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shairport-sync/audio_alsa.c

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/*
* libalsa output driver. This file is part of Shairport.
* Copyright (c) Muffinman, Skaman 2013
* Copyright (c) Mike Brady 2014 -- 2022
* 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 <alsa/asoundlib.h>
#include <inttypes.h>
#include <math.h>
#include <memory.h>
#include <pthread.h>
#include <stdio.h>
#include <unistd.h>
#include "config.h"
#include "activity_monitor.h"
#include "audio.h"
#include "common.h"
enum alsa_backend_mode {
abm_disconnected,
abm_connected,
abm_playing
} alsa_backend_state; // under the control of alsa_mutex
typedef struct {
snd_pcm_format_t alsa_code;
int frame_size;
} format_record;
int output_method_signalled = 0; // for reporting whether it's using mmap or not
int delay_type_notified = -1; // for controlling the reporting of whether the output device can do
// precision delays (e.g. alsa->pulsaudio virtual devices can't)
int use_monotonic_clock = 0; // this value will be set when the hardware is initialised
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 int play(void *buf, int samples, __attribute__((unused)) int sample_type,
__attribute__((unused)) uint32_t timestamp,
__attribute__((unused)) uint64_t playtime);
static void stop(void);
static void flush(void);
static int delay(long *the_delay);
static int stats(uint64_t *raw_measurement_time, uint64_t *corrected_measurement_time,
uint64_t *the_delay, uint64_t *frames_sent_to_dac);
static void *alsa_buffer_monitor_thread_code(void *arg);
static void volume(double vol);
static void do_volume(double vol);
static int prepare(void);
static int do_play(void *buf, int samples);
static void parameters(audio_parameters *info);
static int mute(int do_mute); // returns true if it actually is allowed to use the mute
static double set_volume;
audio_output audio_alsa = {
.name = "alsa",
.help = &help,
.init = &init,
.deinit = &deinit,
.prepare = &prepare,
.start = &start,
.stop = &stop,
.is_running = NULL,
.flush = &flush,
.delay = &delay,
.play = &play,
.stats = &stats, // will also include frames of silence sent to stop
// standby mode
// .rate_info = NULL,
.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 = NULL}; // a function will be provided if it can do hardware volume
pthread_mutex_t alsa_mutex = PTHREAD_MUTEX_INITIALIZER;
pthread_mutex_t alsa_mixer_mutex = PTHREAD_MUTEX_INITIALIZER;
pthread_t alsa_buffer_monitor_thread;
// for deciding when to activate mute
// there are two sources of requests to mute -- the backend itself, e.g. when it
// is flushing
// and the player, e.g. when volume goes down to -144, i.e. mute.
// we may not be allowed to use hardware mute, so we must reflect that too.
int mute_requested_externally = 0;
int mute_requested_internally = 0;
// for tracking how long the output device has stalled
uint64_t stall_monitor_start_time; // zero if not initialised / not started /
// zeroed by flush
long stall_monitor_frame_count; // set to delay at start of time, incremented by
// any writes
uint64_t stall_monitor_error_threshold; // if the time is longer than this, it's
// an error
snd_output_t *output = NULL;
int frame_size; // in bytes for interleaved stereo
int alsa_device_initialised; // boolean to ensure the initialisation is only
// done once
yndk_type precision_delay_available_status =
YNDK_DONT_KNOW; // initially, we don't know if the device can do precision delay
snd_pcm_t *alsa_handle = NULL;
int alsa_handle_status =
ENODEV; // if alsa_handle is NULL, this should say why with a unix error code
snd_pcm_hw_params_t *alsa_params = NULL;
snd_pcm_sw_params_t *alsa_swparams = NULL;
snd_ctl_t *ctl = NULL;
snd_ctl_elem_id_t *elem_id = NULL;
snd_mixer_t *alsa_mix_handle = NULL;
snd_mixer_elem_t *alsa_mix_elem = NULL;
snd_mixer_selem_id_t *alsa_mix_sid = NULL;
long alsa_mix_minv, alsa_mix_maxv;
long alsa_mix_mindb, alsa_mix_maxdb;
char *alsa_out_dev = "default";
char *alsa_mix_dev = NULL;
char *alsa_mix_ctrl = NULL;
int alsa_mix_index = 0;
int has_softvol = 0;
int64_t dither_random_number_store = 0;
int volume_set_request = 0; // set when an external request is made to set the volume.
int mixer_volume_setting_gives_mute = 0; // set when it is discovered that
// particular mixer volume setting
// causes a mute.
long alsa_mix_mute; // setting the volume to this value mutes output, if
// mixer_volume_setting_gives_mute is true
int volume_based_mute_is_active =
0; // set when muting is being done by a setting the volume to a magic value
// use this to allow the use of snd_pcm_writei or snd_pcm_mmap_writei
snd_pcm_sframes_t (*alsa_pcm_write)(snd_pcm_t *, const void *, snd_pcm_uframes_t) = snd_pcm_writei;
void handle_unfixable_error(int errorCode) {
if (config.unfixable_error_reported == 0) {
config.unfixable_error_reported = 1;
char messageString[1024];
messageString[0] = '\0';
snprintf(messageString, sizeof(messageString), "output_device_error_%d", errorCode);
if (config.cmd_unfixable) {
command_execute(config.cmd_unfixable, messageString, 1);
} else {
die("An unrecoverable error, \"output_device_error_%d\", has been "
"detected. Doing an emergency exit, as no run_this_if_an_unfixable_error_is_detected "
"program.",
errorCode);
}
}
}
static int precision_delay_and_status(snd_pcm_state_t *state, snd_pcm_sframes_t *delay,
yndk_type *using_update_timestamps);
static int standard_delay_and_status(snd_pcm_state_t *state, snd_pcm_sframes_t *delay,
yndk_type *using_update_timestamps);
// use this to allow the use of standard or precision delay calculations, with standard the, uh,
// standard.
int (*delay_and_status)(snd_pcm_state_t *state, snd_pcm_sframes_t *delay,
yndk_type *using_update_timestamps) = standard_delay_and_status;
// this will return true if the DAC can return precision delay information and false if not
// if it is not yet known, it will test the output device to find out
// note -- once it has done the test, it decides -- even if the delay comes back with
// "don't know", it will take that as a "No" and remember it.
// If you want it to check again, set precision_delay_available_status to YNDK_DONT_KNOW
// first.
static int precision_delay_available() {
if (precision_delay_available_status == YNDK_DONT_KNOW) {
// this is very crude -- if the device is a hardware device, then it's assumed the delay is
// precise
const char *output_device_name = snd_pcm_name(alsa_handle);
int is_a_real_hardware_device = 0;
if (output_device_name != NULL)
is_a_real_hardware_device = (strstr(output_device_name, "hw:") == output_device_name);
// The criteria as to whether precision delay is available
// is whether the device driver returns non-zero update timestamps
// If it does, and the device is a hardware device (i.e. its name begins with "hw:"),
// it is considered that precision delay is available. Otherwise, it's considered to be
// unavailable.
// To test, we play a silence buffer (fairly large to avoid underflow)
// and then we check the delay return. It will tell us if it
// was able to use the (non-zero) update timestamps
int frames_of_silence = 4410;
size_t size_of_silence_buffer = frames_of_silence * frame_size;
void *silence = malloc(size_of_silence_buffer);
if (silence == NULL) {
debug(1, "alsa: precision_delay_available -- failed to "
"allocate memory for a "
"silent frame buffer.");
} else {
pthread_cleanup_push(malloc_cleanup, silence);
int use_dither = 0;
if ((alsa_mix_ctrl == NULL) && (config.ignore_volume_control == 0) &&
(config.airplay_volume != 0.0))
use_dither = 1;
dither_random_number_store =
generate_zero_frames(silence, frames_of_silence, config.output_format,
use_dither, // i.e. with dither
dither_random_number_store);
do_play(silence, frames_of_silence);
pthread_cleanup_pop(1);
// now we can get the delay, and we'll note if it uses update timestamps
yndk_type uses_update_timestamps;
snd_pcm_state_t state;
snd_pcm_sframes_t delay;
int ret = precision_delay_and_status(&state, &delay, &uses_update_timestamps);
// debug(3,"alsa: precision_delay_available asking for delay and status with a return status
// of %d, a delay of %ld and a uses_update_timestamps of %d.", ret, delay,
// uses_update_timestamps);
if (ret == 0) {
if ((uses_update_timestamps == YNDK_YES) && (is_a_real_hardware_device)) {
precision_delay_available_status = YNDK_YES;
debug(2, "alsa: precision delay timing is available.");
} else {
if ((uses_update_timestamps == YNDK_YES) && (!is_a_real_hardware_device)) {
debug(2, "alsa: precision delay timing is not available because it's not definitely a "
"hardware device.");
} else {
debug(2, "alsa: precision delay timing is not available.");
}
precision_delay_available_status = YNDK_NO;
}
}
}
}
return (precision_delay_available_status == YNDK_YES);
}
int alsa_characteristics_already_listed = 0;
snd_pcm_uframes_t period_size_requested, buffer_size_requested;
int set_period_size_request, set_buffer_size_request;
uint64_t frames_sent_for_playing;
// set to true if there has been a discontinuity between the last reported frames_sent_for_playing
// and the present reported frames_sent_for_playing
// Note that it will be set when the device is opened, as any previous figures for
// frames_sent_for_playing (which Shairport Sync might hold) would be invalid.
