Add send-side BWE to WebRtcVoiceEngine under a finch experiment.

webrtc/call/bitrate_estimator_tests.cc

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 #include <functional>
@@ skipping 172 matching lines @@
 
       if (receive_audio) {
         AudioReceiveStream::Config receive_config;
         receive_config.rtp.remote_ssrc = test_->video_send_config_.rtp.ssrcs[0];
         // Bogus non-default id to prevent hitting a RTC_DCHECK when creating
         // the AudioReceiveStream. Every receive stream has to correspond to
         // an underlying channel id.
         receive_config.voe_channel_id = 0;
         receive_config.rtp.extensions.push_back(
             RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId));
-        receive_config.combined_audio_video_bwe = true;
         audio_receive_stream_ =
             test_->receiver_call_->CreateAudioReceiveStream(receive_config);
       } else {
         VideoReceiveStream::Decoder decoder;
         decoder.decoder = &fake_decoder_;
         decoder.payload_type =
             test_->video_send_config_.encoder_settings.payload_type;
         decoder.payload_name =
             test_->video_send_config_.encoder_settings.payload_name;
         test_->receive_config_.decoders.clear();
@@ skipping 62 matching lines @@
 
 TEST_F(BitrateEstimatorTest, InstantiatesTOFPerDefaultForVideo) {
   video_send_config_.rtp.extensions.push_back(
       RtpExtension(RtpExtension::kTOffset, kTOFExtensionId));
   receiver_log_.PushExpectedLogLine(kSingleStreamLog);
   receiver_log_.PushExpectedLogLine(kSingleStreamLog);
   streams_.push_back(new Stream(this, false));
   EXPECT_TRUE(receiver_log_.Wait());
 }
 
-TEST_F(BitrateEstimatorTest, ImmediatelySwitchToASTForAudio) {

[the sun, 2016/01/26 10:26:55]

Do you have more cleanup to do if AST is now deprecated?

[stefan-webrtc, 2016/01/26 11:40:54]

There is a lot of clean up that can be done, but only for audio at the moment.
I'm leaving most of it for a follow-up. 

-  video_send_config_.rtp.extensions.push_back(
-      RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId));
-  receiver_log_.PushExpectedLogLine(kSingleStreamLog);
-  receiver_log_.PushExpectedLogLine(kSingleStreamLog);
-  receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
-  receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
-  streams_.push_back(new Stream(this, true));
-  EXPECT_TRUE(receiver_log_.Wait());
-}
-
 TEST_F(BitrateEstimatorTest, ImmediatelySwitchToASTForVideo) {
   video_send_config_.rtp.extensions.push_back(
       RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId));
   receiver_log_.PushExpectedLogLine(kSingleStreamLog);
   receiver_log_.PushExpectedLogLine(kSingleStreamLog);
   receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
   receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
   streams_.push_back(new Stream(this, false));
   EXPECT_TRUE(receiver_log_.Wait());
 }
 
-TEST_F(BitrateEstimatorTest, SwitchesToASTForAudio) {
-  receiver_log_.PushExpectedLogLine(kSingleStreamLog);
-  receiver_log_.PushExpectedLogLine(kSingleStreamLog);
-  streams_.push_back(new Stream(this, true));
-  EXPECT_TRUE(receiver_log_.Wait());
-
-  video_send_config_.rtp.extensions.push_back(
-      RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId));
-  receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
-  receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
-  streams_.push_back(new Stream(this, true));
-  EXPECT_TRUE(receiver_log_.Wait());
-}
-
 TEST_F(BitrateEstimatorTest, SwitchesToASTForVideo) {
   video_send_config_.rtp.extensions.push_back(
       RtpExtension(RtpExtension::kTOffset, kTOFExtensionId));
   receiver_log_.PushExpectedLogLine(kSingleStreamLog);
   receiver_log_.PushExpectedLogLine(kSingleStreamLog);
   streams_.push_back(new Stream(this, false));
   EXPECT_TRUE(receiver_log_.Wait());
 
   video_send_config_.rtp.extensions[0] =
       RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId);
@@ skipping 22 matching lines @@
       RtpExtension(RtpExtension::kTOffset, kTOFExtensionId);
   receiver_log_.PushExpectedLogLine(
       "WrappingBitrateEstimator: Switching to transmission time offset RBE.");
   receiver_log_.PushExpectedLogLine(kSingleStreamLog);
   streams_.push_back(new Stream(this, false));
   streams_[0]->StopSending();
   streams_[1]->StopSending();
   EXPECT_TRUE(receiver_log_.Wait());
 }
 }  // namespace webrtc