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module-rtp: Improve rtp_audio_flush_packets() logging

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This commit is contained in:
Carlos Rafael Giani 2026-01-19 21:03:48 +01:00
parent 9f0dc9c1af
commit 5d7f21f130
1 changed files with 41 additions and 8 deletions
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49
src/modules/module-rtp/audio.c
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@ -62,6 +62,7 @@ static void rtp_audio_process_playback(void *data)
if (impl->direct_timestamp) {
uint32_t num_samples_to_read;
uint32_t read_index;
/* In direct timestamp mode, the focus lies on synchronized playback, not
* on a constant latency. The ring buffer fill level is not of interest
@ -106,13 +107,11 @@ static void rtp_audio_process_playback(void *data)
* quantity tracks how many samples are actually available. */
if (impl->io_position) {
uint32_t dummy_read_index;
/* Shift clock position by stream delay to compensate
* for processing and output delay. */
timestamp = impl->io_position->clock.position + device_delay;
spa_ringbuffer_read_update(&impl->ring, timestamp);
avail = spa_ringbuffer_get_read_index(&impl->ring, &dummy_read_index);
avail = spa_ringbuffer_get_read_index(&impl->ring, &read_index);
} else {
/* In the unlikely case that no spa_io_position pointer
* was passed yet by PipeWire to this node, resort to a
@ -121,6 +120,7 @@ static void rtp_audio_process_playback(void *data)
* but _something_ is needed as read index until the
* spa_io_position is available. */
avail = spa_ringbuffer_get_read_index(&impl->ring, &timestamp);
read_index = timestamp;
}
/* If avail is 0, it means that the ring buffer is empty. <0 means
@ -128,8 +128,34 @@ static void rtp_audio_process_playback(void *data)
* is ahead of the RTP data (this can happen when the PTP master
* changes for example). And in cases where only a little bit of
* data is left, it is important to not try to use more than what
* is actually available. */
num_samples_to_read = (avail > 0) ? SPA_MIN((uint32_t)avail, wanted) : 0;
* is actually available.
* Overruns would happen if the write pointer is further ahead than
* what the ringbuffer size actually allows. This too can happen
* if the PTP time jumps. No actual buffer overflow would happen
* then, since the write operations always apply modulo to the
* timestamps to wrap around the ringbuffer borders.
*/
bool has_underrun = (avail < 0);
bool has_overrun = !has_underrun && ((uint32_t)avail) > impl->actual_max_buffer_size;
num_samples_to_read = has_underrun ? 0 : SPA_MIN((uint32_t)avail, wanted);
/* Do some additional logging in the under/overrun cases. */
if (SPA_UNLIKELY(pw_log_topic_enabled(SPA_LOG_LEVEL_TRACE, PW_LOG_TOPIC_DEFAULT)))
{
uint32_t write_index;
int32_t filled = spa_ringbuffer_get_write_index(&impl->ring, &write_index);
if (has_underrun) {
pw_log_trace("Direct timestamp mode: Read index underrun: write_index: %"
PRIu32 ", read_index: %" PRIu32 ", wanted: %u - filled: %" PRIi32,
write_index, read_index, wanted, filled);
} else if (has_overrun) {
pw_log_trace("Direct timestamp mode: Read index overrun: write_index: %"
PRIu32 ", read_index: %" PRIu32 ", wanted: %u - filled: %" PRIi32
", buffer size: %u", write_index, read_index, wanted, filled,
impl->actual_max_buffer_size);
}
}
if (num_samples_to_read > 0) {
spa_ringbuffer_read_data(&impl->ring,
@ -498,20 +524,27 @@ static void rtp_audio_flush_packets(struct impl *impl, uint32_t num_packets, uin
iov[0].iov_len = sizeof(header);
while (num_packets > 0) {
uint32_t rtp_timestamp;
if (impl->marker_on_first && impl->first)
header.m = 1;
else
header.m = 0;
rtp_timestamp = impl->ts_offset + (set_timestamp ? set_timestamp : timestamp);
header.sequence_number = htons(impl->seq);
header.timestamp = htonl(impl->ts_offset + (set_timestamp ? set_timestamp : timestamp));
header.timestamp = htonl(rtp_timestamp);
set_iovec(&impl->ring,
impl->buffer, impl->actual_max_buffer_size,
((uint64_t)timestamp * stride) % impl->actual_max_buffer_size,
&iov[1], tosend * stride);
pw_log_trace("sending %d packet:%d ts_offset:%d timestamp:%d",
tosend, num_packets, impl->ts_offset, timestamp);
pw_log_trace("sending %d packet:%d ts_offset:%d timestamp:%u (%f s)",
tosend, num_packets, impl->ts_offset, timestamp,
(double)timestamp * impl->io_position->clock.rate.num /
impl->io_position->clock.rate.denom);
rtp_stream_emit_send_packet(impl, iov, 3);