int frames_sent_break_occurred;
static void help(void) {
printf(" -d output-device set the output device, default is \"default\".\n"
" -c mixer-control set the mixer control name, default is to use no mixer.\n"
" -m mixer-device set the mixer device, default is the output device.\n"
" -i mixer-index set the mixer index, default is 0.\n");
int r = system("if [ -d /proc/asound ] ; then echo \" hardware output devices:\" ; ls -al "
"/proc/asound/ 2>/dev/null | grep '\\->' | tr -s ' ' | cut -d ' ' -f 9 | while "
"read line; do echo \" \\\"hw:$line\\\"\" ; done ; fi");
if (r != 0)
debug(2, "error %d executing a script to list alsa hardware device names", r);
}
void set_alsa_out_dev(char *dev) { alsa_out_dev = dev; } // ugh -- not static!
// assuming pthread cancellation is disabled
// returns zero of all is okay, a Unx error code if there's a problem
static int open_mixer() {
int response = 0;
if (alsa_mix_ctrl != NULL) {
debug(3, "Open Mixer");
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 ((response = snd_mixer_open(&alsa_mix_handle, 0)) < 0) {
debug(1, "Failed to open mixer");
} else {
debug(3, "Mixer device name is \"%s\".", alsa_mix_dev);
if ((response = snd_mixer_attach(alsa_mix_handle, alsa_mix_dev)) < 0) {
debug(1, "Failed to attach mixer");
} else {
if ((response = snd_mixer_selem_register(alsa_mix_handle, NULL, NULL)) < 0) {
debug(1, "Failed to register mixer element");
} else {
if ((response = snd_mixer_load(alsa_mix_handle)) < 0) {
debug(1, "Failed to load mixer element");
} else {
debug(3, "Mixer control is \"%s\",%d.", alsa_mix_ctrl, alsa_mix_index);
alsa_mix_elem = snd_mixer_find_selem(alsa_mix_handle, alsa_mix_sid);
if (!alsa_mix_elem) {
warn("failed to find mixer control \"%s\",%d.", alsa_mix_ctrl, alsa_mix_index);
response = -ENXIO; // don't let this be ENODEV!
}
}
}
}
}
}
return response;
}
// assuming pthread cancellation is disabled
static int close_mixer() {
int ret = 0;
if (alsa_mix_handle) {
ret = snd_mixer_close(alsa_mix_handle);
alsa_mix_handle = NULL;
}
return ret;
}
// assuming pthread cancellation is disabled
static int do_snd_mixer_selem_set_playback_dB_all(snd_mixer_elem_t *mix_elem, double vol) {
int response = 0;
if ((response = snd_mixer_selem_set_playback_dB_all(mix_elem, vol, 0)) != 0) {
debug(1, "Can't set playback volume accurately to %f dB.", vol);
if ((response = snd_mixer_selem_set_playback_dB_all(mix_elem, vol, -1)) != 0)
if ((response = snd_mixer_selem_set_playback_dB_all(mix_elem, vol, 1)) != 0)
debug(1, "Could not set playback dB volume on the mixer.");
}
return response;
}
// This array is a sequence of the output rates to be tried if automatic speed selection is
// requested.
// There is no benefit to upconverting the frame rate, other than for compatibility.
// The lowest rate that the DAC is capable of is chosen.
unsigned int auto_speed_output_rates[] = {
44100,
88200,
176400,
352800,
};
// This array is of all the formats known to Shairport Sync, in order of the SPS_FORMAT definitions,
// with their equivalent alsa codes and their frame sizes.
// If just one format is requested, then its entry is searched for in the array and checked on the
// device
// If auto format is requested, then each entry in turn is tried until a working format is found.
// So, it should be in the search order.
format_record fr[] = {
{SND_PCM_FORMAT_UNKNOWN, 0}, // unknown
{SND_PCM_FORMAT_S8, 2}, {SND_PCM_FORMAT_U8, 2}, {SND_PCM_FORMAT_S16, 4},
{SND_PCM_FORMAT_S16_LE, 4}, {SND_PCM_FORMAT_S16_BE, 4}, {SND_PCM_FORMAT_S24, 8},
{SND_PCM_FORMAT_S24_LE, 8}, {SND_PCM_FORMAT_S24_BE, 8}, {SND_PCM_FORMAT_S24_3LE, 6},
{SND_PCM_FORMAT_S24_3BE, 6}, {SND_PCM_FORMAT_S32, 8}, {SND_PCM_FORMAT_S32_LE, 8},
{SND_PCM_FORMAT_S32_BE, 8}, {SND_PCM_FORMAT_UNKNOWN, 0}, // auto
{SND_PCM_FORMAT_UNKNOWN, 0}, // illegal
};
// This array is the sequence of formats to be tried if automatic selection of the format is
// requested.
// Ideally, audio should pass through Shairport Sync unaltered, apart from occasional interpolation.
// If the user chooses a hardware mixer, then audio could go straight through, unaltered, as signed
// 16 bit stereo.
// However, the user might, at any point, select an option that requires modification, such as
// stereo to mono mixing,
// additional volume attenuation, convolution, and so on. For this reason,
// we look for the greatest depth the DAC is capable of, since upconverting it is completely
// lossless.
// If audio processing is required, then the dither that must be added will
// be added at the lowest possible level.
// Hence, selecting the greatest bit depth is always either beneficial or neutral.
sps_format_t auto_format_check_sequence[] = {
SPS_FORMAT_S32, SPS_FORMAT_S32_LE, SPS_FORMAT_S32_BE, SPS_FORMAT_S24, SPS_FORMAT_S24_LE,
SPS_FORMAT_S24_BE, SPS_FORMAT_S24_3LE, SPS_FORMAT_S24_3BE, SPS_FORMAT_S16, SPS_FORMAT_S16_LE,
SPS_FORMAT_S16_BE, SPS_FORMAT_S8, SPS_FORMAT_U8,
};
// assuming pthread cancellation is disabled
// if do_auto_setting is true and auto format or auto speed has been requested,
// select the settings as appropriate and store them
static int actual_open_alsa_device(int do_auto_setup) {
// 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
actual_sample_rate; // this will be given the rate requested and will be given the actual rate
// snd_pcm_uframes_t frames = 441 * 10;
snd_pcm_uframes_t 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);
// EHOSTDOWN seems to signify that it's a PipeWire pseudo device that can't be accessed by this
// user. So, try the first device ALSA device and log it.
