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obs-studio/libobs/media-io/audio-io.c

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2013-10-01 04:37:13 +02:00
/******************************************************************************
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
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(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>
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#include <inttypes.h>
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#include "../util/threading.h"
#include "../util/darray.h"
#include "../util/circlebuf.h"
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#include "../util/platform.h"
#include "audio-io.h"
#include "audio-resampler.h"
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/* #define DEBUG_AUDIO */
#define nop() do {int invalid = 0;} while(0)
struct audio_input {
struct audio_convert_info conversion;
audio_resampler_t resampler;
Implement encoder interface (still preliminary) - Implement OBS encoder interface. It was previously incomplete, but now is reaching some level of completion, though probably should still be considered preliminary. I had originally implemented it so that encoders only have a 'reset' function to reset their parameters, but I felt that having both a 'start' and 'stop' function would be useful. Encoders are now assigned to a specific video/audio media output each rather than implicitely assigned to the main obs video/audio contexts. This allows separate encoder contexts that aren't necessarily assigned to the main video/audio context (which is useful for things such as recording specific sources). Will probably have to do this for regular obs outputs as well. When creating an encoder, you must now explicitely state whether that encoder is an audio or video encoder. Audio and video can optionally be automatically converted depending on what the encoder specifies. When something 'attaches' to an encoder, the first attachment starts the encoder, and the encoder automatically attaches to the media output context associated with it. Subsequent attachments won't have the same effect, they will just start receiving the same encoder data when the next keyframe plays (along with SEI if any). When detaching from the encoder, the last detachment will fully stop the encoder and detach the encoder from the media output context associated with the encoder. SEI must actually be exported separately; because new encoder attachments may not always be at the beginning of the stream, the first keyframe they get must have that SEI data in it. If the encoder has SEI data, it needs only add one small function to simply query that SEI data, and then that data will be handled automatically by libobs for all subsequent encoder attachments. - Implement x264 encoder plugin, move x264 files to separate plugin to separate necessary dependencies. - Change video/audio frame output structures to not use const qualifiers to prevent issues with non-const function usage elsewhere. This was an issue when writing the x264 encoder, as the x264 encoder expects non-const frame data. Change stagesurf_map to return a non-const data type to prevent this as well. - Change full range parameter of video scaler to be an enum rather than boolean
2014-03-17 00:21:34 +01:00
void (*callback)(void *param, struct audio_data *data);
void *param;
};
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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;
/* states whether this line is still being used. if not, then when the
* buffer is depleted, it's destroyed */
bool alive;
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);
}
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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;
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DARRAY(uint8_t) mix_buffers[MAX_AV_PLANES];
bool initialized;
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pthread_mutex_t line_mutex;
struct audio_line *first_line;
pthread_mutex_t input_mutex;
DARRAY(struct audio_input) inputs;
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};
static inline void audio_output_removeline(struct audio_output *audio,
struct audio_line *line)
{
pthread_mutex_lock(&audio->line_mutex);
*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(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 size_t ts_diff_frames(audio_t audio, uint64_t ts1, uint64_t ts2)
