Add the ability to read/write to WAV files in FakeAudioDevice

webrtc/call/call_perf_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.
  */
 
@@ skipping 128 matching lines @@
                                       float video_ntp_speed,
                                       float video_rtp_speed,
                                       float audio_rtp_speed) {
   const char* kSyncGroup = "av_sync";
   const uint32_t kAudioSendSsrc = 1234;
   const uint32_t kAudioRecvSsrc = 5678;
 
   metrics::Reset();
   VoiceEngine* voice_engine = VoiceEngine::Create();
   VoEBase* voe_base = VoEBase::GetInterface(voice_engine);
-  FakeAudioDevice fake_audio_device(audio_rtp_speed, 48000, 256);
+  FakeAudioDevice fake_audio_device(
+      FakeAudioDevice::CreatePulsedNoiseCapturer(256, 48000),
+      FakeAudioDevice::CreateDiscarder(48000), audio_rtp_speed);

[kwiberg-webrtc, 2017/03/13 10:18:21]

This looks very nice!

Hmm. You don't have to, but call sites like this one might get even easier to
read if CreateDiscarder was called CreateDiscardRenderer instead---in general,
if the capturers had "Capturer" in their names, and the renderers "Renderer".

[oprypin_webrtc, 2017/03/13 13:55:14]

Changed to CreateDiscardRenderer, but it's hard to do the same for
'CreateWavFileWriter' because if I include 'Renderer' there, it would be
confusing as it does not render WAV files...

[kwiberg-webrtc, 2017/03/13 14:22:26]

Fair point.

   EXPECT_EQ(0, voe_base->Init(&fake_audio_device, nullptr, decoder_factory_));
   VoEBase::ChannelConfig config;
   config.enable_voice_pacing = true;
   int send_channel_id = voe_base->CreateChannel(config);
   int recv_channel_id = voe_base->CreateChannel();
 
   AudioState::Config send_audio_state_config;
   send_audio_state_config.voice_engine = voice_engine;
   send_audio_state_config.audio_mixer = AudioMixerImpl::Create();
   Call::Config sender_config(&event_log_);
@@ skipping 576 matching lines @@
     uint32_t last_set_bitrate_kbps_;
     VideoSendStream* send_stream_;
     test::FrameGeneratorCapturer* frame_generator_;
     VideoEncoderConfig encoder_config_;
   } test;
 
   RunBaseTest(&test);
 }
 
 }  // namespace webrtc