if ((ret == -EHOSTDOWN) && (strcmp(alsa_out_dev, "default") == 0)) {
ret = snd_pcm_open(&alsa_handle, "hw:0", SND_PCM_STREAM_PLAYBACK, 0);
if ((ret == 0) || (ret == -EBUSY)) {
// being busy should be okay
inform("the default ALSA device is inaccessible -- \"hw:0\" used instead.", alsa_out_dev);
set_alsa_out_dev("hw:0");
}
}
if (ret == 0) {
if (alsa_handle_status == -EBUSY)
warn("The output device \"%s\" is no longer busy and will be used by Shairport Sync.",
alsa_out_dev);
alsa_handle_status = ret; // all cool
} else {
alsa_handle = NULL; // to be sure to be sure
if (ret == -EBUSY) {
if (alsa_handle_status != -EBUSY)
warn("The output device \"%s\" is busy and can't be used by Shairport Sync at present.",
alsa_out_dev);
debug(2, "the alsa output_device \"%s\" is busy.", alsa_out_dev);
}
alsa_handle_status = ret;
frames_sent_break_occurred = 1;
return ret;
}
snd_pcm_hw_params_alloca(&alsa_params);
snd_pcm_sw_params_alloca(&alsa_swparams);
ret = snd_pcm_hw_params_any(alsa_handle, alsa_params);
if (ret < 0) {
die("audio_alsa: Broken configuration for device \"%s\": no configurations "
"available",
alsa_out_dev);
return ret;
}
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(3, "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(3, "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) {
die("audio_alsa: Access type not available for device \"%s\": %s", alsa_out_dev,
snd_strerror(ret));
return ret;
}
ret = snd_pcm_hw_params_set_channels(alsa_handle, alsa_params, 2);
if (ret < 0) {
die("audio_alsa: Channels count (2) not available for device \"%s\": %s", alsa_out_dev,
snd_strerror(ret));
return ret;
}
snd_pcm_format_t sf;
if ((do_auto_setup == 0) || (config.output_format_auto_requested == 0)) { // no auto format
if ((config.output_format > SPS_FORMAT_UNKNOWN) && (config.output_format < SPS_FORMAT_AUTO)) {
sf = fr[config.output_format].alsa_code;
frame_size = fr[config.output_format].frame_size;
} else {
warn("alsa: unexpected output format %d. Set to S16_LE.", config.output_format);
config.output_format = SPS_FORMAT_S16_LE;
sf = fr[config.output_format].alsa_code;
frame_size = fr[config.output_format].frame_size;
}
ret = snd_pcm_hw_params_set_format(alsa_handle, alsa_params, sf);
if (ret < 0) {
die("audio_alsa: Alsa sample format %d not available for device \"%s\": %s", sf, alsa_out_dev,
snd_strerror(ret));
return ret;
}
} else { // auto format
int number_of_formats_to_try;
sps_format_t *formats;
formats = auto_format_check_sequence;
number_of_formats_to_try = sizeof(auto_format_check_sequence) / sizeof(sps_format_t);
int i = 0;
int format_found = 0;
sps_format_t trial_format = SPS_FORMAT_UNKNOWN;
while ((i < number_of_formats_to_try) && (format_found == 0)) {
trial_format = formats[i];
sf = fr[trial_format].alsa_code;
frame_size = fr[trial_format].frame_size;
ret = snd_pcm_hw_params_set_format(alsa_handle, alsa_params, sf);
if (ret == 0)
format_found = 1;
else
i++;
}
if (ret == 0) {
config.output_format = trial_format;
debug(2, "alsa: output format chosen is \"%s\".",
sps_format_description_string(config.output_format));
} else {
die("audio_alsa: Could not automatically set the output format for device \"%s\": %s",
alsa_out_dev, snd_strerror(ret));
return ret;
}
}
if ((do_auto_setup == 0) || (config.output_rate_auto_requested == 0)) { // no auto format
actual_sample_rate =
config.output_rate; // this is the requested rate -- it'll be changed to the actual rate
ret = snd_pcm_hw_params_set_rate_near(alsa_handle, alsa_params, &actual_sample_rate, &dir);
if (ret < 0) {
die("audio_alsa: The frame rate of %i frames per second is not available for playback: %s",
config.output_rate, snd_strerror(ret));
return ret;
}
} else {
int number_of_speeds_to_try;
unsigned int *speeds;
speeds = auto_speed_output_rates;
number_of_speeds_to_try = sizeof(auto_speed_output_rates) / sizeof(int);
int i = 0;
int speed_found = 0;
while ((i < number_of_speeds_to_try) && (speed_found == 0)) {
actual_sample_rate = speeds[i];
ret = snd_pcm_hw_params_set_rate_near(alsa_handle, alsa_params, &actual_sample_rate, &dir);
if (ret == 0) {
speed_found = 1;
if (actual_sample_rate != speeds[i])
die("The output DAC can not be set to %d frames per second (fps). The nearest speed "
"available is %d fps.",
speeds[i], actual_sample_rate);
} else {
i++;
}
}
if (ret == 0) {
config.output_rate = actual_sample_rate;
debug(2, "alsa: output speed chosen is %d.", config.output_rate);
} else {
die("audio_alsa: Could not automatically set the output rate for device \"%s\": %s",
alsa_out_dev, snd_strerror(ret));
return ret;
}
}
if (set_period_size_request != 0) {
debug(1, "Attempting to set the period size to %lu", period_size_requested);
ret = snd_pcm_hw_params_set_period_size_near(alsa_handle, alsa_params, &period_size_requested,
&dir);
if (ret < 0) {
warn("audio_alsa: cannot set period size of %lu: %s", period_size_requested,
snd_strerror(ret));
return ret;
} else {
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) {
warn("audio_alsa: cannot set buffer size of %lu: %s", buffer_size_requested,
snd_strerror(ret));
return ret;
} else {
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_requested, actual_buffer_size);
}
}
ret = snd_pcm_hw_params(alsa_handle, alsa_params);
if (ret < 0) {
die("audio_alsa: Unable to set hw parameters for device \"%s\": %s.", alsa_out_dev,
snd_strerror(ret));
return ret;
}
// check parameters after attempting to set them
if (set_period_size_request != 0) {
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) {
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_requested, actual_buffer_size);
}
if (actual_sample_rate != config.output_rate) {
die("Can't set the output DAC to the requested frame rate of %d fps.", config.output_rate);
return -EINVAL;
}
use_monotonic_clock = snd_pcm_hw_params_is_monotonic(alsa_params);
ret = snd_pcm_hw_params_get_buffer_size(alsa_params, &actual_buffer_length);
if (ret < 0) {
warn("audio_alsa: Unable to get hw buffer length for device \"%s\": %s.", alsa_out_dev,
snd_strerror(ret));
return ret;
}
ret = snd_pcm_sw_params_current(alsa_handle, alsa_swparams);
if (ret < 0) {
warn("audio_alsa: Unable to get current sw parameters for device \"%s\": "
"%s.",
alsa_out_dev, snd_strerror(ret));
return ret;
}
ret = snd_pcm_sw_params_set_tstamp_mode(alsa_handle, alsa_swparams, SND_PCM_TSTAMP_ENABLE);
if (ret < 0) {
warn("audio_alsa: Can't enable timestamp mode of device: \"%s\": %s.", alsa_out_dev,
snd_strerror(ret));
return ret;
}
/* write the sw parameters */
ret = snd_pcm_sw_params(alsa_handle, alsa_swparams);
if (ret < 0) {
warn("audio_alsa: Unable to set software parameters of device: \"%s\": %s.", alsa_out_dev,
snd_strerror(ret));
return ret;
}
ret = snd_pcm_prepare(alsa_handle);
if (ret < 0) {
warn("audio_alsa: Unable to prepare the device: \"%s\": %s.", alsa_out_dev, snd_strerror(ret));
return 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 (config.use_precision_timing == YNA_YES)
delay_and_status = precision_delay_and_status;
else if (config.use_precision_timing == YNA_AUTO) {
if (precision_delay_available()) {
delay_and_status = precision_delay_and_status;
debug(2, "alsa: precision timing selected for \"auto\" mode");
}
}
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) && (uval2 != 0))
// watch for a divide by zero too!
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 int open_alsa_device(int do_auto_setup) {
int result;
int oldState;
pthread_setcancelstate(PTHREAD_CANCEL_DISABLE, &oldState); // make this un-cancellable
result = actual_open_alsa_device(do_auto_setup);
pthread_setcancelstate(oldState, NULL);
return result;
}
static int prepare_mixer() {
int response = 0;
// do any alsa device initialisation (general case)
// at present, this is only needed if a hardware mixer is being used
// if there's a hardware mixer, it needs to be initialised before use
if (alsa_mix_ctrl == NULL) {
audio_alsa.volume = NULL;
audio_alsa.parameters = NULL;
audio_alsa.mute = NULL;
} else {
debug(2, "alsa: hardware mixer prepare");
int oldState;
pthread_setcancelstate(PTHREAD_CANCEL_DISABLE, &oldState); // make this un-cancellable
if (alsa_mix_dev == NULL)
alsa_mix_dev = alsa_out_dev;
// Now, start trying to initialise the alsa device with the settings
// obtained
pthread_cleanup_debug_mutex_lock(&alsa_mixer_mutex, 1000, 1);
if (open_mixer() == 0) {
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) {
// For instance, the Raspberry Pi does this
debug(2, "Lowest dB value is a mute");
mixer_volume_setting_gives_mute = 1;
alsa_mix_mute = SND_CTL_TLV_DB_GAIN_MUTE; // this may not be
// necessary -- it's
// always
// going to be SND_CTL_TLV_DB_GAIN_MUTE, right?
