Cleaned up the WebRtcAudioCapturer a bit.

content/renderer/media/webrtc_local_audio_track_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/synchronization/waitable_event.h"
 #include "base/test/test_timeouts.h"
 #include "content/renderer/media/rtc_media_constraints.h"
 #include "content/renderer/media/webrtc_audio_capturer.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "content/renderer/media/webrtc_local_audio_source_provider.h"
@@ skipping 144 matching lines @@
   MOCK_METHOD1(OnSetFormat, void(const media::AudioParameters& params));
 };
 
 }  // namespace
 
 class WebRtcLocalAudioTrackTest : public ::testing::Test {
  protected:
   virtual void SetUp() OVERRIDE {
     params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
                   media::CHANNEL_LAYOUT_STEREO, 2, 0, 48000, 16, 480);
-    capturer_ = WebRtcAudioCapturer::CreateCapturer();
+    capturer_ = WebRtcAudioCapturer::CreateCapturer(-1, StreamDeviceInfo());
     capturer_source_ = new MockCapturerSource(capturer_.get());
-    EXPECT_CALL(*capturer_source_.get(), OnInitialize(_, capturer_.get(), 0))
+    EXPECT_CALL(*capturer_source_.get(), OnInitialize(_, capturer_.get(), -1))
         .WillOnce(Return());
-    capturer_->SetCapturerSource(capturer_source_,
-                                 params_.channel_layout(),
-                                 params_.sample_rate(),
-                                 params_.effects());
+    capturer_->SetCapturerSourceForTesting(capturer_source_, params_);
   }
 
   media::AudioParameters params_;
   scoped_refptr<MockCapturerSource> capturer_source_;
   scoped_refptr<WebRtcAudioCapturer> capturer_;
 };
 
 // Creates a capturer and audio track, fakes its audio thread, and
 // connect/disconnect the sink to the audio track on the fly, the sink should
 // get data callback when the track is connected to the capturer but not when
@@ skipping 266 matching lines @@
           &constraints);
   static_cast<WebRtcLocalAudioSourceProvider*>(
       track->audio_source_provider())->SetSinkParamsForTesting(params_);
   track->Start();
 
   // Setting new source to the capturer and the track should still get packets.
   scoped_refptr<MockCapturerSource> new_source(
       new MockCapturerSource(capturer_.get()));
   EXPECT_CALL(*capturer_source_.get(), OnStop());
   EXPECT_CALL(*new_source.get(), SetAutomaticGainControl(true));
-  EXPECT_CALL(*new_source.get(), OnInitialize(_, capturer_.get(), 0))
+  EXPECT_CALL(*new_source.get(), OnInitialize(_, capturer_.get(), -1))
       .WillOnce(Return());
   EXPECT_CALL(*new_source.get(), OnStart());
-  capturer_->SetCapturerSource(new_source,
-                               params_.channel_layout(),
-                               params_.sample_rate(),
-                               params_.effects());
+  capturer_->SetCapturerSourceForTesting(new_source, params_);
 
   // Stop the track.
   EXPECT_CALL(*new_source.get(), OnStop());
   capturer_->Stop();
 }
 
 // Create a new capturer with new source, connect it to a new audio track.
 TEST_F(WebRtcLocalAudioTrackTest, ConnectTracksToDifferentCapturers) {
   // Setup the first audio track and start it.
   EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true));
@@ skipping 17 matching lines @@
   EXPECT_CALL(
       *sink_1.get(),
       CaptureData(
           kNumberOfNetworkChannelsForTrack1, 48000, 2, _, 0, 0, false, false))
       .Times(AnyNumber()).WillRepeatedly(Return());
   EXPECT_CALL(*sink_1.get(), OnSetFormat(_)).Times(AnyNumber());
   track_1->AddSink(sink_1.get());
 
   // Create a new capturer with new source with different audio format.
   scoped_refptr<WebRtcAudioCapturer> new_capturer(
-      WebRtcAudioCapturer::CreateCapturer());
+      WebRtcAudioCapturer::CreateCapturer(-1, StreamDeviceInfo()));
   scoped_refptr<MockCapturerSource> new_source(
       new MockCapturerSource(new_capturer.get()));
-  EXPECT_CALL(*new_source.get(), OnInitialize(_, new_capturer.get(), 0));
-  new_capturer->SetCapturerSource(new_source,
-                                  media::CHANNEL_LAYOUT_MONO,
-                                  44100,
-                                  media::AudioParameters::NO_EFFECTS);
+  EXPECT_CALL(*new_source.get(), OnInitialize(_, new_capturer.get(), -1));
+  media::AudioParameters new_param(
+      media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
+      media::CHANNEL_LAYOUT_MONO, 44100, 16, 441);
+  new_capturer->SetCapturerSourceForTesting(new_source, new_param);
 
