Variable audio bitrate.

webrtc/call/rampup_tests.h

 /*
  *  Copyright (c) 2014 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.
  */
 
@@ skipping 46 matching lines @@
                     const std::string& units) const;
   void TriggerTestDone();
 
   rtc::Event event_;
   Clock* const clock_;
   FakeNetworkPipe::Config forward_transport_config_;
   const size_t num_video_streams_;
   const size_t num_audio_streams_;
   const bool rtx_;
   const bool red_;
+  Call* sender_call_;
   VideoSendStream* send_stream_;
   test::PacketTransport* send_transport_;
 
  private:
   typedef std::map<uint32_t, uint32_t> SsrcMap;
 
   Call::Config GetSenderCallConfig() override;
   void OnVideoStreamsCreated(
       VideoSendStream* send_stream,
       const std::vector<VideoReceiveStream*>& receive_streams) override;
@@ skipping 15 matching lines @@
   int64_t test_start_ms_;
   int64_t ramp_up_finished_ms_;
 
   const std::string extension_type_;
   std::vector<uint32_t> video_ssrcs_;
   std::vector<uint32_t> video_rtx_ssrcs_;
   std::vector<uint32_t> audio_ssrcs_;
   SsrcMap rtx_ssrc_map_;
 
   rtc::PlatformThread poller_thread_;
-  Call* sender_call_;
 };
 
 class RampUpDownUpTester : public RampUpTester {
  public:
   RampUpDownUpTester(size_t num_video_streams,
                      size_t num_audio_streams,
                      unsigned int start_bitrate_bps,
                      const std::string& extension_type,
                      bool rtx,
                      bool red);
@@ skipping 14 matching lines @@
   std::string GetModifierString() const;
   void EvolveTestState(int bitrate_bps, bool suspended);
 
   TestStates test_state_;
   int64_t state_start_ms_;
   int64_t interval_start_ms_;
   int sent_bytes_;
 };
 }  // namespace webrtc
 #endif  // WEBRTC_CALL_RAMPUP_TESTS_H_