e; - } - taile->next = node; - } - return 1; + if (*rr_head == NULL) { + *rr_head = node; + } else { + struct rr_list *e = *rr_head, *taile; + for (; e; e = e->next) { + // already in list - don't add + if (e->e == rr) { + free(node); + return 0; + } + if (e->next == NULL) + taile = e; + } + taile->next = node; + } + return 1; } -#define FILL_RR_ENTRY(rr, _name, _type) \ - rr->name = _name; \ - rr->type = _type; \ - rr->ttl = DEFAULT_TTL; \ - rr->cache_flush = 1; \ - rr->rr_class = 1; +#define FILL_RR_ENTRY(rr, _name, _type) \ + rr->name = _name; \ + rr->type = _type; \ + rr->ttl = DEFAULT_TTL; \ + rr->cache_flush = 1; \ + rr->rr_class = 1; struct rr_entry *rr_create_a(uint8_t *name, uint32_t addr) { - DECL_MALLOC_ZERO_STRUCT(rr, rr_entry); - FILL_RR_ENTRY(rr, name, RR_A); - rr->data.A.addr = addr; - rr->ttl = DEFAULT_TTL_FOR_RECORD_WITH_HOSTNAME; // 120 seconds -- see RFC 6762 Section 10 - return rr; + DECL_MALLOC_ZERO_STRUCT(rr, rr_entry); + FILL_RR_ENTRY(rr, name, RR_A); + rr->data.A.addr = addr; + rr->ttl = DEFAULT_TTL_FOR_RECORD_WITH_HOSTNAME; // 120 seconds -- see RFC 6762 Section 10 + return rr; } struct rr_entry *rr_create_aaaa(uint8_t *name, struct in6_addr *addr) { - DECL_MALLOC_ZERO_STRUCT(rr, rr_entry); - FILL_RR_ENTRY(rr, name, RR_AAAA); - rr->data.AAAA.addr = addr; - rr->ttl = DEFAULT_TTL_FOR_RECORD_WITH_HOSTNAME; // 120 seconds -- see RFC 6762 Section 10 - return rr; + DECL_MALLOC_ZERO_STRUCT(rr, rr_entry); + FILL_RR_ENTRY(rr, name, RR_AAAA); + rr->data.AAAA.addr = addr; + rr->ttl = DEFAULT_TTL_FOR_RECORD_WITH_HOSTNAME; // 120 seconds -- see RFC 6762 Section 10 + return rr; } struct rr_entry *rr_create_srv(uint8_t *name, uint16_t port, uint8_t *target) { - DECL_MALLOC_ZERO_STRUCT(rr, rr_entry); - FILL_RR_ENTRY(rr, name, RR_SRV); - rr->data.SRV.port = port; - rr->data.SRV.target = target; - return rr; + DECL_MALLOC_ZERO_STRUCT(rr, rr_entry); + FILL_RR_ENTRY(rr, name, RR_SRV); + rr->data.SRV.port = port; + rr->data.SRV.target = target; + return rr; } struct rr_entry *rr_create_ptr(uint8_t *name, struct rr_entry *d_rr) { - DECL_MALLOC_ZERO_STRUCT(rr, rr_entry); - FILL_RR_ENTRY(rr, name, RR_PTR); - rr->cache_flush = 0; // PTRs shouldn't have their cache flush bit set - rr->data.PTR.entry = d_rr; - return rr; + DECL_MALLOC_ZERO_STRUCT(rr, rr_entry); + FILL_RR_ENTRY(rr, name, RR_PTR); + rr->cache_flush = 0; // PTRs shouldn't have their cache flush bit set + rr->data.PTR.entry = d_rr; + return rr; } struct rr_entry *rr_create(uint8_t *name, enum rr_type type) { - DECL_MALLOC_ZERO_STRUCT(rr, rr_entry); - FILL_RR_ENTRY(rr, name, type); - return rr; + DECL_MALLOC_ZERO_STRUCT(rr, rr_entry); + FILL_RR_ENTRY(rr, name, type); + return rr; } void rr_set_nsec(struct rr_entry *rr_nsec, enum rr_type type) { - assert(rr_nsec->type = RR_NSEC); - assert((type / 8) < sizeof(rr_nsec->data.NSEC.bitmap)); + assert(rr_nsec->type = RR_NSEC); + assert((type / 8) < sizeof(rr_nsec->data.NSEC.bitmap)); - rr_nsec->data.NSEC.bitmap[ type / 8 ] = 1 << (7 - (type % 8)); + rr_nsec->data.NSEC.bitmap[type / 8] = 1 << (7 - (type % 8)); } void rr_add_txt(struct rr_entry *rr_txt, const char *txt) { - struct rr_data_txt *txt_rec; - assert(rr_txt->type == RR_TXT); + struct rr_data_txt *txt_rec; + assert(rr_txt->type == RR_TXT); - txt_rec = &rr_txt->data.TXT; + txt_rec = &rr_txt->data.TXT; - // is current data filled? - if (txt_rec->txt == NULL) { - txt_rec->txt = create_label(txt); - return; - } + // is current data filled? + if (txt_rec->txt == NULL) { + txt_rec->txt = create_label(txt); + return; + } - // find the last node - for (; txt_rec->next; txt_rec = txt_rec->next); + // find the last node + for (; txt_rec->next; txt_rec = txt_rec->next) + ; - // create a new empty node - txt_rec->next = malloc(sizeof(struct rr_data_txt)); + // create a new empty node + txt_rec->next = malloc(sizeof(struct rr_data_txt)); - txt_rec = txt_rec->next; - txt_rec->txt = create_label(txt); - txt_rec->next = NULL; + txt_rec = txt_rec->next; + txt_rec->txt = create_label(txt); + txt_rec->next = NULL; } // adds a record to an rr_group void rr_group_add(struct rr_group **group, struct rr_entry *rr) { - struct rr_group *g; + struct rr_group *g; - assert(rr != NULL); + assert(rr != NULL); - if (*group) { - g = rr_group_find(*group, rr->name); - if (g) { - rr_list_append(&g->rr, rr); - return; - } - } + if (*group) { + g = rr_group_find(*group, rr->name); + if (g) { + rr_list_append(&g->rr, rr); + return; + } + } - MALLOC_ZERO_STRUCT(g, rr_group); - g->name = dup_nlabel(rr->name); - rr_list_append(&g->rr, rr); + MALLOC_ZERO_STRUCT(g, rr_group); + g->name = dup_nlabel(rr->name); + rr_list_append(&g->rr, rr); - // prepend to list - g->next = *group; - *group = g; + // prepend to list + g->next = *group; + *group = g; } // finds a rr_group matching the given name -struct rr_group *rr_group_find(struct rr_group* g, uint8_t *name) { - for (; g; g = g->next) { - if (cmp_nlabel(g->name, name) == 0) - return g; - } - return NULL; +struct rr_group *rr_group_find(struct rr_group *g, uint8_t *name) { + for (; g; g = g->next) { + if (cmp_nlabel(g->name, name) == 0) + return g; + } + return NULL; } struct rr_entry *rr_entry_find(struct rr_list *rr_list, uint8_t *name, uint16_t type) { - struct rr_list *rr = rr_list; - for (; rr; rr = rr->next) { - if (rr->e->type == type && cmp_nlabel(rr->e->name, name) == 0) - return rr->e; - } - return NULL; + struct rr_list *rr = rr_list; + for (; rr; rr = rr->next) { + if (rr->e->type == type && cmp_nlabel(rr->e->name, name) == 0) + return rr->e; + } + return NULL; } // looks for a matching entry in rr_list // if entry is a PTR, we need to check if the PTR target also matches struct rr_entry *rr_entry_match(struct rr_list *rr_list, struct rr_entry *entry) { - struct rr_list *rr = rr_list; - for (; rr; rr = rr->next) { - if (rr->e->type == entry->type && cmp_nlabel(rr->e->name, entry->name) == 0) { - if (entry->type != RR_PTR) { - return rr->e; - } else if (cmp_nlabel(MDNS_RR_GET_PTR_NAME(entry), MDNS_RR_GET_PTR_NAME(rr->e)) == 0) { - // if it's a PTR, we need to make sure PTR target also matches - return rr->e; - } - } - } - return NULL; + struct rr_list *rr = rr_list; + for (; rr; rr = rr->next) { + if (rr->e->type == entry->type && cmp_nlabel(rr->e->name, entry->name) == 0) { + if (entry->type != RR_PTR) { + return rr->e; + } else if (cmp_nlabel(MDNS_RR_GET_PTR_NAME(entry), MDNS_RR_GET_PTR_NAME(rr->e)) == 0) { + // if it's a PTR, we need to make sure PTR target also matches + return rr->e; + } + } + } + return NULL; } void rr_group_destroy(struct rr_group *group) { - struct rr_group *g = group; + struct rr_group *g = group; - while (g) { - struct rr_group *nextg = g->next; - free(g->name); - rr_list_destroy(g->rr, 1); - free(g); - g = nextg; - } + while (g) { + struct rr_group *nextg = g->next; + free(g->name); + rr_list_destroy(g->rr, 1); + free(g); + g = nextg; + } } uint8_t *mdns_write_u16(uint8_t *ptr, const uint16_t v) { - *ptr++ = (uint8_t) (v >> 8) & 0xFF; - *ptr++ = (uint8_t) (v >> 0) & 0xFF; - return ptr; + *ptr++ = (uint8_t)(v >> 8) & 0xFF; + *ptr++ = (uint8_t)(v >> 0) & 0xFF; + return ptr; } uint8_t *mdns_write_u32(uint8_t *ptr, const uint32_t v) { - *ptr++ = (uint8_t) (v >> 24) & 0xFF; - *ptr++ = (uint8_t) (v >> 16) & 0xFF; - *ptr++ = (uint8_t) (v >> 8) & 0xFF; - *ptr++ = (uint8_t) (v >> 0) & 0xFF; - return ptr; + *ptr++ = (uint8_t)(v >> 24) & 0xFF; + *ptr++ = (uint8_t)(v >> 16) & 0xFF; + *ptr++ = (uint8_t)(v >> 8) & 0xFF; + *ptr++ = (uint8_t)(v >> 0) & 0xFF; + return ptr; } uint16_t mdns_read_u16(const uint8_t *ptr) { - return ((ptr[0] & 0xFF) << 8) | - ((ptr[1] & 0xFF) << 0); + return ((ptr[0] & 0xFF) << 8) | ((ptr[1] & 0xFF) << 0); } uint32_t mdns_read_u32(const uint8_t *ptr) { - return ((ptr[0] & 0xFF) << 24) | - ((ptr[1] & 0xFF) << 16) | - ((ptr[2] & 0xFF) << 8) | - ((ptr[3] & 0xFF) << 0); + return ((ptr[0] & 0xFF) << 24) | ((ptr[1] & 0xFF) << 16) | ((ptr[2] & 0xFF) << 8) | + ((ptr[3] & 0xFF) << 0); } // initialize the packet for reply // clears the packet of list structures but not its list items void mdns_init_reply(struct mdns_pkt *pkt, uint16_t