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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 -- 2019
* 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 "common.h"
#include "activity_monitor.h"
#include "audio.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;
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);
static void stop(void);
static void flush(void);
int delay(long *the_delay);
int get_rate_information(uint64_t *elapsed_time, uint64_t *frames_played);
void *alsa_buffer_monitor_thread_code(void *arg);
static void volume(double vol);
void do_volume(double vol);
int prepare(void);
static void parameters(audio_parameters *info);
int mute(int do_mute); // returns true if it actually is allowed to use the mute
static double set_volume;
static 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 precison delays (e.g. alsa->pulsaudio virtual devices can't)
int use_monotonic_clock = 0; // this value will be set when the hardware is initialised
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,
.rate_info = &get_rate_information,
.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
static pthread_mutex_t alsa_mutex = PTHREAD_MUTEX_INITIALIZER;
static 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
static 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
snd_pcm_t *alsa_handle = NULL;
static snd_pcm_hw_params_t *alsa_params = NULL;
static snd_pcm_sw_params_t *alsa_swparams = 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;
int64_t dither_random_number_store = 0;
static 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;
int precision_delay_and_status(snd_pcm_state_t *state, snd_pcm_sframes_t *delay, enum yndk_type *using_update_timestamps);
int standard_delay_and_status(snd_pcm_state_t *state, snd_pcm_sframes_t *delay, enum 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, enum yndk_type *using_update_timestamps) = standard_delay_and_status;
int precision_delay_available() {
// 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);
return (strstr(output_device_name,"hw:") == output_device_name);
}
// 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 uint64_t measurement_start_time;
static uint64_t frames_played_at_measurement_start_time;
static uint64_t measurement_time;
static uint64_t frames_played_at_measurement_time;
volatile uint64_t most_recent_write_time;
static uint64_t frames_sent_for_playing;
static uint64_t frame_index;
static int measurement_data_is_valid;
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; }
// assuming pthread cancellation is disabled
int open_mixer() {
int response = 0;
if (hardware_mixer) {
debug(3, "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) {
debug(1, "Failed to open mixer");
response = -1;
} else {
debug(3, "Mixer device name is \"%s\".", alsa_mix_dev);
if ((snd_mixer_attach(alsa_mix_handle, alsa_mix_dev)) < 0) {
debug(1, "Failed to attach mixer");
response = -2;
} else {
if ((snd_mixer_selem_register(alsa_mix_handle, NULL, NULL)) < 0) {
debug(1, "Failed to register mixer element");
response = -3;
} else {
ret = snd_mixer_load(alsa_mix_handle);
if (ret < 0) {
debug(1, "Failed to load mixer element");
response = -4;
} else {
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) {
warn("failed to find mixer control \"%s\".", alsa_mix_ctrl);
response = -5;
} else {
response = 1; // we found a hardware mixer and successfully opened it
}
}
}
}
}
}
return response;
}
// assuming pthread cancellation is disabled
void close_mixer() {
if (alsa_mix_handle) {
snd_mixer_close(alsa_mix_handle);
alsa_mix_handle = NULL;
}
}
// assuming pthread cancellation is disabled
void do_snd_mixer_selem_set_playback_dB_all(snd_mixer_elem_t *mix_elem, double vol) {
if (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 (snd_mixer_selem_set_playback_dB_all(mix_elem, vol, -1) != 0)
if (snd_mixer_selem_set_playback_dB_all(mix_elem, vol, 1) != 0)
debug(1, "Could not set playback dB volume on the mixer.");
}
}
void actual_close_alsa_device() {
debug(1, "actual close");
if (alsa_handle) {
int derr;
if ((derr = snd_pcm_hw_free(alsa_handle)))
debug(1, "Error %d (\"%s\") freeing the output device hardware while "
"closing it.",
derr, snd_strerror(derr));
if ((derr = snd_pcm_close(alsa_handle)))
debug(1, "Error %d (\"%s\") closing the output device.", derr, snd_strerror(derr));
alsa_handle = NULL;
}
}
// 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_S24,4},
{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.
enum 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
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);
if (ret < 0) {
if (ret == -ENOENT) {
warn("the alsa output_device \"%s\" can not be found.", alsa_out_dev);
} else {
char errorstring[1024];
strerror_r(-ret, (char *)errorstring, sizeof(errorstring));
warn("alsa: error %d (\"%s\") opening alsa device \"%s\".", ret, (char *)errorstring, alsa_out_dev);
}
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) {
warn("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) {
warn("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) {
warn("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;
enum 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;
enum 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(1,"alsa: output format chosen is \"%s\".",sps_format_description_string(config.output_format));
} else {
warn("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) {
warn("audio_alsa: Rate %iHz 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])
warn("Speed requested: %d. Speed available: %d.",speeds[i],actual_sample_rate);
} else {
i++;
}
}
if (ret == 0) {
config.output_rate = actual_sample_rate;
debug(1,"alsa: output speed chosen is %d.",config.output_rate);
} else {
warn("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) {
warn("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) {
warn("Can't set the D/A converter to sample rate %d.", 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)
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;
}
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;
}
int do_alsa_device_init_if_needed() {
int response = 0;
// do any alsa device initialisation (general case) if needed
// 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 first use
if (alsa_device_initialised == 0) {
alsa_device_initialised = 1;
if (hardware_mixer) {
debug(2, "alsa: hardware mixer init");
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() == 1) {
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 = ¶meters; // likewise the parameters stuff
if (alsa_mix_mindb == SND_CTL_TLV_DB_GAIN_MUTE) {
// For instance, the Raspberry Pi does this
debug(1, "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 (snd_ctl_open(&ctl, alsa_mix_dev, 0) < 0) {
warn("Cannot open control \"%s\"", alsa_mix_dev);
response = -1;
}
if (snd_ctl_elem_id_malloc(&elem_id) < 0) {
debug(1, "Cannot allocate memory for control \"%s\"", alsa_mix_dev);
elem_id = NULL;
response = -2;
} 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 = ¶meters; // 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 = ¶meters; // 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_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.");
}
close_mixer();
}
debug_mutex_unlock(&alsa_mixer_mutex, 3); // release the mutex
pthread_cleanup_pop(0);
pthread_setcancelstate(oldState, NULL);
}
}
return response;
}
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.audio_backend_silence_threshold =
0.040; // start sending silent frames if the delay goes below this time
config.audio_backend_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,
"stall_monitor_error_threshold is 0x%" PRIx64 ", with alsa_maximum_stall_time of %f sec.",
stall_monitor_error_threshold, 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;
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;