Imported Upstream version 0.13.2+dsfg1
This commit is contained in:
Sebastian Ramacher
commit
fb3990e9e5
2036 changed files with 287360 additions and 0 deletions
474
libobs/obs-audio.c
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474
libobs/obs-audio.c
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/******************************************************************************
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Copyright (C) 2015 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 <inttypes.h>
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#include "obs-internal.h"
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struct ts_info {
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uint64_t start;
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uint64_t end;
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};
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#define DEBUG_AUDIO 0
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#define MAX_BUFFERING_TICKS 45
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static void push_audio_tree(obs_source_t *parent, obs_source_t *source, void *p)
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{
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struct obs_core_audio *audio = p;
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if (da_find(audio->render_order, &source, 0) == DARRAY_INVALID) {
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obs_source_addref(source);
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da_push_back(audio->render_order, &source);
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}
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UNUSED_PARAMETER(parent);
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}
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static inline size_t convert_time_to_frames(size_t sample_rate, uint64_t t)
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{
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return (size_t)(t * (uint64_t)sample_rate / 1000000000ULL);
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}
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static inline void mix_audio(struct audio_output_data *mixes,
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obs_source_t *source, size_t channels, size_t sample_rate,
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struct ts_info *ts)
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{
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size_t total_floats = AUDIO_OUTPUT_FRAMES;
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size_t start_point = 0;
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if (source->audio_ts < ts->start || ts->end <= source->audio_ts)
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return;
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if (source->audio_ts != ts->start) {
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start_point = convert_time_to_frames(sample_rate,
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source->audio_ts - ts->start);
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if (start_point == AUDIO_OUTPUT_FRAMES)
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return;
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total_floats -= start_point;
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}
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for (size_t mix_idx = 0; mix_idx < MAX_AUDIO_MIXES; mix_idx++) {
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for (size_t ch = 0; ch < channels; ch++) {
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register float *mix = mixes[mix_idx].data[ch];
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register float *aud =
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source->audio_output_buf[mix_idx][ch];
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register float *end;
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mix += start_point;
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end = aud + total_floats;
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while (aud < end)
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*(mix++) += *(aud++);
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}
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}
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}
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static void ignore_audio(obs_source_t *source, size_t channels,
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size_t sample_rate)
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{
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size_t num_floats = source->audio_input_buf[0].size / sizeof(float);
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if (num_floats) {
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for (size_t ch = 0; ch < channels; ch++)
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circlebuf_pop_front(&source->audio_input_buf[ch], NULL,
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source->audio_input_buf[ch].size);
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source->last_audio_input_buf_size = 0;
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source->audio_ts += (uint64_t)num_floats * 1000000000ULL /
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(uint64_t)sample_rate;
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}
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}
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static bool discard_if_stopped(obs_source_t *source, size_t channels)
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{
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size_t last_size;
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size_t size;
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last_size = source->last_audio_input_buf_size;
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size = source->audio_input_buf[0].size;
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if (!size)
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return false;
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/* if perpetually pending data, it means the audio has stopped,
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* so clear the audio data */
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if (last_size == size) {
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for (size_t ch = 0; ch < channels; ch++)
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circlebuf_pop_front(&source->audio_input_buf[ch], NULL,
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source->audio_input_buf[ch].size);
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source->audio_ts = 0;
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source->last_audio_input_buf_size = 0;
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#if DEBUG_AUDIO == 1
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blog(LOG_DEBUG, "source audio data appears to have "
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"stopped, clearing");
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#endif
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return true;
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} else {
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source->last_audio_input_buf_size = size;
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return false;
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}
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}
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#define MAX_AUDIO_SIZE (AUDIO_OUTPUT_FRAMES * sizeof(float))
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static inline void discard_audio(struct obs_core_audio *audio,
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obs_source_t *source, size_t channels, size_t sample_rate,
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struct ts_info *ts)
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{
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size_t total_floats = AUDIO_OUTPUT_FRAMES;
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size_t size;
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#if DEBUG_AUDIO == 1
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bool is_audio_source = source->info.output_flags & OBS_SOURCE_AUDIO;
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#endif
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if (source->info.audio_render) {
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source->audio_ts = 0;
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return;
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}
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if (ts->end <= source->audio_ts) {
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#if DEBUG_AUDIO == 1
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blog(LOG_DEBUG, "can't discard, source "
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"timestamp (%"PRIu64") >= "
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"end timestamp (%"PRIu64")",
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source->audio_ts, ts->end);
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#endif
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return;
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}
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if (source->audio_ts < (ts->start - 1)) {
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if (source->audio_pending &&
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source->audio_input_buf[0].size < MAX_AUDIO_SIZE &&
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discard_if_stopped(source, channels))
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return;
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#if DEBUG_AUDIO == 1
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if (is_audio_source) {
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blog(LOG_DEBUG, "can't discard, source "
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"timestamp (%"PRIu64") < "
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"start timestamp (%"PRIu64")",
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source->audio_ts, ts->start);