id) { - // copy transaction ID - pkt->id = id; + // copy transaction ID + pkt->id = id; - // response flags - pkt->flags = MDNS_FLAG_RESP | MDNS_FLAG_AA; + // response flags + pkt->flags = MDNS_FLAG_RESP | MDNS_FLAG_AA; - rr_list_destroy(pkt->rr_qn, 0); - rr_list_destroy(pkt->rr_ans, 0); - rr_list_destroy(pkt->rr_auth, 0); - rr_list_destroy(pkt->rr_add, 0); + rr_list_destroy(pkt->rr_qn, 0); + rr_list_destroy(pkt->rr_ans, 0); + rr_list_destroy(pkt->rr_auth, 0); + rr_list_destroy(pkt->rr_add, 0); - pkt->rr_qn = NULL; - pkt->rr_ans = NULL; - pkt->rr_auth = NULL; - pkt->rr_add = NULL; + pkt->rr_qn = NULL; + pkt->rr_ans = NULL; + pkt->rr_auth = NULL; + pkt->rr_add = NULL; - pkt->num_qn = 0; - pkt->num_ans_rr = 0; - pkt->num_auth_rr = 0; - pkt->num_add_rr = 0; + pkt->num_qn = 0; + pkt->num_ans_rr = 0; + pkt->num_auth_rr = 0; + pkt->num_add_rr = 0; } // destroys an mdns_pkt struct, including its contents void mdns_pkt_destroy(struct mdns_pkt *p) { - rr_list_destroy(p->rr_qn, 1); - rr_list_destroy(p->rr_ans, 1); - rr_list_destroy(p->rr_auth, 1); - rr_list_destroy(p->rr_add, 1); + rr_list_destroy(p->rr_qn, 1); + rr_list_destroy(p->rr_ans, 1); + rr_list_destroy(p->rr_auth, 1); + rr_list_destroy(p->rr_add, 1); - free(p); + free(p); } - // parse the MDNS questions section // stores the parsed data in the given mdns_pkt struct -static size_t mdns_parse_qn(uint8_t *pkt_buf, size_t pkt_len, size_t off, - struct mdns_pkt *pkt) { - const uint8_t *p = pkt_buf + off; - struct rr_entry *rr; - uint8_t *name; +static size_t mdns_parse_qn(uint8_t *pkt_buf, size_t pkt_len, size_t off, struct mdns_pkt *pkt) { + const uint8_t *p = pkt_buf + off; + struct rr_entry *rr; + uint8_t *name; - assert(pkt != NULL); + assert(pkt != NULL); - rr = malloc(sizeof(struct rr_entry)); - memset(rr, 0, sizeof(struct rr_entry)); + rr = malloc(sizeof(struct rr_entry)); + memset(rr, 0, sizeof(struct rr_entry)); - name = uncompress_nlabel(pkt_buf, pkt_len, off); - p += label_len(pkt_buf, pkt_len, off); - rr->name = name; + name = uncompress_nlabel(pkt_buf, pkt_len, off); + p += label_len(pkt_buf, pkt_len, off); + rr->name = name; - rr->type = mdns_read_u16(p); - p += sizeof(uint16_t); + rr->type = mdns_read_u16(p); + p += sizeof(uint16_t); - rr->unicast_query = (*p & 0x80) == 0x80; - rr->rr_class = mdns_read_u16(p) & ~0x80; - p += sizeof(uint16_t); + rr->unicast_query = (*p & 0x80) == 0x80; + rr->rr_class = mdns_read_u16(p) & ~0x80; + p += sizeof(uint16_t); - rr_list_append(&pkt->rr_qn, rr); + rr_list_append(&pkt->rr_qn, rr); - return p - (pkt_buf + off); + return p - (pkt_buf + off); } // parse the MDNS RR section // stores the parsed data in the given mdns_pkt struct -static size_t mdns_parse_rr(uint8_t *pkt_buf, size_t pkt_len, size_t off, - struct mdns_pkt *pkt) { - const uint8_t *p = pkt_buf + off; - const uint8_t *e = pkt_buf + pkt_len; - struct rr_entry *rr; - uint8_t *name; - size_t rr_data_len = 0; - struct rr_data_txt *txt_rec; - int parse_error = 0; +static size_t mdns_parse_rr(uint8_t *pkt_buf, size_t pkt_len, size_t off, struct mdns_pkt *pkt) { + const uint8_t *p = pkt_buf + off; + const uint8_t *e = pkt_buf + pkt_len; + struct rr_entry *rr; + uint8_t *name; + size_t rr_data_len = 0; + struct rr_data_txt *txt_rec; + int parse_error = 0; - assert(pkt != NULL); + assert(pkt != NULL); - if (off > pkt_len) - return 0; + if (off > pkt_len) + return 0; - rr = malloc(sizeof(struct rr_entry)); - memset(rr, 0, sizeof(struct rr_entry)); + rr = malloc(sizeof(struct rr_entry)); + memset(rr, 0, sizeof(struct rr_entry)); - name = uncompress_nlabel(pkt_buf, pkt_len, off); - p += label_len(pkt_buf, pkt_len, off); - rr->name = name; + name = uncompress_nlabel(pkt_buf, pkt_len, off); + p += label_len(pkt_buf, pkt_len, off); + rr->name = name; - rr->type = mdns_read_u16(p); - p += sizeof(uint16_t); + rr->type = mdns_read_u16(p); + p += sizeof(uint16_t); - rr->cache_flush = (*p & 0x80) == 0x80; - rr->rr_class = mdns_read_u16(p) & ~0x80; - p += sizeof(uint16_t); + rr->cache_flush = (*p & 0x80) == 0x80; + rr->rr_class = mdns_read_u16(p) & ~0x80; + p += sizeof(uint16_t); - rr->ttl = mdns_read_u32(p); - p += sizeof(uint32_t); + rr->ttl = mdns_read_u32(p); + p += sizeof(uint32_t); - // RR data - rr_data_len = mdns_read_u16(p); - p += sizeof(uint16_t); + // RR data + rr_data_len = mdns_read_u16(p); + p += sizeof(uint16_t); - if (p + rr_data_len > e) { - DEBUG_PRINTF("rr_data_len goes beyond packet buffer: %lu > %lu\n", rr_data_len, e - p); - rr_entry_destroy(rr); - return 0; - } + if (p + rr_data_len > e) { + DEBUG_PRINTF("rr_data_len goes beyond packet buffer: %lu > %lu\n", rr_data_len, e - p); + rr_entry_destroy(rr); + return 0; + } - e = p + rr_data_len; + e = p + rr_data_len; - // see if we can parse the RR data - switch (rr->type) { - case RR_A: - if (rr_data_len < sizeof(uint32_t)) { - DEBUG_PRINTF("invalid rr_data_len=%lu for A record\n", rr_data_len); - parse_error = 1; - break; - } - rr->data.A.addr = ntohl(mdns_read_u32(p)); /* addr already in net order */ - p += sizeof(uint32_t); - break; + // see if we can parse the RR data + switch (rr->type) { + case RR_A: + if (rr_data_len < sizeof(uint32_t)) { + DEBUG_PRINTF("invalid rr_data_len=%lu for A record\n", rr_data_len); + parse_error = 1; + break; + } + rr->data.A.addr = ntohl(mdns_read_u32(p)); /* addr already in net order */ + p += sizeof(uint32_t); + break; - case RR_AAAA: - if (rr_data_len < sizeof(struct in6_addr)) { - DEBUG_PRINTF("invalid rr_data_len=%lu for AAAA record\n", rr_data_len); - parse_error = 1; - break; - } - rr->data.AAAA.addr = malloc(sizeof(struct in6_addr)); - int i; - for (i = 0; i < sizeof(struct in6_addr); i++) - rr->data.AAAA.addr->s6_addr[i] = p[i]; - p += sizeof(struct in6_addr); - break; + case RR_AAAA: + if (rr_data_len < sizeof(struct in6_addr)) { + DEBUG_PRINTF("invalid rr_data_len=%lu for AAAA record\n", rr_data_len); + parse_error = 1; + break; + } + rr->data.AAAA.addr = malloc(sizeof(struct in6_addr)); + int i; + for (i = 0; i < sizeof(struct in6_addr); i++) + rr->data.AAAA.addr->s6_addr[i] = p[i]; + p += sizeof(struct in6_addr); + break; - case RR_PTR: - rr->data.PTR.name = uncompress_nlabel(pkt_buf, pkt_len, p - pkt_buf); - if (rr->data.PTR.name == NULL) { - DEBUG_PRINTF("unable to parse/uncompress label for PTR name\n"); - parse_error = 1; - break; - } - p += rr_data_len; - break; + case RR_PTR: + rr->data.PTR.name = uncompress_nlabel(pkt_buf, pkt_len, p - pkt_buf); + if (rr->data.PTR.name == NULL) { + DEBUG_PRINTF("unable to parse/uncompress label for PTR name\n"); + parse_error = 1; + break; + } + p += rr_data_len; + break; - case RR_TXT: - txt_rec = &rr->data.TXT; + case RR_TXT: + txt_rec = &rr->data.TXT; - // not supposed to happen, but we should handle it - if (rr_data_len == 0) { - DEBUG_PRINTF("WARN: rr_data_len for TXT is 0\n"); - txt_rec->txt = create_label(""); - break; - } + // not supposed to happen, but we should handle it + if (rr_data_len == 0) { + DEBUG_PRINTF("WARN: rr_data_len for TXT is 0\n"); + txt_rec->txt = create_label(""); + break; + } - while (1) { - txt_rec->txt = copy_label(pkt_buf, pkt_len, p - pkt_buf); - if (txt_rec->txt == NULL) { - DEBUG_PRINTF("unable to copy label for TXT record\n"); - parse_error = 1; - break; - } - p += txt_rec->txt[0] + 1; + while (1) { + txt_rec->txt = copy_label(pkt_buf, pkt_len, p - pkt_buf); + if (txt_rec->txt == NULL) { + DEBUG_PRINTF("unable to copy label for TXT record\n"); + parse_error = 1; + break; + } + p += txt_rec->txt[0] + 1; - if (p >= e) - break; + if (p >= e) + break; - // allocate another record - txt_rec->next = malloc(sizeof(struct rr_data_txt)); - txt_rec = txt_rec->next; - txt_rec->next = NULL; - } - break; + // allocate another record + txt_rec->next = malloc(sizeof(struct rr_data_txt)); + txt_rec = txt_rec->next; + txt_rec->next = NULL; + } + break; - default: - // skip to end of RR data - p = e; - } + default: + // skip to end of RR data + p = e; + } - // if there was a parse error, destroy partial rr_entry - if (parse_error) { - rr_entry_destroy(rr); - return 0; - } + // if there was a parse error, destroy partial rr_entry + if (parse_error) { + rr_entry_destroy(rr); + return 0; + } - rr_list_append(&pkt->rr_ans, rr); + rr_list_append(&pkt->rr_ans, rr); - return p - (pkt_buf + off); + return p - (pkt_buf + off); } // parse a MDNS packet into an mdns_pkt struct struct mdns_pkt *mdns_parse_pkt(uint8_t *pkt_buf, size_t pkt_len) { - uint8_t *p = pkt_buf; - size_t off; - struct mdns_pkt *pkt; - int i; + uint8_t *p = pkt_buf; + size_t off; + struct mdns_pkt *pkt; + int i; - if (pkt_len < 12) - return NULL; + if (pkt_len < 12) + return NULL; - MALLOC_ZERO_STRUCT(pkt, mdns_pkt); + MALLOC_ZERO_STRUCT(pkt, mdns_pkt); - // parse header - pkt->id = mdns_read_u16(p); p += sizeof(uint16_t); - pkt->flags = mdns_read_u16(p); p += sizeof(uint16_t); - pkt->num_qn = mdns_read_u16(p); p += sizeof(uint16_t); - pkt->num_ans_rr = mdns_read_u16(p); p += sizeof(uint16_t); - pkt->num_auth_rr = mdns_read_u16(p); p += sizeof(uint16_t); - pkt->num_add_rr = mdns_read_u16(p); p += sizeof(uint16_t); + // parse header + pkt->id = mdns_read_u16(p); + p += sizeof(uint16_t); + pkt->flags = mdns_read_u16(p); + p += sizeof(uint16_t); + pkt->num_qn = mdns_read_u16(p); + p += sizeof(uint16_t); + pkt->num_ans_rr = mdns_read_u16(p); + p += sizeof(uint16_t); + pkt->num_auth_rr = mdns_read_u16(p); + p += sizeof(uint16_t); + pkt->num_add_rr = mdns_read_u16(p); + p += sizeof(uint16_t); - off = p - pkt_buf; + off = p - pkt_buf; - // parse questions - for (i = 0; i < pkt->num_qn; i++) { - size_t l = mdns_parse_qn(pkt_buf, pkt_len, off, pkt); - if (! l) { - DEBUG_PRINTF("error parsing question #%d\n", i); - mdns_pkt_destroy(pkt); - return NULL; - } + // parse questions + for (i = 0; i < pkt->num_qn; i++) { + size_t l = mdns_parse_qn(pkt_buf, pkt_len, off, pkt); + if (!l) { + DEBUG_PRINTF("error parsing question #%d\n", i); + mdns_pkt_destroy(pkt); + return NULL; + } - off += l; - } + off += l; + } - // parse answer RRs - for (i = 0; i < pkt->num_ans_rr; i++) { - size_t l = mdns_parse_rr(pkt_buf, pkt_len, off, pkt); - if (! l) { - DEBUG_PRINTF("error parsing answer #%d\n", i); - mdns_pkt_destroy(pkt); - return NULL; - } + // parse answer RRs + for (i = 0; i < pkt->num_ans_rr; i++) { + size_t l = mdns_parse_rr(pkt_buf, pkt_len, off, pkt); + if (!l) { + DEBUG_PRINTF("error parsing answer #%d\n", i); + mdns_pkt_destroy(pkt); + return NULL; + } - off += l; - } + off += l; + } - // TODO: parse the authority and additional RR sections + // TODO: parse the authority and additional RR sections - return pkt; + return pkt; } // encodes a name (label) into a packet using the name compression scheme // encoded names will be added to the compression list for subsequent use -static size_t mdns_encode_name(uint8_t *pkt_buf, size_t pkt_len, size_t off, - const uint8_t *name, struct name_comp *comp) { - struct name_comp *c, *c_tail = NULL; - uint8_t *p = pkt_buf + off; - size_t len = 0; +static size_t mdns_encode_name(uint8_t *pkt_buf, size_t pkt_len, size_t off, const uint8_t *name, + struct name_comp *comp) { + struct name_comp *c, *c_tail = NULL; + uint8_t *p = pkt_buf + off; + size_t len = 0; - if (name) { - while (*name) { - // find match for compression - for (c = comp; c; c = c->next) { - if (cmp_nlabel(name, c->label) == 0) { - mdns_write_u16(p, 0xC000 | (c->pos & ~0xC000)); - return len + sizeof(uint16_t); - } - - if (c->next == NULL) - c_tail = c; - } - - // copy this segment - int segment_len = *name + 1; - strncpy((char *) p, (char *) name, segment_len); - - // cache the name for subsequent compression - DECL_MALLOC_ZERO_STRUCT(new_c, name_comp); - - new_c->label = (uint8_t *) name; - new_c->pos = p - pkt_buf; - c_tail->next = new_c; - - // advance to next name segment - p += segment_len; - len += segment_len; - name += segment_len; - } + if (name) { + while (*name) { + // find match for compression + for (c = comp; c; c = c->next) { + if (cmp_nlabel(name, c->label) == 0) { + mdns_write_u16(p, 0xC000 | (c->pos & ~0xC000)); + return len + sizeof(uint16_t); } - *p = '\0'; // root "label" - len += 1; + if (c->next == NULL) + c_tail = c; + } - return len; + // copy this segment + int segment_len = *name + 1; + strncpy((char *)p, (char *)name, segment_len); + + // cache the name for subsequent compression + DECL_MALLOC_ZERO_STRUCT(new_c, name_comp); + + new_c->label = (uint8_t *)name; + new_c->pos = p - pkt_buf; + c_tail->next = new_c; + + // advance to next name segment + p += segment_len; + len += segment_len; + name += segment_len; + } + } + + *p = '\0'; // root "label" + len += 1; + + return len; } // encodes an RR entry at the given offset // returns the size of the entire RR entry -static size_t mdns_encode_rr(uint8_t *pkt_buf, size_t pkt_len, size_t off, - struct rr_entry *rr, struct name_comp *comp) { - uint8_t *p = pkt_buf + off, *p_data; - size_t l; - struct rr_data_txt *txt_rec; - uint8_t *label; - int i; +static size_t mdns_encode_rr(uint8_t *pkt_buf, size_t pkt_len, size_t off, struct rr_entry *rr, + struct name_comp *comp) { + uint8_t *p = pkt_buf + off, *p_data; + size_t l; + struct rr_data_txt *txt_rec; + uint8_t *label; + int i; - assert(off < pkt_len); + assert(off < pkt_len); - // name - l = mdns_encode_name(pkt_buf, pkt_len, off, rr->name, comp); - assert(l != 0); - p += l; + // name + l = mdns_encode_name(pkt_buf, pkt_len, off, rr->name, comp); + assert(l != 0); + p += l; - // type - p = mdns_write_u16(p, rr->type); + // type + p = mdns_write_u16(p, rr->type); - // class & cache flush - p = mdns_write_u16(p, (rr->rr_class & ~0x8000) | (rr->cache_flush << 15)); + // class & cache flush + p = mdns_write_u16(p, (rr->rr_class & ~0x8000) | (rr->cache_flush << 15)); - // TTL - p = mdns_write_u32(p, rr->ttl); + // TTL + p = mdns_write_u32(p, rr->ttl); - // data length (filled in later) - p += sizeof(uint16_t); + // data length (filled in later) + p += sizeof(uint16_t); - // start of data marker - p_data = p; + // start of data marker + p_data = p; - switch (rr->type) { - case RR_A: - /* htonl() needed coz addr already in net order */ - p = mdns_write_u32(p, htonl(rr->data.A.addr)); - break; + switch (rr->type) { + case RR_A: + /* htonl() needed coz addr already in net order */ + p = mdns_write_u32(p, htonl(rr->data.A.addr)); + break; - case RR_AAAA: - for (i = 0; i < sizeof(struct in6_addr); i++) - *p++ = rr->data.AAAA.addr->s6_addr[i]; - break; + case RR_AAAA: + for (i = 0; i < sizeof(struct in6_addr); i++) + *p++ = rr->data.AAAA.addr->s6_addr[i]; + break; - case RR_PTR: - label = rr->data.PTR.name ? - rr->data.PTR.name : - rr->data.PTR.entry->name; - p += mdns_encode_name(pkt_buf, pkt_len, p - pkt_buf, label, comp); - break; + case RR_PTR: + label = rr->data.PTR.name ? rr->data.PTR.name : rr->data.PTR.entry->name; + p += mdns_encode_name(pkt_buf, pkt_len, p - pkt_buf, label, comp); + break; - case RR_TXT: - txt_rec = &rr->data.TXT; - for (; txt_rec; txt_rec = txt_rec->next) { - int len = txt_rec->txt[0] + 1; - strncpy((char *) p, (char *) txt_rec->txt, len); - p += len; - } - break; + case RR_TXT: + txt_rec = &rr->data.TXT; + for (; txt_rec; txt_rec = txt_rec->next) { + int len = txt_rec->txt[0] + 1; + strncpy((char *)p, (char *)txt_rec->txt, len); + p += len; + } + break; - case RR_SRV: - p = mdns_write_u16(p, rr->data.SRV.priority); + case RR_SRV: + p = mdns_write_u16(p, rr->data.SRV.priority); - p = mdns_write_u16(p, rr->data.SRV.weight); + p = mdns_write_u16(p, rr->data.SRV.weight); - p = mdns_write_u16(p, rr->data.SRV.port); + p = mdns_write_u16(p, rr->data.SRV.port); - p += mdns_encode_name(pkt_buf, pkt_len, p - pkt_buf, - rr->data.SRV.target, comp); - break; + p += mdns_encode_name(pkt_buf, pkt_len, p - pkt_buf, rr->data.SRV.target, comp); + break; - case RR_NSEC: - p += mdns_encode_name(pkt_buf, pkt_len, p - pkt_buf, - rr->name, comp); + case RR_NSEC: + p += mdns_encode_name(pkt_buf, pkt_len, p - pkt_buf, rr->name, comp); - *p++ = 0; // bitmap window/block number + *p++ = 0; // bitmap window/block number - *p++ = sizeof(rr->data.NSEC.bitmap); // bitmap length + *p++ = sizeof(rr->data.NSEC.bitmap); // bitmap length - for (i = 0; i < sizeof(rr->data.NSEC.bitmap); i++) - *p++ = rr->data.NSEC.bitmap[i]; + for (i = 0; i < sizeof(rr->data.NSEC.bitmap); i++) + *p++ = rr->data.NSEC.bitmap[i]; - break; + break; - default: - DEBUG_PRINTF("unhandled rr type 0x%02x\n", rr->type); - } + default: + DEBUG_PRINTF("unhandled rr type 0x%02x\n", rr->type); + } - // calculate data length based on p - l = p - p_data; + // calculate data length based on p + l = p - p_data; - // fill in the length - mdns_write_u16(p - l - sizeof(uint16_t), l); + // fill in the length + mdns_write_u16(p - l - sizeof(uint16_t), l); - return p - pkt_buf - off; + return p - pkt_buf - off; } // encodes a MDNS packet from the given mdns_pkt struct into a buffer // returns the size of the entire MDNS packet size_t mdns_encode_pkt(struct mdns_pkt *answer, uint8_t *pkt_buf, size_t pkt_len) { - struct name_comp *comp; - uint8_t *p = pkt_buf; - //uint8_t *e = pkt_buf + pkt_len; - size_t off; - int i; + struct name_comp *comp; + uint8_t *p = pkt_buf; + // uint8_t *e = pkt_buf + pkt_len; + size_t off; + int i; - assert(answer != NULL); - assert(pkt_len >= 12); + assert(answer != NULL); + assert(pkt_len >= 12); - if (p == NULL) - return -1; + if (p == NULL) + return -1; - // this is an Answer - number of qns should be zero - assert(answer->num_qn == 0); + // this is an Answer - number of qns should be zero + assert(answer->num_qn == 0); - p = mdns_write_u16(p, answer->id); - p = mdns_write_u16(p, answer->flags); - p = mdns_write_u16(p, answer->num_qn); - p = mdns_write_u16(p, answer->num_ans_rr); - p = mdns_write_u16(p, answer->num_auth_rr); - p = mdns_write_u16(p, answer->num_add_rr); + p = mdns_write_u16(p, answer->id); + p = mdns_write_u16(p, answer->flags); + p = mdns_write_u16(p, answer->num_qn); + p = mdns_write_u16(p, answer->num_ans_rr); + p = mdns_write_u16(p, answer->num_auth_rr); + p = mdns_write_u16(p, answer->num_add_rr); - off = p - pkt_buf; + off = p - pkt_buf; - // allocate list for name compression - comp = malloc(sizeof(struct name_comp)); - if (comp == NULL) - return -1; - memset(comp, 0, sizeof(struct name_comp)); + // allocate list for name compression + comp = malloc(sizeof(struct name_comp)); + if (comp == NULL) + return -1; + memset(comp, 0, sizeof(struct name_comp)); - // dummy entry - comp->label = (uint8_t *) ""; - comp->pos = 0; + // dummy entry + comp->label = (uint8_t *)""; + comp->pos = 0; - // skip encoding of qn + // skip encoding of qn - struct rr_list *rr_set[] = { - answer->rr_ans, - answer->rr_auth, - answer->rr_add - }; + struct rr_list *rr_set[] = {answer->rr_ans, answer->rr_auth, answer->rr_add}; - // encode answer, authority and additional RRs - for (i = 0; i < sizeof(rr_set) / sizeof(rr_set[0]); i++) { - struct rr_list *rr = rr_set[i]; - for (; rr; rr = rr->next) { - size_t l = mdns_encode_rr(pkt_buf, pkt_len, off, rr->e, comp); - off += l; + // encode answer, authority and additional RRs + for (i = 0; i < sizeof(rr_set) / sizeof(rr_set[0]); i++) { + struct rr_list *rr = rr_set[i]; + for (; rr; rr = rr->next) { + size_t l = mdns_encode_rr(pkt_buf, pkt_len, off, rr->e, comp); + off += l; - if (off >= pkt_len) { - DEBUG_PRINTF("packet buffer too small\n"); - return -1; - } - } + if (off >= pkt_len) { + DEBUG_PRINTF("packet buffer too small\n"); + return -1; + } + } + } - } + // free name compression list + while (comp) { + struct name_comp *c = comp->next; + free(comp); + comp = c; + } - // free name compression list - while (comp) { - struct name_comp *c = comp->next; - free(comp); - comp = c; - } - - return off; + return off; } //******************************************************// @@ -1064,621 +1063,599 @@ size_t mdns_encode_pkt(struct mdns_pkt *answer, uint8_t *pkt_buf, size_t pkt_len #define PACKET_SIZE 65536 -#define SERVICES_DNS_SD_NLABEL \ - ((uint8_t *) "\x09_services\x07_dns-sd\x04_udp\x05local") +#define SERVICES_DNS_SD_NLABEL ((uint8_t *)"\x09_services\x07_dns-sd\x04_udp\x05local") struct mdnsd { - pthread_mutex_t data_lock; - int sockfd; - int notify_pipe[2]; - int stop_flag; + pthread_mutex_t data_lock; + int sockfd; + int notify_pipe[2]; + int stop_flag; - struct rr_group *group; - struct rr_list *announce; - struct rr_list *services; - uint8_t *hostname; + struct rr_group *group; + struct rr_list *announce; + struct rr_list *services; + uint8_t *hostname; }; struct mdns_service { - struct rr_list *entries; + struct rr_list *entries; }; ///////////////////////////////// - - static int create_recv_sock() { - int sd = socket(AF_INET, SOCK_DGRAM, 0); - if (sd < 0) { - log_message(LOG_ERR, "recv socket(): %m"); - return sd; - } + int sd = socket(AF_INET, SOCK_DGRAM, 0); + if (sd < 0) { + log_message(LOG_ERR, "recv socket(): %m"); + return sd; + } - int r = -1; + int r = -1; - int on = 1; - if ((r = setsockopt(sd, SOL_SOCKET, SO_REUSEADDR, (char *) &on, sizeof(on))) < 0) { - log_message(LOG_ERR, "recv setsockopt(SO_REUSEADDR): %m"); - return r; - } + int on = 1; + if ((r = setsockopt(sd, SOL_SOCKET, SO_REUSEADDR, (char *)&on, sizeof(on))) < 0) { + log_message(LOG_ERR, "recv setsockopt(SO_REUSEADDR): %m"); + return r; + } - /* bind to an address */ - struct sockaddr_in serveraddr; - memset(&serveraddr, 0, sizeof(serveraddr)); - serveraddr.sin_family = AF_INET; - serveraddr.sin_port = htons(MDNS_PORT); - serveraddr.sin_addr.s_addr = htonl(INADDR_ANY); /* receive multicast */ - if ((r = bind(sd, (struct sockaddr *)&serveraddr, sizeof(serveraddr))) < 0) { - log_message(LOG_ERR, "recv bind(): %m"); - } + /* bind to an address */ + struct sockaddr_in serveraddr; + memset(&serveraddr, 0, sizeof(serveraddr)); + serveraddr.sin_family = AF_INET; + serveraddr.sin_port = htons(MDNS_PORT); + serveraddr.sin_addr.s_addr = htonl(INADDR_ANY); /* receive multicast */ + if ((r = bind(sd, (struct sockaddr *)&serveraddr, sizeof(serveraddr))) < 0) { + log_message(LOG_ERR, "recv bind(): %m"); + } - // add membership to receiving socket - struct ip_mreq mreq; - memset(&mreq, 0, sizeof(struct ip_mreq)); - mreq.imr_interface.s_addr = htonl(INADDR_ANY); - mreq.imr_multiaddr.s_addr = inet_addr(MDNS_ADDR); - if ((r = setsockopt(sd, IPPROTO_IP, IP_ADD_MEMBERSHIP, (char *) &mreq, sizeof(mreq))) < 0) { - log_message(LOG_ERR, "recv setsockopt(IP_ADD_MEMBERSHIP): %m"); - return r; - } - - // enable loopback in case someone else needs the data - if ((r = setsockopt(sd, IPPROTO_IP, IP_MULTICAST_LOOP, (char *) &on, sizeof(on))) < 0) { - log_message(LOG_ERR, "recv setsockopt(IP_MULTICAST_LOOP): %m"); - return r; - } + // add membership to receiving socket + struct ip_mreq mreq; + memset(&mreq, 0, sizeof(struct ip_mreq)); + mreq.imr_interface.s_addr = htonl(INADDR_ANY); + mreq.imr_multiaddr.s_addr = inet_addr(MDNS_ADDR); + if ((r = setsockopt(sd, IPPROTO_IP, IP_ADD_MEMBERSHIP, (char *)&mreq, sizeof(mreq))) < 0) { + log_message(LOG_ERR, "recv setsockopt(IP_ADD_MEMBERSHIP): %m"); + return r; + } + // enable loopback in case someone else needs the data + if ((r = setsockopt(sd, IPPROTO_IP, IP_MULTICAST_LOOP, (char *)&on, sizeof(on))) < 0) { + log_message(LOG_ERR, "recv setsockopt(IP_MULTICAST_LOOP): %m"); + return r; + } #ifdef IP_PKTINFO - if ((r = setsockopt(sd, SOL_IP, IP_PKTINFO, (char *) &on, sizeof(on))) < 0) { - log_message(LOG_ERR, "recv setsockopt(IP_PKTINFO): %m"); - return r; - } + if ((r = setsockopt(sd, SOL_IP, IP_PKTINFO, (char *)&on, sizeof(on))) < 0) { + log_message(LOG_ERR, "recv setsockopt(IP_PKTINFO): %m"); + return r; + } #endif - return sd; + return sd; } static ssize_t send_packet(int fd, const void *data, size_t len) { - static struct sockaddr_in toaddr; - if (toaddr.sin_family != AF_INET) { - memset(&toaddr, 0, sizeof(struct sockaddr_in)); - toaddr.sin_family = AF_INET; - toaddr.sin_port = htons(MDNS_PORT); - toaddr.sin_addr.s_addr = inet_addr(MDNS_ADDR); - } + static struct sockaddr_in toaddr; + if (toaddr.sin_family != AF_INET) { + memset(&toaddr, 0, sizeof(struct sockaddr_in)); + toaddr.sin_family = AF_INET; + toaddr.sin_port = htons(MDNS_PORT); + toaddr.sin_addr.s_addr = inet_addr(MDNS_ADDR); + } - return sendto(fd, data, len, 0, (struct sockaddr *) &toaddr, sizeof(struct sockaddr_in)); + return sendto(fd, data, len, 0, (struct sockaddr *)&toaddr, sizeof(struct sockaddr_in)); } - // populate the specified list which matches the RR name and type // type can be RR_ANY, which populates all entries EXCEPT RR_NSEC -static int populate_answers(struct mdnsd *svr, struct rr_list **rr_head, uint8_t *name, enum rr_type type) { - int num_ans = 0; +static int populate_answers(struct mdnsd *svr, struct rr_list **rr_head, uint8_t *name, + enum rr_type type) { + int num_ans = 0; - // check if we have the records - pthread_mutex_lock(&svr->data_lock); - struct rr_group *ans_grp = rr_group_find(svr->group, name); - if (ans_grp == NULL) { - pthread_mutex_unlock(&svr->data_lock); - return num_ans; - } + // check if we have the records + pthread_mutex_lock(&svr->data_lock); + struct rr_group *ans_grp = rr_group_find(svr->group, name); + if (ans_grp == NULL) { + pthread_mutex_unlock(&svr->data_lock); + return num_ans; + } - // decide which records should go into answers - struct rr_list *n = ans_grp->rr; - for (; n; n = n->next) { - // exclude NSEC for RR_ANY - if (type == RR_ANY && n->e->type == RR_NSEC) - continue; + // decide which records should go into answers + struct rr_list *n = ans_grp->rr; + for (; n; n = n->next) { + // exclude NSEC for RR_ANY + if (type == RR_ANY && n->e->type == RR_NSEC) + continue; - if ((type == n->e->type || type == RR_ANY) && cmp_nlabel(name, n->e->name) == 0) { - num_ans += rr_list_append(rr_head, n->e); - } - } + if ((type == n->e->type || type == RR_ANY) && cmp_nlabel(name, n->e->name) == 0) { + num_ans += rr_list_append(rr_head, n->e); + } + } - pthread_mutex_unlock(&svr->data_lock); + pthread_mutex_unlock(&svr->data_lock); - return num_ans; + return num_ans; } // given a list of RRs, look up related records and add them static void add_related_rr(struct mdnsd *svr, struct rr_list *list, struct mdns_pkt *reply) { - for (; list; list = list->next) { - struct rr_entry *ans = list->e; + for (; list; list = list->next) { + struct rr_entry *ans = list->e; - switch (ans->type) { - case RR_PTR: - // target host A, AAAA records - reply->num_add_rr += populate_answers(svr, &reply->rr_add, - MDNS_RR_GET_PTR_NAME(ans), RR_ANY); - break; + switch (ans->type) { + case RR_PTR: + // target host A, AAAA records + reply->num_add_rr += populate_answers(svr, &reply->rr_add, MDNS_RR_GET_PTR_NAME(ans), RR_ANY); + break; - case RR_SRV: - // target host A, AAAA records - reply->num_add_rr += populate_answers(svr, &reply->rr_add, - ans->data.SRV.target, RR_ANY); + case RR_SRV: + // target host A, AAAA records + reply->num_add_rr += populate_answers(svr, &reply->rr_add, ans->data.SRV.target, RR_ANY); - // perhaps TXT records of the same name? - // if we use RR_ANY, we risk pulling in the same RR_SRV - reply->num_add_rr += populate_answers(svr, &reply->rr_add, - ans->name, RR_TXT); - break; + // perhaps TXT records of the same name? + // if we use RR_ANY, we risk pulling in the same RR_SRV + reply->num_add_rr += populate_answers(svr, &reply->rr_add, ans->name, RR_TXT); + break; - case RR_A: - case RR_AAAA: - reply->num_add_rr += populate_answers(svr, &reply->rr_add, - ans->name, RR_NSEC); - break; + case RR_A: + case RR_AAAA: + reply->num_add_rr += populate_answers(svr, &reply->rr_add, ans->name, RR_NSEC); + break; - default: - // nothing to add - break; - } - } + default: + // nothing to add + break; + } + } } // creates an announce packet given the type name PTR static void announce_srv(struct mdnsd *svr, struct mdns_pkt *reply, uint8_t *name) { - mdns_init_reply(reply, 0); + mdns_init_reply(reply, 0); - reply->num_ans_rr += populate_answers(svr, &reply->rr_ans, name, RR_PTR); + reply->num_ans_rr += populate_answers(svr, &reply->rr_ans, name, RR_PTR); - // remember to add the services dns-sd PTR too - reply->num_ans_rr += populate_answers(svr, &reply->rr_ans, - SERVICES_DNS_SD_NLABEL, RR_PTR); + // remember to add the services dns-sd PTR too + reply->num_ans_rr += populate_answers(svr, &reply->rr_ans, SERVICES_DNS_SD_NLABEL, RR_PTR); - // see if we can match additional records for answers - add_related_rr(svr, reply->rr_ans, reply); + // see if we can match additional records for answers + add_related_rr(svr, reply->rr_ans, reply); - // additional records for additional records - add_related_rr(svr, reply->rr_add, reply); + // additional records for additional records + add_related_rr(svr, reply->rr_add, reply); } // processes the incoming MDNS packet // returns >0 if processed, 0 otherwise static int process_mdns_pkt(struct mdnsd *svr, struct mdns_pkt *pkt, struct mdns_pkt *reply) { - int i; + int i; - assert(pkt != NULL); + assert(pkt != NULL); - // is it standard query? - if ((pkt->flags & MDNS_FLAG_RESP) == 0 && - MDNS_FLAG_GET_OPCODE(pkt->flags) == 0) { - mdns_init_reply(reply, pkt->id); + // is it standard query? + if ((pkt->flags & MDNS_FLAG_RESP) == 0 && MDNS_FLAG_GET_OPCODE(pkt->flags) == 0) { + mdns_init_reply(reply, pkt->id); - DEBUG_PRINTF("flags = %04x, qn = %d, ans = %d, add = %d\n", - pkt->flags, - pkt->num_qn, - pkt->num_ans_rr, - pkt->num_add_rr); + DEBUG_PRINTF("flags = %04x, qn = %d, ans = %d, add = %d\n", pkt->flags, pkt->num_qn, + pkt->num_ans_rr, pkt->num_add_rr); - // loop through questions - struct rr_list *qnl = pkt->rr_qn; - for (i = 0; i < pkt->num_qn; i++, qnl = qnl->next) { - struct rr_entry *qn = qnl->e; - int num_ans_added = 0; + // loop through questions + struct rr_list *qnl = pkt->rr_qn; + for (i = 0; i < pkt->num_qn; i++, qnl = qnl->next) { + struct rr_entry *qn = qnl->e; + int num_ans_added = 0; - char *namestr = nlabel_to_str(qn->name); - DEBUG_PRINTF("qn #%d: type %s (%02x) %s - ", i, rr_get_type_name(qn->type), qn->type, namestr); - free(namestr); + char *namestr = nlabel_to_str(qn->name); + DEBUG_PRINTF("qn #%d: type %s (%02x) %s - ", i, rr_get_type_name(qn->type), qn->type, + namestr); + free(namestr); - // check if it's a unicast query - we ignore those - if (qn->unicast_query) { - DEBUG_PRINTF("skipping unicast query\n"); - continue; - } + // check if it's a unicast query - we ignore those + if (qn->unicast_query) { + DEBUG_PRINTF("skipping unicast query\n"); + continue; + } - num_ans_added = populate_answers(svr, &reply->rr_ans, qn->name, qn->type); - reply->num_ans_rr += num_ans_added; + num_ans_added = populate_answers(svr, &reply->rr_ans, qn->name, qn->type); + reply->num_ans_rr += num_ans_added; - DEBUG_PRINTF("added %d answers\n", num_ans_added); - } + DEBUG_PRINTF("added %d answers\n", num_ans_added); + } - // remove our replies if they were already in their answers - struct rr_list *ans = NULL, *prev_ans = NULL; - for (ans = reply->rr_ans; ans; ) { - struct rr_list *next_ans = ans->next; - struct rr_entry *known_ans = rr_entry_match(pkt->rr_ans, ans->e); + // remove our replies if they were already in their answers + struct rr_list *ans = NULL, *prev_ans = NULL; + for (ans = reply->rr_ans; ans;) { + struct rr_list *next_ans = ans->next; + struct rr_entry *known_ans = rr_entry_match(pkt->rr_ans, ans->e); - // discard answers that have at least half of the actual TTL - if (known_ans != NULL && known_ans->ttl >= ans->e->ttl / 2) { - char *namestr = nlabel_to_str(ans->e->name); - DEBUG_PRINTF("removing answer for %s\n", namestr); - free(namestr); + // discard answers that have at least half of the actual TTL + if (known_ans != NULL && known_ans->ttl >= ans->e->ttl / 2) { + char *namestr = nlabel_to_str(ans->e->name); + DEBUG_PRINTF("removing answer for %s\n", namestr); + free(namestr); - // check if list item is head - if (prev_ans == NULL) - reply->rr_ans = ans->next; - else - prev_ans->next = ans->next; - free(ans); + // check if list item is head + if (prev_ans == NULL) + reply->rr_ans = ans->next; + else + prev_ans->next = ans->next; + free(ans); - ans = prev_ans; + ans = prev_ans; - // adjust answer count - reply->num_ans_rr--; - } + // adjust answer count + reply->num_ans_rr--; + } - prev_ans = ans; - ans = next_ans; - } + prev_ans = ans; + ans = next_ans; + } + // see if we can match additional records for answers + add_related_rr(svr, reply->rr_ans, reply); - // see if we can match additional records for answers - add_related_rr(svr, reply->rr_ans, reply); + // additional records for additional records + add_related_rr(svr, reply->rr_add, reply); - // additional records for additional records - add_related_rr(svr, reply->rr_add, reply); + DEBUG_PRINTF("\n"); - DEBUG_PRINTF("\n"); + return reply->num_ans_rr; + } - return reply->num_ans_rr; - } - - return 0; + return 0; } int create_pipe(int handles[2]) { #ifdef _WIN32 - SOCKET sock = socket(AF_INET, SOCK_STREAM, 0); - if (sock == INVALID_SOCKET) { - return -1; - } - struct sockaddr_in serv_addr; - memset(&serv_addr, 0, sizeof(serv_addr)); - serv_addr.sin_family = AF_INET; - serv_addr.sin_port = htons(0); - serv_addr.sin_addr.s_addr = htonl(INADDR_LOOPBACK); - if (bind(sock, (struct sockaddr*)&serv_addr, sizeof(serv_addr)) == SOCKET_ERROR) { - closesocket(sock); - return -1; - } - if (listen(sock, 1) == SOCKET_ERROR) { - closesocket(sock); - return -1; - } - int len = sizeof(serv_addr); - if (getsockname(sock, (SOCKADDR*)&serv_addr, &len) == SOCKET_ERROR) { - closesocket(sock); - return -1; - } - if ((handles[1] = socket(PF_INET, SOCK_STREAM, 0)) == INVALID_SOCKET) { - closesocket(sock); - return -1; - } - if (connect(handles[1], (struct sockaddr*)&serv_addr, len) == SOCKET_ERROR) { - closesocket(sock); - return -1; - } - if ((handles[0] = accept(sock, (struct sockaddr*)&serv_addr, &len)) == INVALID_SOCKET) { - closesocket((SOCKET)handles[1]); - handles[1] = INVALID_SOCKET; - closesocket(sock); - return -1; - } - closesocket(sock); - return 0; + SOCKET sock = socket(AF_INET, SOCK_STREAM, 0); + if (sock == INVALID_SOCKET) { + return -1; + } + struct sockaddr_in serv_addr; + memset(&serv_addr, 0, sizeof(serv_addr)); + serv_addr.sin_family = AF_INET; + serv_addr.sin_port = htons(0); + serv_addr.sin_addr.s_addr = htonl(INADDR_LOOPBACK); + if (bind(sock, (struct sockaddr *)&serv_addr, sizeof(serv_addr)) == SOCKET_ERROR) { + closesocket(sock); + return -1; + } + if (listen(sock, 1) == SOCKET_ERROR) { + closesocket(sock); + return -1; + } + int len = sizeof(serv_addr); + if (getsockname(sock, (SOCKADDR *)&serv_addr, &len) == SOCKET_ERROR) { + closesocket(sock); + return -1; + } + if ((handles[1] = socket(PF_INET, SOCK_STREAM, 0)) == INVALID_SOCKET) { + closesocket(sock); + return -1; + } + if (connect(handles[1], (struct sockaddr *)&serv_addr, len) == SOCKET_ERROR) { + closesocket(sock); + return -1; + } + if ((handles[0] = accept(sock, (struct sockaddr *)&serv_addr, &len)) == INVALID_SOCKET) { + closesocket((SOCKET)handles[1]); + handles[1] = INVALID_SOCKET; + closesocket(sock); + return -1; + } + closesocket(sock); + return 0; #else - return pipe(handles); + return pipe(handles); #endif } -int read_pipe(int s, char* buf, int len) { +int read_pipe(int s, char *buf, int len) { #ifdef _WIN32 - int ret = recv(s, buf, len, 0); - if (ret < 0 && WSAGetLastError() == WSAECONNRESET) { - ret = 0; - } - return ret; + int ret = recv(s, buf, len, 0); + if (ret < 0 && WSAGetLastError() == WSAECONNRESET) { + ret = 0; + } + return ret; #else - return read(s, buf, len); + return read(s, buf, len); #endif } -int write_pipe(int s, char* buf, int len) { +int write_pipe(int s, char *buf, int len) { #ifdef _WIN32 - return send(s, buf, len, 0); + return send(s, buf, len, 0); #else - return write(s, buf, len); + return write(s, buf, len); #endif } int close_pipe(int s) { #ifdef _WIN32 - return closesocket(s); + return closesocket(s); #else - return close(s); + return close(s); #endif } // main loop to receive, process and send out MDNS replies // also handles MDNS service announces static void main_loop(struct mdnsd *svr) { - fd_set sockfd_set; - int max_fd = svr->sockfd; - char notify_buf[2]; // buffer for reading of notify_pipe + fd_set sockfd_set; + int max_fd = svr->sockfd; + char notify_buf[2]; // buffer for reading of notify_pipe - void *pkt_buffer = malloc(PACKET_SIZE); + void *pkt_buffer = malloc(PACKET_SIZE); - if (svr->notify_pipe[0] > max_fd) - max_fd = svr->notify_pipe[0]; + if (svr->notify_pipe[0] > max_fd) + max_fd = svr->notify_pipe[0]; - struct mdns_pkt *mdns_reply = malloc(sizeof(struct mdns_pkt)); - memset(mdns_reply, 0, sizeof(struct mdns_pkt)); + struct mdns_pkt *mdns_reply = malloc(sizeof(struct mdns_pkt)); + memset(mdns_reply, 0, sizeof(struct mdns_pkt)); - while (! svr->stop_flag) { - FD_ZERO(&sockfd_set); - FD_SET(svr->sockfd, &sockfd_set); - FD_SET(svr->notify_pipe[0], &sockfd_set); - select(max_fd + 1, &sockfd_set, NULL, NULL, NULL); + while (!svr->stop_flag) { + FD_ZERO(&sockfd_set); + FD_SET(svr->sockfd, &sockfd_set); + FD_SET(svr->notify_pipe[0], &sockfd_set); + select(max_fd + 1, &sockfd_set, NULL, NULL, NULL); - if (FD_ISSET(svr->notify_pipe[0], &sockfd_set)) { - // flush the notify_pipe - read_pipe(svr->notify_pipe[0], (char*)¬ify_buf, 1); - } else if (FD_ISSET(svr->sockfd, &sockfd_set)) { - struct sockaddr_in fromaddr; - socklen_t sockaddr_size = sizeof(struct sockaddr_in); + if (FD_ISSET(svr->notify_pipe[0], &sockfd_set)) { + // flush the notify_pipe + read_pipe(svr->notify_pipe[0], (char *)¬ify_buf, 1); + } else if (FD_ISSET(svr->sockfd, &sockfd_set)) { + struct sockaddr_in fromaddr; + socklen_t sockaddr_size = sizeof(struct sockaddr_in); - ssize_t recvsize = recvfrom(svr->sockfd, pkt_buffer, PACKET_SIZE, 0, - (struct sockaddr *) &fromaddr, &sockaddr_size); - if (recvsize < 0) { - log_message(LOG_ERR, "recv(): %m"); - } + ssize_t recvsize = recvfrom(svr->sockfd, pkt_buffer, PACKET_SIZE, 0, + (struct sockaddr *)&fromaddr, &sockaddr_size); + if (recvsize < 0) { + log_message(LOG_ERR, "recv(): %m"); + } - DEBUG_PRINTF("data from=%s size=%ld\n", inet_ntoa(fromaddr.sin_addr), (long) recvsize); - struct mdns_pkt *mdns = mdns_parse_pkt(pkt_buffer, recvsize); - if (mdns != NULL) { - if (process_mdns_pkt(svr, mdns, mdns_reply)) { - size_t replylen = mdns_encode_pkt(mdns_reply, pkt_buffer, PACKET_SIZE); - send_packet(svr->sockfd, pkt_buffer, replylen); - } else if (mdns->num_qn == 0) { - DEBUG_PRINTF("(no questions in packet)\n\n"); - } - - mdns_pkt_destroy(mdns); - } - } - - // send out announces - while (1) { - struct rr_entry *ann_e = NULL; - - // extract from head of list - pthread_mutex_lock(&svr->data_lock); - if (svr->announce) - ann_e = rr_list_remove(&svr->announce, svr->announce->e); - pthread_mutex_unlock(&svr->data_lock); - - if (! ann_e) - break; - - char *namestr = nlabel_to_str(ann_e->name); - DEBUG_PRINTF("sending announce for %s\n", namestr); - free(namestr); - - announce_srv(svr, mdns_reply, ann_e->name); - - if (mdns_reply->num_ans_rr > 0) { - size_t replylen = mdns_encode_pkt(mdns_reply, pkt_buffer, PACKET_SIZE); - send_packet(svr->sockfd, pkt_buffer, replylen); - } - } + DEBUG_PRINTF("data from=%s size=%ld\n", inet_ntoa(fromaddr.sin_addr), (long)recvsize); + struct mdns_pkt *mdns = mdns_parse_pkt(pkt_buffer, recvsize); + if (mdns != NULL) { + if (process_mdns_pkt(svr, mdns, mdns_reply)) { + size_t replylen = mdns_encode_pkt(mdns_reply, pkt_buffer, PACKET_SIZE); + send_packet(svr->sockfd, pkt_buffer, replylen); + } else if (mdns->num_qn == 0) { + DEBUG_PRINTF("(no questions in packet)\n\n"); } - // main thread terminating. send out "goodbye packets" for services - mdns_init_reply(mdns_reply, 0); + mdns_pkt_destroy(mdns); + } + } - pthread_mutex_lock(&svr->data_lock); - struct rr_list *svc_le = svr->services; - for (; svc_le; svc_le = svc_le->next) { - // set TTL to zero - svc_le->e->ttl = 0; - mdns_reply->num_ans_rr += rr_list_append(&mdns_reply->rr_ans, svc_le->e); - } - pthread_mutex_unlock(&svr->data_lock); + // send out announces + while (1) { + struct rr_entry *ann_e = NULL; - // send out packet - if (mdns_reply->num_ans_rr > 0) { - size_t replylen = mdns_encode_pkt(mdns_reply, pkt_buffer, PACKET_SIZE); - send_packet(svr->sockfd, pkt_buffer, replylen); - } + // extract from head of list + pthread_mutex_lock(&svr->data_lock); + if (svr->announce) + ann_e = rr_list_remove(&svr->announce, svr->announce->e); + pthread_mutex_unlock(&svr->data_lock); - // destroy packet - mdns_init_reply(mdns_reply, 0); - free(mdns_reply); + if (!ann_e) + break; - free(pkt_buffer); + char *namestr = nlabel_to_str(ann_e->name); + DEBUG_PRINTF("sending announce for %s\n", namestr); + free(namestr); - close_pipe(svr->sockfd); + announce_srv(svr, mdns_reply, ann_e->name); - svr->stop_flag = 2; + if (mdns_reply->num_ans_rr > 0) { + size_t replylen = mdns_encode_pkt(mdns_reply, pkt_buffer, PACKET_SIZE); + send_packet(svr->sockfd, pkt_buffer, replylen); + } + } + } + + // main thread terminating. send out "goodbye packets" for services + mdns_init_reply(mdns_reply, 0); + + pthread_mutex_lock(&svr->data_lock); + struct rr_list *svc_le = svr->services; + for (; svc_le; svc_le = svc_le->next) { + // set TTL to zero + svc_le->e->ttl = 0; + mdns_reply->num_ans_rr += rr_list_append(&mdns_reply->rr_ans, svc_le->e); + } + pthread_mutex_unlock(&svr->data_lock); + + // send out packet + if (mdns_reply->num_ans_rr > 0) { + size_t replylen = mdns_encode_pkt(mdns_reply, pkt_buffer, PACKET_SIZE); + send_packet(svr->sockfd, pkt_buffer, replylen); + } + + // destroy packet + mdns_init_reply(mdns_reply, 0); + free(mdns_reply); + + free(pkt_buffer); + + close_pipe(svr->sockfd); + + svr->stop_flag = 2; } ///////////////////////////////////////////////////// - void mdnsd_set_hostname(struct mdnsd *svr, const char *hostname, uint32_t ip) { - struct rr_entry *a_e = NULL, - *nsec_e = NULL; + struct rr_entry *a_e = NULL, *nsec_e = NULL; - // currently can't be called twice - // dont ask me what happens if the IP changes - assert(svr->hostname == NULL); + // currently can't be called twice + // dont ask me what happens if the IP changes + assert(svr->hostname == NULL); - a_e = rr_create_a(create_nlabel(hostname), ip); // 120 seconds automatically + a_e = rr_create_a(create_nlabel(hostname), ip); // 120 seconds automatically - nsec_e = rr_create(create_nlabel(hostname), RR_NSEC); - nsec_e->ttl = DEFAULT_TTL_FOR_RECORD_WITH_HOSTNAME; // set to 120 seconds (default is 4500) - rr_set_nsec(nsec_e, RR_A); + nsec_e = rr_create(create_nlabel(hostname), RR_NSEC); + nsec_e->ttl = DEFAULT_TTL_FOR_RECORD_WITH_HOSTNAME; // set to 120 seconds (default is 4500) + rr_set_nsec(nsec_e, RR_A); - pthread_mutex_lock(&svr->data_lock); - svr->hostname = create_nlabel(hostname); - rr_group_add(&svr->group, a_e); - rr_group_add(&svr->group, nsec_e); - pthread_mutex_unlock(&svr->data_lock); + pthread_mutex_lock(&svr->data_lock); + svr->hostname = create_nlabel(hostname); + rr_group_add(&svr->group, a_e); + rr_group_add(&svr->group, nsec_e); + pthread_mutex_unlock(&svr->data_lock); } -void mdnsd_set_hostname_v6(struct mdnsd *svr, const char *hostname, struct in6_addr *addr) -{ - struct rr_entry *aaaa_e = NULL, - *nsec_e = NULL; +void mdnsd_set_hostname_v6(struct mdnsd *svr, const char *hostname, struct in6_addr *addr) { + struct rr_entry *aaaa_e = NULL, *nsec_e = NULL; - // currently can't be called twice - // dont ask me what happens if the IP changes - assert(svr->hostname == NULL); + // currently can't be called twice + // dont ask me what happens if the IP changes + assert(svr->hostname == NULL); - aaaa_e = rr_create_aaaa(create_nlabel(hostname), addr); // 120 seconds automatically + aaaa_e = rr_create_aaaa(create_nlabel(hostname), addr); // 120 seconds automatically - nsec_e = rr_create(create_nlabel(hostname), RR_NSEC); - nsec_e->ttl = DEFAULT_TTL_FOR_RECORD_WITH_HOSTNAME; // set to 120 seconds (default is 4500) - rr_set_nsec(nsec_e, RR_AAAA); + nsec_e = rr_create(create_nlabel(hostname), RR_NSEC); + nsec_e->ttl = DEFAULT_TTL_FOR_RECORD_WITH_HOSTNAME; // set to 120 seconds (default is 4500) + rr_set_nsec(nsec_e, RR_AAAA); - pthread_mutex_lock(&svr->data_lock); - svr->hostname = create_nlabel(hostname); - rr_group_add(&svr->group, aaaa_e); - rr_group_add(&svr->group, nsec_e); - pthread_mutex_unlock(&svr->data_lock); + pthread_mutex_lock(&svr->data_lock); + svr->hostname = create_nlabel(hostname); + rr_group_add(&svr->group, aaaa_e); + rr_group_add(&svr->group, nsec_e); + pthread_mutex_unlock(&svr->data_lock); } void mdnsd_add_rr(struct mdnsd *svr, struct rr_entry *rr) { - pthread_mutex_lock(&svr->data_lock); - rr_group_add(&svr->group, rr); - pthread_mutex_unlock(&svr->data_lock); + pthread_mutex_lock(&svr->data_lock); + rr_group_add(&svr->group, rr); + pthread_mutex_unlock(&svr->data_lock); } struct mdns_service *mdnsd_register_svc(struct mdnsd *svr, const char *instance_name, - const char *type, uint16_t port, const char *hostname, const char *txt[]) { - struct rr_entry *txt_e = NULL, - *srv_e = NULL, - *ptr_e = NULL, - *bptr_e = NULL; - uint8_t *target; - uint8_t *inst_nlabel, *type_nlabel, *nlabel; - struct mdns_service *service = malloc(sizeof(struct mdns_service)); - memset(service, 0, sizeof(struct mdns_service)); + const char *type, uint16_t port, const char *hostname, + const char *txt[]) { + struct rr_entry *txt_e = NULL, *srv_e = NULL, *ptr_e = NULL, *bptr_e = NULL; + uint8_t *target; + uint8_t *inst_nlabel, *type_nlabel, *nlabel; + struct mdns_service *service = malloc(sizeof(struct mdns_service)); + memset(service, 0, sizeof(struct mdns_service)); - // combine service name - type_nlabel = create_nlabel(type); - inst_nlabel = create_label(instance_name); - nlabel = join_nlabel(inst_nlabel, type_nlabel); + // combine service name + type_nlabel = create_nlabel(type); + inst_nlabel = create_label(instance_name); + nlabel = join_nlabel(inst_nlabel, type_nlabel); - // create TXT record - if (txt && *txt) { - txt_e = rr_create(dup_nlabel(nlabel), RR_TXT); // automatically 4500 seconds - rr_list_append(&service->entries, txt_e); + // create TXT record + if (txt && *txt) { + txt_e = rr_create(dup_nlabel(nlabel), RR_TXT); // automatically 4500 seconds + rr_list_append(&service->entries, txt_e); - // add TXTs - for (; *txt; txt++) - rr_add_txt(txt_e, *txt); - } + // add TXTs + for (; *txt; txt++) + rr_add_txt(txt_e, *txt); + } - // create SRV record - assert(hostname || svr->hostname); // either one as target - target = hostname ? - create_nlabel(hostname) : - dup_nlabel(svr->hostname); + // create SRV record + assert(hostname || svr->hostname); // either one as target + target = hostname ? create_nlabel(hostname) : dup_nlabel(svr->hostname); - srv_e = rr_create_srv(dup_nlabel(nlabel), port, target); // automatically 4500 seconds - rr_list_append(&service->entries, srv_e); + srv_e = rr_create_srv(dup_nlabel(nlabel), port, target); // automatically 4500 seconds + rr_list_append(&service->entries, srv_e); - // create PTR record for type - ptr_e = rr_create_ptr(type_nlabel, srv_e); // automatically 4500 seconds + // create PTR record for type + ptr_e = rr_create_ptr(type_nlabel, srv_e); // automatically 4500 seconds - // create services PTR record for type - // this enables the type to show up as a "service" - bptr_e = rr_create_ptr(dup_nlabel(SERVICES_DNS_SD_NLABEL), ptr_e); // automatically 4500 seconds + // create services PTR record for type + // this enables the type to show up as a "service" + bptr_e = rr_create_ptr(dup_nlabel(SERVICES_DNS_SD_NLABEL), ptr_e); // automatically 4500 seconds - // modify lists here - pthread_mutex_lock(&svr->data_lock); + // modify lists here + pthread_mutex_lock(&svr->data_lock); - if (txt_e) - rr_group_add(&svr->group, txt_e); - rr_group_add(&svr->group, srv_e); - rr_group_add(&svr->group, ptr_e); - rr_group_add(&svr->group, bptr_e); + if (txt_e) + rr_group_add(&svr->group, txt_e); + rr_group_add(&svr->group, srv_e); + rr_group_add(&svr->group, ptr_e); + rr_group_add(&svr->group, bptr_e); - // append PTR entry to announce list - rr_list_append(&svr->announce, ptr_e); - rr_list_append(&svr->services, ptr_e); + // append PTR entry to announce list + rr_list_append(&svr->announce, ptr_e); + rr_list_append(&svr->services, ptr_e); - pthread_mutex_unlock(&svr->data_lock); + pthread_mutex_unlock(&svr->data_lock); - // don't free type_nlabel - it's with the PTR record - free(nlabel); - free(inst_nlabel); + // don't free type_nlabel - it's with the PTR record + free(nlabel); + free(inst_nlabel); - // notify server - write_pipe(svr->notify_pipe[1], ".", 1); + // notify server + write_pipe(svr->notify_pipe[1], ".", 1); - return service; + return service; } void mdns_service_destroy(struct