{
double diff = ts_to_frames(audio, ts1) - ts_to_frames(audio, ts2);
return (size_t)positive_round(diff);
}
static size_t ts_diff_bytes(audio_t audio, uint64_t ts1, uint64_t ts2)
{
return ts_diff_frames(audio, ts1, ts2) * audio->block_size;
}
/* unless the value is 3+ hours worth of frames, this won't overflow */
static inline uint64_t conv_frames_to_time(audio_t audio, uint32_t frames)
{
return (uint64_t)frames * 1000000000ULL /
(uint64_t)audio->info.samples_per_sec;
}
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/* ------------------------------------------------------------------------- */
/* 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 = 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);*/
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;
}
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#ifndef CLAMP
#define CLAMP(val, minval, maxval) \
((val > maxval) ? maxval : ((val < minval) ? minval : val))
#endif
#define MIN_S8 -128
#define MAX_S8 127
#define MIN_S16 -32767
#define MAX_S16 32767
#define MIN_S32 -2147483647
#define MAX_S32 2147483647
#define MIX_BUFFER_SIZE 256
/* TODO: optimize mixing */
static void mix_u8(uint8_t *mix, struct circlebuf *buf, size_t size)
{
uint8_t vals[MIX_BUFFER_SIZE];
register int16_t mix_val;
while (size) {
size_t pop_count = min_size(size, sizeof(vals));
size -= pop_count;
circlebuf_pop_front(buf, vals, pop_count);
for (size_t i = 0; i < pop_count; i++) {
mix_val = (int16_t)*mix - 128;
mix_val += (int16_t)vals[i] - 128;
mix_val = CLAMP(mix_val, MIN_S8, MAX_S8) + 128;
*(mix++) = (uint8_t)mix_val;
}
}
}
static void mix_s16(uint8_t *mix_in, struct circlebuf *buf, size_t size)
{
int16_t *mix = (int16_t*)mix_in;
int16_t vals[MIX_BUFFER_SIZE];
register int32_t mix_val;
while (size) {
size_t pop_count = min_size(size, sizeof(vals));
size -= pop_count;
circlebuf_pop_front(buf, vals, pop_count);
pop_count /= sizeof(int16_t);
for (size_t i = 0; i < pop_count; i++) {
mix_val = (int32_t)*mix;
mix_val += (int32_t)vals[i];
*(mix++) = (int16_t)CLAMP(mix_val, MIN_S16, MAX_S16);
}
}
}
static void mix_s32(uint8_t *mix_in, struct circlebuf *buf, size_t size)
{
int32_t *mix = (int32_t*)mix_in;
int32_t vals[MIX_BUFFER_SIZE];
register int64_t mix_val;
while (size) {
size_t pop_count = min_size(size, sizeof(vals));
size -= pop_count;
circlebuf_pop_front(buf, vals, pop_count);
pop_count /= sizeof(int32_t);
for (size_t i = 0; i < pop_count; i++) {
mix_val = (int64_t)*mix;
mix_val += (int64_t)vals[i];
*(mix++) = (int32_t)CLAMP(mix_val, MIN_S32, MAX_S32);
}
}
}
static void mix_float(uint8_t *mix_in, struct circlebuf *buf, size_t size)
{
float *mix = (float*)mix_in;
float vals[MIX_BUFFER_SIZE];
register float mix_val;
while (size) {
size_t pop_count = min_size(size, sizeof(vals));
size -= pop_count;
circlebuf_pop_front(buf, vals, pop_count);
pop_count /= sizeof(float);
for (size_t i = 0; i < pop_count; i++) {
mix_val = *mix + vals[i];
*(mix++) = CLAMP(mix_val, -1.0f, 1.0f);
}
}
}
static inline void mix_audio(enum audio_format format,
uint8_t *mix, struct circlebuf *buf, size_t size)
{
switch (format) {
case AUDIO_FORMAT_UNKNOWN:
break;
case AUDIO_FORMAT_U8BIT:
case AUDIO_FORMAT_U8BIT_PLANAR:
mix_u8(mix, buf, size); break;
case AUDIO_FORMAT_16BIT:
case AUDIO_FORMAT_16BIT_PLANAR:
mix_s16(mix, buf, size); break;
case AUDIO_FORMAT_32BIT:
case AUDIO_FORMAT_32BIT_PLANAR:
mix_s32(mix, buf, size); break;
case AUDIO_FORMAT_FLOAT:
case AUDIO_FORMAT_FLOAT_PLANAR:
mix_float(mix, buf, size); break;
}
}
static inline bool mix_audio_line(struct audio_output *audio,
struct audio_line *line, size_t size, uint64_t timestamp)
{
size_t time_offset = 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);
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mix_audio(audio->info.format,
audio->mix_buffers[i].array + time_offset,
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&line->buffers[i], pop_size);
}
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,
