audio/audio_state_unittest.cc - src - Git at Google

 /*
  *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */

 #include <memory>
 #include <vector>

 #include "audio/audio_state.h"
 #include "call/test/mock_audio_send_stream.h"
 #include "modules/audio_device/include/mock_audio_device.h"
 #include "modules/audio_mixer/audio_mixer_impl.h"
 #include "modules/audio_processing/include/mock_audio_processing.h"
 #include "rtc_base/ref_counted_object.h"
 #include "test/gtest.h"

 namespace webrtc {
 namespace test {
 namespace {

 constexpr int kSampleRate = 16000;
 constexpr int kNumberOfChannels = 1;

 struct ConfigHelper {
   ConfigHelper() : audio_mixer(AudioMixerImpl::Create()) {
     audio_state_config.audio_mixer = audio_mixer;
     audio_state_config.audio_processing =
         new rtc::RefCountedObject<testing::NiceMock<MockAudioProcessing>>();
     audio_state_config.audio_device_module =
         new rtc::RefCountedObject<MockAudioDeviceModule>();
   }
   AudioState::Config& config() { return audio_state_config; }
   rtc::scoped_refptr<AudioMixer> mixer() { return audio_mixer; }

  private:
   AudioState::Config audio_state_config;
   rtc::scoped_refptr<AudioMixer> audio_mixer;
 };

 class FakeAudioSource : public AudioMixer::Source {
  public:
   // TODO(aleloi): Valid overrides commented out, because the gmock
   // methods don't use any override declarations, and we want to avoid
   // warnings from -Winconsistent-missing-override. See
   // http://crbug.com/428099.
   int Ssrc() const /*override*/ { return 0; }

   int PreferredSampleRate() const /*override*/ { return kSampleRate; }

   MOCK_METHOD2(GetAudioFrameWithInfo,
                AudioFrameInfo(int sample_rate_hz, AudioFrame* audio_frame));
 };

 std::vector<int16_t> Create10msSilentTestData(int sample_rate_hz,
                                               size_t num_channels) {
   const int samples_per_channel = sample_rate_hz / 100;
   std::vector<int16_t> audio_data(samples_per_channel * num_channels, 0);
   return audio_data;
 }

 std::vector<int16_t> Create10msTestData(int sample_rate_hz,
                                         size_t num_channels) {
   const int samples_per_channel = sample_rate_hz / 100;
   std::vector<int16_t> audio_data(samples_per_channel * num_channels, 0);
   // Fill the first channel with a 1kHz sine wave.
   const float inc = (2 * 3.14159265f * 1000) / sample_rate_hz;
   float w = 0.f;
   for (int i = 0; i < samples_per_channel; ++i) {
     audio_data[i * num_channels] = static_cast<int16_t>(32767.f * std::sin(w));
     w += inc;
   }
   return audio_data;
 }

 std::vector<uint32_t> ComputeChannelLevels(AudioFrame* audio_frame) {
   const size_t num_channels = audio_frame->num_channels_;
   const size_t samples_per_channel = audio_frame->samples_per_channel_;
   std::vector<uint32_t> levels(num_channels, 0);
   for (size_t i = 0; i < samples_per_channel; ++i) {
     for (size_t j = 0; j < num_channels; ++j) {
       levels[j] += std::abs(audio_frame->data()[i * num_channels + j]);
     }
   }
   return levels;
 }
 }  // namespace

 TEST(AudioStateTest, Create) {
   ConfigHelper helper;
   auto audio_state = AudioState::Create(helper.config());
   EXPECT_TRUE(audio_state.get());
 }

 TEST(AudioStateTest, ConstructDestruct) {
   ConfigHelper helper;
   rtc::scoped_refptr<internal::AudioState> audio_state(
       new rtc::RefCountedObject<internal::AudioState>(helper.config()));
 }

 TEST(AudioStateTest, RecordedAudioArrivesAtSingleStream) {
   ConfigHelper helper;
   rtc::scoped_refptr<internal::AudioState> audio_state(
       new rtc::RefCountedObject<internal::AudioState>(helper.config()));

   MockAudioSendStream stream;
   audio_state->AddSendingStream(&stream, 8000, 2);

