RTCP: implement reduced size RTCP for audio
reduced-size RTCP, i.e. not prefixing RTCP packets with either a sender report or receiver report has been implemented for a long time but only for video.
This CL adds it for audio as well. This reduces the size of audio NACKs (16 bytes, typically one NACK per packet) sent by not prefixing it with a receiver report (32 bytes).
Other packets are not affected as e.g. transport-cc feedback does not add a RR even though that is technically required.
The effect on NACK can be tested by running Chromium with
--disable-webrtc-encryption --force-fieldtrials=WebRTC-FakeNetworkReceiveConfig/loss_percent:5/
against this fiddle negotiating audio nack:
https://jsfiddle.net/fippo/8ubtLnfx/1/
BUG=webrtc:340041654
Change-Id: I06fb94742ff1b6f9a464c404bfc53913f23498d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350269
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42330}
diff --git a/audio/audio_receive_stream_unittest.cc b/audio/audio_receive_stream_unittest.cc
index 451d5f9..25e7e7b 100644
--- a/audio/audio_receive_stream_unittest.cc
+++ b/audio/audio_receive_stream_unittest.cc
@@ -120,6 +120,7 @@
channel_receive_ = new ::testing::StrictMock<MockChannelReceive>();
EXPECT_CALL(*channel_receive_, SetNACKStatus(true, 15)).Times(1);
+ EXPECT_CALL(*channel_receive_, SetRtcpMode(_)).Times(1);
EXPECT_CALL(*channel_receive_,
RegisterReceiverCongestionControlObjects(&packet_router_))
.Times(1);
@@ -208,9 +209,10 @@
AudioReceiveStreamInterface::Config config;
config.rtp.remote_ssrc = kRemoteSsrc;
config.rtp.local_ssrc = kLocalSsrc;
+ config.rtp.rtcp_mode = RtcpMode::kOff;
EXPECT_EQ(
"{rtp: {remote_ssrc: 1234, local_ssrc: 5678, nack: "
- "{rtp_history_ms: 0}}, "
+ "{rtp_history_ms: 0}, rtcp: off}, "
"rtcp_send_transport: null}",
config.ToString());
}
