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[client/idd] network: discard stale audio after stalls

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Keep PCM and its clock sample in cancellable datagrams so recovery
cannot replay audio buffered before a Wi-Fi stall.

Drop blocked real-time packets with an explicit discontinuity and make
a newer reconnect supersede stale audio authorities.
This commit is contained in:
Geoffrey McRae
2026-10-08 22:51:47 +11:00
parent b1dda92f09
commit 0b0f227c08
11 changed files with 341 additions and 700 deletions
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20
client/tests/audio_test.c
View File

@@ -593,7 +593,7 @@ static void testPlaybackRetry(void)
atomic_store(&b.setupOK, true);
uint8_t frames[480 * 2 * 2] = { 0 };
usleep(10000);
p.events->playbackData(p.eventOpaque, 11, frames, 480, NULL);
p.events->playbackData(p.eventOpaque, 11, frames, 480, NULL, false);
waitFor(&b.setupN, failed + 1);
for (unsigned int i = 0; i < WAIT_MS; ++i)
@@ -605,7 +605,7 @@ static void testPlaybackRetry(void)
}
CHECK(atomic_load(&audio.playback.streamGeneration) == 11);
for (unsigned int i = 0; i < 8; ++i)
p.events->playbackData(p.eventOpaque, 11, frames, 480, NULL);
p.events->playbackData(p.eventOpaque, 11, frames, 480, NULL, false);
waitFor(&b.playStartN, 1);
LG_AudioFailureFn fail = b.playFail;
@@ -617,7 +617,7 @@ static void testPlaybackRetry(void)
const unsigned int retry = atomic_load(&b.setupN);
usleep(10000);
p.events->playbackData(p.eventOpaque, 11, frames, 480, NULL);
p.events->playbackData(p.eventOpaque, 11, frames, 480, NULL, false);
waitFor(&b.setupN, retry + 1);
p.events->playbackStop(p.eventOpaque, 11);
@@ -679,7 +679,7 @@ static void testPlaybackRecovery(void)
CHECK(ringbuffer_consume(audio.playback.buffer, NULL, 16) == 16);
CHECK(ringbuffer_getCount(audio.playback.buffer) == -16);
uint8_t frames[1200 * 2 * 2] = { 0 };
CHECK(playbackData(frames, 1200, NULL, arrival) ==
CHECK(playbackData(frames, 1200, NULL, false, arrival) ==
PLAYBACK_DATA_PROCESSED);
CHECK(source->sourcePacketDurationSec < 0.010);
@@ -690,12 +690,12 @@ static void testPlaybackRecovery(void)
CHECK(fabs(atomic_load(&audio.playback.backendResampleRatio) -
source->lastClockRatio) < 0.000001);
CHECK(playbackData(frames, 1200, NULL, arrival + 1000) ==
CHECK(playbackData(frames, 1200, NULL, false, arrival + 1000) ==
PLAYBACK_DATA_PROCESSED);
CHECK(source->sourcePacketDurationSec < 0.010);
CHECK(source->sourcePacketRecovering);
CHECK(playbackData(frames, 480, NULL,
CHECK(playbackData(frames, 480, NULL, false,
arrival + INT64_C(10000000)) == PLAYBACK_DATA_PROCESSED);
CHECK(!source->sourcePacketRecovering);
CHECK(fabs(source->sourcePacketDurationSec - 0.010) < 0.000001);
@@ -733,7 +733,7 @@ static void testPlaybackStableUnderrun(void)
uint8_t frames[64 * 2 * 2];
uint8_t output[64 * 2 * 2];
memset(frames, 0x5a, sizeof(frames));
CHECK(playbackData(frames, 64, NULL, arrival) ==
CHECK(playbackData(frames, 64, NULL, false, arrival) ==
PLAYBACK_DATA_PROCESSED);
const int recoveredOccupancy =
@@ -765,7 +765,7 @@ static void testPlaybackStableUnderrun(void)
source->devicePositionOffsetFrames;
const int64_t secondPacketArrival = arrival +
llrint(64.0e9 / format.sampleRate);
CHECK(playbackData(frames, 64, NULL,
CHECK(playbackData(frames, 64, NULL, false,
secondPacketArrival) == PLAYBACK_DATA_PROCESSED);
CHECK(source->devicePositionOffsetFrames == recoveredOffset);
CHECK(ringbuffer_getCount(audio.playback.buffer) ==
@@ -852,7 +852,7 @@ static void testPlaybackResume(void)
CHECK(atomic_load(&b.setupN) == setupN);
uint8_t input[480 * 2 * 2] = { 0 };
p.events->playbackData(p.eventOpaque, 12, input, 480, NULL);
p.events->playbackData(p.eventOpaque, 12, input, 480, NULL, false);
CHECK(atomic_load(&audio.playback.streamGeneration) == 12);
CHECK(playbackGetState() == STREAM_STATE_RUN);
CHECK(atomic_load(&audio.playback.underruns) == 0);
@@ -883,7 +883,7 @@ static void testPlaybackResume(void)
p.events->playbackStart(p.eventOpaque, 13, &format, NULL);
CHECK(atomic_load(&audio.playback.streamGeneration) == 13);
p.events->playbackData(p.eventOpaque, 13, input, 480, NULL);
p.events->playbackData(p.eventOpaque, 13, input, 480, NULL, false);
CHECK(playbackGetState() == STREAM_STATE_RUN);
CHECK(source->deviceClock.time == arrival);
CHECK(source->deviceClock.position == idleFrames);

16
client/tests/lgmp_audio_test.c
View File

@@ -150,21 +150,23 @@ static unsigned audioSampleBytes(LG_AudioSampleFormat format)
}
static void playbackData(void * opaque, uint32_t generation,
const void * data, size_t frames, const LG_AudioClock * clock)
const void * data, size_t frames, const LG_AudioClock * clock,
bool discontinuity)
{
struct Trace * trace = opaque;
(void)discontinuity;
pthread_mutex_lock(&trace->mutex);
trace->playGeneration = generation;
trace->playFrames = frames;
const size_t bytes = frames * trace->playFormat.channelCount *
trace->playGeneration = generation;
trace->playFrames = frames;
const size_t bytes = frames * trace->playFormat.channelCount *
audioSampleBytes(trace->playFormat.sampleFormat);
const size_t copy = bytes < sizeof(trace->playData) ?
const size_t copy = bytes < sizeof(trace->playData) ?
bytes : sizeof(trace->playData);
memset(trace->playData, 0, sizeof(trace->playData));
memcpy(trace->playData, data, copy);
trace->playClockValid = clock != NULL;
trace->playClockValid = clock != NULL;
if (clock)
trace->playClock = *clock;
trace->playClock = *clock;
pthread_mutex_unlock(&trace->mutex);
traceEvent(trace, TRACE_PLAY_DATA, generation);
atomic_fetch_add_explicit(&trace->playDataCount, 1,