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shairport-sync/audio_alsa.c
T
Jörg Krause 41ebf30a08 Fix usage of ALSA softvol plugin 
Somewhere in the version 3.x the softvol plugin got broken as the volume
change is not applied anymore. Turned out for the softvol plugin no
`volume()` and `parameters()` are defined.
2017-11-20 17:27:47 +01:00

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/*
* libalsa output driver. This file is part of Shairport.
* Copyright (c) Muffinman, Skaman 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.
*/
#define ALSA_PCM_NEW_HW_PARAMS_API
#include "audio.h"
#include "common.h"
#include <alsa/asoundlib.h>
#include <math.h>
#include <memory.h>
#include <pthread.h>
#include <stdio.h>
#include <unistd.h>
static void help(void);
static int init(int argc, char **argv);
static void deinit(void);
static void start(int i_sample_rate, int i_sample_format);
static void play(short buf[], int samples);
static void stop(void);
static void flush(void);
int delay(long *the_delay);
static void volume(double vol);
void do_volume(double vol);
void do_mute(int request);
static void linear_volume(double vol);
static void parameters(audio_parameters *info);
static void mute(int do_mute);
static double set_volume;
static int output_method_signalled = 0;
audio_output audio_alsa = {
.name = "alsa",
.help = &help,
.init = &init,
.deinit = &deinit,
.start = &start,
.stop = &stop,
.flush = &flush,
.delay = &delay,
.play = &play,
.mute = NULL, // a function will be provided if it can, and is allowed to, do hardware mute
.volume = NULL, // a function will be provided if it can do hardware volume
.parameters = &parameters};
static pthread_mutex_t alsa_mutex = PTHREAD_MUTEX_INITIALIZER;
static unsigned int desired_sample_rate;
static enum sps_format_t sample_format;
static snd_pcm_t *alsa_handle = NULL;
static snd_pcm_hw_params_t *alsa_params = NULL;
static snd_ctl_t *ctl = NULL;
static snd_ctl_elem_id_t *elem_id = NULL;
static snd_mixer_t *alsa_mix_handle = NULL;
static snd_mixer_elem_t *alsa_mix_elem = NULL;
static snd_mixer_selem_id_t *alsa_mix_sid = NULL;
static long alsa_mix_minv, alsa_mix_maxv;
static long alsa_mix_mindb, alsa_mix_maxdb;
static char *alsa_out_dev = "default";
static char *alsa_mix_dev = NULL;
static char *alsa_mix_ctrl = "Master";
static int alsa_mix_index = 0;
static int hardware_mixer = 0;
static int has_softvol = 0;
static int volume_set_request = 0; // set when an external request is made to set the volume.
int mute_request_pending = 0; // set when an external request is made to mute or unmute.
int overriding_mute_state_requested = 0; // 1 = mute; 0 = unmute requested
static snd_pcm_sframes_t (*alsa_pcm_write)(snd_pcm_t *, const void *,
snd_pcm_uframes_t) = snd_pcm_writei;
static int play_number;
static int64_t accumulated_delay, accumulated_da_delay;
int alsa_characteristics_already_listed = 0;
static snd_pcm_uframes_t period_size_requested, buffer_size_requested;
static int set_period_size_request, set_buffer_size_request;
static void help(void) {
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");
}
void set_alsa_out_dev(char *dev) { alsa_out_dev = dev; }
int open_mixer() {
if (hardware_mixer) {
debug(2, "Open Mixer");
int ret = 0;
snd_mixer_selem_id_alloca(&alsa_mix_sid);
snd_mixer_selem_id_set_index(alsa_mix_sid, alsa_mix_index);
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");
debug(3, "Mixer device name is \"%s\".", alsa_mix_dev);
if ((snd_mixer_attach(alsa_mix_handle, alsa_mix_dev)) < 0)
die("Failed to attach mixer");
if ((snd_mixer_selem_register(alsa_mix_handle, NULL, NULL)) < 0)
die("Failed to register mixer element");
ret = snd_mixer_load(alsa_mix_handle);
if (ret < 0)
die("Failed to load mixer element");
debug(3, "Mixer Control name is \"%s\".", alsa_mix_ctrl);
alsa_mix_elem = snd_mixer_find_selem(alsa_mix_handle, alsa_mix_sid);
if (!alsa_mix_elem)
die("Failed to find mixer element");
return 1;
} else {
return 0;
}
}
void close_mixer() {
if (alsa_mix_handle) {
snd_mixer_close(alsa_mix_handle);
alsa_mix_handle = NULL;
}
}
#define RELEASE_ALSA_MUTEX_AND_DIE(X) \
pthread_mutex_unlock(&alsa_mutex); \
die(X);
static int init(int argc, char **argv) {
pthread_mutex_lock(&alsa_mutex);
// debug(2,"audio_alsa init called.");
const char *str;
int value;
double dvalue;
// set up default values first
set_period_size_request = 0;
set_buffer_size_request = 0;
config.alsa_use_playback_switch_for_mute = 0; // don't use it by default
config.audio_backend_latency_offset = 0;
config.audio_backend_buffer_desired_length = 0.15;
// get settings from settings file first, allow them to be overridden by
// command line options
// do the "general" audio options. Note, these options are in the "general" stanza!
