| commit | 3e9af7fe059af739d11bf8693669ff48d50efcfb | [log] [tgz] |
|---|---|---|
| author | Marina Ciocea <marinaciocea@webrtc.org> | Wed Apr 01 07:46:16 2020 +0200 |
| committer | Commit Bot <commit-bot@chromium.org> | Wed Apr 01 08:15:53 2020 +0000 |
| tree | 0e8e5def898be1eedf0f188bc46e4787cf68021c | |
| parent | 784630f0e63e6cc11cb44eb11db578d913e3d37d [diff] [blame] |
Insert audio frame transformer between depacketizer and decoder. The frame transformer is passed from RTPReceiverInterface through the library to be eventually set in ChannelReceive, where the frame transformation will occur in the follow-up CL. Insertable Streams Web API explainer: https://github.com/alvestrand/webrtc-media-streams/blob/master/explainer.md Design doc for WebRTC library changes: http://doc/1eiLkjNUkRy2FssCPLUp6eH08BZuXXoHfbbBP1ZN7EVk Bug: webrtc:11380 Change-Id: I5af06d1431047ef50d00e304cf95e92a832b4220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171872 Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30956}
diff --git a/audio/audio_receive_stream_unittest.cc b/audio/audio_receive_stream_unittest.cc index 0b2cae5..186eb1c 100644 --- a/audio/audio_receive_stream_unittest.cc +++ b/audio/audio_receive_stream_unittest.cc
@@ -100,6 +100,8 @@ .WillRepeatedly(Invoke([](const std::map<int, SdpAudioFormat>& codecs) { EXPECT_THAT(codecs, ::testing::IsEmpty()); })); + EXPECT_CALL(*channel_receive_, SetDepacketizerToDecoderFrameTransformer(_)) + .Times(1); stream_config_.rtp.local_ssrc = kLocalSsrc; stream_config_.rtp.remote_ssrc = kRemoteSsrc;