| Index: webrtc/audio/test/audio_end_to_end_test.h
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| diff --git a/webrtc/audio/test/low_bandwidth_audio_test.h b/webrtc/audio/test/audio_end_to_end_test.h
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| similarity index 64%
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| rename from webrtc/audio/test/low_bandwidth_audio_test.h
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| rename to webrtc/audio/test/audio_end_to_end_test.h
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| index ae75707f66d3b64d3f7b1d707ec8dab8d2b34db1..d14b7a108f6785541f17cb61907568ca4a7022dc 100644
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| --- a/webrtc/audio/test/low_bandwidth_audio_test.h
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| +++ b/webrtc/audio/test/audio_end_to_end_test.h
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| @@ -7,28 +7,28 @@
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|   *  in the file PATENTS.  All contributing project authors may
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|   *  be found in the AUTHORS file in the root of the source tree.
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|   */
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| -#ifndef WEBRTC_AUDIO_TEST_LOW_BANDWIDTH_AUDIO_TEST_H_
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| -#define WEBRTC_AUDIO_TEST_LOW_BANDWIDTH_AUDIO_TEST_H_
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| +#ifndef WEBRTC_AUDIO_TEST_AUDIO_END_TO_END_TEST_H_
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| +#define WEBRTC_AUDIO_TEST_AUDIO_END_TO_END_TEST_H_
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|  
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|  #include <memory>
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|  #include <string>
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|  #include <vector>
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|  
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|  #include "webrtc/test/call_test.h"
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| -#include "webrtc/test/fake_audio_device.h"
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|  
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|  namespace webrtc {
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|  namespace test {
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|  
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| -class AudioQualityTest : public test::EndToEndTest {
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| +class AudioEndToEndTest : public test::EndToEndTest {
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|   public:
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| -  AudioQualityTest();
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| +  AudioEndToEndTest();
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|  
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|   protected:
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| -  virtual std::string AudioInputFile();
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| -  virtual std::string AudioOutputFile();
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| +  test::FakeAudioDevice* send_audio_device() { return send_audio_device_; }
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| +  const AudioSendStream* send_stream() const { return send_stream_; }
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| +  const AudioReceiveStream* receive_stream() const { return receive_stream_; }
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|  
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| -  virtual FakeNetworkPipe::Config GetNetworkPipeConfig();
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| +  virtual FakeNetworkPipe::Config GetNetworkPipeConfig() const;
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|  
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|    size_t GetNumVideoStreams() const override;
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|    size_t GetNumAudioStreams() const override;
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| @@ -50,15 +50,19 @@ class AudioQualityTest : public test::EndToEndTest {
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|    void ModifyAudioConfigs(
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|        AudioSendStream::Config* send_config,
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|        std::vector<AudioReceiveStream::Config>* receive_configs) override;
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| +  void OnAudioStreamsCreated(
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| +      AudioSendStream* send_stream,
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| +      const std::vector<AudioReceiveStream*>& receive_streams) override;
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|  
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|    void PerformTest() override;
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| -  void OnTestFinished() override;
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|  
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|   private:
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| -  test::FakeAudioDevice* send_audio_device_;
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| +  test::FakeAudioDevice* send_audio_device_ = nullptr;
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| +  AudioSendStream* send_stream_ = nullptr;
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| +  AudioReceiveStream* receive_stream_ = nullptr;
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|  };
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|  
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|  }  // namespace test
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|  }  // namespace webrtc
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|  
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| -#endif  // WEBRTC_AUDIO_TEST_LOW_BANDWIDTH_AUDIO_TEST_H_
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| +#endif  // WEBRTC_AUDIO_TEST_AUDIO_END_TO_END_TEST_H_
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| 
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