parse_general_audio_options();
if (config.cfg != NULL) {
/* 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;
}
/* Get the disable_synchronization setting. */
if (config_lookup_string(config.cfg, "alsa.disable_synchronization", &str)) {
if (strcasecmp(str, "no") == 0)
config.no_sync = 0;
else if (strcasecmp(str, "yes") == 0)
config.no_sync = 1;
else {
pthread_mutex_unlock(&alsa_mutex);
die("Invalid disable_synchronization option choice \"%s\". It should be \"yes\" or \"no\"");
}
}
/* Get the mute_using_playback_switch setting. */
if (config_lookup_string(config.cfg, "alsa.mute_using_playback_switch", &str)) {
if (strcasecmp(str, "no") == 0)
config.alsa_use_playback_switch_for_mute = 0;
else if (strcasecmp(str, "yes") == 0)
config.alsa_use_playback_switch_for_mute = 1;
else {
pthread_mutex_unlock(&alsa_mutex);
die("Invalid mute_use_playback_switch option choice \"%s\". It should be \"yes\" or "
"\"no\"");
}
}
/* Get the output format, using the same names as aplay does*/
if (config_lookup_string(config.cfg, "alsa.output_format", &str)) {
if (strcasecmp(str, "S16") == 0)
config.output_format = SPS_FORMAT_S16;
else if (strcasecmp(str, "S24") == 0)
config.output_format = SPS_FORMAT_S24;
else if (strcasecmp(str, "S24_3LE") == 0)
config.output_format = SPS_FORMAT_S24_3LE;
else if (strcasecmp(str, "S24_3BE") == 0)
config.output_format = SPS_FORMAT_S24_3BE;
else if (strcasecmp(str, "S32") == 0)
config.output_format = SPS_FORMAT_S32;
else if (strcasecmp(str, "U8") == 0)
config.output_format = SPS_FORMAT_U8;
else if (strcasecmp(str, "S8") == 0)
config.output_format = SPS_FORMAT_S8;
else {
pthread_mutex_unlock(&alsa_mutex);
die("Invalid output format \"%s\". It should be \"U8\", \"S8\", \"S16\", \"S24\", "
"\"S24_3LE\", \"S24_3BE\" or "
"\"S32\"",
str);
}
}
/* Get the output rate, which must be a multiple of 44,100*/
if (config_lookup_int(config.cfg, "alsa.output_rate", &value)) {
debug(1, "Value read for output rate is %d.", value);
switch (value) {
case 44100:
case 88200:
case 176400:
case 352800:
config.output_rate = value;
break;
default:
pthread_mutex_unlock(&alsa_mutex);
die("Invalid output rate \"%d\". It should be a multiple of 44,100 up to 352,800", value);
}
}
/* Get the use_mmap_if_available setting. */
if (config_lookup_string(config.cfg, "alsa.use_mmap_if_available", &str)) {
if (strcasecmp(str, "no") == 0)
config.no_mmap = 1;
else if (strcasecmp(str, "yes") == 0)
config.no_mmap = 0;
else {
pthread_mutex_unlock(&alsa_mutex);
die("Invalid use_mmap_if_available option choice \"%s\". It should be \"yes\" or \"no\"");
}
}
/* Get the optional period size value */
if (config_lookup_int(config.cfg, "alsa.period_size", &value)) {
set_period_size_request = 1;
debug(1, "Value read for period size is %d.", value);
if (value < 0) {
pthread_mutex_unlock(&alsa_mutex);
die("Invalid alsa period size setting \"%d\". It "
"must be greater than 0.",
value);
} else {
period_size_requested = value;
}
}
/* Get the optional buffer size value */
if (config_lookup_int(config.cfg, "alsa.buffer_size", &value)) {
set_buffer_size_request = 1;
debug(1, "Value read for buffer size is %d.", value);
if (value < 0) {
pthread_mutex_unlock(&alsa_mutex);
die("Invalid alsa buffer size setting \"%d\". It "
"must be greater than 0.",
value);
} else {
buffer_size_requested = value;
}
}
}
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':
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();
pthread_mutex_unlock(&alsa_mutex);
die("Invalid audio option -%c specified", opt);
}
}
if (optind < argc) {
pthread_mutex_unlock(&alsa_mutex);
