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 #include "third_party/WebKit/public/web/WebHeap.h"
  17
  18 namespace content {
  19
  20 class WebRtcLocalAudioSourceProviderTest : public testing::Test {
  21  protected:
  22   virtual void SetUp() override {
  23     source_params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
  24                          media::CHANNEL_LAYOUT_MONO, 1, 48000, 16, 480);
  25     sink_params_.Reset(
  26         media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
  27         media::CHANNEL_LAYOUT_STEREO, 2, 44100, 16,
  28         WebRtcLocalAudioSourceProvider::kWebAudioRenderBufferSize);
  29     const int length =
  30         source_params_.frames_per_buffer() * source_params_.channels();
  31     source_data_.reset(new int16[length]);
  32     sink_bus_ = media::AudioBus::Create(sink_params_);
  33     MockMediaConstraintFactory constraint_factory;
  34       scoped_refptr<WebRtcAudioCapturer> capturer(
  35         WebRtcAudioCapturer::CreateCapturer(
  36             -1, StreamDeviceInfo(),
  37             constraint_factory.CreateWebMediaConstraints(), NULL, NULL));
  38     scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
  39         WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
  40     scoped_ptr<WebRtcLocalAudioTrack> native_track(
  41         new WebRtcLocalAudioTrack(adapter.get(), capturer, NULL));
  42     blink::WebMediaStreamSource audio_source;
  43     audio_source.initialize(base::UTF8ToUTF16("dummy_source_id"),
  44                             blink::WebMediaStreamSource::TypeAudio,
  45                             base::UTF8ToUTF16("dummy_source_name"));
  46     blink_track_.initialize(blink::WebString::fromUTF8("audio_track"),
  47                             audio_source);
  48     blink_track_.setExtraData(native_track.release());
  49     source_provider_.reset(new WebRtcLocalAudioSourceProvider(blink_track_));
  50     source_provider_->SetSinkParamsForTesting(sink_params_);
  51     source_provider_->OnSetFormat(source_params_);
  52   }
  53
  54   virtual void TearDown() override {
  55     source_provider_.reset();
  56     blink_track_.reset();
  57     blink::WebHeap::collectAllGarbageForTesting();
  58   }
  59
  60   media::AudioParameters source_params_;
  61   scoped_ptr<int16[]> source_data_;
  62   media::AudioParameters sink_params_;
  63   scoped_ptr<media::AudioBus> sink_bus_;
  64   blink::WebMediaStreamTrack blink_track_;
  65   scoped_ptr<WebRtcLocalAudioSourceProvider> source_provider_;
  66 };
  67
  68 TEST_F(WebRtcLocalAudioSourceProviderTest, VerifyDataFlow) {
  69   // Point the WebVector into memory owned by |sink_bus_|.
  70   blink::WebVector<float*> audio_data(
  71       static_cast<size_t>(sink_bus_->channels()));
  72   for (size_t i = 0; i < audio_data.size(); ++i)
  73     audio_data[i] = sink_bus_->channel(i);
  74
  75   // Enable the |source_provider_| by asking for data. This will inject
  76   // source_params_.frames_per_buffer() of zero into the resampler since there
  77   // no available data in the FIFO.
  78   source_provider_->provideInput(audio_data, sink_params_.frames_per_buffer());
  79   EXPECT_TRUE(sink_bus_->channel(0)[0] == 0);
  80
  81   // Set the value of source data to be 1.
  82   const int length =
  83       source_params_.frames_per_buffer() * source_params_.channels();
  84   std::fill(source_data_.get(), source_data_.get() + length, 1);
  85
  86   // Deliver data to |source_provider_|.
  87   source_provider_->OnData(source_data_.get(),
  88                            source_params_.sample_rate(),
  89                            source_params_.channels(),
  90                            source_params_.frames_per_buffer());
  91
  92   // Consume the first packet in the resampler, which contains only zero.
  93   // And the consumption of the data will trigger pulling the real packet from
  94   // the source provider FIFO into the resampler.
  95   // Note that we need to count in the provideInput() call a few lines above.
  96   for (int i = sink_params_.frames_per_buffer();
  97        i < source_params_.frames_per_buffer();
  98        i += sink_params_.frames_per_buffer()) {
  99     sink_bus_->Zero();
 100     source_provider_->provideInput(audio_data,
 101                                    sink_params_.frames_per_buffer());
 102     EXPECT_DOUBLE_EQ(0.0, sink_bus_->channel(0)[0]);
 103     EXPECT_DOUBLE_EQ(0.0, sink_bus_->channel(1)[0]);
 104   }
 105
 106   // Prepare the second packet for featching.
 107   source_provider_->OnData(source_data_.get(),
 108                            source_params_.sample_rate(),
 109                            source_params_.channels(),
 110                            source_params_.frames_per_buffer());
 111
 112   // Verify the packets.
 113   for (int i = 0; i < source_params_.frames_per_buffer();
 114        i += sink_params_.frames_per_buffer()) {
 115     sink_bus_->Zero();
 116     source_provider_->provideInput(audio_data,
 117                                    sink_params_.frames_per_buffer());
 118     EXPECT_GT(sink_bus_->channel(0)[0], 0);
 119     EXPECT_GT(sink_bus_->channel(1)[0], 0);
 120     EXPECT_DOUBLE_EQ(sink_bus_->channel(0)[0], sink_bus_->channel(1)[0]);
 121   }
 122 }
 123
 124 TEST_F(WebRtcLocalAudioSourceProviderTest,
 125        DeleteSourceProviderBeforeStoppingTrack) {
 126   source_provider_.reset();
 127
 128   // Stop the audio track.
 129   WebRtcLocalAudioTrack* native_track = static_cast<WebRtcLocalAudioTrack*>(
 130       MediaStreamTrack::GetTrack(blink_track_));
 131   native_track->Stop();
 132 }
 133
 134 TEST_F(WebRtcLocalAudioSourceProviderTest,
 135        StopTrackBeforeDeletingSourceProvider) {
 136   // Stop the audio track.
 137   WebRtcLocalAudioTrack* native_track = static_cast<WebRtcLocalAudioTrack*>(
 138       MediaStreamTrack::GetTrack(blink_track_));
 139   native_track->Stop();
 140
 141   // Delete the source provider.
 142   source_provider_.reset();
 143 }
 144
 145 }  // namespace content