502 lines
13 KiB
C
502 lines
13 KiB
C
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/******************************************************************************
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Copyright (C) 2013 by Hugh Bailey <obs.jim@gmail.com>
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This program is free software: you can redistribute it and/or modify
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it under the terms of the GNU General Public License as published by
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the Free Software Foundation, either version 2 of the License, or
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(at your option) any later version.
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This program is distributed in the hope that it will be useful,
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but WITHOUT ANY WARRANTY; without even the implied warranty of
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MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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GNU General Public License for more details.
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You should have received a copy of the GNU General Public License
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along with this program. If not, see <http://www.gnu.org/licenses/>.
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******************************************************************************/
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#include <math.h>
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#include <inttypes.h>
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#include "../util/threading.h"
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#include "../util/darray.h"
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#include "../util/circlebuf.h"
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#include "../util/platform.h"
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#include "../util/profiler.h"
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#include "audio-io.h"
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#include "audio-resampler.h"
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extern profiler_name_store_t *obs_get_profiler_name_store(void);
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/* #define DEBUG_AUDIO */
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#define nop() do {int invalid = 0;} while(0)
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struct audio_input {
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struct audio_convert_info conversion;
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audio_resampler_t *resampler;
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audio_output_callback_t callback;
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void *param;
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};
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static inline void audio_input_free(struct audio_input *input)
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{
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audio_resampler_destroy(input->resampler);
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}
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struct audio_mix {
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DARRAY(struct audio_input) inputs;
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float buffer[MAX_AUDIO_CHANNELS][AUDIO_OUTPUT_FRAMES];
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};
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struct audio_output {
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struct audio_output_info info;
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size_t block_size;
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size_t channels;
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size_t planes;
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pthread_t thread;
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os_event_t *stop_event;
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bool initialized;
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audio_input_callback_t input_cb;
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void *input_param;
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pthread_mutex_t input_mutex;
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struct audio_mix mixes[MAX_AUDIO_MIXES];
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};
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/* ------------------------------------------------------------------------- */
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/* the following functions are used to calculate frame offsets based upon
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* timestamps. this will actually work accurately as long as you handle the
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* values correctly */
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static inline double ts_to_frames(const audio_t *audio, uint64_t ts)
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{
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double audio_offset_d = (double)ts;
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audio_offset_d /= 1000000000.0;
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audio_offset_d *= (double)audio->info.samples_per_sec;
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return audio_offset_d;
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}
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static inline double positive_round(double val)
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{
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return floor(val+0.5);
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}
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static int64_t ts_diff_frames(const audio_t *audio, uint64_t ts1, uint64_t ts2)
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{
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double diff = ts_to_frames(audio, ts1) - ts_to_frames(audio, ts2);
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return (int64_t)positive_round(diff);
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}
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static int64_t ts_diff_bytes(const audio_t *audio, uint64_t ts1, uint64_t ts2)
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{
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return ts_diff_frames(audio, ts1, ts2) * (int64_t)audio->block_size;
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}
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/* ------------------------------------------------------------------------- */
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static inline uint64_t min_uint64(uint64_t a, uint64_t b)
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{
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return a < b ? a : b;
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}
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static inline size_t min_size(size_t a, size_t b)
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{
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return a < b ? a : b;
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}
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#ifndef CLAMP
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#define CLAMP(val, minval, maxval) \
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((val > maxval) ? maxval : ((val < minval) ? minval : val))
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#endif
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static bool resample_audio_output(struct audio_input *input,
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struct audio_data *data)
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{
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bool success = true;
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if (input->resampler) {
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uint8_t *output[MAX_AV_PLANES];
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uint32_t frames;
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uint64_t offset;
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memset(output, 0, sizeof(output));
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success = audio_resampler_resample(input->resampler,
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output, &frames, &offset,
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(const uint8_t *const *)data->data,
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data->frames);
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for (size_t i = 0; i < MAX_AV_PLANES; i++)
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data->data[i] = output[i];
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data->frames = frames;
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data->timestamp -= offset;
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}
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return success;
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}
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static inline void do_audio_output(struct audio_output *audio,
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size_t mix_idx, uint64_t timestamp, uint32_t frames)
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{
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struct audio_mix *mix = &audio->mixes[mix_idx];
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struct audio_data data;