Implement encoder interface (still preliminary) - Implement OBS encoder interface. It was previously incomplete, but now is reaching some level of completion, though probably should still be considered preliminary. I had originally implemented it so that encoders only have a 'reset' function to reset their parameters, but I felt that having both a 'start' and 'stop' function would be useful. Encoders are now assigned to a specific video/audio media output each rather than implicitely assigned to the main obs video/audio contexts. This allows separate encoder contexts that aren't necessarily assigned to the main video/audio context (which is useful for things such as recording specific sources). Will probably have to do this for regular obs outputs as well. When creating an encoder, you must now explicitely state whether that encoder is an audio or video encoder. Audio and video can optionally be automatically converted depending on what the encoder specifies. When something 'attaches' to an encoder, the first attachment starts the encoder, and the encoder automatically attaches to the media output context associated with it. Subsequent attachments won't have the same effect, they will just start receiving the same encoder data when the next keyframe plays (along with SEI if any). When detaching from the encoder, the last detachment will fully stop the encoder and detach the encoder from the media output context associated with the encoder. SEI must actually be exported separately; because new encoder attachments may not always be at the beginning of the stream, the first keyframe they get must have that SEI data in it. If the encoder has SEI data, it needs only add one small function to simply query that SEI data, and then that data will be handled automatically by libobs for all subsequent encoder attachments. - Implement x264 encoder plugin, move x264 files to separate plugin to separate necessary dependencies. - Change video/audio frame output structures to not use const qualifiers to prevent issues with non-const function usage elsewhere. This was an issue when writing the x264 encoder, as the x264 encoder expects non-const frame data. Change stagesurf_map to return a non-const data type to prevent this as well. - Change full range parameter of video scaler to be an enum rather than boolean
2014-03-17 00:21:34 +01:00
(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,
uint64_t timestamp, uint32_t frames)
{
struct audio_data data;
for (size_t i = 0; i < MAX_AV_PLANES; i++)
data.data[i] = audio->mix_buffers[i].array;
data.frames = frames;
data.timestamp = timestamp;
data.volume = 1.0f;
pthread_mutex_lock(&audio->input_mutex);
for (size_t i = 0; i < audio->inputs.num; i++) {
struct audio_input *input = audio->inputs.array+i;
if (resample_audio_output(input, &data))
input->callback(input->param, &data);
}
pthread_mutex_unlock(&audio->input_mutex);
}
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 i = 0; i < audio->planes; i++) {
da_resize(audio->mix_buffers[i], bytes);
memset(audio->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;
}
if (mix_audio_line(audio, line, bytes, prev_time))
line->base_timestamp = audio_time;
pthread_mutex_unlock(&line->mutex);
line = next;
}
/* output */
do_audio_output(audio, prev_time, frames);
return audio_time;
}
/* sample audio 40 times a second */
#define AUDIO_WAIT_TIME (1000/40)
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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;
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while (os_event_try(audio->stop_event) == EAGAIN) {
os_sleep_ms(AUDIO_WAIT_TIME);
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);
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}
return NULL;
}
/* ------------------------------------------------------------------------- */
static size_t audio_get_input_idx(audio_t video,
Implement encoder interface (still preliminary) - Implement OBS encoder interface. It was previously incomplete, but now is reaching some level of completion, though probably should still be considered preliminary. I had originally implemented it so that encoders only have a 'reset' function to reset their parameters, but I felt that having both a 'start' and 'stop' function would be useful. Encoders are now assigned to a specific video/audio media output each rather than implicitely assigned to the main obs video/audio contexts. This allows separate encoder contexts that aren't necessarily assigned to the main video/audio context (which is useful for things such as recording specific sources). Will probably have to do this for regular obs outputs as well. When creating an encoder, you must now explicitely state whether that encoder is an audio or video encoder. Audio and video can optionally be automatically converted depending on what the encoder specifies. When something 'attaches' to an