Use only first payload timestamp for RTCP SR generation for audio
Since now RTP rate is set correctly for audio, there's no need to
use the very last data packet rtp/capture timestamps for generating
RTCP SR packets.
Using only one (first) packet timestamp eliminates the jitter between
rtp and capture timestamps for audio. This jitter comes from the fact
that capture timestamp for audio is unknown and we generate bogus
timestamp at arbitrary, non-constant offset from the real capture time.
Bug: webrtc:9905
Change-Id: I855556184cfe994be39ab7780836a050f5a38c35
Reviewed-on: https://webrtc-review.googlesource.com/c/108580
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25430}diff --git a/audio/channel_send.cc b/audio/channel_send.cc
index cc8b1f7..e3d3472 100644
--- a/audio/channel_send.cc
+++ b/audio/channel_send.cc
@@ -333,6 +333,7 @@
// This call will trigger Transport::SendPacket() from the RTP/RTCP module.
if (!_rtpRtcpModule->SendOutgoingData((FrameType&)frameType, payloadType,
timeStamp,
+ // TODO(https://bugs.webrtc.org/9905):
// Leaving the time when this frame was
// received from the capture device as
// undefined for voice for now.
diff --git a/modules/rtp_rtcp/source/rtcp_sender.cc b/modules/rtp_rtcp/source/rtcp_sender.cc
index 67fdf34..b2aad1e 100644
--- a/modules/rtp_rtcp/source/rtcp_sender.cc
+++ b/modules/rtp_rtcp/source/rtcp_sender.cc
@@ -258,6 +258,15 @@
int64_t capture_time_ms,
int8_t payload_type) {
rtc::CritScope lock(&critical_section_rtcp_sender_);
+ // Workaround for https://bugs.webrtc.org/9905
+ // Only very first SetLastRtpTime for audio should update
+ // last_frame_capture_time_ms_ and last_payload_type_.
+ // This eliminates jitter between last rtp and capture timestamps.
+ // TODO(https://bugs.webrtc.org/9905): remove once the bug is fixed.
+ if (capture_time_ms < 0 && last_frame_capture_time_ms_ > 0 &&
+ payload_type != -1 && last_payload_type_ == payload_type) {
+ return;
+ }
// For compatibility with clients who don't set payload type correctly on all
// calls.
if (payload_type != -1) {
