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Detect audio time values within OS time thresh. 

If audio timestamps are within the operating system timing threshold,
always use those values directly as a timestamp, and do not apply the
regular jump checks and timing adjustments that normally occur.

This potentially fixes an issue with plugins that use OS timestamps
directly as timestamp values for their audio samples, and bypasses the
timing conversions to system time for the audio line and uses it
directly as the timestamp value.  It prevents those calculations from
potentially affecting the audio timestamp value when OS timestamps are
used.

For example, if the first set of audio samples from the audio source
came in delayed, while the subsequent samples were not delayed, those
first samples could have potentially inadvertently triggered the timing
adjustments, which would affect all subsequent audio samples.
master
jp9000 2014-10-19 09:21:39 -07:00
parent 1c40ce6332
commit 381afe0493
1 changed files with 20 additions and 16 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

36
libobs/obs-source.c
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@ -607,14 +607,15 @@ static inline uint64_t conv_frames_to_time(size_t frames)
/* maximum timestamp variance in nanoseconds */
#define MAX_TS_VAR 2000000000ULL
static inline void reset_audio_timing(obs_source_t *source, uint64_t timestamp)
static inline void reset_audio_timing(obs_source_t *source, uint64_t timestamp,
uint64_t os_time)
{
source->timing_set = true;
source->timing_adjust = os_gettime_ns() - timestamp;
source->timing_adjust = os_time - timestamp;
}
static inline void handle_ts_jump(obs_source_t *source, uint64_t expected,
uint64_t ts, uint64_t diff)
uint64_t ts, uint64_t diff, uint64_t os_time)
{
blog(LOG_DEBUG, "Timestamp for source '%s' jumped by '%"PRIu64"', "
"expected value %"PRIu64", input value %"PRIu64,
@ -622,7 +623,7 @@ static inline void handle_ts_jump(obs_source_t *source, uint64_t expected,
/* if has video, ignore audio data until reset */
if (!(source->info.output_flags & OBS_SOURCE_ASYNC))
reset_audio_timing(source, ts);
reset_audio_timing(source, ts, os_time);
}
#define VOL_MIN -96.0f
@ -707,31 +708,34 @@ static void obs_source_update_volume_level(obs_source_t *source,
}
}
static inline uint64_t uint64_diff(uint64_t ts1, uint64_t ts2)
{
return (ts1 < ts2) ? (ts2 - ts1) : (ts1 - ts2);
}
static void source_output_audio_line(obs_source_t *source,
const struct audio_data *data)
{
struct audio_data in = *data;
uint64_t diff;
uint64_t os_time = os_gettime_ns();
if (!source->timing_set) {
reset_audio_timing(source, in.timestamp);
/* detects 'directly' set timestamps as long as they're within
* a certain threshold */
if (uint64_diff(in.timestamp, os_time) < MAX_TS_VAR) {
source->timing_adjust = 0;
source->timing_set = true;
/* 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 if (!source->timing_set) {
reset_audio_timing(source, in.timestamp, os_time);
} else if (source->next_audio_ts_min != 0) {
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);
diff = uint64_diff(source->next_audio_ts_min, in.timestamp);
/* smooth audio if within threshold */
if (diff > MAX_TS_VAR)
handle_ts_jump(source, source->next_audio_ts_min,
in.timestamp, diff);
in.timestamp, diff, os_time);
else if (diff < TS_SMOOTHING_THRESHOLD)
in.timestamp = source->next_audio_ts_min;
}