Cleaned up the WebRtcAudioCapturer a bit.

content/renderer/media/webrtc_audio_capturer_unittest.cc

 // Copyright 2013 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "base/logging.h"
 #include "content/renderer/media/rtc_media_constraints.h"
 #include "content/renderer/media/webrtc_audio_capturer.h"
 #include "content/renderer/media/webrtc_local_audio_track.h"
 #include "media/audio/audio_parameters.h"
 #include "media/base/audio_bus.h"
@@ skipping 76 matching lines @@
  protected:
   WebRtcAudioCapturerTest()
 #if defined(OS_ANDROID)
       : params_(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
                 media::CHANNEL_LAYOUT_STEREO, 48000, 16, 960) {
     // Android works with a buffer size bigger than 20ms.
 #else
       : params_(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
                 media::CHANNEL_LAYOUT_STEREO, 48000, 16, 128) {
 #endif
-    capturer_ = WebRtcAudioCapturer::CreateCapturer();
-    capturer_->Initialize(-1, params_.channel_layout(), params_.sample_rate(),
-                          params_.frames_per_buffer(), 0, std::string(), 0, 0,
-                          params_.effects());
+    capturer_ = WebRtcAudioCapturer::CreateCapturer(
+        -1,
+        StreamDeviceInfo(MEDIA_DEVICE_AUDIO_CAPTURE,
+                         "", "", params_.sample_rate(),
+                         params_.channel_layout(),
+                         params_.frames_per_buffer()),
+        NULL);
     capturer_source_ = new MockCapturerSource();
-    EXPECT_CALL(*capturer_source_.get(), Initialize(_, capturer_.get(), 0));
-    capturer_->SetCapturerSource(capturer_source_,
-                                 params_.channel_layout(),
-                                 params_.sample_rate(),
-                                 params_.effects());
+    EXPECT_CALL(*capturer_source_.get(), Initialize(_, capturer_.get(), -1));
+    capturer_->SetCapturerSourceForTesting(capturer_source_, params_);
 
     EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true));
     EXPECT_CALL(*capturer_source_.get(), Start());
     RTCMediaConstraints constraints;
     ApplyFixedAudioConstraints(&constraints);
     track_ = WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL,
                                            NULL, &constraints);
     static_cast<WebRtcLocalAudioSourceProvider*>(
         track_->audio_source_provider())->SetSinkParamsForTesting(params_);
     track_->Start();
@@ skipping 48 matching lines @@
   EXPECT_EQ(cached_delay.InMilliseconds(), delay_ms);
   EXPECT_EQ(cached_volume, expected_volume_value);
   EXPECT_EQ(cached_key_pressed, key_pressed);
 
   track_->RemoveSink(sink.get());
   EXPECT_CALL(*capturer_source_.get(), Stop());
   capturer_->Stop();
 }
 
 }  // namespace content