die("Invalid audio argument: %s", argv[optind]);
}
debug(1, "alsa output device name is \"%s\".", alsa_out_dev);
if (hardware_mixer) {
if (alsa_mix_dev == NULL)
alsa_mix_dev = alsa_out_dev;
// Open mixer
open_mixer();
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_get_playback_dB_range(alsa_mix_elem, &alsa_mix_mindb, &alsa_mix_maxdb) ==
0) {
audio_alsa.volume = &volume; // insert the volume function now we know it can do dB stuff
audio_alsa.parameters = &parameters; // likewise the parameters stuff
if (alsa_mix_mindb == SND_CTL_TLV_DB_GAIN_MUTE) {
// Raspberry Pi does this
debug(1, "Lowest dB value is a mute -- try minimum volume +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 minimum volume + 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);
} else {
// use the linear scale and do the db conversion ourselves
debug(1, "note: the hardware mixer specified -- \"%s\" -- does not have "
"a dB volume scale, so it can't be used. Trying software "
"volume control.",
alsa_mix_ctrl);
if (snd_ctl_open(&ctl, alsa_mix_dev, 0) < 0) {
pthread_mutex_unlock(&alsa_mutex);
die("Cannot open control \"%s\"", alsa_mix_dev);
}
if (snd_ctl_elem_id_malloc(&elem_id) < 0) {
pthread_mutex_unlock(&alsa_mutex);
die("Cannot allocate memory for control \"%s\"", alsa_mix_dev);
}
snd_ctl_elem_id_set_interface(elem_id, SND_CTL_ELEM_IFACE_MIXER);
snd_ctl_elem_id_set_name(elem_id, alsa_mix_ctrl);
if (snd_ctl_get_dB_range(ctl, elem_id, &alsa_mix_mindb, &alsa_mix_maxdb) == 0) {
debug(1, "Volume control \"%s\" has dB volume from %f to %f.", alsa_mix_ctrl,
(1.0 * alsa_mix_mindb) / 100.0, (1.0 * alsa_mix_maxdb) / 100.0);
has_softvol = 1;
audio_alsa.volume = &volume; // insert the volume function now we know it can do dB stuff
audio_alsa.parameters = &parameters; // likewise the parameters stuff
} else {
debug(1, "Cannot get the dB range from the volume control \"%s\"", alsa_mix_ctrl);
}
/*
debug(1, "Min and max volumes are %d and
%d.",alsa_mix_minv,alsa_mix_maxv);
alsa_mix_maxdb = 0;
if ((alsa_mix_maxv!=0) && (alsa_mix_minv!=0))
alsa_mix_mindb =
-20*100*(log10(alsa_mix_maxv*1.0)-log10(alsa_mix_minv*1.0));
else if (alsa_mix_maxv!=0)
alsa_mix_mindb = -20*100*log10(alsa_mix_maxv*1.0);
audio_alsa.volume = &linear_volume; // insert the linear volume function
audio_alsa.parameters = &parameters; // likewise the parameters stuff
debug(1,"Max and min dB calculated are %d and
%d.",alsa_mix_maxdb,alsa_mix_mindb);
*/
}
}
if ((config.alsa_use_playback_switch_for_mute == 1) &&
(snd_mixer_selem_has_playback_switch(alsa_mix_elem))) {
audio_alsa.mute = &mute; // insert the mute function now we know it can do muting stuff
// debug(1, "Has mixer and mute ability we will use.");
} else {
// debug(1, "Has mixer but not using hardware mute.");
}
close_mixer();
} else {
// debug(1, "Has no mixer and thus no hardware mute.");
}
alsa_mix_handle = NULL;
pthread_mutex_unlock(&alsa_mutex);
return 0;
}
static void deinit(void) {
// debug(2,"audio_alsa deinit called.");
stop();
}
int open_alsa_device(void) {
// the alsa mutex is already acquired when this is called
const snd_pcm_uframes_t minimal_buffer_headroom =
352 * 2; // we accept this much headroom in the hardware buffer, but we'll
// accept less
const snd_pcm_uframes_t requested_buffer_headroom =
minimal_buffer_headroom + 2048; // we ask for this much headroom in the
// hardware buffer, but we'll accept less
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, actual_buffer_length;
snd_pcm_access_t access;
// ensure no calls are made to the alsa device enquiring about the buffer length if
// synchronisation is disabled.