   // Setup the second audio track, connect it to the new capturer and start it.
   EXPECT_CALL(*new_source.get(), SetAutomaticGainControl(true));
   EXPECT_CALL(*new_source.get(), OnStart());
   scoped_refptr<WebRtcLocalAudioTrack> track_2 =
       WebRtcLocalAudioTrack::Create(std::string(), new_capturer, NULL, NULL,
           &constraints);
   static_cast<WebRtcLocalAudioSourceProvider*>(
       track_2->audio_source_provider())->SetSinkParamsForTesting(params_);
   track_2->Start();
 
   // Connect a number of network channels to the |track_2|.
   static const int kNumberOfNetworkChannelsForTrack2 = 3;
   for (int i = 0; i < kNumberOfNetworkChannelsForTrack2; ++i) {
     static_cast<webrtc::AudioTrackInterface*>(track_2.get())->
         GetRenderer()->AddChannel(i);
   }
   // Verify the data flow by connecting the |sink_2| to |track_2|.
   scoped_ptr<MockMediaStreamAudioSink> sink_2(new MockMediaStreamAudioSink());
   base::WaitableEvent event(false, false);
   EXPECT_CALL(
       *sink_2,
       CaptureData(
-          kNumberOfNetworkChannelsForTrack2, 44100, 1, _, 0, 0, false, false))
+          kNumberOfNetworkChannelsForTrack2, new_param.sample_rate(),
+          new_param.channels(), _, 0, 0, false, false))
       .Times(AnyNumber()).WillRepeatedly(Return());
   EXPECT_CALL(*sink_2, OnSetFormat(_)).WillOnce(SignalEvent(&event));
   track_2->AddSink(sink_2.get());
   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
 
   // Stopping the new source will stop the second track.
   event.Reset();
   EXPECT_CALL(*new_source.get(), OnStop())
       .Times(1).WillOnce(SignalEvent(&event));
   new_capturer->Stop();
   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
 
   // Stop the capturer of the first audio track.
   EXPECT_CALL(*capturer_source_.get(), OnStop());
   capturer_->Stop();
 }
 
 
 // Make sure a audio track can deliver packets with a buffer size smaller than
 // 10ms when it is not connected with a peer connection.
 TEST_F(WebRtcLocalAudioTrackTest, TrackWorkWithSmallBufferSize) {
   // Setup a capturer which works with a buffer size smaller than 10ms.
   media::AudioParameters params(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
                                 media::CHANNEL_LAYOUT_STEREO, 48000, 16, 128);
 
   // Create a capturer with new source which works with the format above.
   scoped_refptr<WebRtcAudioCapturer> capturer(
-      WebRtcAudioCapturer::CreateCapturer());
+      WebRtcAudioCapturer::CreateCapturer(
+          -1, StreamDeviceInfo(MEDIA_DEVICE_AUDIO_CAPTURE,
+                               "", "", params.sample_rate(),
+                               params.channel_layout(),
+                               params.frames_per_buffer())));
   scoped_refptr<MockCapturerSource> source(
       new MockCapturerSource(capturer.get()));
-  capturer->Initialize(-1, params.channel_layout(), params.sample_rate(),
-                       params.frames_per_buffer(), 0, std::string(), 0, 0,
-                       params.effects());
-
-  EXPECT_CALL(*source.get(), OnInitialize(_, capturer.get(), 0));
-  capturer->SetCapturerSource(source, params.channel_layout(),
-                              params.sample_rate(), params.effects());
+  EXPECT_CALL(*source.get(), OnInitialize(_, capturer.get(), -1));
+  capturer->SetCapturerSourceForTesting(source, params);
 
   // Setup a audio track, connect it to the capturer and start it.
   EXPECT_CALL(*source.get(), SetAutomaticGainControl(true));
   EXPECT_CALL(*source.get(), OnStart());
   RTCMediaConstraints constraints;
   scoped_refptr<WebRtcLocalAudioTrack> track =
       WebRtcLocalAudioTrack::Create(std::string(), capturer, NULL, NULL,
                                     &constraints);
   static_cast<WebRtcLocalAudioSourceProvider*>(
       track->audio_source_provider())->SetSinkParamsForTesting(params);
@@ skipping 15 matching lines @@
       .Times(AtLeast(1)).WillRepeatedly(SignalEvent(&event));
   track->AddSink(sink.get());
   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
 
   // Stopping the new source will stop the second track.
   EXPECT_CALL(*source, OnStop()).Times(1);
   capturer->Stop();
 }
 
 }  // namespace content