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pthread_mutex_lock(&audio->input_mutex);
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for (size_t i = mix->inputs.num; i > 0; i--) {
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struct audio_input *input = mix->inputs.array+(i-1);
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for (size_t i = 0; i < audio->planes; i++)
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data.data[i] = (uint8_t*)mix->buffer[i];
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data.frames = frames;
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data.timestamp = timestamp;
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if (resample_audio_output(input, &data))
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input->callback(input->param, mix_idx, &data);
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}
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pthread_mutex_unlock(&audio->input_mutex);
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}
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static inline void clamp_audio_output(struct audio_output *audio, size_t bytes)
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{
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size_t float_size = bytes / sizeof(float);
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for (size_t mix_idx = 0; mix_idx < MAX_AUDIO_MIXES; mix_idx++) {
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struct audio_mix *mix = &audio->mixes[mix_idx];
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/* do not process mixing if a specific mix is inactive */
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if (!mix->inputs.num)
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continue;
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for (size_t plane = 0; plane < audio->planes; plane++) {
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float *mix_data = mix->buffer[plane];
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float *mix_end = &mix_data[float_size];
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while (mix_data < mix_end) {
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float val = *mix_data;
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val = (val > 1.0f) ? 1.0f : val;
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val = (val < -1.0f) ? -1.0f : val;
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*(mix_data++) = val;
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}
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}
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}
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}
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static void input_and_output(struct audio_output *audio,
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uint64_t audio_time, uint64_t prev_time)
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{
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size_t bytes = AUDIO_OUTPUT_FRAMES * audio->block_size;
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struct audio_output_data data[MAX_AUDIO_MIXES];
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uint32_t active_mixes = 0;
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uint64_t new_ts = 0;
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bool success;
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memset(data, 0, sizeof(data));
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#ifdef DEBUG_AUDIO
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blog(LOG_DEBUG, "audio_time: %llu, prev_time: %llu, bytes: %lu",
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audio_time, prev_time, bytes);
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#endif
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/* get mixers */
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pthread_mutex_lock(&audio->input_mutex);
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for (size_t i = 0; i < MAX_AUDIO_MIXES; i++) {
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if (audio->mixes[i].inputs.num)
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active_mixes |= (1 << i);
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}
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pthread_mutex_unlock(&audio->input_mutex);
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/* clear mix buffers */
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for (size_t mix_idx = 0; mix_idx < MAX_AUDIO_MIXES; mix_idx++) {
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struct audio_mix *mix = &audio->mixes[mix_idx];
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memset(mix->buffer[0], 0, AUDIO_OUTPUT_FRAMES *
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MAX_AUDIO_CHANNELS * sizeof(float));
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for (size_t i = 0; i < audio->planes; i++)
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data[mix_idx].data[i] = mix->buffer[i];
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}
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/* get new audio data */
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success = audio->input_cb(audio->input_param, prev_time, audio_time,
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&new_ts, active_mixes, data);
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if (!success)
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return;
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/* clamps audio data to -1.0..1.0 */
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clamp_audio_output(audio, bytes);
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/* output */
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for (size_t i = 0; i < MAX_AUDIO_MIXES; i++)
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do_audio_output(audio, i, new_ts, AUDIO_OUTPUT_FRAMES);
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}
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static void *audio_thread(void *param)
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{
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struct audio_output *audio = param;
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size_t rate = audio->info.samples_per_sec;
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uint64_t samples = 0;
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uint64_t start_time = os_gettime_ns();
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uint64_t prev_time = start_time;
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uint64_t audio_time = prev_time;
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uint32_t audio_wait_time =
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(uint32_t)(audio_frames_to_ns(rate, AUDIO_OUTPUT_FRAMES) /
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1000000);
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os_set_thread_name("audio-io: audio thread");
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const char *audio_thread_name =
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profile_store_name(obs_get_profiler_name_store(),
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"audio_thread(%s)", audio->info.name);
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while (os_event_try(audio->stop_event) == EAGAIN) {
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uint64_t cur_time;
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os_sleep_ms(audio_wait_time);
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profile_start(audio_thread_name);
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cur_time = os_gettime_ns();
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while (audio_time <= cur_time) {
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samples += AUDIO_OUTPUT_FRAMES;
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audio_time = start_time +
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audio_frames_to_ns(rate, samples);
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input_and_output(audio, audio_time, prev_time);
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prev_time = audio_time;
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}
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profile_end(audio_thread_name);
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profile_reenable_thread();
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}
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return NULL;
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}
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/* ------------------------------------------------------------------------- */
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static size_t audio_get_input_idx(const audio_t *audio, size_t mix_idx,
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audio_output_callback_t callback, void *param)
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{
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const struct audio_mix *mix = &audio->mixes[mix_idx];
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for (size_t i = 0; i < mix->inputs.num; i++) {
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struct audio_input *input = mix->inputs.array+i;
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if (input->callback == callback && input->param == param)
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return i;
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}
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return DARRAY_INVALID;
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}
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static inline bool audio_input_init(struct audio_input *input,