if (config.no_sync != 0)
audio_alsa.delay = NULL;
// ensure no calls are made to the alsa device enquiring about the buffer length if
// synchronisation is disabled.
if (config.no_sync != 0)
audio_alsa.delay = NULL;
ret = snd_pcm_open(&alsa_handle, alsa_out_dev, SND_PCM_STREAM_PLAYBACK, 0);
if (ret < 0)
return (ret);
snd_pcm_hw_params_alloca(&alsa_params);
ret = snd_pcm_hw_params_any(alsa_handle, alsa_params);
if (ret < 0) {
pthread_mutex_unlock(&alsa_mutex);
;
die("audio_alsa: Broken configuration for device \"%s\": no configurations "
"available",
alsa_out_dev);
}
if ((config.no_mmap == 0) &&
(snd_pcm_hw_params_set_access(alsa_handle, alsa_params, SND_PCM_ACCESS_MMAP_INTERLEAVED) >=
0)) {
if (output_method_signalled == 0) {
debug(1, "Output written using MMAP");
output_method_signalled = 1;
}
access = SND_PCM_ACCESS_MMAP_INTERLEAVED;
alsa_pcm_write = snd_pcm_mmap_writei;
} else {
if (output_method_signalled == 0) {
debug(1, "Output written with RW");
output_method_signalled = 1;
}
access = SND_PCM_ACCESS_RW_INTERLEAVED;
alsa_pcm_write = snd_pcm_writei;
}
ret = snd_pcm_hw_params_set_access(alsa_handle, alsa_params, access);
if (ret < 0) {
pthread_mutex_unlock(&alsa_mutex);
die("audio_alsa: Access type not available for device \"%s\": %s", alsa_out_dev,
snd_strerror(ret));
}
snd_pcm_format_t sf;
switch (sample_format) {
case SPS_FORMAT_S8:
sf = SND_PCM_FORMAT_S8;
break;
case SPS_FORMAT_U8:
sf = SND_PCM_FORMAT_U8;
break;
case SPS_FORMAT_S16:
sf = SND_PCM_FORMAT_S16;
break;
case SPS_FORMAT_S24:
sf = SND_PCM_FORMAT_S24;
break;
case SPS_FORMAT_S24_3LE:
sf = SND_PCM_FORMAT_S24_3LE;
break;
case SPS_FORMAT_S24_3BE:
sf = SND_PCM_FORMAT_S24_3BE;
break;
case SPS_FORMAT_S32:
sf = SND_PCM_FORMAT_S32;
break;
default:
pthread_mutex_unlock(&alsa_mutex);
die("Unsupported output format at audio_alsa.c");
}
ret = snd_pcm_hw_params_set_format(alsa_handle, alsa_params, sf);
if (ret < 0) {
pthread_mutex_unlock(&alsa_mutex);
die("audio_alsa: Sample format %d not available for device \"%s\": %s", sample_format,
alsa_out_dev, snd_strerror(ret));
}
ret = snd_pcm_hw_params_set_channels(alsa_handle, alsa_params, 2);
if (ret < 0) {
pthread_mutex_unlock(&alsa_mutex);
die("audio_alsa: Channels count (2) not available for device \"%s\": %s", alsa_out_dev,
snd_strerror(ret));
}
ret = snd_pcm_hw_params_set_rate_near(alsa_handle, alsa_params, &my_sample_rate, &dir);
if (ret < 0) {
die("audio_alsa: Rate %iHz not available for playback: %s", desired_sample_rate,
snd_strerror(ret));
}
if (set_period_size_request != 0) {
debug(1, "Attempting to set the period size");
ret = snd_pcm_hw_params_set_period_size_near(alsa_handle, alsa_params, &period_size_requested,
&dir);
if (ret < 0) {
pthread_mutex_unlock(&alsa_mutex);
die("audio_alsa: cannot set period size of %lu: %s", period_size_requested,
snd_strerror(ret));
snd_pcm_uframes_t actual_period_size;
snd_pcm_hw_params_get_period_size(alsa_params, &actual_period_size, &dir);
if (actual_period_size != period_size_requested)
inform("Actual period size set to a different value than requested. Requested: %lu, actual "
"setting: %lu",
period_size_requested, actual_period_size);
}
}
if (set_buffer_size_request != 0) {
debug(1, "Attempting to set the buffer size to %lu", buffer_size_requested);
ret = snd_pcm_hw_params_set_buffer_size_near(alsa_handle, alsa_params, &buffer_size_requested);
