root/content/renderer/media/webrtc_local_audio_source_provider_unittest.cc

DEFINITIONS

1. SetUp
2. TEST_F
3. TEST_F
4. TEST_F

// 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 "base/strings/utf_string_conversions.h"
#include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
#include "content/renderer/media/webrtc_audio_capturer.h"
#include "content/renderer/media/webrtc_local_audio_source_provider.h"
#include "content/renderer/media/webrtc_local_audio_track.h"
#include "media/audio/audio_parameters.h"
#include "media/base/audio_bus.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "third_party/WebKit/public/platform/WebMediaConstraints.h"
#include "third_party/WebKit/public/platform/WebMediaStreamTrack.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_);
    blink::WebMediaConstraints constraints;
    scoped_refptr<WebRtcAudioCapturer> capturer(
        WebRtcAudioCapturer::CreateCapturer(-1, StreamDeviceInfo(),
                                            constraints, NULL));
    scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
        WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
    scoped_ptr<WebRtcLocalAudioTrack> native_track(
        new WebRtcLocalAudioTrack(adapter, capturer, NULL));
    blink::WebMediaStreamSource audio_source;
    audio_source.initialize(base::UTF8ToUTF16("dummy_source_id"),
                            blink::WebMediaStreamSource::TypeAudio,
                            base::UTF8ToUTF16("dummy_source_name"));
    blink_track_.initialize(blink::WebString::fromUTF8("audio_track"),
                            audio_source);
    blink_track_.setExtraData(native_track.release());
    source_provider_.reset(new WebRtcLocalAudioSourceProvider(blink_track_));
    source_provider_->SetSinkParamsForTesting(sink_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_;
  blink::WebMediaStreamTrack blink_track_;
  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_|.
  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_->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]);
  }
}

TEST_F(WebRtcLocalAudioSourceProviderTest,
       DeleteSourceProviderBeforeStoppingTrack) {
  source_provider_.reset();

  // Stop the audio track.
  WebRtcLocalAudioTrack* native_track = static_cast<WebRtcLocalAudioTrack*>(
      MediaStreamTrack::GetTrack(blink_track_));
  native_track->Stop();
}

TEST_F(WebRtcLocalAudioSourceProviderTest,
       StopTrackBeforeDeletingSourceProvider) {
  // Stop the audio track.
  WebRtcLocalAudioTrack* native_track = static_cast<WebRtcLocalAudioTrack*>(
      MediaStreamTrack::GetTrack(blink_track_));
  native_track->Stop();

  // Delete the source provider.
  source_provider_.reset();
}

}  // namespace content