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obs-studio/libobs/media-io/audio-io.c
jp9000 ff6cf508e5 libobs/media-io: Improve audio data logging 
This improves logging for when audio data insertion is way out of bounds
or is getting cut off in the front due to a bad negative sync offset.

Instead of throwing out a log message for every time this happens with
each piece of data, it now states when the out of bounds or cutoff has
started and stopped only.
2015-08-27 16:33:07 -07:00

874 lines
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/******************************************************************************
Copyright (C) 2013 by Hugh Bailey <obs.jim@gmail.com>
This program is free software: you can redistribute it and/or modify
it under the terms of the GNU General Public License as published by
the Free Software Foundation, either version 2 of the License, or
(at your option) any later version.
This program is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
GNU General Public License for more details.
You should have received a copy of the GNU General Public License
along with this program. If not, see <http://www.gnu.org/licenses/>.
******************************************************************************/
#include <math.h>
#include <inttypes.h>
#include "../util/threading.h"
#include "../util/darray.h"
#include "../util/circlebuf.h"
#include "../util/platform.h"
#include "../util/profiler.h"
#include "audio-io.h"
#include "audio-resampler.h"
extern profiler_name_store_t *obs_get_profiler_name_store(void);
/* #define DEBUG_AUDIO */
#define nop() do {int invalid = 0;} while(0)
struct audio_input {
struct audio_convert_info conversion;
audio_resampler_t *resampler;
audio_output_callback_t callback;
void *param;
};
static inline void audio_input_free(struct audio_input *input)
{
audio_resampler_destroy(input->resampler);
}
struct audio_line {
char *name;
struct audio_output *audio;
struct circlebuf buffers[MAX_AV_PLANES];
pthread_mutex_t mutex;
DARRAY(uint8_t) volume_buffers[MAX_AV_PLANES];
uint64_t base_timestamp;
uint64_t last_timestamp;
uint64_t next_ts_min;
/* specifies which mixes this line applies to via bits */
uint32_t mixers;
/* states whether this line is still being used. if not, then when the
* buffer is depleted, it's destroyed */
bool alive;
/* gets set when audio is getting cut off in the front of the buffer */
bool audio_getting_cut_off;
/* gets set when audio data is being inserted way outside of bounds of
* the circular buffer */
bool audio_data_out_of_bounds;
struct audio_line **prev_next;
struct audio_line *next;
};
static inline void audio_line_destroy_data(struct audio_line *line)
{
for (size_t i = 0; i < MAX_AV_PLANES; i++) {
circlebuf_free(&line->buffers[i]);
da_free(line->volume_buffers[i]);
}
pthread_mutex_destroy(&line->mutex);
bfree(line->name);
bfree(line);
}
struct audio_mix {
DARRAY(struct audio_input) inputs;
DARRAY(uint8_t) mix_buffers[MAX_AV_PLANES];
};
struct audio_output {
struct audio_output_info info;
size_t block_size;
size_t channels;
size_t planes;
pthread_t thread;
os_event_t *stop_event;
bool initialized;
pthread_mutex_t line_mutex;
struct audio_line *first_line;
pthread_mutex_t input_mutex;
struct audio_mix mixes[MAX_AUDIO_MIXES];
};
static inline void audio_output_removeline(struct audio_output *audio,
struct audio_line *line)
{
pthread_mutex_lock(&audio->line_mutex);
if (line->prev_next)
*line->prev_next = line->next;
if (line->next)
line->next->prev_next = line->prev_next;
pthread_mutex_unlock(&audio->line_mutex);
audio_line_destroy_data(line);
}
/* ------------------------------------------------------------------------- */
/* the following functions are used to calculate frame offsets based upon
* timestamps. this will actually work accurately as long as you handle the
* values correctly */
static inline double ts_to_frames(const audio_t *audio, uint64_t ts)
