Test and fix for huge bwe drop after alr state.

webrtc/audio/test/low_bandwidth_audio_test.h

 /*
  *  Copyright (c) 2017 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.
  */
 #ifndef WEBRTC_AUDIO_TEST_LOW_BANDWIDTH_AUDIO_TEST_H_
@@ skipping 33 matching lines @@
   test::PacketTransport* CreateSendTransport(Call* sender_call) override;
   test::PacketTransport* CreateReceiveTransport() override;
 
   void ModifyAudioConfigs(
       AudioSendStream::Config* send_config,
       std::vector<AudioReceiveStream::Config>* receive_configs) override;
 
   void PerformTest() override;
   void OnTestFinished() override;
 
+  Call* sender_call_;

[holmer, 2017/06/19 08:53:49]

AFAICT this can still be private?

[tschumi, 2017/06/19 10:36:03]

I think i could move it in to the NoBandwidthDropAfterDtx class.
Then it could be private. But this is the AudioQualityTest.

[holmer, 2017/06/19 11:51:53]

Ok, maybe worth doing? Or protected is enough?

[tschumi, 2017/06/19 12:08:48]

Ok moved sender_call_ to NoBandwidthDropAfterDtx.

+
  private:
   test::FakeAudioDevice* send_audio_device_;
 };
 
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_TEST_LOW_BANDWIDTH_AUDIO_TEST_H_