   EXPECT_CALL(
       stream,
       SendAudioDataForMock(testing::AllOf(
           testing::Field(&AudioFrame::sample_rate_hz_, testing::Eq(8000)),
           testing::Field(&AudioFrame::num_channels_, testing::Eq(2u)))))
       .WillOnce(
           // Verify that channels are not swapped by default.
           testing::Invoke([](AudioFrame* audio_frame) {
             auto levels = ComputeChannelLevels(audio_frame);
             EXPECT_LT(0u, levels[0]);
             EXPECT_EQ(0u, levels[1]);
           }));
   MockAudioProcessing* ap =
       static_cast<MockAudioProcessing*>(audio_state->audio_processing());
   EXPECT_CALL(*ap, set_stream_delay_ms(0));
   EXPECT_CALL(*ap, set_stream_key_pressed(false));
   EXPECT_CALL(*ap, ProcessStream(testing::_));

   constexpr int kSampleRate = 16000;
   constexpr size_t kNumChannels = 2;
   auto audio_data = Create10msTestData(kSampleRate, kNumChannels);
   uint32_t new_mic_level = 667;
   audio_state->audio_transport()->RecordedDataIsAvailable(
       &audio_data[0], kSampleRate / 100, kNumChannels * 2, kNumChannels,
       kSampleRate, 0, 0, 0, false, new_mic_level);
   EXPECT_EQ(667u, new_mic_level);

   audio_state->RemoveSendingStream(&stream);
 }

 TEST(AudioStateTest, RecordedAudioArrivesAtMultipleStreams) {
   ConfigHelper helper;
   rtc::scoped_refptr<internal::AudioState> audio_state(
       new rtc::RefCountedObject<internal::AudioState>(helper.config()));

   MockAudioSendStream stream_1;
   MockAudioSendStream stream_2;
   audio_state->AddSendingStream(&stream_1, 8001, 2);
   audio_state->AddSendingStream(&stream_2, 32000, 1);

   EXPECT_CALL(
       stream_1,
       SendAudioDataForMock(testing::AllOf(
           testing::Field(&AudioFrame::sample_rate_hz_, testing::Eq(16000)),
           testing::Field(&AudioFrame::num_channels_, testing::Eq(1u)))))
       .WillOnce(
           // Verify that there is output signal.
           testing::Invoke([](AudioFrame* audio_frame) {
             auto levels = ComputeChannelLevels(audio_frame);
             EXPECT_LT(0u, levels[0]);
           }));
   EXPECT_CALL(
       stream_2,
       SendAudioDataForMock(testing::AllOf(
           testing::Field(&AudioFrame::sample_rate_hz_, testing::Eq(16000)),
           testing::Field(&AudioFrame::num_channels_, testing::Eq(1u)))))
       .WillOnce(
           // Verify that there is output signal.
           testing::Invoke([](AudioFrame* audio_frame) {
             auto levels = ComputeChannelLevels(audio_frame);
             EXPECT_LT(0u, levels[0]);
           }));
   MockAudioProcessing* ap =
       static_cast<MockAudioProcessing*>(audio_state->audio_processing());
   EXPECT_CALL(*ap, set_stream_delay_ms(5));
   EXPECT_CALL(*ap, set_stream_key_pressed(true));
   EXPECT_CALL(*ap, ProcessStream(testing::_));

   constexpr int kSampleRate = 16000;
   constexpr size_t kNumChannels = 1;
   auto audio_data = Create10msTestData(kSampleRate, kNumChannels);
   uint32_t new_mic_level = 667;
   audio_state->audio_transport()->RecordedDataIsAvailable(
       &audio_data[0], kSampleRate / 100, kNumChannels * 2, kNumChannels,
       kSampleRate, 5, 0, 0, true, new_mic_level);
   EXPECT_EQ(667u, new_mic_level);

   audio_state->RemoveSendingStream(&stream_1);
   audio_state->RemoveSendingStream(&stream_2);
 }

 TEST(AudioStateTest, EnableChannelSwap) {
   constexpr int kSampleRate = 16000;
   constexpr size_t kNumChannels = 2;

   ConfigHelper helper;
   rtc::scoped_refptr<internal::AudioState> audio_state(
       new rtc::RefCountedObject<internal::AudioState>(helper.config()));

   audio_state->SetStereoChannelSwapping(true);

   MockAudioSendStream stream;
   audio_state->AddSendingStream(&stream, kSampleRate, kNumChannels);

   EXPECT_CALL(stream, SendAudioDataForMock(testing::_))
       .WillOnce(
           // Verify that channels are swapped.
           testing::Invoke([](AudioFrame* audio_frame) {
             auto levels = ComputeChannelLevels(audio_frame);
             EXPECT_EQ(0u, levels[0]);
             EXPECT_LT(0u, levels[1]);
           }));

   auto audio_data = Create10msTestData(kSampleRate, kNumChannels);
   uint32_t new_mic_level = 667;
   audio_state->audio_transport()->RecordedDataIsAvailable(
       &audio_data[0], kSampleRate / 100, kNumChannels * 2, kNumChannels,
       kSampleRate, 0, 0, 0, false, new_mic_level);
   EXPECT_EQ(667u, new_mic_level);

   audio_state->RemoveSendingStream(&stream);
 }

 TEST(AudioStateTest, InputLevelStats) {
   constexpr int kSampleRate = 16000;
   constexpr size_t kNumChannels = 1;

