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Set RtpRtcp config receive_only in voe::ChannelReceive

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Followup to https://webrtc-review.googlesource.com/c/103803

Bug: webrtc:9801
Change-Id: I8467aea241f0470aa116639ddcf5f7ef9a3cc711
Reviewed-on: https://webrtc-review.googlesource.com/c/106860
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25501}
This commit is contained in:
Niels Möller 2018-10-31 15:25:26 +01:00 committed by Commit Bot
parent aed3070ad4
commit fd1a2fb8e4
2 changed files with 11 additions and 4 deletions
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6
audio/channel_receive.cc
View file

@ -299,9 +299,7 @@ ChannelReceive::ChannelReceive(
rtp_receive_statistics_->EnableRetransmitDetection(remote_ssrc_, true);
RtpRtcp::Configuration configuration;
configuration.audio = true;
// TODO(nisse): Also set receiver_only = true, but that seems to break RTT
// estimation, resulting in test failures for
// PeerConnectionIntegrationTest.GetCaptureStartNtpTimeWithOldStatsApi
configuration.receiver_only = true;
configuration.outgoing_transport = rtcp_send_transport;
configuration.receive_statistics = rtp_receive_statistics_.get();
@ -779,6 +777,8 @@ int64_t ChannelReceive::GetRTT() const {
int64_t avg_rtt = 0;
int64_t max_rtt = 0;
int64_t min_rtt = 0;
// TODO(nisse): This method computes RTT based on sender reports, even though
// a receive stream is not supposed to do that.
if (_rtpRtcpModule->RTT(remote_ssrc_, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
0) {
return 0;

9
modules/rtp_rtcp/source/rtcp_receiver.cc
View file

@ -504,7 +504,14 @@ void RTCPReceiver::HandleReportBlock(const ReportBlock& report_block,
// If no SR has been received yet, the field is set to zero.
// Receiver rtp_rtcp module is not expected to calculate rtt using
// Sender Reports even if it accidentally can.
if (!receiver_only_ && send_time_ntp != 0) {
// TODO(nisse): Use this way to determine the RTT only when |receiver_only_|
// is false. However, that currently breaks the tests of the
// googCaptureStartNtpTimeMs stat for audio receive streams. To fix, either
// delete all dependencies on RTT measurements for audio receive streams, or
// ensure that audio receive streams that need RTT and stats that depend on it
// are configured with an associated audio send stream.
if (send_time_ntp != 0) {
uint32_t delay_ntp = report_block.delay_since_last_sr();
// Local NTP time.
uint32_t receive_time_ntp =