Add generic interfaces for the sinks of the media stream audio track

content/renderer/media/webrtc_local_audio_source_provider_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/webrtc_local_audio_source_provider.h"
 #include "media/audio/audio_parameters.h"
 #include "media/base/audio_bus.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
 namespace content {
 
 class WebRtcLocalAudioSourceProviderTest : public testing::Test {
  protected:
   virtual void SetUp() OVERRIDE {
     source_params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
                          media::CHANNEL_LAYOUT_MONO, 1, 0, 48000, 16, 480);
     sink_params_.Reset(
         media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
         media::CHANNEL_LAYOUT_STEREO, 2, 0, 44100, 16,
         WebRtcLocalAudioSourceProvider::kWebAudioRenderBufferSize);
     const int length =
         source_params_.frames_per_buffer() * source_params_.channels();
     source_data_.reset(new int16[length]);
     sink_bus_ = media::AudioBus::Create(sink_params_);
     source_provider_.reset(new WebRtcLocalAudioSourceProvider());
     source_provider_->SetSinkParamsForTesting(sink_params_);
-    source_provider_->SetCaptureFormat(source_params_);
+    source_provider_->OnSetFormat(source_params_);
   }
 
   media::AudioParameters source_params_;
   scoped_ptr<int16[]> source_data_;
   media::AudioParameters sink_params_;
   scoped_ptr<media::AudioBus> sink_bus_;
   scoped_ptr<WebRtcLocalAudioSourceProvider> source_provider_;
 };
 
 TEST_F(WebRtcLocalAudioSourceProviderTest, VerifyDataFlow) {
   // Point the WebVector into memory owned by |sink_bus_|.
   blink::WebVector<float*> audio_data(
       static_cast<size_t>(sink_bus_->channels()));
   for (size_t i = 0; i < audio_data.size(); ++i)
     audio_data[i] = sink_bus_->channel(i);
 
   // Enable the |source_provider_| by asking for data. This will inject
   // source_params_.frames_per_buffer() of zero into the resampler since there
   // no available data in the FIFO.
   source_provider_->provideInput(audio_data, sink_params_.frames_per_buffer());
   EXPECT_TRUE(sink_bus_->channel(0)[0] == 0);
 
   // Set the value of source data to be 1.
   const int length =
       source_params_.frames_per_buffer() * source_params_.channels();
   std::fill(source_data_.get(), source_data_.get() + length, 1);
 
   // Deliver data to |source_provider_|.
-  std::vector<int> voe_channels;
-  source_provider_->CaptureData(voe_channels,
-                                source_data_.get(),
-                                source_params_.sample_rate(),
-                                source_params_.channels(),
-                                source_params_.frames_per_buffer(),
-                                0, 0, false, false);
+  source_provider_->OnData(source_data_.get(),
+                           source_params_.sample_rate(),
+                           source_params_.channels(),
+                           source_params_.frames_per_buffer());
 
   // Consume the first packet in the resampler, which contains only zero.
   // And the consumption of the data will trigger pulling the real packet from
   // the source provider FIFO into the resampler.
   // Note that we need to count in the provideInput() call a few lines above.
   for (int i = sink_params_.frames_per_buffer();
        i < source_params_.frames_per_buffer();
        i += sink_params_.frames_per_buffer()) {
     sink_bus_->Zero();
     source_provider_->provideInput(audio_data,
                                    sink_params_.frames_per_buffer());
     EXPECT_DOUBLE_EQ(0.0, sink_bus_->channel(0)[0]);
     EXPECT_DOUBLE_EQ(0.0, sink_bus_->channel(1)[0]);
   }
 
   // Prepare the second packet for featching.
-  source_provider_->CaptureData(voe_channels,
-                                source_data_.get(),
-                                source_params_.sample_rate(),
-                                source_params_.channels(),
-                                source_params_.frames_per_buffer(),
-                                0, 0, false, false);
+  source_provider_->OnData(source_data_.get(),
+                           source_params_.sample_rate(),
+                           source_params_.channels(),
+                           source_params_.frames_per_buffer());
 
   // Verify the packets.
   for (int i = 0; i < source_params_.frames_per_buffer();
        i += sink_params_.frames_per_buffer()) {
     sink_bus_->Zero();
     source_provider_->provideInput(audio_data,
                                    sink_params_.frames_per_buffer());
     EXPECT_GT(sink_bus_->channel(0)[0], 0);
     EXPECT_GT(sink_bus_->channel(1)[0], 0);
     EXPECT_DOUBLE_EQ(sink_bus_->channel(0)[0], sink_bus_->channel(1)[0]);
   }
 }
 
 }  // namespace content