// debug(1, "Try minimum volume + 1 as lowest true attenuation
// value");
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(3, "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
warn("The hardware mixer specified -- \"%s\" -- does not have "
"a dB volume scale.",
alsa_mix_ctrl);
if ((response = snd_ctl_open(&ctl, alsa_mix_dev, 0)) < 0) {
warn("Cannot open control \"%s\"", alsa_mix_dev);
}
if ((response = snd_ctl_elem_id_malloc(&elem_id)) < 0) {
debug(1, "Cannot allocate memory for control \"%s\"", alsa_mix_dev);
elem_id = NULL;
} else {
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,
"alsa: hardware mixer \"%s\" selected, with 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);
}
}
}
}
if (((config.alsa_use_hardware_mute == 1) &&
(snd_mixer_selem_has_playback_switch(alsa_mix_elem))) ||
mixer_volume_setting_gives_mute) {
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.");
}
if (response == 0)
response = close_mixer();
}
debug_mutex_unlock(&alsa_mixer_mutex, 3); // release the mutex
pthread_cleanup_pop(0);
pthread_setcancelstate(oldState, NULL);
}
return response;
}
static int alsa_device_init() { return prepare_mixer(); }
static int init(int argc, char **argv) {
// for debugging
snd_output_stdio_attach(&output, stdout, 0);
// debug(2,"audio_alsa init called.");
int response = 0; // this will be what we return to the caller.
alsa_device_initialised = 0;
const char *str;
int value;
// double dvalue;
// set up default values first
alsa_backend_state = abm_disconnected; // startup state
debug(2, "alsa: init() -- alsa_backend_state => abm_disconnected.");
set_period_size_request = 0;
set_buffer_size_request = 0;
config.alsa_use_hardware_mute = 0; // don't use it by default
config.audio_backend_latency_offset = 0;
config.audio_backend_buffer_desired_length = 0.200;
config.audio_backend_buffer_interpolation_threshold_in_seconds =
0.120; // below this, basic interpolation will be used to save time.
config.alsa_maximum_stall_time = 0.200; // 200 milliseconds -- if it takes longer, it's a problem
config.disable_standby_mode_silence_threshold =
0.040; // start sending silent frames if the delay goes below this time
config.disable_standby_mode_silence_scan_interval = 0.004; // check silence threshold this often
stall_monitor_error_threshold =
(uint64_t)1000000 * config.alsa_maximum_stall_time; // stall time max to microseconds;
stall_monitor_error_threshold = (stall_monitor_error_threshold << 32) / 1000000; // now in fp form
debug(1, "alsa: alsa_maximum_stall_time of %f sec.", config.alsa_maximum_stall_time);
stall_monitor_start_time = 0;
stall_monitor_frame_count = 0;
config.disable_standby_mode = disable_standby_off;
config.keep_dac_busy = 0;
config.use_precision_timing = YNA_AUTO;
// 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) {
double dvalue;
/* 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;
}
// Get the Mixer Control Index
if (config_lookup_int(config.cfg, "alsa.mixer_control_index", &value)) {
alsa_mix_index = value;
}
/* 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 {
warn("Invalid disable_synchronization option choice \"%s\". It should "
"be \"yes\" or "
"\"no\". It is set to \"no\".");
config.no_sync = 0;
}
}
/* Get the mute_using_playback_switch setting. */
if (config_lookup_string(config.cfg, "alsa.mute_using_playback_switch", &str)) {
inform("The alsa \"mute_using_playback_switch\" setting is deprecated. "
"Please use the \"use_hardware_mute_if_available\" setting instead.");
if (strcasecmp(str, "no") == 0)
config.alsa_use_hardware_mute = 0;
else if (strcasecmp(str, "yes") == 0)
config.alsa_use_hardware_mute = 1;
else {
warn("Invalid mute_using_playback_switch option choice \"%s\". It "
"should be \"yes\" or "
"\"no\". It is set to \"no\".");
config.alsa_use_hardware_mute = 0;
}
}
/* Get the use_hardware_mute_if_available setting. */
if (config_lookup_string(config.cfg, "alsa.use_hardware_mute_if_available", &str)) {
if (strcasecmp(str, "no") == 0)
config.alsa_use_hardware_mute = 0;
else if (strcasecmp(str, "yes") == 0)
config.alsa_use_hardware_mute = 1;
else {
warn("Invalid use_hardware_mute_if_available option choice \"%s\". It "
"should be \"yes\" or "
"\"no\". It is set to \"no\".");
config.alsa_use_hardware_mute = 0;
}
}
/* Get the output format, using the same names as aplay does*/
if (config_lookup_string(config.cfg, "alsa.output_format", &str)) {
int temp_output_format_auto_requested = config.output_format_auto_requested;
config.output_format_auto_requested = 0; // assume a valid format will be given.
if (strcasecmp(str, "S16") == 0)
config.output_format = SPS_FORMAT_S16;
else if (strcasecmp(str, "S16_LE") == 0)
config.output_format = SPS_FORMAT_S16_LE;
else if (strcasecmp(str, "S16_BE") == 0)
config.output_format = SPS_FORMAT_S16_BE;
else if (strcasecmp(str, "S24") == 0)
config.output_format = SPS_FORMAT_S24;
else if (strcasecmp(str, "S24_LE") == 0)
config.output_format = SPS_FORMAT_S24_LE;
else if (strcasecmp(str, "S24_BE") == 0)
config.output_format = SPS_FORMAT_S24_BE;
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, "S32_LE") == 0)
config.output_format = SPS_FORMAT_S32_LE;
else if (strcasecmp(str, "S32_BE") == 0)
config.output_format = SPS_FORMAT_S32_BE;
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 if (strcasecmp(str, "auto") == 0)
config.output_format_auto_requested = 1;
else {
config.output_format_auto_requested =
temp_output_format_auto_requested; // format was invalid; recall the original setting
warn("Invalid output format \"%s\". It should be \"auto\", \"U8\", \"S8\", "
"\"S16\", \"S24\", \"S24_LE\", \"S24_BE\", "
"\"S24_3LE\", \"S24_3BE\" or "
"\"S32\", \"S32_LE\", \"S32_BE\". It remains set to \"%s\".",
str,
config.output_format_auto_requested == 1
? "auto"
: sps_format_description_string(config.output_format));
}
}
if (config_lookup_string(config.cfg, "alsa.output_rate", &str)) {
if (strcasecmp(str, "auto") == 0) {
config.output_rate_auto_requested = 1;
} else {
if (config.output_rate_auto_requested == 1)
warn("Invalid output rate \"%s\". It should be \"auto\", 44100, 88200, 176400 or 352800. "
"It remains set to \"auto\". Note: numbers should not be placed in quotes.",
str);
else
warn("Invalid output rate \"%s\". It should be \"auto\", 44100, 88200, 176400 or 352800. "
"It remains set to %d. Note: numbers should not be placed in quotes.",
str, config.output_rate);
}
}
/* Get the output rate, which must be a multiple of 44,100*/
if (config_lookup_int(config.cfg, "alsa.output_rate", &value)) {
debug(1, "alsa output rate is %d frames per second", value);
switch (value) {
case 44100:
case 88200:
case 176400:
case 352800:
config.output_rate = value;
config.output_rate_auto_requested = 0;
break;
default:
if (config.output_rate_auto_requested == 1)
warn("Invalid output rate \"%d\". It should be \"auto\", 44100, 88200, 176400 or 352800. "
"It remains set to \"auto\".",
value);
else
warn("Invalid output rate \"%d\".It should be \"auto\", 44100, 88200, 176400 or 352800. "
"It remains set to %d.",
value, config.output_rate);
}
}
/* 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 {
warn("Invalid use_mmap_if_available option choice \"%s\". It should be "
