content/renderer/media/webrtc/webrtc_local_audio_track_adapter_unittest.cc

   1 // Copyright 2014 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 "content/renderer/media/mock_media_constraint_factory.h"
   6 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
   7 #include "content/renderer/media/webrtc_audio_capturer.h"
   8 #include "content/renderer/media/webrtc_local_audio_track.h"
   9 #include "testing/gmock/include/gmock/gmock.h"
  10 #include "testing/gtest/include/gtest/gtest.h"
  11 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
  12
  13 using ::testing::_;
  14 using ::testing::AnyNumber;
  15
  16 namespace content {
  17
  18 namespace {
  19
  20 class MockWebRtcAudioSink : public webrtc::AudioTrackSinkInterface {
  21  public:
  22   MockWebRtcAudioSink() {}
  23   ~MockWebRtcAudioSink() {}
  24   MOCK_METHOD5(OnData, void(const void* audio_data,
  25                             int bits_per_sample,
  26                             int sample_rate,
  27                             int number_of_channels,
  28                             int number_of_frames));
  29 };
  30
  31 }  // namespace
  32
  33 class WebRtcLocalAudioTrackAdapterTest : public ::testing::Test {
  34  public:
  35   WebRtcLocalAudioTrackAdapterTest()
  36       : params_(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
  37                 media::CHANNEL_LAYOUT_STEREO, 48000, 16, 480),
  38         adapter_(WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)) {
  39     MockMediaConstraintFactory constraint_factory;
  40     capturer_ = WebRtcAudioCapturer::CreateCapturer(
  41         -1, StreamDeviceInfo(MEDIA_DEVICE_AUDIO_CAPTURE, "", ""),
  42         constraint_factory.CreateWebMediaConstraints(), NULL, NULL);
  43     track_.reset(new WebRtcLocalAudioTrack(adapter_.get(), capturer_, NULL));
  44   }
  45
  46  protected:
  47   void SetUp() override {
  48     track_->OnSetFormat(params_);
  49     EXPECT_TRUE(track_->GetAudioAdapter()->enabled());
  50   }
  51
  52   media::AudioParameters params_;
  53   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_;
  54   scoped_refptr<WebRtcAudioCapturer> capturer_;
  55   scoped_ptr<WebRtcLocalAudioTrack> track_;
  56 };
  57
  58 // Adds and Removes a WebRtcAudioSink to a local audio track.
  59 TEST_F(WebRtcLocalAudioTrackAdapterTest, AddAndRemoveSink) {
  60   // Add a sink to the webrtc track.
  61   scoped_ptr<MockWebRtcAudioSink> sink(new MockWebRtcAudioSink());
  62   webrtc::AudioTrackInterface* webrtc_track =
  63       static_cast<webrtc::AudioTrackInterface*>(adapter_.get());
  64   webrtc_track->AddSink(sink.get());
  65
  66   // Send a packet via |track_| and the data should reach the sink of the
  67   // |adapter_|.
  68   const scoped_ptr<media::AudioBus> audio_bus =
  69       media::AudioBus::Create(params_);
  70   // While this test is not checking the signal data being passed around, the
  71   // implementation in WebRtcLocalAudioTrack reads the data for its signal level
  72   // computation.  Initialize all samples to zero to make the memory sanitizer
  73   // happy.
  74   audio_bus->Zero();
  75
  76   base::TimeTicks estimated_capture_time = base::TimeTicks::Now();
  77   EXPECT_CALL(*sink,
  78               OnData(_, 16, params_.sample_rate(), params_.channels(),
  79                      params_.frames_per_buffer()));
  80   track_->Capture(*audio_bus, estimated_capture_time, false);
  81
  82   // Remove the sink from the webrtc track.
  83   webrtc_track->RemoveSink(sink.get());
  84   sink.reset();
  85
  86   // Verify that no more callback gets into the sink.
  87   estimated_capture_time +=
  88       params_.frames_per_buffer() * base::TimeDelta::FromSeconds(1) /
  89           params_.sample_rate();
  90   track_->Capture(*audio_bus, estimated_capture_time, false);
  91 }
  92
  93 TEST_F(WebRtcLocalAudioTrackAdapterTest, GetSignalLevel) {
  94   webrtc::AudioTrackInterface* webrtc_track =
  95       static_cast<webrtc::AudioTrackInterface*>(adapter_.get());
  96   int signal_level = 0;
  97   EXPECT_TRUE(webrtc_track->GetSignalLevel(&signal_level));
  98 }
  99
 100 }  // namespace content