{
double audio_offset_d = (double)ts;
audio_offset_d /= 1000000000.0;
audio_offset_d *= (double)audio->info.samples_per_sec;
return audio_offset_d;
}
static inline double positive_round(double val)
{
return floor(val+0.5);
}
static int64_t ts_diff_frames(const audio_t *audio, uint64_t ts1, uint64_t ts2)
{
double diff = ts_to_frames(audio, ts1) - ts_to_frames(audio, ts2);
return (int64_t)positive_round(diff);
}
static int64_t ts_diff_bytes(const audio_t *audio, uint64_t ts1, uint64_t ts2)
{
return ts_diff_frames(audio, ts1, ts2) * (int64_t)audio->block_size;
}
/* unless the value is 3+ hours worth of frames, this won't overflow */
static inline uint64_t conv_frames_to_time(const audio_t *audio,
uint32_t frames)
{
return (uint64_t)frames * 1000000000ULL /
(uint64_t)audio->info.samples_per_sec;
}
/* ------------------------------------------------------------------------- */
/* this only really happens with the very initial data insertion. can be
* ignored safely. */
static inline void clear_excess_audio_data(struct audio_line *line,
uint64_t prev_time)
{
size_t size = (size_t)ts_diff_bytes(line->audio, prev_time,
line->base_timestamp);
/*blog(LOG_DEBUG, "Excess audio data for audio line '%s', somehow "
"audio data went back in time by %"PRIu32" bytes. "
"prev_time: %"PRIu64", line->base_timestamp: %"PRIu64,
line->name, (uint32_t)size,
prev_time, line->base_timestamp);*/
if (!line->audio_getting_cut_off) {
blog(LOG_WARNING, "Audio line '%s' audio data currently "
"getting cut off. This could be due to a "
"negative sync offset that's larger than "
"the current audio buffering time.",
line->name);
line->audio_getting_cut_off = true;
}
for (size_t i = 0; i < line->audio->planes; i++) {
size_t clear_size = (size < line->buffers[i].size) ?
size : line->buffers[i].size;
circlebuf_pop_front(&line->buffers[i], NULL, clear_size);
}
}
static inline uint64_t min_uint64(uint64_t a, uint64_t b)
{
return a < b ? a : b;
}
static inline size_t min_size(size_t a, size_t b)
{
return a < b ? a : b;
}
#ifndef CLAMP
#define CLAMP(val, minval, maxval) \
((val > maxval) ? maxval : ((val < minval) ? minval : val))
#endif
#define MIX_BUFFER_SIZE 256
/* TODO: optimize mixing */
static void mix_float(struct audio_output *audio, struct audio_line *line,
size_t size, size_t time_offset, size_t plane)
{
float *mixes[MAX_AUDIO_MIXES];
float vals[MIX_BUFFER_SIZE];
for (size_t mix_idx = 0; mix_idx < MAX_AUDIO_MIXES; mix_idx++) {
uint8_t *bytes = audio->mixes[mix_idx].mix_buffers[plane].array;
mixes[mix_idx] = (float*)&bytes[time_offset];
}
while (size) {
size_t pop_count = min_size(size, sizeof(vals));
size -= pop_count;
circlebuf_pop_front(&line->buffers[plane], vals, pop_count);
pop_count /= sizeof(float);
for (size_t mix_idx = 0; mix_idx < MAX_AUDIO_MIXES; mix_idx++) {
/* only include this audio line in this mix if it's set
* via the line's 'mixes' variable */
if ((line->mixers & (1 << mix_idx)) == 0)
continue;
for (size_t i = 0; i < pop_count; i++) {
*(mixes[mix_idx]++) += vals[i];
}
}
}
}
static inline bool mix_audio_line(struct audio_output *audio,
struct audio_line *line, size_t size, uint64_t timestamp)
{
size_t time_offset = (size_t)ts_diff_bytes(audio,
line->base_timestamp, timestamp);
if (time_offset > size)
return false;
size -= time_offset;
#ifdef DEBUG_AUDIO
blog(LOG_DEBUG, "shaved off %lu bytes", size);
#endif
for (size_t i = 0; i < audio->planes; i++) {
size_t pop_size = min_size(size, line->buffers[i].size);
mix_float(audio, line, pop_size, time_offset, i);
}
return true;
}
static bool resample_audio_output(struct audio_input *input,
struct audio_data *data)
{
bool success = true;
if (input->resampler) {
uint8_t *output[MAX_AV_PLANES];
uint32_t frames;
uint64_t offset;
memset(output, 0, sizeof(output));
success = audio_resampler_resample(input->resampler,
output, &frames, &offset,
(const uint8_t *const *)data->data,
data->frames);