mdns_service *srv) { - assert(srv != NULL); - rr_list_destroy(srv->entries, 0); - free(srv); + assert(srv != NULL); + rr_list_destroy(srv->entries, 0); + free(srv); } struct mdnsd *mdnsd_start() { - pthread_t tid; - pthread_attr_t attr; + pthread_t tid; + pthread_attr_t attr; - struct mdnsd *server = malloc(sizeof(struct mdnsd)); - memset(server, 0, sizeof(struct mdnsd)); + struct mdnsd *server = malloc(sizeof(struct mdnsd)); + memset(server, 0, sizeof(struct mdnsd)); - if (create_pipe(server->notify_pipe) != 0) { - log_message(LOG_ERR, "pipe(): %m\n"); - free(server); - return NULL; - } + if (create_pipe(server->notify_pipe) != 0) { + log_message(LOG_ERR, "pipe(): %m\n"); + free(server); + return NULL; + } - server->sockfd = create_recv_sock(); - if (server->sockfd < 0) { - log_message(LOG_ERR, "unable to create recv socket"); - free(server); - return NULL; - } + server->sockfd = create_recv_sock(); + if (server->sockfd < 0) { + log_message(LOG_ERR, "unable to create recv socket"); + free(server); + return NULL; + } - pthread_mutex_init(&server->data_lock, NULL); + pthread_mutex_init(&server->data_lock, NULL); - // init thread - pthread_attr_init(&attr); - pthread_attr_setdetachstate(&attr, PTHREAD_CREATE_DETACHED); + // init thread + pthread_attr_init(&attr); + pthread_attr_setdetachstate(&attr, PTHREAD_CREATE_DETACHED); - if (pthread_create(&tid, &attr, (void *(*)(void *)) main_loop, (void *) server) != 0) { - pthread_mutex_destroy(&server->data_lock); - free(server); - return NULL; - } + if (pthread_create(&tid, &attr, (void *(*)(void *))main_loop, (void *)server) != 0) { + pthread_mutex_destroy(&server->data_lock); + free(server); + return NULL; + } - return server; + return server; } void mdnsd_stop(struct mdnsd *s) { - assert(s != NULL); + assert(s != NULL); - struct timeval tv = { - .tv_sec = 0, - .tv_usec = 500 * 1000, - }; + struct timeval tv = { + .tv_sec = 0, .tv_usec = 500 * 1000, + }; - s->stop_flag = 1; - write_pipe(s->notify_pipe[1], ".", 1); + s->stop_flag = 1; + write_pipe(s->notify_pipe[1], ".", 1); - while (s->stop_flag != 2) - select(0, NULL, NULL, NULL, &tv); + while (s->stop_flag != 2) + select(0, NULL, NULL, NULL, &tv); - close_pipe(s->notify_pipe[0]); - close_pipe(s->notify_pipe[1]); + close_pipe(s->notify_pipe[0]); + close_pipe(s->notify_pipe[1]); - pthread_mutex_destroy(&s->data_lock); - rr_group_destroy(s->group); - rr_list_destroy(s->announce, 0); - rr_list_destroy(s->services, 0); + pthread_mutex_destroy(&s->data_lock); + rr_group_destroy(s->group); + rr_list_destroy(s->announce, 0); + rr_list_destroy(s->services, 0); - if (s->hostname) - free(s->hostname); + if (s->hostname) + free(s->hostname); - free(s); + free(s); } diff --git a/tinysvcmdns.h b/tinysvcmdns.h index 98f26fac..c32c4430 100644 --- a/tinysvcmdns.h +++ b/tinysvcmdns.h @@ -45,120 +45,118 @@ #include #endif -#define MALLOC_ZERO_STRUCT(x, type) \ - x = malloc(sizeof(struct type)); \ - memset(x, 0, sizeof(struct type)); - -#define DECL_MALLOC_ZERO_STRUCT(x, type) \ - struct type * MALLOC_ZERO_STRUCT(x, type) - +#define MALLOC_ZERO_STRUCT(x, type) \ + x = malloc(sizeof(struct type)); \ + memset(x, 0, sizeof(struct type)); +#define DECL_MALLOC_ZERO_STRUCT(x, type) struct type *MALLOC_ZERO_STRUCT(x, type) struct rr_data_srv { - uint16_t priority; - uint16_t weight; - uint16_t port; - uint8_t *target; // host + uint16_t priority; + uint16_t weight; + uint16_t port; + uint8_t *target; // host }; struct rr_data_txt { - struct rr_data_txt *next; - uint8_t *txt; + struct rr_data_txt *next; + uint8_t *txt; }; struct rr_data_nsec { - //uint8_t *name; // same as record + // uint8_t *name; // same as record - // NSEC occupies the 47th bit, 5 bytes - //uint8_t bitmap_len; // = 5 - uint8_t bitmap[5]; // network order: first byte contains LSB + // NSEC occupies the 47th bit, 5 bytes + // uint8_t bitmap_len; // = 5 + uint8_t bitmap[5]; // network order: first byte contains LSB }; struct rr_data_ptr { - uint8_t *name; // NULL if entry is to be used - struct rr_entry *entry; + uint8_t *name; // NULL if entry is to be used + struct rr_entry *entry; }; struct rr_data_a { - uint32_t addr; + uint32_t addr; }; struct rr_data_aaaa { - struct in6_addr *addr; + struct in6_addr *addr; }; struct rr_entry { - uint8_t *name; + uint8_t *name; - enum rr_type { - RR_A = 0x01, - RR_PTR = 0x0C, - RR_TXT = 0x10, - RR_AAAA = 0x1C, - RR_SRV = 0x21, - RR_NSEC = 0x2F, - RR_ANY = 0xFF, - } type; + enum rr_type { + RR_A = 0x01, + RR_PTR = 0x0C, + RR_TXT = 0x10, + RR_AAAA = 0x1C, + RR_SRV = 0x21, + RR_NSEC = 0x2F, + RR_ANY = 0xFF, + } type; - uint32_t ttl; + uint32_t ttl; - // for use in Questions only - char unicast_query; + // for use in Questions only + char unicast_query; - // for use in Answers only - char cache_flush; + // for use in Answers only + char cache_flush; - uint16_t rr_class; + uint16_t rr_class; - // RR data - union { - struct rr_data_nsec NSEC; - struct rr_data_srv SRV; - struct rr_data_txt TXT; - struct rr_data_ptr PTR; - struct rr_data_a A; - struct rr_data_aaaa AAAA; - } data; + // RR data + union { + struct rr_data_nsec NSEC; + struct rr_data_srv SRV; + struct rr_data_txt TXT; + struct rr_data_ptr PTR; + struct rr_data_a A; + struct rr_data_aaaa AAAA; + } data; }; struct rr_list { - struct rr_entry *e; - struct rr_list *next; + struct rr_entry *e; + struct rr_list *next; }; struct rr_group { - uint8_t *name; + uint8_t *name; - struct rr_list *rr; + struct rr_list *rr; - struct rr_group *next; + struct rr_group *next; }; -#define MDNS_FLAG_RESP (1 << 15) // Query=0 / Response=1 -#define MDNS_FLAG_AA (1 << 10) // Authoritative -#define MDNS_FLAG_TC (1 << 9) // TrunCation -#define MDNS_FLAG_RD (1 << 8) // Recursion Desired -#define MDNS_FLAG_RA (1 << 7) // Recursion Available -#define MDNS_FLAG_Z (1 << 6) // Reserved (zero) +#define MDNS_FLAG_RESP (1 << 15) // Query=0 / Response=1 +#define MDNS_FLAG_AA (1 << 10) // Authoritative +#define MDNS_FLAG_TC (1 << 9) // TrunCation +#define MDNS_FLAG_RD (1 << 8) // Recursion Desired +#define MDNS_FLAG_RA (1 << 7) // Recursion Available +#define MDNS_FLAG_Z (1 << 6) // Reserved (zero) -#define MDNS_FLAG_GET_RCODE(x) (x & 0x0F) +#define MDNS_FLAG_GET_RCODE(x) (x & 0x0F) #define MDNS_FLAG_GET_OPCODE(x) ((x >> 11) & 0x0F) // gets the PTR target name, either from "name" member or "entry" member -#define MDNS_RR_GET_PTR_NAME(rr) (rr->data.PTR.name != NULL ? rr->data.PTR.name : rr->data.PTR.entry->name) +#define MDNS_RR_GET_PTR_NAME(rr) \ + (rr->data.PTR.name != NULL ? rr->data.PTR.name : rr->data.PTR.entry->name) struct mdns_pkt { - uint16_t id; // transaction ID - uint16_t flags; - uint16_t num_qn; - uint16_t num_ans_rr; - uint16_t num_auth_rr; - uint16_t num_add_rr; + uint16_t id; // transaction ID + uint16_t flags; + uint16_t num_qn; + uint16_t num_ans_rr; + uint16_t num_auth_rr; + uint16_t num_add_rr; - struct rr_list *rr_qn; // questions - struct rr_list *rr_ans; // answer RRs - struct rr_list *rr_auth; // authority RRs - struct rr_list *rr_add; // additional RRs + struct rr_list *rr_qn; // questions + struct rr_list *rr_ans; // answer RRs + struct rr_list *rr_auth; // authority RRs + struct rr_list *rr_add; // additional RRs }; struct mdns_pkt *mdns_parse_pkt(uint8_t *pkt_buf, size_t pkt_len); @@ -197,7 +195,7 @@ uint8_t *join_nlabel(const uint8_t *n1, const uint8_t *n2); // compares 2 names static inline int cmp_nlabel(const uint8_t *L1, const uint8_t *L2) { - return strcmp((char *) L1, (char *) L2); + return strcmp((char *)L1, (char *)L2); } //******************************************************// @@ -225,10 +223,10 @@ void mdnsd_add_rr(struct mdnsd *svr, struct rr_entry *rr); // registers a service with the MDNS responder instance struct mdns_service *mdnsd_register_svc(struct mdnsd *svr, const char *instance_name, - const char *type, uint16_t port, const char *hostname, const char *txt[]); + const char *type, uint16_t port, const char *hostname, + const char *txt[]); // destroys the mdns_service struct returned by mdnsd_register_svc() void mdns_service_destroy(struct mdns_service *srv); - #endif // _TINYSVCMDNS_H