content/renderer/media/webrtc_local_audio_source_provider_unittest.cc

   1 // Copyright 2013 The Chromium Authors. All rights reserved.
   2 // Use of this source code is governed by a BSD-style license that can be
   3 // found in the LICENSE file.
   4
   5 #include "base/logging.h"
   6 #include "base/strings/utf_string_conversions.h"
   7 #include "content/renderer/media/mock_media_constraint_factory.h"
   8 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
   9 #include "content/renderer/media/webrtc_audio_capturer.h"
  10 #include "content/renderer/media/webrtc_local_audio_source_provider.h"
  11 #include "content/renderer/media/webrtc_local_audio_track.h"
  12 #include "media/audio/audio_parameters.h"
  13 #include "media/base/audio_bus.h"
  14 #include "testing/gtest/include/gtest/gtest.h"
  15 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h"
  16
  17 namespace content {
  18
  19 class WebRtcLocalAudioSourceProviderTest : public testing::Test {
  20  protected:
  21   virtual void SetUp() OVERRIDE {
  22     source_params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
  23                          media::CHANNEL_LAYOUT_MONO, 1, 0, 48000, 16, 480);
  24     sink_params_.Reset(
  25         media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
  26         media::CHANNEL_LAYOUT_STEREO, 2, 0, 44100, 16,
  27         WebRtcLocalAudioSourceProvider::kWebAudioRenderBufferSize);
  28     const int length =
  29         source_params_.frames_per_buffer() * source_params_.channels();
  30     source_data_.reset(new int16[length]);
  31     sink_bus_ = media::AudioBus::Create(sink_params_);
  32     MockMediaConstraintFactory constraint_factory;
  33       scoped_refptr<WebRtcAudioCapturer> capturer(
  34         WebRtcAudioCapturer::CreateCapturer(
  35             -1, StreamDeviceInfo(),
  36             constraint_factory.CreateWebMediaConstraints(), NULL, NULL));
  37     scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
  38         WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
  39     scoped_ptr<WebRtcLocalAudioTrack> native_track(
  40         new WebRtcLocalAudioTrack(adapter, capturer, NULL));
  41     blink::WebMediaStreamSource audio_source;
  42     audio_source.initialize(base::UTF8ToUTF16("dummy_source_id"),
  43                             blink::WebMediaStreamSource::TypeAudio,
  44                             base::UTF8ToUTF16("dummy_source_name"));
  45     blink_track_.initialize(blink::WebString::fromUTF8("audio_track"),
  46                             audio_source);
  47     blink_track_.setExtraData(native_track.release());
  48     source_provider_.reset(new WebRtcLocalAudioSourceProvider(blink_track_));
  49     source_provider_->SetSinkParamsForTesting(sink_params_);
  50     source_provider_->OnSetFormat(source_params_);
  51   }
  52
  53   media::AudioParameters source_params_;
  54   scoped_ptr<int16[]> source_data_;
  55   media::AudioParameters sink_params_;
  56   scoped_ptr<media::AudioBus> sink_bus_;
  57   blink::WebMediaStreamTrack blink_track_;
  58   scoped_ptr<WebRtcLocalAudioSourceProvider> source_provider_;
  59 };
  60
  61 TEST_F(WebRtcLocalAudioSourceProviderTest, VerifyDataFlow) {
  62   // Point the WebVector into memory owned by |sink_bus_|.
  63   blink::WebVector<float*> audio_data(
  64       static_cast<size_t>(sink_bus_->channels()));
  65   for (size_t i = 0; i < audio_data.size(); ++i)
  66     audio_data[i] = sink_bus_->channel(i);
  67
  68   // Enable the |source_provider_| by asking for data. This will inject
  69   // source_params_.frames_per_buffer() of zero into the resampler since there
  70   // no available data in the FIFO.
  71   source_provider_->provideInput(audio_data, sink_params_.frames_per_buffer());
  72   EXPECT_TRUE(sink_bus_->channel(0)[0] == 0);
  73
  74   // Set the value of source data to be 1.
  75   const int length =
  76       source_params_.frames_per_buffer() * source_params_.channels();
  77   std::fill(source_data_.get(), source_data_.get() + length, 1);
  78
  79   // Deliver data to |source_provider_|.
  80   source_provider_->OnData(source_data_.get(),
  81                            source_params_.sample_rate(),
  82                            source_params_.channels(),
  83                            source_params_.frames_per_buffer());
  84
  85   // Consume the first packet in the resampler, which contains only zero.
  86   // And the consumption of the data will trigger pulling the real packet from
  87   // the source provider FIFO into the resampler.
  88   // Note that we need to count in the provideInput() call a few lines above.
  89   for (int i = sink_params_.frames_per_buffer();
  90        i < source_params_.frames_per_buffer();
  91        i += sink_params_.frames_per_buffer()) {
  92     sink_bus_->Zero();
  93     source_provider_->provideInput(audio_data,
  94                                    sink_params_.frames_per_buffer());
  95     EXPECT_DOUBLE_EQ(0.0, sink_bus_->channel(0)[0]);
  96     EXPECT_DOUBLE_EQ(0.0, sink_bus_->channel(1)[0]);
  97   }
  98
  99   // Prepare the second packet for featching.
 100   source_provider_->OnData(source_data_.get(),
 101                            source_params_.sample_rate(),
 102                            source_params_.channels(),
 103                            source_params_.frames_per_buffer());
 104
 105   // Verify the packets.
 106   for (int i = 0; i < source_params_.frames_per_buffer();
 107        i += sink_params_.frames_per_buffer()) {
 108     sink_bus_->Zero();
 109     source_provider_->provideInput(audio_data,
 110                                    sink_params_.frames_per_buffer());
 111     EXPECT_GT(sink_bus_->channel(0)[0], 0);
 112     EXPECT_GT(sink_bus_->channel(1)[0], 0);
 113     EXPECT_DOUBLE_EQ(sink_bus_->channel(0)[0], sink_bus_->channel(1)[0]);
 114   }
 115 }
 116
 117 TEST_F(WebRtcLocalAudioSourceProviderTest,
 118        DeleteSourceProviderBeforeStoppingTrack) {
 119   source_provider_.reset();
 120
 121   // Stop the audio track.
 122   WebRtcLocalAudioTrack* native_track = static_cast<WebRtcLocalAudioTrack*>(
 123       MediaStreamTrack::GetTrack(blink_track_));
 124   native_track->Stop();
 125 }
 126
 127 TEST_F(WebRtcLocalAudioSourceProviderTest,
 128        StopTrackBeforeDeletingSourceProvider) {
 129   // Stop the audio track.
 130   WebRtcLocalAudioTrack* native_track = static_cast<WebRtcLocalAudioTrack*>(
 131       MediaStreamTrack::GetTrack(blink_track_));
 132   native_track->Stop();
 133
 134   // Delete the source provider.
 135   source_provider_.reset();
 136 }
 137
 138 }  // namespace content