"\"yes\" or \"no\". "
"It remains set to \"yes\".");
config.no_mmap = 0;
}
}
/* 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) {
warn("Invalid alsa period size setting \"%d\". It "
"must be greater than 0. No setting is made.",
value);
set_period_size_request = 0;
} 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) {
warn("Invalid alsa buffer size setting \"%d\". It "
"must be greater than 0. No setting is made.",
value);
set_buffer_size_request = 0;
} else {
buffer_size_requested = value;
}
}
/* Get the optional alsa_maximum_stall_time setting. */
if (config_lookup_float(config.cfg, "alsa.maximum_stall_time", &dvalue)) {
if (dvalue < 0.0) {
warn("Invalid alsa maximum write time setting \"%f\". It "
"must be greater than 0. Default is \"%f\". No setting is made.",
dvalue, config.alsa_maximum_stall_time);
} else {
config.alsa_maximum_stall_time = dvalue;
}
}
/* Get the optional disable_standby_mode_silence_threshold setting. */
if (config_lookup_float(config.cfg, "alsa.disable_standby_mode_silence_threshold", &dvalue)) {
if (dvalue < 0.0) {
warn("Invalid alsa disable_standby_mode_silence_threshold setting \"%f\". It "
"must be greater than 0. Default is \"%f\". No setting is made.",
dvalue, config.disable_standby_mode_silence_threshold);
} else {
config.disable_standby_mode_silence_threshold = dvalue;
}
}
/* Get the optional disable_standby_mode_silence_scan_interval setting. */
if (config_lookup_float(config.cfg, "alsa.disable_standby_mode_silence_scan_interval",
&dvalue)) {
if (dvalue < 0.0) {
warn("Invalid alsa disable_standby_mode_silence_scan_interval setting \"%f\". It "
"must be greater than 0. Default is \"%f\". No setting is made.",
dvalue, config.disable_standby_mode_silence_scan_interval);
} else {
config.disable_standby_mode_silence_scan_interval = dvalue;
}
}
/* Get the optional disable_standby_mode setting. */
if (config_lookup_string(config.cfg, "alsa.disable_standby_mode", &str)) {
if ((strcasecmp(str, "no") == 0) || (strcasecmp(str, "off") == 0) ||
(strcasecmp(str, "never") == 0))
config.disable_standby_mode = disable_standby_off;
else if ((strcasecmp(str, "yes") == 0) || (strcasecmp(str, "on") == 0) ||
(strcasecmp(str, "always") == 0)) {
config.disable_standby_mode = disable_standby_always;
config.keep_dac_busy = 1;
} else if (strcasecmp(str, "auto") == 0)
config.disable_standby_mode = disable_standby_auto;
else {
warn("Invalid disable_standby_mode option choice \"%s\". It should be "
"\"always\", \"auto\" or \"never\". "
"It remains set to \"never\".",
str);
}
}
if (config_lookup_string(config.cfg, "alsa.use_precision_timing", &str)) {
if ((strcasecmp(str, "no") == 0) || (strcasecmp(str, "off") == 0) ||
(strcasecmp(str, "never") == 0))
config.use_precision_timing = YNA_NO;
else if ((strcasecmp(str, "yes") == 0) || (strcasecmp(str, "on") == 0) ||
(strcasecmp(str, "always") == 0)) {
config.use_precision_timing = YNA_YES;
config.keep_dac_busy = 1;
} else if (strcasecmp(str, "auto") == 0)
config.use_precision_timing = YNA_AUTO;
else {
warn("Invalid use_precision_timing option choice \"%s\". It should be "
"\"yes\", \"auto\" or \"no\". "
"It remains set to \"%s\".",
config.use_precision_timing == YNA_NO ? "no"
: config.use_precision_timing == YNA_AUTO ? "auto"
: "yes");
}
}
debug(1, "alsa: disable_standby_mode is \"%s\".",
config.disable_standby_mode == disable_standby_off ? "never"
: config.disable_standby_mode == disable_standby_always ? "always"
: "auto");
debug(1, "alsa: disable_standby_mode_silence_threshold is %f seconds.",
config.disable_standby_mode_silence_threshold);
debug(1, "alsa: disable_standby_mode_silence_scan_interval is %f seconds.",
config.disable_standby_mode_silence_scan_interval);
}
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;
break;
case 'i':
alsa_mix_index = strtol(optarg, NULL, 10);
break;
default:
warn("Invalid audio option \"-%c\" specified -- ignored.", opt);
help();
}
}
if (optind < argc) {
warn("Invalid audio argument: \"%s\" -- ignored", argv[optind]);
}
debug(1, "alsa: output device name is \"%s\".", alsa_out_dev);
// so, now, if the option to keep the DAC running has been selected, start a
// thread to monitor the
// length of the queue
// if the queue gets too short, stuff it with silence
pthread_create(&alsa_buffer_monitor_thread, NULL, &alsa_buffer_monitor_thread_code, NULL);
return response;
}
static void deinit(void) {
int oldState;
pthread_setcancelstate(PTHREAD_CANCEL_DISABLE, &oldState); // make this un-cancellable
// debug(2,"audio_alsa deinit called.");
stop();
debug(2, "Cancel buffer monitor thread.");
pthread_cancel(alsa_buffer_monitor_thread);
debug(3, "Join buffer monitor thread.");
pthread_join(alsa_buffer_monitor_thread, NULL);
pthread_setcancelstate(oldState, NULL);
}
static int set_mute_state() {
int response = 1; // some problem expected, e.g. no mixer or not allowed to use it or disconnected
int oldState;
pthread_setcancelstate(PTHREAD_CANCEL_DISABLE, &oldState); // make this un-cancellable
pthread_cleanup_debug_mutex_lock(&alsa_mixer_mutex, 10000, 0);
if ((alsa_backend_state != abm_disconnected) && (config.alsa_use_hardware_mute == 1) &&
(open_mixer() == 0)) {
response = 0; // okay if actually using the mute facility
debug(2, "alsa: actually set_mute_state");
int mute = 0;
if ((mute_requested_externally != 0) || (mute_requested_internally != 0))
mute = 1;
if (mute == 1) {
debug(2, "alsa: hardware mute switched on");
if (snd_mixer_selem_has_playback_switch(alsa_mix_elem))
snd_mixer_selem_set_playback_switch_all(alsa_mix_elem, 0);
else {
volume_based_mute_is_active = 1;
do_snd_mixer_selem_set_playback_dB_all(alsa_mix_elem, alsa_mix_mute);
}
} else {
debug(2, "alsa: hardware mute switched off");
if (snd_mixer_selem_has_playback_switch(alsa_mix_elem))
snd_mixer_selem_set_playback_switch_all(alsa_mix_elem, 1);
else {
volume_based_mute_is_active = 0;
do_snd_mixer_selem_set_playback_dB_all(alsa_mix_elem, set_volume);
}
}
close_mixer();
}
debug_mutex_unlock(&alsa_mixer_mutex, 3); // release the mutex
pthread_cleanup_pop(0); // release the mutex
pthread_setcancelstate(oldState, NULL);
return response;
}
static void start(__attribute__((unused)) int i_sample_rate,
__attribute__((unused)) int i_sample_format) {
debug(3, "audio_alsa start called.");
// frame_index = 0;
// measurement_data_is_valid = 0;
stall_monitor_start_time = 0;
stall_monitor_frame_count = 0;
if (alsa_device_initialised == 0) {
debug(1, "alsa: start() calling alsa_device_init.");
alsa_device_init();
alsa_device_initialised = 1;
}
}
static int standard_delay_and_status(snd_pcm_state_t *state, snd_pcm_sframes_t *delay,
yndk_type *using_update_timestamps) {
int ret = alsa_handle_status;
if (using_update_timestamps)
*using_update_timestamps = YNDK_NO;
snd_pcm_state_t state_temp = SND_PCM_STATE_DISCONNECTED;
snd_pcm_sframes_t delay_temp = 0;
if (alsa_handle != NULL) {
state_temp = snd_pcm_state(alsa_handle);
if ((state_temp == SND_PCM_STATE_RUNNING) || (state_temp == SND_PCM_STATE_DRAINING)) {
ret = snd_pcm_delay(alsa_handle, &delay_temp);
} else {
// not running, thus no delay information, thus can't check for frame
// rates
// frame_index = 0; // we'll be starting over...