   ConfigHelper helper;
   rtc::scoped_refptr<internal::AudioState> audio_state(
       new rtc::RefCountedObject<internal::AudioState>(helper.config()));

   // Push a silent buffer -> Level stats should be zeros except for duration.
   {
     auto audio_data = Create10msSilentTestData(kSampleRate, kNumChannels);
     uint32_t new_mic_level = 667;
     audio_state->audio_transport()->RecordedDataIsAvailable(
         &audio_data[0], kSampleRate / 100, kNumChannels * 2, kNumChannels,
         kSampleRate, 0, 0, 0, false, new_mic_level);
     auto stats = audio_state->GetAudioInputStats();
     EXPECT_EQ(0, stats.audio_level);
     EXPECT_THAT(stats.total_energy, testing::DoubleEq(0.0));
     EXPECT_THAT(stats.total_duration, testing::DoubleEq(0.01));
   }

   // Push 10 non-silent buffers -> Level stats should be non-zero.
   {
     auto audio_data = Create10msTestData(kSampleRate, kNumChannels);
     uint32_t new_mic_level = 667;
     for (int i = 0; i < 10; ++i) {
       audio_state->audio_transport()->RecordedDataIsAvailable(
           &audio_data[0], kSampleRate / 100, kNumChannels * 2, kNumChannels,
           kSampleRate, 0, 0, 0, false, new_mic_level);
     }
     auto stats = audio_state->GetAudioInputStats();
     EXPECT_EQ(32767, stats.audio_level);
     EXPECT_THAT(stats.total_energy, testing::DoubleEq(0.01));
     EXPECT_THAT(stats.total_duration, testing::DoubleEq(0.11));
   }
 }

 TEST(AudioStateTest,
      QueryingTransportForAudioShouldResultInGetAudioCallOnMixerSource) {
   ConfigHelper helper;
   auto audio_state = AudioState::Create(helper.config());

   FakeAudioSource fake_source;
   helper.mixer()->AddSource(&fake_source);

   EXPECT_CALL(fake_source, GetAudioFrameWithInfo(testing::_, testing::_))
       .WillOnce(
           testing::Invoke([](int sample_rate_hz, AudioFrame* audio_frame) {
             audio_frame->sample_rate_hz_ = sample_rate_hz;
             audio_frame->samples_per_channel_ = sample_rate_hz / 100;
             audio_frame->num_channels_ = kNumberOfChannels;
             return AudioMixer::Source::AudioFrameInfo::kNormal;
           }));

   int16_t audio_buffer[kSampleRate / 100 * kNumberOfChannels];
   size_t n_samples_out;
   int64_t elapsed_time_ms;
   int64_t ntp_time_ms;
   audio_state->audio_transport()->NeedMorePlayData(
       kSampleRate / 100, kNumberOfChannels * 2, kNumberOfChannels, kSampleRate,
       audio_buffer, n_samples_out, &elapsed_time_ms, &ntp_time_ms);
 }
 }  // namespace test
 }  // namespace webrtc
	/*
	* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
	*
	* Use of this source code is governed by a BSD-style license
	* that can be found in the LICENSE file in the root of the source
	* tree. An additional intellectual property rights grant can be found
	* in the file PATENTS. All contributing project authors may
	* be found in the AUTHORS file in the root of the source tree.
	*/

	#include <memory>
	#include <vector>

	#include "audio/audio_state.h"
	#include "call/test/mock_audio_send_stream.h"
	#include "modules/audio_device/include/mock_audio_device.h"
	#include "modules/audio_mixer/audio_mixer_impl.h"
	#include "modules/audio_processing/include/mock_audio_processing.h"
	#include "rtc_base/ref_counted_object.h"
	#include "test/gtest.h"

	namespace webrtc {
	namespace test {
	namespace {

	constexpr int kSampleRate = 16000;
	constexpr int kNumberOfChannels = 1;

	struct ConfigHelper {
	ConfigHelper() : audio_mixer(AudioMixerImpl::Create()) {
	audio_state_config.audio_mixer = audio_mixer;
	audio_state_config.audio_processing =
	new rtc::RefCountedObject<testing::NiceMock<MockAudioProcessing>>();
	audio_state_config.audio_device_module =
	new rtc::RefCountedObject<MockAudioDeviceModule>();
	}
	AudioState::Config& config() { return audio_state_config; }
	rtc::scoped_refptr<AudioMixer> mixer() { return audio_mixer; }

