commit ee5ccbc57fcbd14032ec28dd1bbd74f1baaf4324
author Niels Möller <nisse@webrtc.org> Wed Mar 06 16:47:29 2019 +0100
committer Commit Bot <commit-bot@chromium.org> Wed Mar 06 17:15:00 2019 +0000
tree b890a364a7b51e7bc17ba65636313f5b72e6840a
parent 232b3fda921a475e873f09cc58fbc8ceffdbe4ac

Move ownership of RTPSenderAudio to ChannelSend.

This change takes out responsibility for packetization from the
RtpRtcp class, and deletes the method RtpRtcp::SendOutgoingData.

Video packetization was similarly moved in cl
https://webrtc-review.googlesource.com/c/src/+/123187

Bug: webrtc:7135
Change-Id: I0953125a5ca22a2ce51761b83693e0bb8ea74cd8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125721
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@{#27000}

audio/audio_send_stream_unittest.cc

diff --git a/audio/audio_send_stream_unittest.cc b/audio/audio_send_stream_unittest.cc
index 1d4d5b7..df5ee30 100644
--- a/audio/audio_send_stream_unittest.cc
+++ b/audio/audio_send_stream_unittest.cc
@@ -450,6 +450,7 @@
             stolen_encoder = std::move(*encoder);
             return true;
           }));
+  EXPECT_CALL(*helper.channel_send(), RegisterCngPayloadType(105, 8000));
 
   auto send_stream = helper.CreateAudioSendStream();
 
@@ -501,6 +502,8 @@
   EXPECT_CALL(*helper.channel_send(), SetEncoderForMock(_, _))
       .WillOnce(Return());
 
+  EXPECT_CALL(*helper.channel_send(), RegisterCngPayloadType(105, 8000));
+
   helper.config().send_codec_spec =
       AudioSendStream::Config::SendCodecSpec(9, kG722Format);
   helper.config().send_codec_spec->cng_payload_type = 105;