for (size_t i = 0; i < MAX_AV_PLANES; i++)
data->data[i] = output[i];
data->frames = frames;
data->timestamp -= offset;
}
return success;
}
static inline void do_audio_output(struct audio_output *audio,
size_t mix_idx, uint64_t timestamp, uint32_t frames)
{
struct audio_mix *mix = &audio->mixes[mix_idx];
struct audio_data data;
for (size_t i = 0; i < MAX_AV_PLANES; i++)
data.data[i] = mix->mix_buffers[i].array;
data.frames = frames;
data.timestamp = timestamp;
data.volume = 1.0f;
pthread_mutex_lock(&audio->input_mutex);
for (size_t i = mix->inputs.num; i > 0; i--) {
struct audio_input *input = mix->inputs.array+(i-1);
if (resample_audio_output(input, &data))
input->callback(input->param, mix_idx, &data);
}
pthread_mutex_unlock(&audio->input_mutex);
}
static inline void clamp_audio_output(struct audio_output *audio, size_t bytes)
{
size_t float_size = bytes / sizeof(float);
for (size_t mix_idx = 0; mix_idx < MAX_AUDIO_MIXES; mix_idx++) {
struct audio_mix *mix = &audio->mixes[mix_idx];
/* do not process mixing if a specific mix is inactive */
if (!mix->inputs.num)
continue;
for (size_t plane = 0; plane < audio->planes; plane++) {
float *mix_data = (float*)mix->mix_buffers[plane].array;
float *mix_end = &mix_data[float_size];
while (mix_data < mix_end) {
float val = *mix_data;
val = (val > 1.0f) ? 1.0f : val;
val = (val < -1.0f) ? -1.0f : val;
*(mix_data++) = val;
}
}
}
}
static uint64_t mix_and_output(struct audio_output *audio, uint64_t audio_time,
uint64_t prev_time)
{
struct audio_line *line = audio->first_line;
uint32_t frames = (uint32_t)ts_diff_frames(audio, audio_time,
prev_time);
size_t bytes = frames * audio->block_size;
#ifdef DEBUG_AUDIO
blog(LOG_DEBUG, "audio_time: %llu, prev_time: %llu, bytes: %lu",
audio_time, prev_time, bytes);
#endif
/* return an adjusted audio_time according to the amount
* of data that was sampled to ensure seamless transmission */
audio_time = prev_time + conv_frames_to_time(audio, frames);
/* resize and clear mix buffers */
for (size_t mix_idx = 0; mix_idx < MAX_AUDIO_MIXES; mix_idx++) {
struct audio_mix *mix = &audio->mixes[mix_idx];
for (size_t i = 0; i < audio->planes; i++) {
da_resize(mix->mix_buffers[i], bytes);
memset(mix->mix_buffers[i].array, 0, bytes);
}
}
/* mix audio lines */
while (line) {
struct audio_line *next = line->next;
/* if line marked for removal, destroy and move to the next */
if (!line->buffers[0].size) {
if (!line->alive) {
audio_output_removeline(audio, line);
line = next;
continue;
}
}
pthread_mutex_lock(&line->mutex);
if (line->buffers[0].size && line->base_timestamp < prev_time) {
clear_excess_audio_data(line, prev_time);
line->base_timestamp = prev_time;
} else if (line->audio_getting_cut_off) {
line->audio_getting_cut_off = false;
blog(LOG_WARNING, "Audio line '%s' audio data no "
"longer getting cut off.",
line->name);
}
if (mix_audio_line(audio, line, bytes, prev_time))
line->base_timestamp = audio_time;
pthread_mutex_unlock(&line->mutex);
line = next;
}
/* clamps audio data to -1.0..1.0 */
clamp_audio_output(audio, bytes);
/* output */
for (size_t i = 0; i < MAX_AUDIO_MIXES; i++)
do_audio_output(audio, i, prev_time, frames);
return audio_time;
}
/* sample audio 40 times a second */
#define AUDIO_WAIT_TIME (1000/40)
static void *audio_thread(void *param)
{
struct audio_output *audio = param;
uint64_t buffer_time = audio->info.buffer_ms * 1000000;
uint64_t prev_time = os_gettime_ns() - buffer_time;
uint64_t audio_time;
os_set_thread_name("audio-io: audio thread");
const char *audio_thread_name =
profile_store_name(obs_get_profiler_name_store(),
"audio_thread(%s)", audio->info.name);
while (os_event_try(audio->stop_event) == EAGAIN) {
os_sleep_ms(AUDIO_WAIT_TIME);
profile_start(audio_thread_name);
pthread_mutex_lock(&audio->line_mutex);
audio_time = os_gettime_ns() - buffer_time;
audio_time = mix_and_output(audio, audio_time, prev_time);
prev_time = audio_time;