// measurement_data_is_valid = 0;
// delay_temp = 0;
ret = 0;
}
} // else {
// debug(1, "alsa_handle is NULL in standard_delay_and_status.");
// }
stall_monitor_start_time = 0; // zero if not initialised / not started / zeroed by flush
stall_monitor_frame_count = 0; // set to delay at start of time, incremented by any writes
if (delay != NULL)
*delay = delay_temp;
if (state != NULL)
*state = state_temp;
return ret;
}
static int precision_delay_and_status(snd_pcm_state_t *state, snd_pcm_sframes_t *delay,
yndk_type *using_update_timestamps) {
snd_pcm_state_t state_temp = SND_PCM_STATE_DISCONNECTED;
snd_pcm_sframes_t delay_temp = 0;
if (using_update_timestamps)
*using_update_timestamps = YNDK_DONT_KNOW;
int ret = alsa_handle_status;
snd_pcm_status_t *alsa_snd_pcm_status;
snd_pcm_status_alloca(&alsa_snd_pcm_status);
struct timespec tn; // time now
snd_htimestamp_t update_timestamp; // actually a struct timespec
if (alsa_handle != NULL) {
ret = snd_pcm_status(alsa_handle, alsa_snd_pcm_status);
if (ret == 0) {
snd_pcm_status_get_htstamp(alsa_snd_pcm_status, &update_timestamp);
/*
// must be 1.1 or later to use snd_pcm_status_get_driver_htstamp
#if SND_LIB_MINOR != 0
snd_htimestamp_t driver_htstamp;
snd_pcm_status_get_driver_htstamp(alsa_snd_pcm_status, &driver_htstamp);
uint64_t driver_htstamp_ns = driver_htstamp.tv_sec;
driver_htstamp_ns = driver_htstamp_ns * 1000000000;
driver_htstamp_ns = driver_htstamp_ns + driver_htstamp.tv_nsec;
debug(1,"driver_htstamp: %f.", driver_htstamp_ns * 0.000000001);
#endif
*/
state_temp = snd_pcm_status_get_state(alsa_snd_pcm_status);
if ((state_temp == SND_PCM_STATE_RUNNING) || (state_temp == SND_PCM_STATE_DRAINING)) {
// uint64_t update_timestamp_ns =
// update_timestamp.tv_sec * (uint64_t)1000000000 + update_timestamp.tv_nsec;
uint64_t update_timestamp_ns = update_timestamp.tv_sec;
update_timestamp_ns = update_timestamp_ns * 1000000000;
update_timestamp_ns = update_timestamp_ns + update_timestamp.tv_nsec;
// if the update_timestamp is zero, we take this to mean that the device doesn't report
// interrupt timings. (It could be that it's not a real hardware device.)
// so we switch to getting the delay the regular way
// i.e. using snd_pcm_delay ()
if (using_update_timestamps) {
if (update_timestamp_ns == 0)
*using_update_timestamps = YNDK_NO;
else
*using_update_timestamps = YNDK_YES;
}
// user information
if (update_timestamp_ns == 0) {
if (delay_type_notified != 1) {
debug(2, "alsa: update timestamps unavailable");
delay_type_notified = 1;
}
} else {
// diagnostic
if (delay_type_notified != 0) {
debug(2, "alsa: update timestamps available");
delay_type_notified = 0;
}
}
if (update_timestamp_ns == 0) {
ret = snd_pcm_delay(alsa_handle, &delay_temp);
} else {
delay_temp = snd_pcm_status_get_delay(alsa_snd_pcm_status);
/*
// It seems that the alsa library uses CLOCK_REALTIME before 1.0.28, even though
// the check for monotonic returns true. Might have to watch out for this.
#if SND_LIB_MINOR == 0 && SND_LIB_SUBMINOR < 28
clock_gettime(CLOCK_REALTIME, &tn);
#else
clock_gettime(CLOCK_MONOTONIC, &tn);
#endif
*/
if (use_monotonic_clock)
clock_gettime(CLOCK_MONOTONIC, &tn);
else
clock_gettime(CLOCK_REALTIME, &tn);
// uint64_t time_now_ns = tn.tv_sec * (uint64_t)1000000000 + tn.tv_nsec;
uint64_t time_now_ns = tn.tv_sec;
time_now_ns = time_now_ns * 1000000000;
time_now_ns = time_now_ns + tn.tv_nsec;
// see if it's stalled
if ((stall_monitor_start_time != 0) && (stall_monitor_frame_count == delay_temp)) {
// hasn't outputted anything since the last call to delay()
if (((update_timestamp_ns - stall_monitor_start_time) >
stall_monitor_error_threshold) ||
((time_now_ns - stall_monitor_start_time) > stall_monitor_error_threshold)) {
debug(2,
"DAC seems to have stalled with time_now_ns: %" PRIX64
", update_timestamp_ns: %" PRIX64 ", stall_monitor_start_time %" PRIX64
", stall_monitor_error_threshold %" PRIX64 ".",
time_now_ns, update_timestamp_ns, stall_monitor_start_time,
stall_monitor_error_threshold);
debug(2,
"DAC seems to have stalled with time_now: %lx,%lx"
", update_timestamp: %lx,%lx, stall_monitor_start_time %" PRIX64
", stall_monitor_error_threshold %" PRIX64 ".",
tn.tv_sec, tn.tv_nsec, update_timestamp.tv_sec, update_timestamp.tv_nsec,
stall_monitor_start_time, stall_monitor_error_threshold);
ret = sps_extra_code_output_stalled;
}
} else {
stall_monitor_start_time = update_timestamp_ns;
stall_monitor_frame_count = delay_temp;
}
if (ret == 0) {
uint64_t delta = time_now_ns - update_timestamp_ns;
// uint64_t frames_played_since_last_interrupt =
// ((uint64_t)config.output_rate * delta) / 1000000000;
uint64_t frames_played_since_last_interrupt = config.output_rate;
frames_played_since_last_interrupt = frames_played_since_last_interrupt * delta;
frames_played_since_last_interrupt = frames_played_since_last_interrupt / 1000000000;
snd_pcm_sframes_t frames_played_since_last_interrupt_sized =
frames_played_since_last_interrupt;
if ((frames_played_since_last_interrupt_sized < 0) ||
((uint64_t)frames_played_since_last_interrupt_sized !=
frames_played_since_last_interrupt))
debug(1,
"overflow resizing frames_played_since_last_interrupt %" PRIx64
" to frames_played_since_last_interrupt %lx.",
frames_played_since_last_interrupt, frames_played_since_last_interrupt_sized);
delay_temp = delay_temp - frames_played_since_last_interrupt_sized;
}
}
} else { // not running, thus no delay information, thus can't check for
// stall
delay_temp = 0;
stall_monitor_start_time = 0; // zero if not initialised / not started / zeroed by flush
stall_monitor_frame_count = 0; // set to delay at start of time, incremented by any writes
// not running, thus no delay information, thus can't check for frame
// rates
// frame_index = 0; // we'll be starting over...
// measurement_data_is_valid = 0;
}
} else {
debug(1, "alsa: can't get device's status.");
}
} else {
debug(2, "alsa_handle is NULL in precision_delay_and_status!");
}
if (delay != NULL)
*delay = delay_temp;
if (state != NULL)
*state = state_temp;
return ret;
}
static int delay(long *the_delay) {
// returns 0 if the device is in a valid state -- SND_PCM_STATE_RUNNING or
// SND_PCM_STATE_PREPARED
// or SND_PCM_STATE_DRAINING
// and returns the actual delay if running or 0 if prepared in *the_delay
// otherwise return an error code
// the error code could be a Unix errno code or a snderror code, or
// the sps_extra_code_output_stalled or the
// sps_extra_code_output_state_cannot_make_ready codes
int ret = 0;
snd_pcm_sframes_t my_delay = 0;
int oldState;
snd_pcm_state_t state;
pthread_setcancelstate(PTHREAD_CANCEL_DISABLE, &oldState); // make this un-cancellable
pthread_cleanup_debug_mutex_lock(&alsa_mutex, 10000, 0);
ret = delay_and_status(&state, &my_delay, NULL);
debug_mutex_unlock(&alsa_mutex, 0);
pthread_cleanup_pop(0);
pthread_setcancelstate(oldState, NULL);
if (the_delay != NULL) // can't imagine why this might happen
*the_delay = my_delay; // note: snd_pcm_sframes_t is a long
return ret;
}
static int stats(uint64_t *raw_measurement_time, uint64_t *corrected_measurement_time,
uint64_t *the_delay, uint64_t *frames_sent_to_dac) {
// returns 0 if the device is in a valid state -- SND_PCM_STATE_RUNNING or
// SND_PCM_STATE_PREPARED
// or SND_PCM_STATE_DRAINING.
