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

502 lines
13 KiB
C

/******************************************************************************
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_mix {
DARRAY(struct audio_input) inputs;
float buffer[MAX_AUDIO_CHANNELS][AUDIO_OUTPUT_FRAMES];
};
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;
audio_input_callback_t input_cb;
void *input_param;
pthread_mutex_t input_mutex;
struct audio_mix mixes[MAX_AUDIO_MIXES];
};
/* ------------------------------------------------------------------------- */
/* 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;
}
/* ------------------------------------------------------------------------- */
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
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;
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);
for (size_t i = 0; i < audio->planes; i++)
data.data[i] = (uint8_t*)mix->buffer[i];
data.frames = frames;
data.timestamp = timestamp;
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 = mix->buffer[plane];
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 void input_and_output(struct audio_output *audio,
uint64_t audio_time, uint64_t prev_time)
{
size_t bytes = AUDIO_OUTPUT_FRAMES * audio->block_size;
struct audio_output_data data[MAX_AUDIO_MIXES];
uint32_t active_mixes = 0;
uint64_t new_ts = 0;
bool success;
memset(data, 0, sizeof(data));
#ifdef DEBUG_AUDIO
blog(LOG_DEBUG, "audio_time: %llu, prev_time: %llu, bytes: %lu",
audio_time, prev_time, bytes);
#endif
/* get mixers */
pthread_mutex_lock(&audio->input_mutex);
for (size_t i = 0; i < MAX_AUDIO_MIXES; i++) {
if (audio->mixes[i].inputs.num)
active_mixes |= (1 << i);
}
pthread_mutex_unlock(&audio->input_mutex);
/* clear mix buffers */
for (size_t mix_idx = 0; mix_idx < MAX_AUDIO_MIXES; mix_idx++) {
struct audio_mix *mix = &audio->mixes[mix_idx];
memset(mix->buffer[0], 0, AUDIO_OUTPUT_FRAMES *
MAX_AUDIO_CHANNELS * sizeof(float));
for (size_t i = 0; i < audio->planes; i++)
data[mix_idx].data[i] = mix->buffer[i];
}
/* get new audio data */
success = audio->input_cb(audio->input_param, prev_time, audio_time,
&new_ts, active_mixes, data);
if (!success)
return;
/* 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, new_ts, AUDIO_OUTPUT_FRAMES);
}
static void *audio_thread(void *param)
{
struct audio_output *audio = param;
size_t rate = audio->info.samples_per_sec;
uint64_t samples = 0;
uint64_t start_time = os_gettime_ns();
uint64_t prev_time = start_time;
uint64_t audio_time = prev_time;
uint32_t audio_wait_time =
(uint32_t)(audio_frames_to_ns(rate, AUDIO_OUTPUT_FRAMES) /
1000000);
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) {
uint64_t cur_time;
os_sleep_ms(audio_wait_time);
profile_start(audio_thread_name);
cur_time = os_gettime_ns();
while (audio_time <= cur_time) {
samples += AUDIO_OUTPUT_FRAMES;
audio_time = start_time +
audio_frames_to_ns(rate, samples);
input_and_output(audio, audio_time, prev_time);
prev_time = audio_time;
}
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));
out->channels = get_audio_channels(info->speakers);
out->planes = planar ? out->channels : 1;
out->input_cb = info->input_callback;
out->input_param= info->input_param;
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->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;
if (!audio)
return;
if (audio->initialized) {
os_event_signal(audio->stop_event);
pthread_join(audio->thread, &thread_ret);
}
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);
da_free(mix->inputs);
}
os_event_destroy(audio->stop_event);
bfree(audio);
}
const struct audio_output_info *audio_output_get_info(const audio_t *audio)
{
return audio ? &audio->info : NULL;
}
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;
}