pthread_mutex_unlock(&audio->line_mutex);
profile_end(audio_thread_name);
profile_reenable_thread();
}
return NULL;
}
/* ------------------------------------------------------------------------- */
static size_t audio_get_input_idx(const audio_t *audio, size_t mix_idx,
audio_output_callback_t callback, void *param)
{
const struct audio_mix *mix = &audio->mixes[mix_idx];
for (size_t i = 0; i < mix->inputs.num; i++) {
struct audio_input *input = mix->inputs.array+i;
if (input->callback == callback && input->param == param)
return i;
}
return DARRAY_INVALID;
}
static inline bool audio_input_init(struct audio_input *input,
struct audio_output *audio)
{
if (input->conversion.format != audio->info.format ||
input->conversion.samples_per_sec != audio->info.samples_per_sec ||
input->conversion.speakers != audio->info.speakers) {
struct resample_info from = {
.format = audio->info.format,
.samples_per_sec = audio->info.samples_per_sec,
.speakers = audio->info.speakers
};
struct resample_info to = {
.format = input->conversion.format,
.samples_per_sec = input->conversion.samples_per_sec,
.speakers = input->conversion.speakers
};
input->resampler = audio_resampler_create(&to, &from);
if (!input->resampler) {
blog(LOG_ERROR, "audio_input_init: Failed to "
"create resampler");
return false;
}
} else {
input->resampler = NULL;
}
return true;
}
bool audio_output_connect(audio_t *audio, size_t mi,
const struct audio_convert_info *conversion,
audio_output_callback_t callback, void *param)
{
bool success = false;
if (!audio || mi >= MAX_AUDIO_MIXES) return false;
pthread_mutex_lock(&audio->input_mutex);
if (audio_get_input_idx(audio, mi, callback, param) == DARRAY_INVALID) {
struct audio_mix *mix = &audio->mixes[mi];
struct audio_input input;
input.callback = callback;
input.param = param;
if (conversion) {
input.conversion = *conversion;
} else {
input.conversion.format = audio->info.format;
input.conversion.speakers = audio->info.speakers;
input.conversion.samples_per_sec =
audio->info.samples_per_sec;
}
if (input.conversion.format == AUDIO_FORMAT_UNKNOWN)
input.conversion.format = audio->info.format;
if (input.conversion.speakers == SPEAKERS_UNKNOWN)
input.conversion.speakers = audio->info.speakers;
if (input.conversion.samples_per_sec == 0)
input.conversion.samples_per_sec =
audio->info.samples_per_sec;
success = audio_input_init(&input, audio);
if (success)
da_push_back(mix->inputs, &input);
}
pthread_mutex_unlock(&audio->input_mutex);
return success;
}
void audio_output_disconnect(audio_t *audio, size_t mix_idx,
audio_output_callback_t callback, void *param)
{
if (!audio || mix_idx >= MAX_AUDIO_MIXES) return;
pthread_mutex_lock(&audio->input_mutex);
size_t idx = audio_get_input_idx(audio, mix_idx, callback, param);
if (idx != DARRAY_INVALID) {
struct audio_mix *mix = &audio->mixes[mix_idx];
audio_input_free(mix->inputs.array+idx);
da_erase(mix->inputs, idx);
}
pthread_mutex_unlock(&audio->input_mutex);
}
static inline bool valid_audio_params(const struct audio_output_info *info)
{
return info->format && info->name && info->samples_per_sec > 0 &&
info->speakers > 0;
}
int audio_output_open(audio_t **audio, struct audio_output_info *info)
{
struct audio_output *out;
pthread_mutexattr_t attr;
bool planar = is_audio_planar(info->format);
if (!valid_audio_params(info))
return AUDIO_OUTPUT_INVALIDPARAM;
out = bzalloc(sizeof(struct audio_output));
if (!out)
goto fail;
memcpy(&out->info, info, sizeof(struct audio_output_info));
pthread_mutex_init_value(&out->line_mutex);
out->channels = get_audio_channels(info->speakers);
out->planes = planar ? out->channels : 1;
out->block_size = (planar ? 1 : out->channels) *
get_audio_bytes_per_channel(info->format);
if (pthread_mutexattr_init(&attr) != 0)
goto fail;
if (pthread_mutexattr_settype(&attr, PTHREAD_MUTEX_RECURSIVE) != 0)
goto fail;
if (pthread_mutex_init(&out->line_mutex, &attr) != 0)
