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[client] audio: ignore jitter after underruns

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Do not learn late delivery from packets that already caused an
underrun and media-clock rebase. Preserve learned jitter only across
continuous playback.
This commit is contained in:
Geoffrey McRae
2026-08-22 17:04:25 +10:00
parent 7476e56ded
commit 5b106015de
3 changed files with 36 additions and 3 deletions
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7
client/src/audio.c
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@@ -924,7 +924,7 @@ static int64_t playbackMapMediaTime(PlaybackSourceData * sourceData,
} }
const int64_t delta = time - sourceData->mediaTime; const int64_t delta = time - sourceData->mediaTime;
if ((!clock || !clock->discontinuity) && if (!*discontinuity && (!clock || !clock->discontinuity) &&
sourceData->mediaClockFromSource == fromSource && sourceData->mediaClockFromSource == fromSource &&
sourceData->mediaPositionValid && sourceData->mediaPositionValid &&
sourceData->mediaPosition == position && delta >= 0 && sourceData->mediaPosition == position && delta >= 0 &&
@@ -937,8 +937,9 @@ static int64_t playbackMapMediaTime(PlaybackSourceData * sourceData,
(now - sourceData->lastArrivalTime) * 1.0e-9; (now - sourceData->lastArrivalTime) * 1.0e-9;
const double lateness = max(arrivalDelta - mediaDelta, 0.0); const double lateness = max(arrivalDelta - mediaDelta, 0.0);
/* Learn harmful late delivery before a buffer underrun rebases the media /* Learn harmful late delivery only while playback remains continuous.
* clock. Early catch-up packets do not require additional reserve. */ * An underrun rebases the media clock and must not become future
* latency. Early catch-up packets do not require additional reserve. */
sourceData->arrivalJitterSec = sourceData->arrivalJitterSec =
min(PLAYBACK_MAX_JITTER_SEC, min(PLAYBACK_MAX_JITTER_SEC,
max(lateness, sourceData->arrivalJitterSec * max(lateness, sourceData->arrivalJitterSec *

1
client/tests/CMakeLists.txt
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@@ -819,6 +819,7 @@ if(ENABLE_AUDIO)
convert convert
provider provider
playback-retry playback-retry
jitter
consent consent
quiesce quiesce
) )

31
client/tests/audio_test.c
View File

@@ -624,6 +624,36 @@ static void testPlaybackRetry(void)
stopAudio(); stopAudio();
} }
static void testPlaybackJitter(void)
{
PlaybackSourceData source = { 0 };
playbackPrepareMediaClock(&source, NULL);
bool discontinuity = false;
playbackMapMediaTime(&source, NULL, 480, 48000, 0,
&discontinuity);
source.inputPosition = 480;
source.lastArrivalTime = 0;
discontinuity = true;
playbackMapMediaTime(&source, NULL, 480, 48000,
INT64_C(110000000), &discontinuity);
CHECK(discontinuity);
CHECK(source.arrivalJitterSec == 0.0);
memset(&source, 0, sizeof(source));
playbackPrepareMediaClock(&source, NULL);
discontinuity = false;
playbackMapMediaTime(&source, NULL, 480, 48000, 0,
&discontinuity);
source.inputPosition = 480;
source.lastArrivalTime = 0;
playbackMapMediaTime(&source, NULL, 480, 48000,
INT64_C(20000000), &discontinuity);
CHECK(source.arrivalJitterSec > 0.009);
CHECK(source.arrivalJitterSec < 0.011);
}
static void testConsent(void) static void testConsent(void)
{ {
reset(); reset();
@@ -750,6 +780,7 @@ static const struct Test tests[] =
{ "convert" , testConvert }, { "convert" , testConvert },
{ "provider" , testProvider }, { "provider" , testProvider },
{ "playback-retry", testPlaybackRetry }, { "playback-retry", testPlaybackRetry },
{ "jitter" , testPlaybackJitter },
{ "consent" , testConsent }, { "consent" , testConsent },
{ "quiesce" , testQuiesce }, { "quiesce" , testQuiesce },
}; };