encoder, the first attachment starts the encoder, and the encoder automatically attaches to the media output context associated with it. Subsequent attachments won't have the same effect, they will just start receiving the same encoder data when the next keyframe plays (along with SEI if any). When detaching from the encoder, the last detachment will fully stop the encoder and detach the encoder from the media output context associated with the encoder. SEI must actually be exported separately; because new encoder attachments may not always be at the beginning of the stream, the first keyframe they get must have that SEI data in it. If the encoder has SEI data, it needs only add one small function to simply query that SEI data, and then that data will be handled automatically by libobs for all subsequent encoder attachments. - Implement x264 encoder plugin, move x264 files to separate plugin to separate necessary dependencies. - Change video/audio frame output structures to not use const qualifiers to prevent issues with non-const function usage elsewhere. This was an issue when writing the x264 encoder, as the x264 encoder expects non-const frame data. Change stagesurf_map to return a non-const data type to prevent this as well. - Change full range parameter of video scaler to be an enum rather than boolean
2014-03-17 00:21:34 +01:00
void (*callback)(void *param, struct audio_data *data),
void *param)
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{
for (size_t i = 0; i < video->inputs.num; i++) {
struct audio_input *input = video->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,
const struct audio_convert_info *conversion,
Implement encoder interface (still preliminary) - Implement OBS encoder interface. It was previously incomplete, but now is reaching some level of completion, though probably should still be considered preliminary. I had originally implemented it so that encoders only have a 'reset' function to reset their parameters, but I felt that having both a 'start' and 'stop' function would be useful. Encoders are now assigned to a specific video/audio media output each rather than implicitely assigned to the main obs video/audio contexts. This allows separate encoder contexts that aren't necessarily assigned to the main video/audio context (which is useful for things such as recording specific sources). Will probably have to do this for regular obs outputs as well. When creating an encoder, you must now explicitely state whether that encoder is an audio or video encoder. Audio and video can optionally be automatically converted depending on what the encoder specifies. When something 'attaches' to an encoder, the first attachment starts the encoder, and the encoder automatically attaches to the media output context associated with it. Subsequent attachments won't have the same effect, they will just start receiving the same encoder data when the next keyframe plays (along with SEI if any). When detaching from the encoder, the last detachment will fully stop the encoder and detach the encoder from the media output context associated with the encoder. SEI must actually be exported separately; because new encoder attachments may not always be at the beginning of the stream, the first keyframe they get must have that SEI data in it. If the encoder has SEI data, it needs only add one small function to simply query that SEI data, and then that data will be handled automatically by libobs for all subsequent encoder attachments. - Implement x264 encoder plugin, move x264 files to separate plugin to separate necessary dependencies. - Change video/audio frame output structures to not use const qualifiers to prevent issues with non-const function usage elsewhere. This was an issue when writing the x264 encoder, as the x264 encoder expects non-const frame data. Change stagesurf_map to return a non-const data type to prevent this as well. - Change full range parameter of video scaler to be an enum rather than boolean
2014-03-17 00:21:34 +01:00
void (*callback)(void *param, struct audio_data *data),
void *param)
{
bool success = false;
2014-02-24 06:39:33 +01:00
if (!audio) return false;
pthread_mutex_lock(&audio->input_mutex);
if (audio_get_input_idx(audio, callback, param) == DARRAY_INVALID) {
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(audio->inputs, &input);
}
pthread_mutex_unlock(&audio->input_mutex);
return success;
2013-10-01 04:37:13 +02:00