	private:
	AudioState::Config audio_state_config;
	rtc::scoped_refptr<AudioMixer> audio_mixer;
	};

	class FakeAudioSource : public AudioMixer::Source {
	public:
	// TODO(aleloi): Valid overrides commented out, because the gmock
	// methods don't use any override declarations, and we want to avoid
	// warnings from -Winconsistent-missing-override. See
	// http://crbug.com/428099.
	int Ssrc() const /override/ { return 0; }

	int PreferredSampleRate() const /override/ { return kSampleRate; }

	MOCK_METHOD2(GetAudioFrameWithInfo,
	AudioFrameInfo(int sample_rate_hz, AudioFrame* audio_frame));
	};

	std::vector<int16_t> Create10msSilentTestData(int sample_rate_hz,
	size_t num_channels) {
	const int samples_per_channel = sample_rate_hz / 100;
	std::vector<int16_t> audio_data(samples_per_channel * num_channels, 0);
	return audio_data;
	}

	std::vector<int16_t> Create10msTestData(int sample_rate_hz,
	size_t num_channels) {
	const int samples_per_channel = sample_rate_hz / 100;
	std::vector<int16_t> audio_data(samples_per_channel * num_channels, 0);
	// Fill the first channel with a 1kHz sine wave.
	const float inc = (2 * 3.14159265f * 1000) / sample_rate_hz;
	float w = 0.f;
	for (int i = 0; i < samples_per_channel; ++i) {
	audio_data[i * num_channels] = static_cast<int16_t>(32767.f * std::sin(w));
	w += inc;
	}
	return audio_data;
	}

	std::vector<uint32_t> ComputeChannelLevels(AudioFrame* audio_frame) {
	const size_t num_channels = audio_frame->num_channels_;
	const size_t samples_per_channel = audio_frame->samples_per_channel_;
	std::vector<uint32_t> levels(num_channels, 0);
	for (size_t i = 0; i < samples_per_channel; ++i) {
	for (size_t j = 0; j < num_channels; ++j) {
	levels[j] += std::abs(audio_frame->data()[i * num_channels + j]);
	}
	}
	return levels;
	}
	} // namespace

	TEST(AudioStateTest, Create) {
	ConfigHelper helper;
	auto audio_state = AudioState::Create(helper.config());
	EXPECT_TRUE(audio_state.get());
	}

	TEST(AudioStateTest, ConstructDestruct) {
	ConfigHelper helper;
	rtc::scoped_refptr<internal::AudioState> audio_state(
	new rtc::RefCountedObject<internal::AudioState>(helper.config()));
	}

	TEST(AudioStateTest, RecordedAudioArrivesAtSingleStream) {
	ConfigHelper helper;
	rtc::scoped_refptr<internal::AudioState> audio_state(
	new rtc::RefCountedObject<internal::AudioState>(helper.config()));

	MockAudioSendStream stream;
	audio_state->AddSendingStream(&stream, 8000, 2);

	EXPECT_CALL(
	stream,
	SendAudioDataForMock(testing::AllOf(
	testing::Field(&AudioFrame::sample_rate_hz_, testing::Eq(8000)),
	testing::Field(&AudioFrame::num_channels_, testing::Eq(2u)))))
	.WillOnce(
	// Verify that channels are not swapped by default.
	testing::Invoke([](AudioFrame* audio_frame) {
	auto levels = ComputeChannelLevels(audio_frame);
	EXPECT_LT(0u, levels[0]);
	EXPECT_EQ(0u, levels[1]);
	}));
	MockAudioProcessing* ap =
	static_cast<MockAudioProcessing*>(audio_state->audio_processing());
	EXPECT_CALL(*ap, set_stream_delay_ms(0));
	EXPECT_CALL(*ap, set_stream_key_pressed(false));
	EXPECT_CALL(*ap, ProcessStream(testing::_));

	constexpr int kSampleRate = 16000;
	constexpr size_t kNumChannels = 2;
	auto audio_data = Create10msTestData(kSampleRate, kNumChannels);
	uint32_t new_mic_level = 667;
	audio_state->audio_transport()->RecordedDataIsAvailable(
	&audio_data[0], kSampleRate / 100, kNumChannels * 2, kNumChannels,
	kSampleRate, 0, 0, 0, false, new_mic_level);
	EXPECT_EQ(667u, new_mic_level);

	audio_state->RemoveSendingStream(&stream);
	}

	TEST(AudioStateTest, RecordedAudioArrivesAtMultipleStreams) {
	ConfigHelper helper;
	rtc::scoped_refptr<internal::AudioState> audio_state(
	new rtc::RefCountedObject<internal::AudioState>(helper.config()));