if (ret < 0) {
pthread_mutex_unlock(&alsa_mutex);
die("audio_alsa: cannot set buffer size of %lu: %s", buffer_size_requested,
snd_strerror(ret));
snd_pcm_uframes_t actual_buffer_size;
snd_pcm_hw_params_get_buffer_size(alsa_params, &actual_buffer_size);
if (actual_buffer_size != buffer_size_requested)
inform("Actual period size set to a different value than requested. Requested: %lu, actual "
"setting: %lu",
buffer_size, actual_buffer_size);
}
}
ret = snd_pcm_hw_params(alsa_handle, alsa_params);
if (ret < 0) {
pthread_mutex_unlock(&alsa_mutex);
die("audio_alsa: Unable to set hw parameters for device \"%s\": %s.", alsa_out_dev,
snd_strerror(ret));
}
if (my_sample_rate != desired_sample_rate) {
pthread_mutex_unlock(&alsa_mutex);
die("Can't set the D/A converter to %d.", desired_sample_rate);
}
ret = snd_pcm_hw_params_get_buffer_size(alsa_params, &actual_buffer_length);
if (ret < 0) {
pthread_mutex_unlock(&alsa_mutex);
die("audio_alsa: Unable to get hw buffer length for device \"%s\": %s.", alsa_out_dev,
snd_strerror(ret));
}
if (actual_buffer_length < config.audio_backend_buffer_desired_length + minimal_buffer_headroom) {
/*
// the dac buffer is too small, so let's try to set it
buffer_size =
config.audio_backend_buffer_desired_length + requested_buffer_headroom;
ret = snd_pcm_hw_params_set_buffer_size_near(alsa_handle, alsa_params,
&buffer_size);
if (ret < 0)
die("audio_alsa: Unable to set hw buffer size to %lu for device \"%s\": "
"%s.",
config.audio_backend_buffer_desired_length +
requested_buffer_headroom,
alsa_out_dev, snd_strerror(ret));
if (config.audio_backend_buffer_desired_length + minimal_buffer_headroom >
buffer_size) {
die("audio_alsa: Can't set hw buffer size to %lu or more for device "
"\"%s\". Requested size: %lu, granted size: %lu.",
config.audio_backend_buffer_desired_length + minimal_buffer_headroom,
alsa_out_dev, config.audio_backend_buffer_desired_length +
requested_buffer_headroom,
buffer_size);
}
*/
debug(1, "The alsa buffer is smaller (%lu bytes) than the desired backend buffer "
"length (%ld) you have chosen.",
actual_buffer_length, config.audio_backend_buffer_desired_length);
}
if (alsa_characteristics_already_listed == 0) {
alsa_characteristics_already_listed = 1;
int log_level = 2; // the level at which debug information should be output
int rc;
snd_pcm_access_t access_type;
snd_pcm_format_t format_type;
snd_pcm_subformat_t subformat_type;
unsigned int val, val2;
unsigned int uval, uval2;
int sval;
int dir;
snd_pcm_uframes_t frames;
debug(log_level, "PCM handle name = '%s'", snd_pcm_name(alsa_handle));
// ret = snd_pcm_hw_params_any(alsa_handle, alsa_params);
// if (ret < 0) {
// die("audio_alsa: Cannpot get configuration for device \"%s\": no
// configurations
//"
// "available",
// alsa_out_dev);
// }
debug(log_level, "alsa device parameters:");
snd_pcm_hw_params_get_access(alsa_params, &access_type);
debug(log_level, " access type = %s", snd_pcm_access_name(access_type));
snd_pcm_hw_params_get_format(alsa_params, &format_type);
debug(log_level, " format = '%s' (%s)", snd_pcm_format_name(format_type),
snd_pcm_format_description(format_type));
snd_pcm_hw_params_get_subformat(alsa_params, &subformat_type);
debug(log_level, " subformat = '%s' (%s)", snd_pcm_subformat_name(subformat_type),
