src/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 "base/command_line.h"
   6 #include "content/public/common/content_switches.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_local_audio_track.h"
  10 #include "testing/gmock/include/gmock/gmock.h"
  11 #include "testing/gtest/include/gtest/gtest.h"
  12 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
  13
  14 using ::testing::_;
  15 using ::testing::AnyNumber;
  16
  17 namespace content {
  18
  19 namespace {
  20
  21 class MockWebRtcAudioSink : public webrtc::AudioTrackSinkInterface {
  22  public:
  23   MockWebRtcAudioSink() {}
  24   ~MockWebRtcAudioSink() {}
  25   MOCK_METHOD5(OnData, void(const void* audio_data,
  26                             int bits_per_sample,
  27                             int sample_rate,
  28                             int number_of_channels,
  29                             int number_of_frames));
  30 };
  31
  32 }  // namespace
  33
  34 class WebRtcLocalAudioTrackAdapterTest : public ::testing::Test {
  35  public:
  36   WebRtcLocalAudioTrackAdapterTest()
  37       : params_(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
  38                 media::CHANNEL_LAYOUT_STEREO, 48000, 16, 480),
  39         adapter_(WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)) {
  40     MockMediaConstraintFactory constraint_factory;
  41     capturer_ = WebRtcAudioCapturer::CreateCapturer(
  42         -1, StreamDeviceInfo(MEDIA_DEVICE_AUDIO_CAPTURE, "", ""),
  43         constraint_factory.CreateWebMediaConstraints(), NULL, NULL);
  44     track_.reset(new WebRtcLocalAudioTrack(adapter_.get(), capturer_, NULL));
  45   }
  46
  47  protected:
  48   virtual void SetUp() OVERRIDE {
  49     track_->OnSetFormat(params_);
  50     EXPECT_TRUE(track_->GetAudioAdapter()->enabled());
  51   }
  52
  53   media::AudioParameters params_;
  54   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_;
  55   scoped_refptr<WebRtcAudioCapturer> capturer_;
  56   scoped_ptr<WebRtcLocalAudioTrack> track_;
  57 };
  58
  59 // Adds and Removes a WebRtcAudioSink to a local audio track.
  60 TEST_F(WebRtcLocalAudioTrackAdapterTest, AddAndRemoveSink) {
  61   // Add a sink to the webrtc track.
  62   scoped_ptr<MockWebRtcAudioSink> sink(new MockWebRtcAudioSink());
  63   webrtc::AudioTrackInterface* webrtc_track =
  64       static_cast<webrtc::AudioTrackInterface*>(adapter_.get());
  65   webrtc_track->AddSink(sink.get());
  66
  67   // Send a packet via |track_| and it data should reach the sink of the
  68   // |adapter_|.
  69   const int length = params_.frames_per_buffer() * params_.channels();
  70   scoped_ptr<int16[]> data(new int16[length]);
  71   // Initialize the data to 0 to avoid Memcheck:Uninitialized warning.
  72   memset(data.get(), 0, length * sizeof(data[0]));
  73
  74   EXPECT_CALL(*sink,
  75               OnData(_, 16, params_.sample_rate(), params_.channels(),
  76                      params_.frames_per_buffer()));
  77   track_->Capture(data.get(), base::TimeDelta(), 255, false, false);
  78
  79   // Remove the sink from the webrtc track.
  80   webrtc_track->RemoveSink(sink.get());
  81   sink.reset();
  82
  83   // Verify that no more callback gets into the sink.
  84   track_->Capture(data.get(), base::TimeDelta(), 255, false, false);
  85 }
  86
  87 TEST_F(WebRtcLocalAudioTrackAdapterTest, GetSignalLevel) {
  88   webrtc::AudioTrackInterface* webrtc_track =
  89       static_cast<webrtc::AudioTrackInterface*>(adapter_.get());
  90   int signal_level = 0;
  91   EXPECT_TRUE(webrtc_track->GetSignalLevel(&signal_level));
  92
  93   // Disable the audio processing in the audio track.
  94   CommandLine::ForCurrentProcess()->AppendSwitch(
  95       switches::kDisableAudioTrackProcessing);
  96   EXPECT_FALSE(webrtc_track->GetSignalLevel(&signal_level));
  97 }
  98
  99 }  // namespace content