commit bd9106d88f70922dc29098bf50b89ba38c7a9ba4
author Philipp Hancke <philipp.hancke@googlemail.com> Thu Oct 14 08:34:53 2021 +0200
committer WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> Thu Nov 04 12:47:48 2021 +0000
tree 5d6d84336cd61a96883c21f8b015318d461e8ef8
parent 3677bbaa32929769c9f012529bf717aac193302c

voice_engine: dont announce rid/rrid header extensions

which do not make sense for audio due to lack of support for RTX.

BUG=webrtc:13279

Change-Id: Ida42d8912bf993f01e0dc5c6ffbdbf4b84495c30
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235061
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35309}

audio/audio_send_stream.cc

diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc
index 7618dde..7d70592 100644
--- a/audio/audio_send_stream.cc
+++ b/audio/audio_send_stream.cc
@@ -311,24 +311,6 @@
     rtp_rtcp_module_->SetMid(new_config.rtp.mid);
   }
 
-  // RID RTP header extension
-  if ((first_time || new_ids.rid != old_ids.rid ||
-       new_ids.repaired_rid != old_ids.repaired_rid ||
-       new_config.rtp.rid != old_config.rtp.rid)) {
-    if (new_ids.rid != 0 || new_ids.repaired_rid != 0) {
-      if (new_config.rtp.rid.empty()) {
-        rtp_rtcp_module_->DeregisterSendRtpHeaderExtension(RtpStreamId::Uri());
-      } else if (new_ids.repaired_rid != 0) {
-        rtp_rtcp_module_->RegisterRtpHeaderExtension(RtpStreamId::Uri(),
-                                                     new_ids.repaired_rid);
-      } else {
-        rtp_rtcp_module_->RegisterRtpHeaderExtension(RtpStreamId::Uri(),
-                                                     new_ids.rid);
-      }
-    }
-    rtp_rtcp_module_->SetRid(new_config.rtp.rid);
-  }
-
   if (first_time || new_ids.abs_capture_time != old_ids.abs_capture_time) {
     absl::string_view uri = AbsoluteCaptureTimeExtension::Uri();
     rtp_rtcp_module_->DeregisterSendRtpHeaderExtension(uri);

audio/audio_send_stream_unittest.cc

diff --git a/audio/audio_send_stream_unittest.cc b/audio/audio_send_stream_unittest.cc
index 6ac5f4e..3c5b142 100644
--- a/audio/audio_send_stream_unittest.cc
+++ b/audio/audio_send_stream_unittest.cc
@@ -246,7 +246,6 @@
           .Times(1);
     }
     EXPECT_CALL(*channel_send_, ResetSenderCongestionControlObjects()).Times(1);
-    EXPECT_CALL(rtp_rtcp_, SetRid(std::string())).Times(1);
   }
 
   void SetupMockForSetupSendCodec(bool expect_set_encoder_call) {

media/engine/webrtc_voice_engine.cc

diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc
index ad72aae..5d4455c 100644
--- a/media/engine/webrtc_voice_engine.cc
+++ b/media/engine/webrtc_voice_engine.cc
@@ -662,12 +662,10 @@
   RTC_DCHECK(signal_thread_checker_.IsCurrent());
   std::vector<webrtc::RtpHeaderExtensionCapability> result;
   int id = 1;
-  for (const auto& uri :
-       {webrtc::RtpExtension::kAudioLevelUri,
-        webrtc::RtpExtension::kAbsSendTimeUri,
-        webrtc::RtpExtension::kTransportSequenceNumberUri,
-        webrtc::RtpExtension::kMidUri, webrtc::RtpExtension::kRidUri,
-        webrtc::RtpExtension::kRepairedRidUri}) {
+  for (const auto& uri : {webrtc::RtpExtension::kAudioLevelUri,
+                          webrtc::RtpExtension::kAbsSendTimeUri,
+                          webrtc::RtpExtension::kTransportSequenceNumberUri,
+                          webrtc::RtpExtension::kMidUri}) {
     result.emplace_back(uri, id++, webrtc::RtpTransceiverDirection::kSendRecv);
   }
   return result;