snd_pcm_subformat_description(subformat_type));
snd_pcm_hw_params_get_channels(alsa_params, &uval);
debug(log_level, " number of channels = %u", uval);
sval = snd_pcm_hw_params_get_sbits(alsa_params);
debug(log_level, " number of significant bits = %d", sval);
snd_pcm_hw_params_get_rate(alsa_params, &uval, &dir);
switch (dir) {
case -1:
debug(log_level, " rate = %u frames per second (<).", uval);
break;
case 0:
debug(log_level, " rate = %u frames per second (precisely).", uval);
break;
case 1:
debug(log_level, " rate = %u frames per second (>).", uval);
break;
}
if (snd_pcm_hw_params_get_rate_numden(alsa_params, &uval, &uval2) == 0)
debug(log_level, " precise (rational) rate = %.3f frames per second (i.e. %u/%u).", uval,
uval2, ((double)uval) / uval2);
else
debug(log_level, " precise (rational) rate information unavailable.");
snd_pcm_hw_params_get_period_time(alsa_params, &uval, &dir);
switch (dir) {
case -1:
debug(log_level, " period_time = %u us (<).", uval);
break;
case 0:
debug(log_level, " period_time = %u us (precisely).", uval);
break;
case 1:
debug(log_level, " period_time = %u us (>).", uval);
break;
}
snd_pcm_hw_params_get_period_size(alsa_params, &frames, &dir);
switch (dir) {
case -1:
debug(log_level, " period_size = %lu frames (<).", frames);
break;
case 0:
debug(log_level, " period_size = %lu frames (precisely).", frames);
break;
case 1:
debug(log_level, " period_size = %lu frames (>).", frames);
break;
}
snd_pcm_hw_params_get_buffer_time(alsa_params, &uval, &dir);
switch (dir) {
case -1:
debug(log_level, " buffer_time = %u us (<).", uval);
break;
case 0:
debug(log_level, " buffer_time = %u us (precisely).", uval);
break;
case 1:
debug(log_level, " buffer_time = %u us (>).", uval);
break;
}
snd_pcm_hw_params_get_buffer_size(alsa_params, &frames);
switch (dir) {
case -1:
debug(log_level, " buffer_size = %lu frames (<).", frames);
break;
case 0:
debug(log_level, " buffer_size = %lu frames (precisely).", frames);
break;
case 1:
debug(log_level, " buffer_size = %lu frames (>).", frames);
break;
}
snd_pcm_hw_params_get_periods(alsa_params, &uval, &dir);
switch (dir) {
case -1:
debug(log_level, " periods_per_buffer = %u (<).", uval);
break;
case 0:
debug(log_level, " periods_per_buffer = %u (precisely).", uval);
break;
case 1:
debug(log_level, " periods_per_buffer = %u (>).", uval);
break;
}
}
return (0);
}
static void start(int i_sample_rate, int i_sample_format) {
// debug(2,"audio_alsa start called.");
if (i_sample_rate == 0)
desired_sample_rate = 44100; // default
else
desired_sample_rate = i_sample_rate; // must be a variable
if (i_sample_format == 0)
sample_format = SPS_FORMAT_S16; // default
else
sample_format = i_sample_format;
}
int delay(long *the_delay) {
// snd_pcm_sframes_t is a signed long -- hence the return of a "long"
int reply;
// debug(3,"audio_alsa delay called.");
if (alsa_handle == NULL) {
return -ENODEV;
} else {
pthread_mutex_lock(&alsa_mutex);
int derr, ignore;
if (snd_pcm_state(alsa_handle) == SND_PCM_STATE_RUNNING) {
*the_delay = 0; // just to see what happens
reply = snd_pcm_delay(alsa_handle, the_delay);
if (reply != 0) {
debug(1, "Error %d in delay(): \"%s\". Delay reported is %d frames.", reply,
snd_strerror(reply), *the_delay);
ignore = snd_pcm_recover(alsa_handle, reply, 1);
}
} else if (snd_pcm_state(alsa_handle) == SND_PCM_STATE_PREPARED) {
