Mike Brady
fd880056fb
Add three parameters to the backend play() function call -- (1) a flag indicating whether the samples are timed or not. If timed, (2) the timestamp and (3) the local time at which the first frame should be heard.
Update missing pipewire library message.
Makefile.am fix.
Update the SHM version and fix the SHM name so that it works with FreeBSD.
Remove redundant (?) AC_HEADER_STDC check.
Fix compilation and installation under FreeBSD -- changes to allow building in a separate directory broke the FreeBSD build process.
Add AC_CHECK_LIB for gcrypt in case PKG_CHECK_MODULES fails to find it.
Find gcrypt using pkg-config
Add code to detect when the RTSP channel goes idle for a period, and use it to reset clocks and timings, etc. so that SPS resumes correctly where the source has gone to sleep while playing a realtime stream and has subsequently woken up.
Don't exit if a UDP Clock Control packet is empty. Also check minimum packet size.
Remove some very experimental code which may cause memory and port leaks.
Strip file path from the filename used in debug messages, information messages, warnings and fatal error messages.
BB fix to Makefile.am -- overwriting the DBUS stuff with MPRIS definitions, duh.
Fix some problems building on FreeBSD and tidy up the use of the "sed" editor.
Fix some problems building on picore.
Fix an uninitialised variable
start using the "nqptp" SMI
85 lines
3.6 KiB
C
85 lines
3.6 KiB
C
/*
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* dummy output driver. This file is part of Shairport.
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* Copyright (c) James Laird 2013
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* All rights reserved.
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*
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* Modifications for audio synchronisation
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* and related work, copyright (c) Mike Brady 2014 -- 2022
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* All rights reserved.
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*
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* Permission is hereby granted, free of charge, to any person
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* obtaining a copy of this software and associated documentation
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* files (the "Software"), to deal in the Software without
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* restriction, including without limitation the rights to use,
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* copy, modify, merge, publish, distribute, sublicense, and/or
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* sell copies of the Software, and to permit persons to whom the
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* Software is furnished to do so, subject to the following conditions:
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*
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* The above copyright notice and this permission notice shall be
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* included in all copies or substantial portions of the Software.
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*
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
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* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES
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* OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND
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* NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT
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* HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY,
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* WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
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* FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR
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* OTHER DEALINGS IN THE SOFTWARE.
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*/
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#include "audio.h"
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#include "common.h"
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#include <inttypes.h> // for PRId64 and friends
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#include <stdio.h>
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#include <sys/time.h>
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#include <unistd.h>
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static int init(__attribute__((unused)) int argc, __attribute__((unused)) char **argv) { return 0; }
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static void deinit(void) {}
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static void start(int sample_rate, __attribute__((unused)) int sample_format) {
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debug(1, "dummy audio output started at %d frames per second.", sample_rate);
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}
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// clang-format off
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// Here is a brief explanation of the parameters:
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// sample_type is 'play_samples_are_untimed' or 'play_samples_are_timed' defined in audio.h.
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// untimed samples are typically silence used by Shairport Sync itself to set up synchronisation or to fill for missing packets of audio -- the timestamp and playtime are zero.
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// timed samples come from the client.
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// timestamp is the RTP timestamp given to the first frame by the client
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// playtime is when the first frame should to be played. It is given in "local time".
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// local time is CLOCK_MONOTONIC_RAW if available, or CLOCK_MONOTONIC otherwise, expressed in nanoseconds as an unsigned 64-bit integer (See get_absolute_time_in_ns() in common.c)
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// clang-format on
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static int play(__attribute__((unused)) void *buf, __attribute__((unused)) int samples,
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int sample_type, uint32_t timestamp, uint64_t playtime) {
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// sample code using some of the extra information
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if (sample_type == play_samples_are_timed) {
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int64_t lead_time = playtime - get_absolute_time_in_ns();
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debug(1, "leadtime for frame %u is %" PRId64 " nanoseconds, i.e. %f seconds.", timestamp,
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lead_time, 0.000000001 * lead_time);
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}
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return 0;
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}
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static void stop(void) { debug(1, "dummy audio stopped\n"); }
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audio_output audio_dummy = {.name = "dummy",
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.help = NULL,
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.init = &init,
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.deinit = &deinit,
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.prepare = NULL,
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.start = &start,
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.stop = &stop,
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.is_running = NULL,
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.flush = NULL,
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.delay = NULL,
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.stats = NULL,
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.play = &play,
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.volume = NULL,
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.parameters = NULL,
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.mute = NULL};
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