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module-rtp: Convert clock pos to RTP time in RTP source and factor in ASRC

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This can easily be overlooked if the RTP rate equals the clock rate, which
is fairly common (for example, RTP rate and clock rate both being 48 kHz).

And, if an ASRC is active, and converting the output of the RTP source
node, the resampler's delay need to be taken into the account as well.
This commit is contained in:
Carlos Rafael Giani 2026-01-19 23:24:04 +01:00
parent e874f705a9
commit 6192c11038
1 changed files with 18 additions and 3 deletions
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21
src/modules/module-rtp/audio.c
View file

@ -22,6 +22,15 @@ static void ringbuffer_clear(struct spa_ringbuffer *rbuf SPA_UNUSED,
memset(iov[1].iov_base, 0, iov[1].iov_len);
}
static inline uint64_t scale_u64(uint64_t val, uint32_t num, uint32_t denom)
{
#if 0
return ((__uint128_t)val * num) / denom;
#else
return (uint64_t)((double)val / denom * num);
#endif
}
static void rtp_audio_process_playback(void *data)
{
struct impl *impl = data;
@ -107,9 +116,15 @@ static void rtp_audio_process_playback(void *data)
* quantity tracks how many samples are actually available. */
if (impl->io_position) {
/* Shift clock position by stream delay to compensate
* for processing and output delay. */
timestamp = impl->io_position->clock.position + device_delay;
uint32_t clock_rate = impl->io_position->clock.rate.denom;
/* Translate the clock position to an RTP timestamp and
* shift it to compensate for device delay and ASRC delay.
* The device delay is scaled along with the clock position,
* since both are expressed in clock sample units, while
* pwt.buffered is expressed in stream time. */
timestamp = scale_u64(impl->io_position->clock.position + device_delay,
impl->rate, clock_rate) + pwt.buffered;
spa_ringbuffer_read_update(&impl->ring, timestamp);
avail = spa_ringbuffer_get_read_index(&impl->ring, &read_index);
} else {