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/webrtc/webrtc_local_audio_track_adapter.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         capturer_(WebRtcAudioCapturer::CreateCapturer(
  40             -1, StreamDeviceInfo(MEDIA_DEVICE_AUDIO_CAPTURE, "", ""),
  41             blink::WebMediaConstraints(), NULL, NULL)),
  42         track_(new WebRtcLocalAudioTrack(adapter_, capturer_, NULL)) {}
  43
  44  protected:
  45   virtual void SetUp() OVERRIDE {
  46     track_->OnSetFormat(params_);
  47     EXPECT_TRUE(track_->GetAudioAdapter()->enabled());
  48   }
  49
  50   media::AudioParameters params_;
  51   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_;
  52   scoped_refptr<WebRtcAudioCapturer> capturer_;
  53   scoped_ptr<WebRtcLocalAudioTrack> track_;
  54 };
  55
  56 // Adds and Removes a WebRtcAudioSink to a local audio track.
  57 TEST_F(WebRtcLocalAudioTrackAdapterTest, AddAndRemoveSink) {
  58   // Add a sink to the webrtc track.
  59   scoped_ptr<MockWebRtcAudioSink> sink(new MockWebRtcAudioSink());
  60   webrtc::AudioTrackInterface* webrtc_track =
  61       static_cast<webrtc::AudioTrackInterface*>(adapter_.get());
  62   webrtc_track->AddSink(sink.get());
  63
  64   // Send a packet via |track_| and it data should reach the sink of the
  65   // |adapter_|.
  66   const int length = params_.frames_per_buffer() * params_.channels();
  67   scoped_ptr<int16[]> data(new int16[length]);
  68   // Initialize the data to 0 to avoid Memcheck:Uninitialized warning.
  69   memset(data.get(), 0, length * sizeof(data[0]));
  70
  71   EXPECT_CALL(*sink,
  72               OnData(_, 16, params_.sample_rate(), params_.channels(),
  73                      params_.frames_per_buffer()));
  74   track_->Capture(data.get(), base::TimeDelta(), 255, false, false);
  75
  76   // Remove the sink from the webrtc track.
  77   webrtc_track->RemoveSink(sink.get());
  78   sink.reset();
  79
  80   // Verify that no more callback gets into the sink.
  81   track_->Capture(data.get(), base::TimeDelta(), 255, false, false);
  82 }
  83
  84 TEST_F(WebRtcLocalAudioTrackAdapterTest, GetSignalLevel) {
  85   webrtc::AudioTrackInterface* webrtc_track =
  86       static_cast<webrtc::AudioTrackInterface*>(adapter_.get());
  87   int signal_level = 0;
  88   EXPECT_FALSE(webrtc_track->GetSignalLevel(&signal_level));
  89
  90   // Enable the audio processing in the audio track.
  91   CommandLine::ForCurrentProcess()->AppendSwitch(
  92       switches::kEnableAudioTrackProcessing);
  93   EXPECT_TRUE(webrtc_track->GetSignalLevel(&signal_level));
  94 }
  95
  96 }  // namespace content