mirrors/pulseaudio
1
0
Fork 0
mirror of https://gitlab.freedesktop.org/pulseaudio/pulseaudio.git synced 2025-11-12 13:30:10 -05:00
Code Issues Projects Releases Packages Wiki Activity Actions

rtp: gstreamer: Account for rounding errors in RTP timestamp conversion

Browse source 
Since the RTP timestamp is converted to time units and back, a small
error can creep up, which then results in a single frame error in where
we place the buffer in the output memblockq. This results in minor
glitches, so we check for and eliminate the error.
This commit is contained in:
Arun Raghavan 2020-10-22 10:12:03 -04:00
parent 5f12dde2f5
commit 314e97d995
1 changed files with 17 additions and 0 deletions
Download patch file Download diff file Expand all files Collapse all files

17
src/modules/rtp/rtp-gstreamer.c
View file

@ -54,6 +54,7 @@ struct pa_rtp_context {
GstElement *appsink;
GstCaps *meta_reference;
bool first_buffer;
uint32_t last_timestamp;
uint8_t *send_buf;
@ -482,6 +483,7 @@ pa_rtp_context* pa_rtp_context_new_recv(int fd, uint8_t payload, const pa_sample
c->fdsem = pa_fdsem_new();
c->ss = *ss;
c->send_buf = NULL;
c->first_buffer = true;
if (!gst_init_check(NULL, NULL, &error)) {
pa_log_error("Could not initialise GStreamer: %s", error->message);
@ -598,6 +600,21 @@ int pa_rtp_recv(pa_rtp_context *c, pa_memchunk *chunk, pa_mempool *pool, uint32_
if (timestamp != GST_CLOCK_TIME_NONE)
pa_timeval_rtstore(tstamp, timestamp / PA_NSEC_PER_USEC, false);
if (c->first_buffer) {
c->first_buffer = false;
c->last_timestamp = *rtp_tstamp;
} else {
/* The RTP clock -> time domain -> RTP clock transformation above might
* add a ±1 rounding error, so let's get rid of that */
uint32_t expected = c->last_timestamp + (uint32_t) (data_len / pa_rtp_context_get_frame_size(c));
int delta = *rtp_tstamp - expected;
if (delta == 1 || delta == -1)
*rtp_tstamp -= delta;
c->last_timestamp = *rtp_tstamp;
}
gst_buffer_list_unref(buf_list);
gst_object_unref(adapter);