Low-bandwidth audio testing

webrtc/test/fake_audio_device.h

 /*
  *  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.
  */
 #ifndef WEBRTC_TEST_FAKE_AUDIO_DEVICE_H_
@@ skipping 71 matching lines @@
       std::string filename, int sampling_frequency_in_hz);
 
   // Returns a Capturer instance that gets its data from a file.
   // Automatically detects sample rate.
   static std::unique_ptr<Capturer> CreateWavFileReader(std::string filename);
 
   // Returns a Renderer instance that writes its data to a file.
   static std::unique_ptr<Renderer> CreateWavFileWriter(
       std::string filename, int sampling_frequency_in_hz);
 
+  // Returns a Renderer instance that writes its data to a file a WAV file,

[stefan-webrtc, 2017/03/20 14:40:10]

"to a WAV file"

[oprypin_webrtc, 2017/03/20 15:09:16]

Done.

+  // cutting off silence at the beginning (not necessarily perfect silence, see
+  // kAmplitudeThreshold) and at the end (only actual 0 samples in this case).
+  static std::unique_ptr<Renderer> CreateBoundedWavFileWriter(
+      std::string filename, int sampling_frequency_in_hz);
+
   // Returns a Renderer instance that does nothing with the audio data.
   static std::unique_ptr<Renderer> CreateDiscardRenderer(
       int sampling_frequency_in_hz);
 
   int32_t Init() override;
   int32_t RegisterAudioCallback(AudioTransport* callback) override;
 
   int32_t StartPlayout() override;
   int32_t StopPlayout() override;
   int32_t StartRecording() override;
@@ skipping 27 matching lines @@
   std::vector<int16_t> playout_buffer_ GUARDED_BY(lock_);
   rtc::BufferT<int16_t> recording_buffer_ GUARDED_BY(lock_);
 
   std::unique_ptr<EventTimerWrapper> tick_;
   rtc::PlatformThread thread_;
 };
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_TEST_FAKE_AUDIO_DEVICE_H_