static void source_output_audio_line(obs_source_t source,
const struct audio_data *data)
{
struct audio_data in = *data;
if (!source->timing_set) {
reset_audio_timing(source, in.timestamp);
/* detects 'directly' set timestamps as long as they're within
* a certain threshold */
if ((source->timing_adjust + MAX_TS_VAR) < MAX_TS_VAR * 2)
source->timing_adjust = 0;
} else {
uint64_t time_diff =
data->timestamp - source->next_audio_timestamp_min;
/* don't need signed because negative will trigger it
* regardless, which is what we want */
if (time_diff > MAX_TIMESTAMP_JUMP) {
blog(LOG_DEBUG, "Audio timestamp for source '%s' "
"jumped by '%lld', resetting audio "
"timing", source->name, time_diff);
reset_audio_timing(source, in.timestamp);
}
}
source->next_audio_timestamp_min = in.timestamp +
conv_frames_to_time(source, in.frames);
in.timestamp += source->timing_adjust;
in.volume = source->volume;
audio_line_output(source->audio_line, &in);
}
static void source_output_audio_line(obs_source_t source,
const struct audio_data *data)
{
struct audio_data in = *data;
uint64_t diff;
if (!source->timing_set) {
reset_audio_timing(source, in.timestamp);
/* detects 'directly' set timestamps as long as they're within
* a certain threshold */
if ((source->timing_adjust + MAX_TS_VAR) < MAX_TS_VAR * 2)
source->timing_adjust = 0;
} else {
diff = in.timestamp - source->next_audio_ts_min;
/* don't need signed because negative will trigger it
* regardless, which is what we want */
if (diff > MAX_TIMESTAMP_JUMP)
handle_ts_jump(source, in.timestamp, diff);
}
source->next_audio_ts_min = in.timestamp +
conv_frames_to_time(source, in.frames);
if (source->audio_reset_ref != 0)
return;
in.timestamp += source->timing_adjust;
in.volume = source->volume;
audio_line_output(source->audio_line, &in);
}
static inline void handle_ts_jump(obs_source_t source, uint64_t expected,
uint64_t ts, uint64_t diff)
{
blog(LOG_DEBUG, "Timestamp for source '%s' jumped by '%"PRIu64"', "
"expected value %"PRIu64", input value %"PRIu64,
source->context.name, diff, expected, ts);
/* if has video, ignore audio data until reset */
if (source->info.output_flags & OBS_SOURCE_ASYNC)
os_atomic_dec_long(&source->av_sync_ref);
else
reset_audio_timing(source, ts);
}
static inline void handle_ts_jump(obs_source_t source, uint64_t ts,
uint64_t diff)
{
uint32_t flags = source->callbacks.get_output_flags(source->data);
blog(LOG_DEBUG, "Timestamp for source '%s' jumped by '%lld', "
"resetting audio timing", source->name, diff);
/* if has video, ignore audio data until reset */
if (flags & SOURCE_ASYNC_VIDEO)
source->audio_reset_ref--;
else
reset_audio_timing(source, ts);
}
static void source_output_audio_line(obs_source_t source,
const struct audio_data *data)
{
struct audio_data in = *data;
uint64_t diff;
if (!source->timing_set) {
reset_audio_timing(source, in.timestamp);
/* detects 'directly' set timestamps as long as they're within
* a certain threshold */
if ((source->timing_adjust + MAX_TS_VAR) < MAX_TS_VAR * 2)
source->timing_adjust = 0;
} else {
bool ts_under = (in.timestamp < source->next_audio_ts_min);
diff = ts_under ?
(source->next_audio_ts_min - in.timestamp) :
(in.timestamp - source->next_audio_ts_min);
/* smooth audio if lower or within threshold */
if (diff > MAX_TIMESTAMP_JUMP)
handle_ts_jump(source, source->next_audio_ts_min,
in.timestamp, diff);
else if (ts_under || diff < TS_SMOOTHING_THRESHOLD)
in.timestamp = source->next_audio_ts_min;
}
source->next_audio_ts_min = in.timestamp +
conv_frames_to_time(in.frames);
if (source->av_sync_ref != 0)
return;
in.timestamp += source->timing_adjust + source->sync_offset;
in.volume = source->user_volume * source->present_volume *
obs->audio.user_volume * obs->audio.present_volume;
audio_line_output(source->audio_line, &in);
}