// returns the actual delay if running or 0 if prepared in *the_delay
// returns the present value of frames_sent_for_playing
// otherwise return a non-zero value
int ret = 0;
*the_delay = 0;
int oldState;
snd_pcm_state_t state;
snd_pcm_sframes_t my_delay = 0; // this initialisation is to silence a clang warning
pthread_setcancelstate(PTHREAD_CANCEL_DISABLE, &oldState); // make this un-cancellable
pthread_cleanup_debug_mutex_lock(&alsa_mutex, 10000, 0);
if (alsa_handle == NULL) {
ret = alsa_handle_status;
} else {
*raw_measurement_time =
get_absolute_time_in_ns(); // this is not conditioned ("disciplined") by NTP
*corrected_measurement_time = get_monotonic_time_in_ns(); // this is ("disciplined") by NTP
ret = delay_and_status(&state, &my_delay, NULL);
}
if (ret == 0)
ret = frames_sent_break_occurred; // will be zero unless an error like an underrun occurred
else
ret = 1; // just indicate there was some kind of a break
frames_sent_break_occurred = 0; // reset it.
if (frames_sent_to_dac != NULL)
*frames_sent_to_dac = frames_sent_for_playing;
debug_mutex_unlock(&alsa_mutex, 0);
pthread_cleanup_pop(0);
pthread_setcancelstate(oldState, NULL);
uint64_t hd = my_delay; // note: snd_pcm_sframes_t is a long
*the_delay = hd;
return ret;
}
static int do_play(void *buf, int samples) {
// assuming the alsa_mutex has been acquired
int ret = 0;
if ((samples != 0) && (buf != NULL)) {
int oldState;
pthread_setcancelstate(PTHREAD_CANCEL_DISABLE, &oldState); // make this un-cancellable
snd_pcm_state_t state;
snd_pcm_sframes_t my_delay;
ret = delay_and_status(&state, &my_delay, NULL);
if (ret == 0) { // will be non-zero if an error or a stall
// just check the state of the DAC
if ((state != SND_PCM_STATE_PREPARED) && (state != SND_PCM_STATE_RUNNING) &&
(state != SND_PCM_STATE_XRUN)) {
debug(1, "alsa: DAC in odd SND_PCM_STATE_* %d prior to writing.", state);
}
snd_pcm_state_t prior_state = state; // keep this for afterwards....
// debug(3, "alsa: write %d frames.", samples);
ret = alsa_pcm_write(alsa_handle, buf, samples);
if (ret > 0)
frames_sent_for_playing += ret; // this is the number of frames accepted
if (ret == samples) {
stall_monitor_frame_count += samples;
} else {
frames_sent_break_occurred = 1; // note than an output error has occurred
if (ret == -EPIPE) { /* underrun */
// It could be that the DAC was in the SND_PCM_STATE_XRUN state before
// sending the samples to be output. If so, it will still be in
// the SND_PCM_STATE_XRUN state after the call and it needs to be recovered.
// The underrun occurred in the past, so flagging an
// error at this point is misleading.
// In fact, having put samples in the buffer, we are about to fix it by now
// issuing a snd_pcm_recover().
// So, if state is SND_PCM_STATE_XRUN now, only report it if the state was
// not SND_PCM_STATE_XRUN prior to the call, i.e. report it only
// if we are not trying to recover from a previous underrun.
if (prior_state == SND_PCM_STATE_XRUN)
debug(1, "alsa: recovering from a previous underrun.");
else
debug(1, "alsa: underrun while writing %d samples to alsa device.", samples);
ret = snd_pcm_recover(alsa_handle, ret, 1);
} else if (ret == -ESTRPIPE) { /* suspended */
if (state != prior_state)
debug(1, "alsa: suspended while writing %d samples to alsa device.", samples);
if ((ret = snd_pcm_resume(alsa_handle)) == -ENOSYS)
ret = snd_pcm_prepare(alsa_handle);
} else if (ret >= 0) {
debug(1, "alsa: only %d of %d samples output.", ret, samples);
}
}
}
pthread_setcancelstate(oldState, NULL);
if (ret < 0) {
char errorstring[1024];
strerror_r(-ret, (char *)errorstring, sizeof(errorstring));
debug(1, "alsa: SND_PCM_STATE_* %d, error %d (\"%s\") writing %d samples to alsa device.",
state, ret, (char *)errorstring, samples);
}
if ((ret == -ENOENT) || (ret == -ENODEV)) // if the device isn't there...
handle_unfixable_error(-ret);
}
return ret;
}
static int do_open(int do_auto_setup) {
int ret = 0;
if (alsa_backend_state != abm_disconnected)
debug(1, "alsa: do_open() -- opening the output device when it is already "
"connected");
if (alsa_handle == NULL) {
// debug(1,"alsa: do_open() -- opening the output device");
ret = open_alsa_device(do_auto_setup);
if (ret == 0) {
mute_requested_internally = 0;
if (audio_alsa.volume)
do_volume(set_volume);
if (audio_alsa.mute) {
debug(2, "do_open() set_mute_state");
set_mute_state(); // the mute_requested_externally flag will have been
// set accordingly
// do_mute(0); // complete unmute
}
frames_sent_break_occurred = 1; // there is a discontinuity with
// any previously-reported frame count
frames_sent_for_playing = 0;
alsa_backend_state = abm_connected; // only do this if it really opened it.
} else {
if ((ret == -ENOENT) || (ret == -ENODEV)) // if the device isn't there...
handle_unfixable_error(-ret);
}
} else {
debug(1, "alsa: do_open() -- output device already open.");
}
return ret;
}
static int do_close() {
debug(2, "alsa: do_close()");
if (alsa_backend_state == abm_disconnected)
debug(1, "alsa: do_close() -- closing the output device when it is already "
"disconnected");
int derr = 0;
if (alsa_handle) {
// debug(1,"alsa: do_close() -- closing the output device");
if ((derr = snd_pcm_drop(alsa_handle)))
debug(1, "Error %d (\"%s\") dropping output device.", derr, snd_strerror(derr));
usleep(10000); // wait for the hardware to do its trick. BTW, this make the function pthread
// cancellable
if ((derr = snd_pcm_hw_free(alsa_handle)))
debug(1, "Error %d (\"%s\") freeing the output device hardware.", derr, snd_strerror(derr));
debug(2, "alsa: do_close() -- closing alsa handle");
if ((derr = snd_pcm_close(alsa_handle)))
debug(1, "Error %d (\"%s\") closing the output device.", derr, snd_strerror(derr));
alsa_handle = NULL;
alsa_handle_status = ENODEV; // no device open
} else {
debug(1, "alsa: do_close() -- output device already closed.");
}
alsa_backend_state = abm_disconnected;
return derr;
}
static int sub_flush() {
if (alsa_backend_state == abm_disconnected)
debug(1, "alsa: do_flush() -- closing the output device when it is already "
"disconnected");
int derr = 0;
if (alsa_handle) {
debug(2, "alsa: do_flush() -- flushing the output device");
frames_sent_break_occurred = 1;
if ((derr = snd_pcm_drop(alsa_handle)))
debug(1, "Error %d (\"%s\") dropping output device.", derr, snd_strerror(derr));
if ((derr = snd_pcm_prepare(alsa_handle)))
debug(1, "Error %d (\"%s\") preparing output device after flush.", derr, snd_strerror(derr));
alsa_backend_state = abm_connected;
} else {
debug(1, "alsa: do_flush() -- output device already closed.");
}
return derr;
}
static int play(void *buf, int samples, __attribute__((unused)) int sample_type,
__attribute__((unused)) uint32_t timestamp,
__attribute__((unused)) uint64_t playtime) {
// play() will change the state of the alsa_backend_mode to abm_playing
// also, if the present alsa_backend_state is abm_disconnected, then first the
// DAC must be
// connected
// debug(3,"audio_alsa play called.");
int ret = 0;
pthread_cleanup_debug_mutex_lock(&alsa_mutex, 50000, 0);
if (alsa_backend_state == abm_disconnected) {
ret = do_open(0); // don't try to auto setup
if (ret == 0)
debug(2, "alsa: play() -- opened output device");
}
if (ret == 0) {
if (alsa_backend_state != abm_playing) {
debug(2, "alsa: play() -- alsa_backend_state => abm_playing");
alsa_backend_state = abm_playing;
// mute_requested_internally = 0; // stop requesting a mute for backend's own
// reasons, which might have been a flush
// debug(2, "play() set_mute_state");
// set_mute_state(); // try to action the request and return a status
// do_mute(0); // unmute for backend's reason
}
ret = do_play(buf, samples);
}
debug_mutex_unlock(&alsa_mutex, 0);
pthread_cleanup_pop(0); // release the mutex
return ret;
}
static int prepare(void) {
// this will leave the DAC open / connected.
int ret = 0;
pthread_cleanup_debug_mutex_lock(&alsa_mutex, 50000, 0);
if (alsa_backend_state == abm_disconnected) {
if (alsa_device_initialised == 0) {
// debug(1, "alsa: prepare() calling alsa_device_init.");
alsa_device_init();
alsa_device_initialised = 1;
}
ret = do_open(1); // do auto setup
if (ret == 0)
debug(2, "alsa: prepare() -- opened output device");
}
debug_mutex_unlock(&alsa_mutex, 0);
pthread_cleanup_pop(0); // release the mutex
return ret;
}
static void flush(void) {
// debug(2,"audio_alsa flush called.");
pthread_cleanup_debug_mutex_lock(&alsa_mutex, 10000, 1);
if (alsa_backend_state != abm_disconnected) { // must be playing or connected...