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}
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#endif
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if (audio->total_buffering_ticks == MAX_BUFFERING_TICKS)
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ignore_audio(source, channels, sample_rate);
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return;
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}
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if (source->audio_ts != ts->start &&
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source->audio_ts != (ts->start - 1)) {
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size_t start_point = convert_time_to_frames(sample_rate,
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source->audio_ts - ts->start);
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if (start_point == AUDIO_OUTPUT_FRAMES) {
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#if DEBUG_AUDIO == 1
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if (is_audio_source)
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blog(LOG_DEBUG, "can't dicard, start point is "
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"at audio frame count");
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#endif
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return;
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}
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total_floats -= start_point;
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}
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size = total_floats * sizeof(float);
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if (source->audio_input_buf[0].size < size) {
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if (discard_if_stopped(source, channels))
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return;
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#if DEBUG_AUDIO == 1
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if (is_audio_source)
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blog(LOG_DEBUG, "can't discard, data still pending");
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#endif
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return;
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}
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for (size_t ch = 0; ch < channels; ch++)
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circlebuf_pop_front(&source->audio_input_buf[ch], NULL, size);
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source->last_audio_input_buf_size = 0;
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#if DEBUG_AUDIO == 1
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if (is_audio_source)
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blog(LOG_DEBUG, "audio discarded, new ts: %"PRIu64,
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ts->end);
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#endif
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source->audio_ts = ts->end;
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}
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static void add_audio_buffering(struct obs_core_audio *audio,
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size_t sample_rate, struct ts_info *ts, uint64_t min_ts)
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{
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struct ts_info new_ts;
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uint64_t offset;
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uint64_t frames;
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size_t total_ms;
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size_t ms;
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int ticks;
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if (audio->total_buffering_ticks == MAX_BUFFERING_TICKS)
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return;
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if (!audio->buffering_wait_ticks)
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audio->buffered_ts = ts->start;
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offset = ts->start - min_ts;
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frames = ns_to_audio_frames(sample_rate, offset);
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ticks = (int)((frames + AUDIO_OUTPUT_FRAMES - 1) / AUDIO_OUTPUT_FRAMES);
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audio->total_buffering_ticks += ticks;
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if (audio->total_buffering_ticks >= MAX_BUFFERING_TICKS) {
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ticks -= audio->total_buffering_ticks - MAX_BUFFERING_TICKS;
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audio->total_buffering_ticks = MAX_BUFFERING_TICKS;
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blog(LOG_WARNING, "Max audio buffering reached!");
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}
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ms = ticks * AUDIO_OUTPUT_FRAMES * 1000 / sample_rate;
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total_ms = audio->total_buffering_ticks * AUDIO_OUTPUT_FRAMES * 1000 /
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sample_rate;
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blog(LOG_INFO, "adding %d milliseconds of audio buffering, total "
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"audio buffering is now %d milliseconds",
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(int)ms, (int)total_ms);
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#if DEBUG_AUDIO == 1
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blog(LOG_DEBUG, "min_ts (%"PRIu64") < start timestamp "
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"(%"PRIu64")", min_ts, ts->start);
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blog(LOG_DEBUG, "old buffered ts: %"PRIu64"-%"PRIu64,
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ts->start, ts->end);
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#endif
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new_ts.start = audio->buffered_ts - audio_frames_to_ns(sample_rate,
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audio->buffering_wait_ticks * AUDIO_OUTPUT_FRAMES);
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while (ticks--) {
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int cur_ticks = ++audio->buffering_wait_ticks;
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new_ts.end = new_ts.start;
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new_ts.start = audio->buffered_ts - audio_frames_to_ns(
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sample_rate,
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cur_ticks * AUDIO_OUTPUT_FRAMES);
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#if DEBUG_AUDIO == 1
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blog(LOG_DEBUG, "add buffered ts: %"PRIu64"-%"PRIu64,
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new_ts.start, new_ts.end);
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#endif
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circlebuf_push_front(&audio->buffered_timestamps, &new_ts,
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sizeof(new_ts));
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}
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*ts = new_ts;
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}
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static bool audio_buffer_insuffient(struct obs_source *source,
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size_t sample_rate, uint64_t min_ts)
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{
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size_t total_floats = AUDIO_OUTPUT_FRAMES;
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size_t size;
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if (source->info.audio_render || source->audio_pending ||
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!source->audio_ts) {
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return false;
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}
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if (source->audio_ts != min_ts &&
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source->audio_ts != (min_ts - 1)) {
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size_t start_point = convert_time_to_frames(sample_rate,
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source->audio_ts - min_ts);
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if (start_point >= AUDIO_OUTPUT_FRAMES)
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return false;
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total_floats -= start_point;
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}
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size = total_floats * sizeof(float);
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if (source->audio_input_buf[0].size < size) {
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source->audio_pending = true;
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return true;
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}
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return false;
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}
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static inline void find_min_ts(struct obs_core_data *data,
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uint64_t *min_ts)
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{
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struct obs_source *source = data->first_audio_source;
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while (source) {
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if (!source->audio_pending && source->audio_ts &&
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source->audio_ts < *min_ts)
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*min_ts = source->audio_ts;
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source = (struct obs_source*)source->next_audio_source;
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}
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}
