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// Copyright 2013 The Chromium Authors. All rights reserved.
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.
#include "base/logging.h"
#include "content/renderer/media/rtc_media_constraints.h"
#include "content/renderer/media/webrtc_audio_capturer.h"
#include "content/renderer/media/webrtc_local_audio_track.h"
#include "media/audio/audio_parameters.h"
#include "media/base/audio_bus.h"
#include "testing/gmock/include/gmock/gmock.h"
#include "testing/gtest/include/gtest/gtest.h"
using ::testing::_;
using ::testing::AtLeast;
namespace content {
namespace {
// TODO(xians): Consolidate the similar methods in different unittests into
// one.
void ApplyFixedAudioConstraints(RTCMediaConstraints* constraints) {
// Constant constraint keys which enables default audio constraints on
// mediastreams with audio.
struct {
const char* key;
const char* value;
} static const kDefaultAudioConstraints[] = {
{ webrtc::MediaConstraintsInterface::kEchoCancellation,
webrtc::MediaConstraintsInterface::kValueTrue },
#if defined(OS_CHROMEOS) || defined(OS_MACOSX)
// Enable the extended filter mode AEC on platforms with known echo issues.
{ webrtc::MediaConstraintsInterface::kExperimentalEchoCancellation,
webrtc::MediaConstraintsInterface::kValueTrue },
#endif
{ webrtc::MediaConstraintsInterface::kAutoGainControl,
webrtc::MediaConstraintsInterface::kValueTrue },
{ webrtc::MediaConstraintsInterface::kExperimentalAutoGainControl,
webrtc::MediaConstraintsInterface::kValueTrue },
{ webrtc::MediaConstraintsInterface::kNoiseSuppression,
webrtc::MediaConstraintsInterface::kValueTrue },
{ webrtc::MediaConstraintsInterface::kHighpassFilter,
webrtc::MediaConstraintsInterface::kValueTrue },
};
for (size_t i = 0; i < ARRAYSIZE_UNSAFE(kDefaultAudioConstraints); ++i) {
constraints->AddMandatory(kDefaultAudioConstraints[i].key,
kDefaultAudioConstraints[i].value, false);
}
}
class MockCapturerSource : public media::AudioCapturerSource {
public:
MockCapturerSource() {}
MOCK_METHOD3(Initialize, void(const media::AudioParameters& params,
CaptureCallback* callback,
int session_id));
MOCK_METHOD0(Start, void());
MOCK_METHOD0(Stop, void());
MOCK_METHOD1(SetVolume, void(double volume));
MOCK_METHOD1(SetAutomaticGainControl, void(bool enable));
protected:
virtual ~MockCapturerSource() {}
};
class MockPeerConnectionAudioSink : public PeerConnectionAudioSink {
public:
MockPeerConnectionAudioSink() {}
~MockPeerConnectionAudioSink() {}
MOCK_METHOD9(OnData, int(const int16* audio_data,
int sample_rate,
int number_of_channels,
int number_of_frames,
const std::vector<int>& channels,
int audio_delay_milliseconds,
int current_volume,
bool need_audio_processing,
bool key_pressed));
MOCK_METHOD1(OnSetFormat, void(const media::AudioParameters& params));
};
} // namespace
class WebRtcAudioCapturerTest : public testing::Test {
protected:
WebRtcAudioCapturerTest()
#if defined(OS_ANDROID)
: params_(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
media::CHANNEL_LAYOUT_STEREO, 48000, 16, 960) {
// Android works with a buffer size bigger than 20ms.
#else
: params_(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
media::CHANNEL_LAYOUT_STEREO, 48000, 16, 128) {
#endif
capturer_ = WebRtcAudioCapturer::CreateCapturer();
capturer_->Initialize(-1, params_.channel_layout(), params_.sample_rate(),
params_.frames_per_buffer(), 0, std::string(), 0, 0,
params_.effects());
capturer_source_ = new MockCapturerSource();
EXPECT_CALL(*capturer_source_.get(), Initialize(_, capturer_.get(), 0));
capturer_->SetCapturerSource(capturer_source_,
params_.channel_layout(),
params_.sample_rate(),
params_.effects());
EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true));
EXPECT_CALL(*capturer_source_.get(), Start());
RTCMediaConstraints constraints;
ApplyFixedAudioConstraints(&constraints);
track_ = WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL,
NULL, &constraints);
static_cast<WebRtcLocalAudioSourceProvider*>(
track_->audio_source_provider())->SetSinkParamsForTesting(params_);
track_->Start();
EXPECT_TRUE(track_->enabled());
}
media::AudioParameters params_;
scoped_refptr<MockCapturerSource> capturer_source_;
scoped_refptr<WebRtcAudioCapturer> capturer_;
scoped_refptr<WebRtcLocalAudioTrack> track_;
};
// Pass the delay value, vollume and key_pressed info via capture callback, and
// those values should be correctly stored and passed to the track.
TEST_F(WebRtcAudioCapturerTest, VerifyAudioParams) {
// Connect a mock sink to the track.
scoped_ptr<MockPeerConnectionAudioSink> sink(
new MockPeerConnectionAudioSink());
track_->AddSink(sink.get());
int delay_ms = 65;
bool key_pressed = true;
double volume = 0.9;
// MaxVolume() in WebRtcAudioCapturer is hard-coded to return 255, we add 0.5
// to do the correct truncation as how the production code does.
int expected_volume_value = volume * capturer_->MaxVolume() + 0.5;
scoped_ptr<media::AudioBus> audio_bus = media::AudioBus::Create(params_);
audio_bus->Zero();
#if defined(OS_ANDROID)
const int expected_buffer_size = params_.sample_rate() / 100;
#else
const int expected_buffer_size = params_.frames_per_buffer();
#endif
bool expected_need_audio_processing = true;
media::AudioCapturerSource::CaptureCallback* callback =
static_cast<media::AudioCapturerSource::CaptureCallback*>(capturer_);
// Verify the sink is getting the correct values.
EXPECT_CALL(*sink, OnSetFormat(_));
EXPECT_CALL(*sink,
OnData(_, params_.sample_rate(), params_.channels(),
expected_buffer_size, _, delay_ms,
expected_volume_value, expected_need_audio_processing,
key_pressed)).Times(AtLeast(1));
callback->Capture(audio_bus.get(), delay_ms, volume, key_pressed);
// Verify the cached values in the capturer fits what we expect.
base::TimeDelta cached_delay;
int cached_volume = !expected_volume_value;
bool cached_key_pressed = !key_pressed;
capturer_->GetAudioProcessingParams(&cached_delay, &cached_volume,
&cached_key_pressed);
EXPECT_EQ(cached_delay.InMilliseconds(), delay_ms);
EXPECT_EQ(cached_volume, expected_volume_value);
EXPECT_EQ(cached_key_pressed, key_pressed);
track_->RemoveSink(sink.get());
EXPECT_CALL(*capturer_source_.get(), Stop());
capturer_->Stop();
}
} // namespace content
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