goto fail;
if (pthread_mutex_init(&out->input_mutex, &attr) != 0)
goto fail;
if (os_event_init(&out->stop_event, OS_EVENT_TYPE_MANUAL) != 0)
goto fail;
if (pthread_create(&out->thread, NULL, audio_thread, out) != 0)
goto fail;
out->initialized = true;
*audio = out;
return AUDIO_OUTPUT_SUCCESS;
fail:
audio_output_close(out);
return AUDIO_OUTPUT_FAIL;
}
void audio_output_close(audio_t *audio)
{
void *thread_ret;
struct audio_line *line;
if (!audio)
return;
if (audio->initialized) {
os_event_signal(audio->stop_event);
pthread_join(audio->thread, &thread_ret);
}
line = audio->first_line;
while (line) {
struct audio_line *next = line->next;
audio_line_destroy_data(line);
line = next;
}
for (size_t mix_idx = 0; mix_idx < MAX_AUDIO_MIXES; mix_idx++) {
struct audio_mix *mix = &audio->mixes[mix_idx];
for (size_t i = 0; i < mix->inputs.num; i++)
audio_input_free(mix->inputs.array+i);
for (size_t i = 0; i < MAX_AV_PLANES; i++)
da_free(mix->mix_buffers[i]);
da_free(mix->inputs);
}
os_event_destroy(audio->stop_event);
pthread_mutex_destroy(&audio->line_mutex);
bfree(audio);
}
audio_line_t *audio_output_create_line(audio_t *audio, const char *name,
uint32_t mixers)
{
if (!audio) return NULL;
struct audio_line *line = bzalloc(sizeof(struct audio_line));
line->alive = true;
line->audio = audio;
line->mixers = mixers;
if (pthread_mutex_init(&line->mutex, NULL) != 0) {
blog(LOG_ERROR, "audio_output_createline: Failed to create "
"mutex");
bfree(line);
return NULL;
}
pthread_mutex_lock(&audio->line_mutex);
if (audio->first_line) {
audio->first_line->prev_next = &line->next;
line->next = audio->first_line;
}
line->prev_next = &audio->first_line;
audio->first_line = line;
pthread_mutex_unlock(&audio->line_mutex);
line->name = bstrdup(name ? name : "(unnamed audio line)");
return line;
}
const struct audio_output_info *audio_output_get_info(const audio_t *audio)
{
return audio ? &audio->info : NULL;
}
void audio_line_destroy(struct audio_line *line)
{
if (line) {
if (!line->buffers[0].size)
audio_output_removeline(line->audio, line);
else
line->alive = false;
}
}
bool audio_output_active(const audio_t *audio)
{
if (!audio) return false;
for (size_t mix_idx = 0; mix_idx < MAX_AUDIO_MIXES; mix_idx++) {
const struct audio_mix *mix = &audio->mixes[mix_idx];
if (mix->inputs.num != 0)
return true;
}
return false;
}
size_t audio_output_get_block_size(const audio_t *audio)
{
return audio ? audio->block_size : 0;
}
size_t audio_output_get_planes(const audio_t *audio)
{
return audio ? audio->planes : 0;
}
size_t audio_output_get_channels(const audio_t *audio)
{
return audio ? audio->channels : 0;
}
uint32_t audio_output_get_sample_rate(const audio_t *audio)
{
return audio ? audio->info.samples_per_sec : 0;
}
/* TODO: optimize these two functions */
static inline void mul_vol_float(float *array, float volume, size_t count)
{
for (size_t i = 0; i < count; i++)
array[i] *= volume;
}
static void audio_line_place_data_pos(struct audio_line *line,
const struct audio_data *data, size_t position)
{
bool planar = line->audio->planes > 1;
size_t total_num = data->frames * (planar ? 1 : line->audio->channels);
size_t total_size = data->frames * line->audio->block_size;
for (size_t i = 0; i < line->audio->planes; i++) {
da_copy_array(line->volume_buffers[i], data->data[i],
total_size);
uint8_t *array = line->volume_buffers[i].array;
switch (line->audio->info.format) {
case AUDIO_FORMAT_FLOAT:
case AUDIO_FORMAT_FLOAT_PLANAR:
mul_vol_float((float*)array, data->volume, total_num);
break;
default:
blog(LOG_ERROR, "audio_line_place_data_pos: "
"Unsupported or unknown format");
break;
}
circlebuf_place(&line->buffers[i], position,
line->volume_buffers[i].array, total_size);
}
}
static inline uint64_t smooth_ts(struct audio_line *line, uint64_t timestamp)
{
if (!line->next_ts_min)
return timestamp;
bool ts_under = (timestamp < line->next_ts_min);
uint64_t diff = ts_under ?
