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   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     sink_bus_ = media::AudioBus::Create(sink_params_);
  30     MockMediaConstraintFactory constraint_factory;
  31     scoped_refptr<WebRtcAudioCapturer> capturer(
  32         WebRtcAudioCapturer::CreateCapturer(
  33             -1, StreamDeviceInfo(),
  34             constraint_factory.CreateWebMediaConstraints(), NULL, NULL));
  35     scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
  36         WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
  37     scoped_ptr<WebRtcLocalAudioTrack> native_track(
  38         new WebRtcLocalAudioTrack(adapter.get(), capturer, NULL));
  39     blink::WebMediaStreamSource audio_source;
  40     audio_source.initialize(base::UTF8ToUTF16("dummy_source_id"),
  41                             blink::WebMediaStreamSource::TypeAudio,
  42                             base::UTF8ToUTF16("dummy_source_name"),
  43                             false /* remote */, true /* readonly */);
  44     blink_track_.initialize(blink::WebString::fromUTF8("audio_track"),
  45                             audio_source);
  46     blink_track_.setExtraData(native_track.release());
  47     source_provider_.reset(new WebRtcLocalAudioSourceProvider(blink_track_));
  48     source_provider_->SetSinkParamsForTesting(sink_params_);
  49     source_provider_->OnSetFormat(source_params_);
  50   }
  51
  52   void TearDown() override {
  53     source_provider_.reset();
  54     blink_track_.reset();
  55     blink::WebHeap::collectAllGarbageForTesting();
  56   }
  57
  58   media::AudioParameters source_params_;
  59   media::AudioParameters sink_params_;
  60   scoped_ptr<media::AudioBus> sink_bus_;
  61   blink::WebMediaStreamTrack blink_track_;
  62   scoped_ptr<WebRtcLocalAudioSourceProvider> source_provider_;
  63 };
  64
  65 TEST_F(WebRtcLocalAudioSourceProviderTest, VerifyDataFlow) {
  66   // Point the WebVector into memory owned by |sink_bus_|.
  67   blink::WebVector<float*> audio_data(
  68       static_cast<size_t>(sink_bus_->channels()));
  69   for (size_t i = 0; i < audio_data.size(); ++i)
  70     audio_data[i] = sink_bus_->channel(i);
  71
  72   // Enable the |source_provider_| by asking for data. This will inject
  73   // source_params_.frames_per_buffer() of zero into the resampler since there
  74   // no available data in the FIFO.
  75   source_provider_->provideInput(audio_data, sink_params_.frames_per_buffer());
  76   EXPECT_TRUE(sink_bus_->channel(0)[0] == 0);
  77
  78   // Create a source AudioBus with channel data filled with non-zero values.
  79   const scoped_ptr<media::AudioBus> source_bus =
  80       media::AudioBus::Create(source_params_);
  81   std::fill(source_bus->channel(0),
  82             source_bus->channel(0) + source_bus->frames(),
  83             0.5f);
  84
  85   // Deliver data to |source_provider_|.
  86   base::TimeTicks estimated_capture_time = base::TimeTicks::Now();
  87   source_provider_->OnData(*source_bus, estimated_capture_time);
  88
  89   // Consume the first packet in the resampler, which contains only zeros.
  90   // And the consumption of the data will trigger pulling the real packet from
  91   // the source provider FIFO into the resampler.
  92   // Note that we need to count in the provideInput() call a few lines above.
  93   for (int i = sink_params_.frames_per_buffer();
  94        i < source_params_.frames_per_buffer();
  95        i += sink_params_.frames_per_buffer()) {
  96     sink_bus_->Zero();
  97     source_provider_->provideInput(audio_data,
  98                                    sink_params_.frames_per_buffer());
  99     EXPECT_DOUBLE_EQ(0.0, sink_bus_->channel(0)[0]);
 100     EXPECT_DOUBLE_EQ(0.0, sink_bus_->channel(1)[0]);
 101   }
 102
 103   // Make a second data delivery.
 104   estimated_capture_time +=
 105       source_bus->frames() * base::TimeDelta::FromSeconds(1) /
 106       source_params_.sample_rate();
 107   source_provider_->OnData(*source_bus, estimated_capture_time);
 108
 109   // Verify that non-zero data samples are present in the results of the
 110   // following calls to provideInput().
 111   for (int i = 0; i < source_params_.frames_per_buffer();
 112        i += sink_params_.frames_per_buffer()) {
 113     sink_bus_->Zero();
 114     source_provider_->provideInput(audio_data,
 115                                    sink_params_.frames_per_buffer());
 116     EXPECT_NEAR(0.5f, sink_bus_->channel(0)[0], 0.001f);
 117     EXPECT_NEAR(0.5f, sink_bus_->channel(1)[0], 0.001f);
 118     EXPECT_DOUBLE_EQ(sink_bus_->channel(0)[0], sink_bus_->channel(1)[0]);
 119   }
 120 }
 121
 122 TEST_F(WebRtcLocalAudioSourceProviderTest,
 123        DeleteSourceProviderBeforeStoppingTrack) {
 124   source_provider_.reset();
 125
 126   // Stop the audio track.
 127   WebRtcLocalAudioTrack* native_track = static_cast<WebRtcLocalAudioTrack*>(
 128       MediaStreamTrack::GetTrack(blink_track_));
 129   native_track->Stop();
 130 }
 131
 132 TEST_F(WebRtcLocalAudioSourceProviderTest,
 133        StopTrackBeforeDeletingSourceProvider) {
 134   // Stop the audio track.
 135   WebRtcLocalAudioTrack* native_track = static_cast<WebRtcLocalAudioTrack*>(
 136       MediaStreamTrack::GetTrack(blink_track_));
 137   native_track->Stop();
 138
 139   // Delete the source provider.
 140   source_provider_.reset();
 141 }
 142
 143 }  // namespace content