*the_delay = 0;
reply = 0; // no error
} else {
if (snd_pcm_state(alsa_handle) == SND_PCM_STATE_XRUN) {
*the_delay = 0;
reply = 0; // no error
} else {
reply = -EIO;
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, 1);
debug(1, "Error preparing after delay error: \"%s\".", snd_strerror(derr));
}
}
pthread_mutex_unlock(&alsa_mutex);
// here, occasionally pretend there's a problem with pcm_get_delay()
// if ((random() % 100000) < 3) // keep it pretty rare
// reply = -EPERM; // pretend something bad has happened
return reply;
}
}
static void play(short buf[], int samples) {
// debug(3,"audio_alsa play called.");
int ret = 0;
if (alsa_handle == NULL) {
pthread_mutex_lock(&alsa_mutex);
ret = open_alsa_device();
if (ret == 0) {
if (audio_alsa.volume)
do_volume(set_volume);
if (audio_alsa.mute)
do_mute(0);
}
pthread_mutex_unlock(&alsa_mutex);
}
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_XRUN) {
if ((err = snd_pcm_prepare(alsa_handle))) {
ignore = snd_pcm_recover(alsa_handle, err, 1);
debug(1, "Error preparing after underrun: \"%s\".", snd_strerror(err));
}
}
if ((snd_pcm_state(alsa_handle) == SND_PCM_STATE_PREPARED) ||
(snd_pcm_state(alsa_handle) == SND_PCM_STATE_RUNNING)) {
if (buf == NULL)
debug(1, "NULL buffer passed to pcm_writei -- skipping it");
if (samples == 0)
debug(1, "empty buffer being passed to pcm_writei -- skipping it");
if ((samples != 0) && (buf != NULL)) {
err = alsa_pcm_write(alsa_handle, (char *)buf, samples);
if (err < 0) {
debug(1, "Error %d writing %d samples in play(): \"%s\".", err, samples,
snd_strerror(err));
ignore = snd_pcm_recover(alsa_handle, err, 1);
}
}
} else {
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, 1);
debug(1, "Error preparing after play error: \"%s\".", snd_strerror(err));
}
}
pthread_mutex_unlock(&alsa_mutex);
}
}
static void flush(void) {
// debug(2,"audio_alsa flush called.");
pthread_mutex_lock(&alsa_mutex);
int derr;
do_mute(1);
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,"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,"Frames successfully dropped.");
*/
/*
if (snd_pcm_state(alsa_handle)==SND_PCM_STATE_PREPARED)
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));
*/
// this is derived from
// http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2latency_8c-example.html#a45
if ((derr = snd_pcm_nonblock(alsa_handle, 0)))
debug(1, "Error %d (\"%s\") unblocking output device.", derr, snd_strerror(derr));
if ((derr = snd_pcm_drain(alsa_handle)))
debug(1, "Error %d (\"%s\") draining output device.", derr, snd_strerror(derr));
if ((derr = snd_pcm_nonblock(alsa_handle, 1)))
debug(1, "Error %d (\"%s\") reblocking output device.", derr, snd_strerror(derr));
if ((derr = snd_pcm_hw_free(alsa_handle)))
debug(1, "Error %d (\"%s\") freeing output device hardware.", derr, snd_strerror(derr));
// flush also closes the device
snd_pcm_close(alsa_handle);
alsa_handle = NULL;
}
pthread_mutex_unlock(&alsa_mutex);
}
static void stop(void) {
// debug(2,"audio_alsa stop called.");
// 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->minimum_volume_dB = alsa_mix_mindb;
info->maximum_volume_dB = alsa_mix_maxdb;
}
void do_volume(double vol) { // caller is assumed to have the alsa_mutex when using this function
debug(3, "Setting volume db to %f.", vol);
