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[client] audio: learn underrun delivery jitter

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This commit is contained in:
Geoffrey McRae
2026-08-12 13:10:11 +10:00
parent b30ef31f34
commit 88b9bc32d1
1 changed files with 21 additions and 22 deletions
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43
client/src/audio.c
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@@ -167,7 +167,6 @@ typedef struct
int64_t mediaTimeMs;
int64_t mediaLocalOrigin;
uint64_t mediaPosition;
int64_t lastPacketTime;
int64_t lastArrivalTime;
double arrivalJitterSec;
double sourcePhaseBaselineSec;
@@ -867,7 +866,6 @@ static void playbackResetMediaClock(PlaybackSourceData * sourceData,
sourceData->mediaTimeMs = 0;
sourceData->mediaLocalOrigin = now;
sourceData->mediaPosition = position;
sourceData->lastPacketTime = INT64_MIN;
sourceData->lastArrivalTime = INT64_MIN;
sourceData->mediaClockValid = true;
sourceData->mediaPositionValid = true;
@@ -916,6 +914,27 @@ static int64_t playbackMapMediaTime(PlaybackSourceData * sourceData,
}
const int64_t delta = time - sourceData->mediaTime;
if ((!clock || !clock->discontinuity) &&
sourceData->mediaClockFromSource == fromSource &&
sourceData->mediaPositionValid &&
sourceData->mediaPosition == position && delta >= 0 &&
delta <= PLAYBACK_TIMESTAMP_DISCONTINUITY_NS &&
sourceData->lastArrivalTime != INT64_MIN &&
now >= sourceData->lastArrivalTime)
{
const double mediaDelta = delta * 1.0e-9;
const double arrivalDelta =
(now - sourceData->lastArrivalTime) * 1.0e-9;
const double lateness = max(arrivalDelta - mediaDelta, 0.0);
/* Learn harmful late delivery before a buffer underrun rebases the media
* clock. Early catch-up packets do not require additional reserve. */
sourceData->arrivalJitterSec =
min(PLAYBACK_MAX_JITTER_SEC,
max(lateness, sourceData->arrivalJitterSec *
exp(-arrivalDelta / PLAYBACK_JITTER_DECAY_SEC)));
}
if (*discontinuity ||
sourceData->mediaClockFromSource != fromSource ||
!sourceData->mediaPositionValid ||
@@ -2150,26 +2169,6 @@ static PlaybackDataResult playbackData(const void * data, size_t frameCount,
sourceData->bufferOverrunPending = false;
}
if (!discontinuity &&
sourceData->lastPacketTime != INT64_MIN &&
sourceData->lastArrivalTime != INT64_MIN)
{
const double mediaDelta =
(packetTime - sourceData->lastPacketTime) * 1.0e-9;
const double arrivalDelta =
(arrivalTime - sourceData->lastArrivalTime) * 1.0e-9;
const double jitter = fabs(arrivalDelta - mediaDelta);
/* Keep a slowly decaying peak rather than feeding arrival jitter into the
* virtual clock. This lets the buffer absorb real delivery jitter while
* the rate controller follows only the source media clock. */
const double decaySec = max(arrivalDelta, 0.0);
sourceData->arrivalJitterSec =
min(PLAYBACK_MAX_JITTER_SEC,
max(jitter, sourceData->arrivalJitterSec *
exp(-decaySec / PLAYBACK_JITTER_DECAY_SEC)));
}
sourceData->lastPacketTime = packetTime;
sourceData->lastArrivalTime = arrivalTime;
const bool sourceRateWasValid =