// do nothing for a flush if config.keep_dac_busy is true
if (config.keep_dac_busy == 0) {
sub_flush();
}
} else {
debug(3, "alsa: flush() -- called on a disconnected alsa backend");
}
debug_mutex_unlock(&alsa_mutex, 3);
pthread_cleanup_pop(0); // release the mutex
}
static void stop(void) {
pthread_cleanup_debug_mutex_lock(&alsa_mutex, 10000, 1);
if (alsa_backend_state != abm_disconnected) { // must be playing or connected...
if (config.keep_dac_busy == 0) {
do_close();
}
} else
debug(3, "alsa: stop() -- called on a disconnected alsa backend");
debug_mutex_unlock(&alsa_mutex, 3);
pthread_cleanup_pop(0); // release the mutex
}
static void parameters(audio_parameters *info) {
info->minimum_volume_dB = alsa_mix_mindb;
info->maximum_volume_dB = alsa_mix_maxdb;
}
static 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);
int oldState;
pthread_setcancelstate(PTHREAD_CANCEL_DISABLE, &oldState); // make this un-cancellable
set_volume = vol;
pthread_cleanup_debug_mutex_lock(&alsa_mixer_mutex, 1000, 1);
if (volume_set_request && (open_mixer() == 0)) {
if (has_softvol) {
if (ctl && elem_id) {
snd_ctl_elem_value_t *value;
long raw;
if (snd_ctl_convert_from_dB(ctl, elem_id, 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 (volume_based_mute_is_active == 0) {
// debug(1,"Set alsa volume.");
do_snd_mixer_selem_set_playback_dB_all(alsa_mix_elem, vol);
} else {
debug(2, "Not setting volume because volume-based mute is active");
}
}
volume_set_request = 0; // any external request that has been made is now satisfied
close_mixer();
}
debug_mutex_unlock(&alsa_mixer_mutex, 3);
pthread_cleanup_pop(0); // release the mutex
pthread_setcancelstate(oldState, NULL);
}
static void volume(double vol) {
volume_set_request = 1; // an external request has been made to set the volume
do_volume(vol);
}
/*
static void linear_volume(double vol) {
debug(2, "Setting linear volume to %f.", vol);
set_volume = vol;
if ((alsa_mix_ctrl == NULL) && 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 int mute(int mute_state_requested) { // these would be for external reasons, not
// because of the
// state of the backend.
mute_requested_externally = mute_state_requested; // request a mute for external reasons
debug(2, "mute(%d) set_mute_state", mute_state_requested);
return set_mute_state();
}
/*
static void alsa_buffer_monitor_thread_cleanup_function(__attribute__((unused)) void
*arg) {
debug(1, "alsa: alsa_buffer_monitor_thread_cleanup_function called.");
}
*/
static void *alsa_buffer_monitor_thread_code(__attribute__((unused)) void *arg) {
int frame_count = 0;
int error_count = 0;
int error_detected = 0;
int okb = -1;
// if too many play errors occur early on, we will turn off the disable standby mode
while (error_detected == 0) {
int keep_dac_busy_has_just_gone_off = 0;
if (okb != config.keep_dac_busy) {
if ((okb != 0) && (config.keep_dac_busy == 0))
keep_dac_busy_has_just_gone_off = 1;
debug(2, "keep_dac_busy is now \"%s\"", config.keep_dac_busy == 0 ? "no" : "yes");
okb = config.keep_dac_busy;
}
if ((config.keep_dac_busy != 0) && (alsa_device_initialised == 0)) {
debug(2, "alsa: alsa_buffer_monitor_thread_code() calling "
"alsa_device_init.");
alsa_device_init();
alsa_device_initialised = 1;
}
int sleep_time_us = (int)(config.disable_standby_mode_silence_scan_interval * 1000000);
pthread_cleanup_debug_mutex_lock(&alsa_mutex, 200000, 0);
// check possible state transitions here
if ((alsa_backend_state == abm_disconnected) && (config.keep_dac_busy != 0)) {
// open the dac and move to abm_connected mode
if (do_open(1) == 0) // no automatic setup of rate and speed if necessary
debug(2, "alsa: alsa_buffer_monitor_thread_code() -- output device opened; "
"alsa_backend_state => abm_connected");
} else if ((alsa_backend_state != abm_disconnected) && (keep_dac_busy_has_just_gone_off != 0)) {
stall_monitor_start_time = 0;
// frame_index = 0;
// measurement_data_is_valid = 0;
debug(2, "alsa: alsa_buffer_monitor_thread_code() -- closing the output "
"device");
do_close();
debug(2, "alsa: alsa_buffer_monitor_thread_code() -- alsa_backend_state "
"=> abm_disconnected");
}
// now, if the backend is not in the abm_disconnected state
// and config.keep_dac_busy is true (at the present, this has to be the case
// to be in the
// abm_connected state in the first place...) then do the silence-filling
// thing, if needed /* only if the output device is capable of precision delay */.
if ((alsa_backend_state != abm_disconnected) &&
(config.keep_dac_busy != 0) /* && precision_delay_available() */) {
int reply;
long buffer_size = 0;
snd_pcm_state_t state;
reply = delay_and_status(&state, &buffer_size, NULL);
if (reply != 0) {
buffer_size = 0;
char errorstring[1024];
strerror_r(-reply, (char *)errorstring, sizeof(errorstring));
debug(1, "alsa: alsa_buffer_monitor_thread_code delay error %d: \"%s\".", reply,
(char *)errorstring);
}
long buffer_size_threshold =
(long)(config.disable_standby_mode_silence_threshold * config.output_rate);
size_t size_of_silence_buffer;
if (buffer_size < buffer_size_threshold) {
int frames_of_silence = 1024;
size_of_silence_buffer = frames_of_silence * frame_size;
void *silence = malloc(size_of_silence_buffer);
if (silence == NULL) {
warn("disable_standby_mode has been turned off because a memory allocation error "
"occurred.");
error_detected = 1;
} else {
int ret;
pthread_cleanup_push(malloc_cleanup, silence);
int use_dither = 0;
if ((alsa_mix_ctrl == NULL) && (config.ignore_volume_control == 0) &&
(config.airplay_volume != 0.0))
use_dither = 1;
dither_random_number_store =
generate_zero_frames(silence, frames_of_silence, config.output_format,
use_dither, // i.e. with dither
dither_random_number_store);
ret = do_play(silence, frames_of_silence);
frame_count++;
pthread_cleanup_pop(1); // free malloced buffer
if (ret < 0) {
error_count++;
char errorstring[1024];
strerror_r(-ret, (char *)errorstring, sizeof(errorstring));
debug(2,
"alsa: alsa_buffer_monitor_thread_code error %d (\"%s\") writing %d samples "
"to alsa device -- %d errors in %d trials.",
ret, (char *)errorstring, frames_of_silence, error_count, frame_count);
if ((error_count > 40) && (frame_count < 100)) {
warn("disable_standby_mode has been turned off because too many underruns "
"occurred. Is Shairport Sync outputting to a virtual device or running in a "
"virtual machine?");
error_detected = 1;
}
}
}
}
}
debug_mutex_unlock(&alsa_mutex, 0);
pthread_cleanup_pop(0); // release the mutex
usleep(sleep_time_us); // has a cancellation point in it
}
pthread_exit(NULL);
}
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