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struct audio_output *audio)
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{
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if (input->conversion.format != audio->info.format ||
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input->conversion.samples_per_sec != audio->info.samples_per_sec ||
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input->conversion.speakers != audio->info.speakers) {
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struct resample_info from = {
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.format = audio->info.format,
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.samples_per_sec = audio->info.samples_per_sec,
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.speakers = audio->info.speakers
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};
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struct resample_info to = {
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.format = input->conversion.format,
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.samples_per_sec = input->conversion.samples_per_sec,
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.speakers = input->conversion.speakers
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};
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input->resampler = audio_resampler_create(&to, &from);
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if (!input->resampler) {
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blog(LOG_ERROR, "audio_input_init: Failed to "
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"create resampler");
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return false;
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}
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} else {
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input->resampler = NULL;
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}
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return true;
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}
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bool audio_output_connect(audio_t *audio, size_t mi,
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const struct audio_convert_info *conversion,
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audio_output_callback_t callback, void *param)
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{
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bool success = false;
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if (!audio || mi >= MAX_AUDIO_MIXES) return false;
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pthread_mutex_lock(&audio->input_mutex);
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if (audio_get_input_idx(audio, mi, callback, param) == DARRAY_INVALID) {
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struct audio_mix *mix = &audio->mixes[mi];
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struct audio_input input;
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input.callback = callback;
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input.param = param;
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if (conversion) {
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input.conversion = *conversion;
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} else {
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input.conversion.format = audio->info.format;
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input.conversion.speakers = audio->info.speakers;
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input.conversion.samples_per_sec =
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audio->info.samples_per_sec;
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}
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if (input.conversion.format == AUDIO_FORMAT_UNKNOWN)
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input.conversion.format = audio->info.format;
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if (input.conversion.speakers == SPEAKERS_UNKNOWN)
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input.conversion.speakers = audio->info.speakers;
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if (input.conversion.samples_per_sec == 0)
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input.conversion.samples_per_sec =
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audio->info.samples_per_sec;
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success = audio_input_init(&input, audio);
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if (success)
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da_push_back(mix->inputs, &input);
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}
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pthread_mutex_unlock(&audio->input_mutex);
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return success;
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}
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void audio_output_disconnect(audio_t *audio, size_t mix_idx,
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audio_output_callback_t callback, void *param)
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{
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if (!audio || mix_idx >= MAX_AUDIO_MIXES) return;
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pthread_mutex_lock(&audio->input_mutex);
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size_t idx = audio_get_input_idx(audio, mix_idx, callback, param);
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if (idx != DARRAY_INVALID) {
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struct audio_mix *mix = &audio->mixes[mix_idx];
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audio_input_free(mix->inputs.array+idx);
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da_erase(mix->inputs, idx);
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}
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pthread_mutex_unlock(&audio->input_mutex);
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}
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static inline bool valid_audio_params(const struct audio_output_info *info)
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{
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return info->format && info->name && info->samples_per_sec > 0 &&
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info->speakers > 0;
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}
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int audio_output_open(audio_t **audio, struct audio_output_info *info)
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{
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struct audio_output *out;
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pthread_mutexattr_t attr;
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bool planar = is_audio_planar(info->format);
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if (!valid_audio_params(info))
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return AUDIO_OUTPUT_INVALIDPARAM;
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out = bzalloc(sizeof(struct audio_output));
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if (!out)
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goto fail;
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memcpy(&out->info, info, sizeof(struct audio_output_info));
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out->channels = get_audio_channels(info->speakers);
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out->planes = planar ? out->channels : 1;
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out->input_cb = info->input_callback;
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out->input_param= info->input_param;
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out->block_size = (planar ? 1 : out->channels) *
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get_audio_bytes_per_channel(info->format);
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if (pthread_mutexattr_init(&attr) != 0)
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goto fail;
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if (pthread_mutexattr_settype(&attr, PTHREAD_MUTEX_RECURSIVE) != 0)
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goto fail;
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if (pthread_mutex_init(&out->input_mutex, &attr) != 0)
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goto fail;
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if (os_event_init(&out->stop_event, OS_EVENT_TYPE_MANUAL) != 0)
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goto fail;
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if (pthread_create(&out->thread, NULL, audio_thread, out) != 0)
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goto fail;
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out->initialized = true;
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*audio = out;
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return AUDIO_OUTPUT_SUCCESS;
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fail:
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audio_output_close(out);
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return AUDIO_OUTPUT_FAIL;
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}
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void audio_output_close(audio_t *audio)
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{
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void *thread_ret;
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||
|
|
||
|
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;
|
||
|
}
|