}
void audio_output_disconnect(audio_t audio,
Implement encoder interface (still preliminary) - Implement OBS encoder interface. It was previously incomplete, but now is reaching some level of completion, though probably should still be considered preliminary. I had originally implemented it so that encoders only have a 'reset' function to reset their parameters, but I felt that having both a 'start' and 'stop' function would be useful. Encoders are now assigned to a specific video/audio media output each rather than implicitely assigned to the main obs video/audio contexts. This allows separate encoder contexts that aren't necessarily assigned to the main video/audio context (which is useful for things such as recording specific sources). Will probably have to do this for regular obs outputs as well. When creating an encoder, you must now explicitely state whether that encoder is an audio or video encoder. Audio and video can optionally be automatically converted depending on what the encoder specifies. When something 'attaches' to an encoder, the first attachment starts the encoder, and the encoder automatically attaches to the media output context associated with it. Subsequent attachments won't have the same effect, they will just start receiving the same encoder data when the next keyframe plays (along with SEI if any). When detaching from the encoder, the last detachment will fully stop the encoder and detach the encoder from the media output context associated with the encoder. SEI must actually be exported separately; because new encoder attachments may not always be at the beginning of the stream, the first keyframe they get must have that SEI data in it. If the encoder has SEI data, it needs only add one small function to simply query that SEI data, and then that data will be handled automatically by libobs for all subsequent encoder attachments. - Implement x264 encoder plugin, move x264 files to separate plugin to separate necessary dependencies. - Change video/audio frame output structures to not use const qualifiers to prevent issues with non-const function usage elsewhere. This was an issue when writing the x264 encoder, as the x264 encoder expects non-const frame data. Change stagesurf_map to return a non-const data type to prevent this as well. - Change full range parameter of video scaler to be an enum rather than boolean
2014-03-17 00:21:34 +01:00
void (*callback)(void *param, struct audio_data *data),
void *param)
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{
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if (!audio) return;
pthread_mutex_lock(&audio->input_mutex);
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size_t idx = audio_get_input_idx(audio, callback, param);
if (idx != DARRAY_INVALID) {
audio_input_free(audio->inputs.array+idx);
da_erase(audio->inputs, idx);
}
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pthread_mutex_unlock(&audio->input_mutex);
2013-10-01 04:37:13 +02:00
}
static inline bool valid_audio_params(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)
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{
struct audio_output *out;
pthread_mutexattr_t attr;
bool planar = is_audio_planar(info->format);
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if (!valid_audio_params(info))
return AUDIO_OUTPUT_INVALIDPARAM;
out = bzalloc(sizeof(struct audio_output));
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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);
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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)
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goto fail;
if (pthread_mutex_init(&out->input_mutex, NULL) != 0)
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goto fail;
if (os_event_init(&out->stop_event, OS_EVENT_TYPE_MANUAL) != 0)
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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 i = 0; i < audio->inputs.num; i++)
audio_input_free(audio->inputs.array+i);
for (size_t i = 0; i < MAX_AV_PLANES; i++)
da_free(audio->mix_buffers[i]);
da_free(audio->inputs);
os_event_destroy(audio->stop_event);
pthread_mutex_destroy(&audio->line_mutex);
bfree(audio);
}
audio_line_t audio_output_createline(audio_t audio, const char *name)
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{
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if (!audio) return NULL;
