commit d2aa8f97f1091f87561d5559a72c27ec73fa4cdb
author Marina Ciocea <marinaciocea@webrtc.org> Tue Mar 31 11:29:56 2020 +0200
committer Commit Bot <commit-bot@chromium.org> Tue Mar 31 11:14:00 2020 +0000
tree 886a42fd7a4ac70b69f6f0c8c3f736a98a7387aa
parent e062c15ce64929f488bd57e5733c6661d11890ed

Insert audio frame transformer between encoder and packetizer.

The frame transformer is passed from RTPSenderInterface through the
library to be eventually set in ChannelSend, 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: I01b2adc3c96b948d182d5401a9a4fe14cf5960a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171870
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30946}

audio/audio_send_stream_unittest.cc

diff --git a/audio/audio_send_stream_unittest.cc b/audio/audio_send_stream_unittest.cc
index de152cd..de1f2fe 100644
--- a/audio/audio_send_stream_unittest.cc
+++ b/audio/audio_send_stream_unittest.cc
@@ -215,6 +215,8 @@
     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_, SetEncoderToPacketizerFrameTransformer(_))
+        .Times(1);
     EXPECT_CALL(rtp_rtcp_, SetExtmapAllowMixed(false)).Times(1);
     EXPECT_CALL(*channel_send_,
                 SetSendAudioLevelIndicationStatus(true, kAudioLevelId))