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Properly clamp audio timing and remove warnings

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If audio was under, it originally did a full reset of the audio timing.
However, resetting the audio timing when this happens is kind of a bad
thing.  It's better just to clamp the value to the expected timestamp to
ensure seamless audio output.

Also, implement audio timestamp smoothing to ensure audio tries to be as
seamless as possible.
This commit is contained in:
jp9000
2014-02-18 14:38:51 -07:00
parent c2f5a88c01
commit 27b851c06e
1 changed files with 19 additions and 9 deletions
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28
libobs/obs-source.c
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@@ -325,9 +325,12 @@ static inline uint64_t conv_frames_to_time(size_t frames)
}
/* maximum "direct" timestamp variance in nanoseconds */
#define MAX_TS_VAR 5000000000ULL
#define MAX_TS_VAR 5000000000ULL
/* maximum time that timestamp can jump in nanoseconds */
#define MAX_TIMESTAMP_JUMP 2000000000ULL
#define MAX_TIMESTAMP_JUMP 2000000000ULL
/* time threshold in nanoseconds to ensure audio timing is as seamless as
* possible */
#define TS_SMOOTHING_THRESHOLD 70000000ULL
static inline void reset_audio_timing(obs_source_t source, uint64_t timetamp)
{
@@ -335,11 +338,12 @@ static inline void reset_audio_timing(obs_source_t source, uint64_t timetamp)
source->timing_adjust = os_gettime_ns() - timetamp;
}
static inline void handle_ts_jump(obs_source_t source, uint64_t ts,
uint64_t diff)
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"', "
"resetting audio timing", source->name, diff);
"expected value %"PRIu64", input value %"PRIu64,
source->name, diff, expected, ts);
/* if has video, ignore audio data until reset */
if (source->info.output_flags & OBS_SOURCE_ASYNC_VIDEO)
@@ -363,12 +367,18 @@ static void source_output_audio_line(obs_source_t source,
source->timing_adjust = 0;
} else {
diff = in.timestamp - source->next_audio_ts_min;
bool ts_under = (in.timestamp < source->next_audio_ts_min);
/* don't need signed because negative will trigger it
* regardless, which is what we want */
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, in.timestamp, diff);
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 +