struct audio_line *line = bzalloc(sizeof(struct audio_line));
line->alive = true;
line->audio = audio;
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_getinfo(audio_t audio)
{
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return audio ? &audio->info : NULL;
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}
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(audio_t audio)
{
if (!audio) return false;
return audio->inputs.num != 0;
}
size_t audio_output_blocksize(audio_t audio)
{
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return audio ? audio->block_size : 0;
}
size_t audio_output_planes(audio_t audio)
{
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return audio ? audio->planes : 0;
}
size_t audio_output_channels(audio_t audio)
{
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return audio ? audio->channels : 0;
}
/* TODO: Optimization of volume multiplication functions */
static inline void mul_vol_u8bit(void *array, float volume, size_t total_num)
{
uint8_t *vals = array;
int32_t vol = (int32_t)(volume * 127.0f);
for (size_t i = 0; i < total_num; i++) {
int32_t val = (int32_t)vals[i] - 128;
int32_t output = val * vol / 127;
vals[i] = (uint8_t)(CLAMP(output, MIN_S8, MAX_S8) + 128);
}
}
static inline void mul_vol_16bit(void *array, float volume, size_t total_num)
{
uint16_t *vals = array;
int64_t vol = (int64_t)(volume * 32767.0f);
for (size_t i = 0; i < total_num; i++) {
int64_t output = (int64_t)vals[i] * vol / 32767;
vals[i] = (int32_t)CLAMP(output, MIN_S16, MAX_S16);
}
}
static inline float conv_24bit_to_float(uint8_t *vals)
{
int32_t val = ((int32_t)vals[0]) |
((int32_t)vals[1] << 8) |
((int32_t)vals[2] << 16);
if ((val & 0x800000) != 0)
val |= 0xFF000000;
return (float)val / 8388607.0f;
}
static inline void conv_float_to_24bit(float fval, uint8_t *vals)
{
int32_t val = (int32_t)(fval * 8388607.0f);
vals[0] = (val) & 0xFF;
vals[1] = (val >> 8) & 0xFF;
vals[2] = (val >> 16) & 0xFF;
}
static inline void mul_vol_24bit(void *array, float volume, size_t total_num)
{
uint8_t *vals = array;
for (size_t i = 0; i < total_num; i++) {
float val = conv_24bit_to_float(vals) * volume;
conv_float_to_24bit(CLAMP(val, -1.0f, 1.0f), vals);
vals += 3;
}
}
static inline void mul_vol_32bit(void *array, float volume, size_t total_num)
{
int32_t *vals = array;
double dvol = (double)volume;
for (size_t i = 0; i < total_num; i++) {
double val = (double)vals[i] / 2147483647.0;
double output = val * dvol;
vals[i] = (int32_t)(CLAMP(output, -1.0, 1.0) * 2147483647.0);
}
}
static inline void mul_vol_float(void *array, float volume, size_t total_num)
{
float *vals = array;
for (size_t i = 0; i < total_num; i++)
vals[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;
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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_U8BIT:
case AUDIO_FORMAT_U8BIT_PLANAR:
mul_vol_u8bit(array, data->volume, total_num);
break;
case AUDIO_FORMAT_16BIT:
case AUDIO_FORMAT_16BIT_PLANAR:
mul_vol_16bit(array, data->volume, total_num);
break;
case AUDIO_FORMAT_32BIT:
case AUDIO_FORMAT_32BIT_PLANAR:
mul_vol_32bit(array, data->volume, total_num);
break;
case AUDIO_FORMAT_FLOAT:
case AUDIO_FORMAT_FLOAT_PLANAR:
mul_vol_float(array, data->volume, total_num);
break;
case AUDIO_FORMAT_UNKNOWN:
blog(LOG_ERROR, "audio_line_place_data_pos: "
"Unknown format");
break;
}
circlebuf_place(&line->buffers[i], position,
line->volume_buffers[i].array, total_size);
}
}
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static void audio_line_place_data(struct audio_line *line,
const struct audio_data *data)
{
size_t pos = ts_diff_bytes(line->audio, data->timestamp,
line->base_timestamp);
#ifdef DEBUG_AUDIO
blog(LOG_DEBUG, "data->timestamp: %llu, line->base_timestamp: %llu, "
"pos: %lu, bytes: %lu, buf size: %lu",
data->timestamp, line->base_timestamp, pos,
data->frames * line->audio->block_size,
line->buffers[0].size);
#endif
audio_line_place_data_pos(line, data, pos);
}
void audio_line_output(audio_line_t line, const struct audio_data *data)
{
/* TODO: prevent insertation of data too far away from expected
* audio timing */
2014-02-24 06:39:33 +01:00
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;
audio_line_place_data(line, data);
} else if (line->base_timestamp <= data->timestamp) {
audio_line_place_data(line, data);
} else {
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);
}
pthread_mutex_unlock(&line->mutex);
}