set_volume = vol;
if (volume_set_request && open_mixer()) {
if (has_softvol) {
if (ctl && elem_id) {
snd_ctl_elem_value_t *value;
long raw;
if (snd_ctl_convert_from_dB(ctl, elem_id, (long)vol, &raw, 0) < 0)
debug(1, "Failed converting dB gain to raw volume value for the "
"software volume control.");
snd_ctl_elem_value_alloca(&value);
snd_ctl_elem_value_set_id(value, elem_id);
snd_ctl_elem_value_set_integer(value, 0, raw);
snd_ctl_elem_value_set_integer(value, 1, raw);
if (snd_ctl_elem_write(ctl, value) < 0)
debug(1, "Failed to set playback dB volume for the software volume "
"control.");
}
} else {
if (snd_mixer_selem_set_playback_dB_all(alsa_mix_elem, vol, 0) != 0) {
debug(1, "Can't set playback volume accurately to %f dB.", vol);
if (snd_mixer_selem_set_playback_dB_all(alsa_mix_elem, vol, -1) != 0)
if (snd_mixer_selem_set_playback_dB_all(alsa_mix_elem, vol, 1) != 0)
debug(1, "Could not set playback dB volume on the mixer.");
}
}
volume_set_request = 0; // any external request that has been made is now satisfied
// debug(1,"Alsa volume actually set.");
close_mixer();
}
}
void volume(double vol) {
pthread_mutex_lock(&alsa_mutex);
volume_set_request = 1; // an external request has been made to set the volume
do_volume(vol);
pthread_mutex_unlock(&alsa_mutex);
}
/*
static void linear_volume(double vol) {
debug(2, "Setting linear volume to %f.", vol);
set_volume = vol;
if (hardware_mixer && alsa_mix_handle) {
double linear_volume = pow(10, vol);
// debug(1,"Linear volume is %f.",linear_volume);
long int_vol = alsa_mix_minv + (alsa_mix_maxv - alsa_mix_minv) * linear_volume;
// debug(1,"Setting volume to %ld, for volume input of %f.",int_vol,vol);
if (alsa_mix_handle) {
if (snd_mixer_selem_set_playback_volume_all(alsa_mix_elem, int_vol) != 0)
die("Failed to set playback volume");
}
}
}
*/
static void mute(int mute_state_requested) {
// debug(1,"External Mute Request: %d",mute_state_requested);
pthread_mutex_lock(&alsa_mutex);
mute_request_pending = 1;
overriding_mute_state_requested = mute_state_requested;
do_mute(mute_state_requested);
pthread_mutex_unlock(&alsa_mutex);
}
void do_mute(int mute_state_requested) {
// if a mute is requested now, then
// if an external mute request is in place, leave everything muted
// otherwise, if an external mute request is pending, action it
// otherwise, action the do_mute request
int local_mute_state_requested =
overriding_mute_state_requested; // go with whatever was asked by the external "mute" call
// The mute state requested will be actioned unless mute_request_pending is set
// If it is set, then that the pending request will be actioned.
// If the hardware isn't there, or we are not allowed to use it, nothing will be done
// The caller must have the alsa mutex
if (config.alsa_use_playback_switch_for_mute == 1) {
if (mute_request_pending == 0)
local_mute_state_requested = mute_state_requested;
if (open_mixer()) {
if (local_mute_state_requested) {
// debug(1,"Playback Switch mute actually done");
snd_mixer_selem_set_playback_switch_all(alsa_mix_elem, 0);
} else if (overriding_mute_state_requested == 0) {
// debug(1,"Playback Switch unmute actually done");
snd_mixer_selem_set_playback_switch_all(alsa_mix_elem, 1);
}
close_mixer();
}
}
mute_request_pending = 0;
}
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