New upstream version 24.0.1+dfsg1
This commit is contained in:
Sebastian Ramacher
parent
b14f9eae6d
commit
5a730d6ec3
842 changed files with 42245 additions and 33385 deletions
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@ -32,7 +32,8 @@ static void push_audio_tree(obs_source_t *parent, obs_source_t *source, void *p)
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if (da_find(audio->render_order, &source, 0) == DARRAY_INVALID) {
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obs_source_t *s = obs_source_get_ref(source);
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if (s) da_push_back(audio->render_order, &s);
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if (s)
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da_push_back(audio->render_order, &s);
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}
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UNUSED_PARAMETER(parent);
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@ -44,8 +45,8 @@ static inline size_t convert_time_to_frames(size_t sample_rate, uint64_t t)
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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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obs_source_t *source, size_t channels,
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size_t sample_rate, 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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@ -54,8 +55,8 @@ static inline void mix_audio(struct audio_output_data *mixes,
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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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start_point = convert_time_to_frames(
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sample_rate, source->audio_ts - ts->start);
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if (start_point == AUDIO_OUTPUT_FRAMES)
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return;
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@ -79,18 +80,18 @@ static inline void mix_audio(struct audio_output_data *mixes,
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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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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->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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(uint64_t)sample_rate;
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}
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}
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@ -112,14 +113,14 @@ static bool discard_if_stopped(obs_source_t *source, size_t channels)
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source->pending_stop = true;
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#if DEBUG_AUDIO == 1
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blog(LOG_DEBUG, "doing pending stop trick: '%s'",
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source->context.name);
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source->context.name);
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#endif
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return true;
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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,
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source->audio_input_buf[ch].size);
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source->audio_input_buf[ch].size);
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source->pending_stop = false;
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source->audio_ts = 0;
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@ -138,8 +139,8 @@ static bool discard_if_stopped(obs_source_t *source, size_t channels)
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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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obs_source_t *source, size_t channels,
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size_t sample_rate, 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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@ -155,10 +156,11 @@ static inline void discard_audio(struct obs_core_audio *audio,
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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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blog(LOG_DEBUG,
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"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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@ -171,10 +173,11 @@ static inline void discard_audio(struct obs_core_audio *audio,
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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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blog(LOG_DEBUG,
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"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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@ -184,8 +187,8 @@ static inline void discard_audio(struct obs_core_audio *audio,
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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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size_t start_point = convert_time_to_frames(
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sample_rate, 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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@ -208,6 +211,7 @@ static inline void discard_audio(struct obs_core_audio *audio,
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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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source->audio_ts = ts->end;
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return;
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}
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@ -218,8 +222,7 @@ static inline void discard_audio(struct obs_core_audio *audio,
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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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blog(LOG_DEBUG, "audio discarded, new ts: %" PRIu64, ts->end);
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#endif
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source->pending_stop = false;
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@ -227,8 +230,8 @@ static inline void discard_audio(struct obs_core_audio *audio,
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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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const char *buffering_name)
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size_t sample_rate, struct ts_info *ts,
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uint64_t min_ts, const char *buffering_name)
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{
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struct ts_info new_ts;
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uint64_t offset;
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@ -257,44 +260,50 @@ static void add_audio_buffering(struct obs_core_audio *audio,
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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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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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" (source: %s)\n",
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(int)ms, (int)total_ms, buffering_name);
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blog(LOG_INFO,
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"adding %d milliseconds of audio buffering, total "
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"audio buffering is now %d milliseconds"
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" (source: %s)\n",
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(int)ms, (int)total_ms, buffering_name);
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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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blog(LOG_DEBUG,
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"min_ts (%" PRIu64 ") < start timestamp "
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"(%" PRIu64 ")",
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min_ts, ts->start);
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blog(LOG_DEBUG, "old buffered ts: %" PRIu64 "-%" PRIu64, ts->start,
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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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new_ts.start =
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audio->buffered_ts -
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audio_frames_to_ns(sample_rate, audio->buffering_wait_ticks *
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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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new_ts.start =
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audio->buffered_ts -
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audio_frames_to_ns(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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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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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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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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@ -304,10 +313,9 @@ static bool audio_buffer_insuffient(struct obs_source *source,
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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 (source->audio_ts != min_ts && source->audio_ts != (min_ts - 1)) {
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size_t start_point = convert_time_to_frames(
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sample_rate, 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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@ -325,39 +333,39 @@ static bool audio_buffer_insuffient(struct obs_source *source,
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}
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static inline const char *find_min_ts(struct obs_core_data *data,
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uint64_t *min_ts)
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uint64_t *min_ts)
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{
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obs_source_t *buffering_source = NULL;
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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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source->audio_ts < *min_ts) {
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*min_ts = source->audio_ts;
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buffering_source = source;
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}
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source = (struct obs_source*)source->next_audio_source;
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source = (struct obs_source *)source->next_audio_source;
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}
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return buffering_source ? obs_source_get_name(buffering_source) : NULL;
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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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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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recalculate |=
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audio_buffer_insuffient(source, sample_rate, 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 const char *calc_min_ts(struct obs_core_data *data,
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size_t sample_rate, uint64_t *min_ts)
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size_t sample_rate, uint64_t *min_ts)
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{
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const char *buffering_name = find_min_ts(data, min_ts);
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if (mark_invalid_sources(data, sample_rate, *min_ts))
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@ -371,9 +379,9 @@ static inline void release_audio_sources(struct obs_core_audio *audio)
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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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bool audio_callback(void *param, uint64_t start_ts_in, uint64_t end_ts_in,
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uint64_t *out_ts, uint32_t mixers,
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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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@ -404,7 +412,7 @@ bool audio_callback(void *param,
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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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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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@ -416,7 +424,7 @@ bool audio_callback(void *param,
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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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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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@ -426,7 +434,7 @@ bool audio_callback(void *param,
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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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audio_size);
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}
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/* ------------------------------------------------ */
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@ -439,7 +447,7 @@ bool audio_callback(void *param,
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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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buffering_name);
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buffering_name);
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/* ------------------------------------------------ */
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/* mix audio */
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@ -454,7 +462,7 @@ bool audio_callback(void *param,
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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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&ts);
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pthread_mutex_unlock(&source->audio_buf_mutex);
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}
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@ -470,7 +478,7 @@ bool audio_callback(void *param,
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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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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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