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static inline bool mark_invalid_sources(struct obs_core_data *data,
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size_t sample_rate, uint64_t min_ts)
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{
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bool recalculate = false;
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struct obs_source *source = data->first_audio_source;
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while (source) {
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recalculate |= audio_buffer_insuffient(source, sample_rate,
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min_ts);
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source = (struct obs_source*)source->next_audio_source;
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}
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return recalculate;
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}
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static inline void calc_min_ts(struct obs_core_data *data,
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size_t sample_rate, uint64_t *min_ts)
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{
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find_min_ts(data, min_ts);
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if (mark_invalid_sources(data, sample_rate, *min_ts))
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find_min_ts(data, min_ts);
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}
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static inline void release_audio_sources(struct obs_core_audio *audio)
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{
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for (size_t i = 0; i < audio->render_order.num; i++)
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obs_source_release(audio->render_order.array[i]);
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}
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bool audio_callback(void *param,
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uint64_t start_ts_in, uint64_t end_ts_in, uint64_t *out_ts,
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uint32_t mixers, struct audio_output_data *mixes)
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{
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struct obs_core_data *data = &obs->data;
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struct obs_core_audio *audio = &obs->audio;
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struct obs_source *source;
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size_t sample_rate = audio_output_get_sample_rate(audio->audio);
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size_t channels = audio_output_get_channels(audio->audio);
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struct ts_info ts = {start_ts_in, end_ts_in};
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size_t audio_size;
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uint64_t min_ts;
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da_resize(audio->render_order, 0);
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da_resize(audio->root_nodes, 0);
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circlebuf_push_back(&audio->buffered_timestamps, &ts, sizeof(ts));
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circlebuf_peek_front(&audio->buffered_timestamps, &ts, sizeof(ts));
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min_ts = ts.start;
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audio_size = AUDIO_OUTPUT_FRAMES * sizeof(float);
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#if DEBUG_AUDIO == 1
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blog(LOG_DEBUG, "ts %llu-%llu", ts.start, ts.end);
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#endif
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/* ------------------------------------------------ */
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/* build audio render order
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* NOTE: these are source channels, not audio channels */
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for (uint32_t i = 0; i < MAX_CHANNELS; i++) {
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obs_source_t *source = obs_get_output_source(i);
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if (source) {
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obs_source_enum_active_tree(source, push_audio_tree,
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audio);
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push_audio_tree(NULL, source, audio);
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da_push_back(audio->root_nodes, &source);
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obs_source_release(source);
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}
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}
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pthread_mutex_lock(&data->audio_sources_mutex);
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source = data->first_audio_source;
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while (source) {
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push_audio_tree(NULL, source, audio);
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source = (struct obs_source*)source->next_audio_source;
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}
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pthread_mutex_unlock(&data->audio_sources_mutex);
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/* ------------------------------------------------ */
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/* render audio data */
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for (size_t i = 0; i < audio->render_order.num; i++) {
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obs_source_t *source = audio->render_order.array[i];
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obs_source_audio_render(source, mixers, channels, sample_rate,
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audio_size);
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}
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/* ------------------------------------------------ */
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/* get minimum audio timestamp */
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pthread_mutex_lock(&data->audio_sources_mutex);
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calc_min_ts(data, sample_rate, &min_ts);
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pthread_mutex_unlock(&data->audio_sources_mutex);
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/* ------------------------------------------------ */
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/* if a source has gone backward in time, buffer */
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if (min_ts < ts.start)
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add_audio_buffering(audio, sample_rate, &ts, min_ts);
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/* ------------------------------------------------ */
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/* mix audio */
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if (!audio->buffering_wait_ticks) {
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for (size_t i = 0; i < audio->root_nodes.num; i++) {
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obs_source_t *source = audio->root_nodes.array[i];
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if (source->audio_pending)
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continue;
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pthread_mutex_lock(&source->audio_buf_mutex);
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if (source->audio_output_buf[0][0] && source->audio_ts)
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mix_audio(mixes, source, channels, sample_rate,
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&ts);
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pthread_mutex_unlock(&source->audio_buf_mutex);
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}
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}
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/* ------------------------------------------------ */
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/* discard audio */
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pthread_mutex_lock(&data->audio_sources_mutex);
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source = data->first_audio_source;
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while (source) {
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pthread_mutex_lock(&source->audio_buf_mutex);
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discard_audio(audio, source, channels, sample_rate, &ts);
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pthread_mutex_unlock(&source->audio_buf_mutex);
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source = (struct obs_source*)source->next_audio_source;
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}
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pthread_mutex_unlock(&data->audio_sources_mutex);
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/* ------------------------------------------------ */
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/* release audio sources */
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release_audio_sources(audio);
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circlebuf_pop_front(&audio->buffered_timestamps, NULL, sizeof(ts));
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*out_ts = ts.start;
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if (audio->buffering_wait_ticks) {
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audio->buffering_wait_ticks--;
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return false;
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}
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UNUSED_PARAMETER(param);
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return true;
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}
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