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module-rtp: Compensate for stream resampler effects in RTP sink node

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This commit is contained in:
Carlos Rafael Giani 2026-01-21 21:50:14 +01:00
parent 6192c11038
commit 642504f5f9
1 changed files with 18 additions and 0 deletions
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18
src/modules/module-rtp/audio.c
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@ -620,6 +620,7 @@ static void rtp_audio_process_capture(void *data)
uint32_t pending, num_queued;
struct spa_io_position *pos;
uint64_t next_nsec, quantum;
struct pw_time pwt;
if (impl->separate_sender) {
/* apply the DLL rate */
@ -637,6 +638,8 @@ static void rtp_audio_process_capture(void *data)
stride = impl->stride;
wanted = size / stride;
pw_stream_get_time_n(impl->stream, &pwt, sizeof(pwt));
filled = spa_ringbuffer_get_write_index(&impl->ring, &expected_timestamp);
pos = impl->io_position;
@ -653,6 +656,21 @@ static void rtp_audio_process_capture(void *data)
impl->sink_resamp_delay = impl->io_rate_match->delay;
impl->sink_quantum = (uint64_t)(pos->clock.duration * SPA_NSEC_PER_SEC / rate);
}
/* Compensate for the stream resampler's delay. */
actual_timestamp -= pwt.buffered;
/* If we got a request for less than quantum worth of samples, it indicates that there
* is a gap created by the resampler. We have to skip it to avoid timestamp discontinuity. */
if (pwt.buffered > 0) {
int32_t ideal_quantum = (int32_t)scale_u64(pos->clock.duration, impl->rate, rate);
if (wanted < ideal_quantum) {
int32_t num_samples_to_skip = ideal_quantum - wanted;
pw_log_info("wanted: %" PRId32 " < ideal quantum: %" PRId32 " - skipping %"
PRId32" samples", wanted, ideal_quantum, num_samples_to_skip);
actual_timestamp += num_samples_to_skip;
}
}
} else {
actual_timestamp = expected_timestamp;
next_nsec = 0;