commit 6298b5689048c81e1aa711d980b7c5e1248f676c
author Sebastian Jansson <srte@webrtc.org> Tue Jan 14 17:55:19 2020 +0100
committer Commit Bot <commit-bot@chromium.org> Wed Jan 15 19:01:50 2020 +0000
tree 94781d9652a7ca53ddf67b2c42c8b924070eb90c
parent c2509fec7c0cff95612a837e043032b5c85322c5

Cleanup: Using RtpRtcp directly from AudioSendStream

This reduces indirection and makes it easier to follow code. It also
fits into a long term strategy of reducing the scope of ChannelSend.

Bug: webrtc:9883
Change-Id: I2661c4aa6c561f7691beaaa289636254f7a58b72
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166042
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30273}

audio/audio_send_stream_unittest.cc

diff --git a/audio/audio_send_stream_unittest.cc b/audio/audio_send_stream_unittest.cc
index 95d7f73..0472366 100644
--- a/audio/audio_send_stream_unittest.cc
+++ b/audio/audio_send_stream_unittest.cc
@@ -195,6 +195,7 @@
     return *static_cast<MockAudioEncoderFactory*>(
         stream_config_.encoder_factory.get());
   }
+  MockRtpRtcp* rtp_rtcp() { return &rtp_rtcp_; }
   MockChannelSend* channel_send() { return channel_send_; }
   RtpTransportControllerSendInterface* transport() { return &rtp_transport_; }
 
@@ -213,15 +214,16 @@
     EXPECT_CALL(rtp_rtcp_, SSRC).WillRepeatedly(Return(kSsrc));
     EXPECT_CALL(*channel_send_, SetRTCP_CNAME(StrEq(kCName))).Times(1);
     EXPECT_CALL(*channel_send_, SetFrameEncryptor(_)).Times(1);
-    EXPECT_CALL(*channel_send_, SetExtmapAllowMixed(false)).Times(1);
+    EXPECT_CALL(rtp_rtcp_, SetExtmapAllowMixed(false)).Times(1);
     EXPECT_CALL(*channel_send_,
                 SetSendAudioLevelIndicationStatus(true, kAudioLevelId))
         .Times(1);
     EXPECT_CALL(rtp_transport_, GetBandwidthObserver())
         .WillRepeatedly(Return(&bandwidth_observer_));
     if (audio_bwe_enabled) {
-      EXPECT_CALL(*channel_send_,
-                  EnableSendTransportSequenceNumber(kTransportSequenceNumberId))
+      EXPECT_CALL(rtp_rtcp_,
+                  RegisterRtpHeaderExtension(TransportSequenceNumber::kUri,
+                                             kTransportSequenceNumberId))
           .Times(1);
       EXPECT_CALL(*channel_send_,
                   RegisterSenderCongestionControlObjects(
@@ -233,7 +235,7 @@
           .Times(1);
     }
     EXPECT_CALL(*channel_send_, ResetSenderCongestionControlObjects()).Times(1);
-    EXPECT_CALL(*channel_send_, SetRid(std::string(), 0, 0)).Times(1);
+    EXPECT_CALL(rtp_rtcp_, SetRid(std::string())).Times(1);
   }
 
   void SetupMockForSetupSendCodec(bool expect_set_encoder_call) {
@@ -705,8 +707,10 @@
   auto send_stream = helper.CreateAudioSendStream();
   auto new_config = helper.config();
   ConfigHelper::AddBweToConfig(&new_config);
-  EXPECT_CALL(*helper.channel_send(),
-              EnableSendTransportSequenceNumber(kTransportSequenceNumberId))
+
+  EXPECT_CALL(*helper.rtp_rtcp(),
+              RegisterRtpHeaderExtension(TransportSequenceNumber::kUri,
+                                         kTransportSequenceNumberId))
       .Times(1);
   {
     ::testing::InSequence seq;