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