(line->next_ts_min - timestamp) :
(timestamp - line->next_ts_min);
#ifdef DEBUG_AUDIO
if (diff >= TS_SMOOTHING_THRESHOLD)
blog(LOG_DEBUG, "above TS smoothing threshold by %"PRIu64,
diff);
#endif
return (diff < TS_SMOOTHING_THRESHOLD) ? line->next_ts_min : timestamp;
}
static bool audio_line_place_data(struct audio_line *line,
const struct audio_data *data)
{
int64_t pos;
uint64_t timestamp = smooth_ts(line, data->timestamp);
pos = ts_diff_bytes(line->audio, timestamp, line->base_timestamp);
if (pos < 0) {
return false;
}
line->next_ts_min =
timestamp + conv_frames_to_time(line->audio, data->frames);
#ifdef DEBUG_AUDIO
blog(LOG_DEBUG, "data->timestamp: %llu, line->base_timestamp: %llu, "
"pos: %lu, bytes: %lu, buf size: %lu",
timestamp, line->base_timestamp, pos,
data->frames * line->audio->block_size,
line->buffers[0].size);
#endif
audio_line_place_data_pos(line, data, (size_t)pos);
return true;
}
#define MAX_DELAY_NS 6000000000ULL
/* prevent insertation of data too far away from expected audio timing */
static inline bool valid_timestamp_range(struct audio_line *line, uint64_t ts)
{
uint64_t buffer_ns = 1000000ULL * line->audio->info.buffer_ms;
uint64_t max_ts = line->base_timestamp + buffer_ns + MAX_DELAY_NS;
return ts >= line->base_timestamp && ts < max_ts;
}
void audio_line_output(audio_line_t *line, const struct audio_data *data)
{
bool inserted_audio = false;
if (!line || !data) return;
pthread_mutex_lock(&line->mutex);
if (!line->buffers[0].size) {
line->base_timestamp = data->timestamp -
line->audio->info.buffer_ms * 1000000;
inserted_audio = audio_line_place_data(line, data);
} else if (valid_timestamp_range(line, data->timestamp)) {
inserted_audio = audio_line_place_data(line, data);
}
if (!inserted_audio) {
if (!line->audio_data_out_of_bounds) {
blog(LOG_WARNING, "Audio line '%s' currently "
"receiving out of bounds audio "
"data. This can sometimes happen "
"if there's a pause in the thread.",
line->name);
line->audio_data_out_of_bounds = true;
}
/*blog(LOG_DEBUG, "Bad timestamp for audio line '%s', "
"data->timestamp: %"PRIu64", "
"line->base_timestamp: %"PRIu64". This can "
"sometimes happen when there's a pause in "
"the threads.", line->name, data->timestamp,
line->base_timestamp);*/
} else if (line->audio_data_out_of_bounds) {
blog(LOG_WARNING, "Audio line '%s' no longer receiving "
"out of bounds audio data.", line->name);
line->audio_data_out_of_bounds = false;
}
pthread_mutex_unlock(&line->mutex);
}
void audio_line_set_mixers(audio_line_t *line, uint32_t mixers)
{
if (!!line)
line->mixers = mixers;
}
uint32_t audio_line_get_mixers(audio_line_t *line)
{
return !!line ? line->mixers : 0;
}
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