	MockAudioSendStream stream_1;
	MockAudioSendStream stream_2;
	audio_state->AddSendingStream(&stream_1, 8001, 2);
	audio_state->AddSendingStream(&stream_2, 32000, 1);

	EXPECT_CALL(
	stream_1,
	SendAudioDataForMock(testing::AllOf(
	testing::Field(&AudioFrame::sample_rate_hz_, testing::Eq(16000)),
	testing::Field(&AudioFrame::num_channels_, testing::Eq(1u)))))
	.WillOnce(
	// Verify that there is output signal.
	testing::Invoke([](AudioFrame* audio_frame) {
	auto levels = ComputeChannelLevels(audio_frame);
	EXPECT_LT(0u, levels[0]);
	}));
	EXPECT_CALL(
	stream_2,
	SendAudioDataForMock(testing::AllOf(
	testing::Field(&AudioFrame::sample_rate_hz_, testing::Eq(16000)),
	testing::Field(&AudioFrame::num_channels_, testing::Eq(1u)))))
	.WillOnce(
	// Verify that there is output signal.
	testing::Invoke([](AudioFrame* audio_frame) {
	auto levels = ComputeChannelLevels(audio_frame);
	EXPECT_LT(0u, levels[0]);
	}));
	MockAudioProcessing* ap =
	static_cast<MockAudioProcessing*>(audio_state->audio_processing());
	EXPECT_CALL(*ap, set_stream_delay_ms(5));
	EXPECT_CALL(*ap, set_stream_key_pressed(true));
	EXPECT_CALL(*ap, ProcessStream(testing::_));

	constexpr int kSampleRate = 16000;
	constexpr size_t kNumChannels = 1;
	auto audio_data = Create10msTestData(kSampleRate, kNumChannels);
	uint32_t new_mic_level = 667;
	audio_state->audio_transport()->RecordedDataIsAvailable(
	&audio_data[0], kSampleRate / 100, kNumChannels * 2, kNumChannels,
	kSampleRate, 5, 0, 0, true, new_mic_level);
	EXPECT_EQ(667u, new_mic_level);

	audio_state->RemoveSendingStream(&stream_1);
	audio_state->RemoveSendingStream(&stream_2);
	}

	TEST(AudioStateTest, EnableChannelSwap) {
	constexpr int kSampleRate = 16000;
	constexpr size_t kNumChannels = 2;

	ConfigHelper helper;
	rtc::scoped_refptr<internal::AudioState> audio_state(
	new rtc::RefCountedObject<internal::AudioState>(helper.config()));

	audio_state->SetStereoChannelSwapping(true);

	MockAudioSendStream stream;
	audio_state->AddSendingStream(&stream, kSampleRate, kNumChannels);

	EXPECT_CALL(stream, SendAudioDataForMock(testing::_))
	.WillOnce(
	// Verify that channels are swapped.
	testing::Invoke([](AudioFrame* audio_frame) {
	auto levels = ComputeChannelLevels(audio_frame);
	EXPECT_EQ(0u, levels[0]);
	EXPECT_LT(0u, levels[1]);
	}));

	auto audio_data = Create10msTestData(kSampleRate, kNumChannels);
	uint32_t new_mic_level = 667;
	audio_state->audio_transport()->RecordedDataIsAvailable(
	&audio_data[0], kSampleRate / 100, kNumChannels * 2, kNumChannels,
	kSampleRate, 0, 0, 0, false, new_mic_level);
	EXPECT_EQ(667u, new_mic_level);

	audio_state->RemoveSendingStream(&stream);
	}

	TEST(AudioStateTest, InputLevelStats) {
	constexpr int kSampleRate = 16000;
	constexpr size_t kNumChannels = 1;

	ConfigHelper helper;
	rtc::scoped_refptr<internal::AudioState> audio_state(
	new rtc::RefCountedObject<internal::AudioState>(helper.config()));

	// Push a silent buffer -> Level stats should be zeros except for duration.
	{
	auto audio_data = Create10msSilentTestData(kSampleRate, kNumChannels);
	uint32_t new_mic_level = 667;
	audio_state->audio_transport()->RecordedDataIsAvailable(
	&audio_data[0], kSampleRate / 100, kNumChannels * 2, kNumChannels,
	kSampleRate, 0, 0, 0, false, new_mic_level);
	auto stats = audio_state->GetAudioInputStats();
	EXPECT_EQ(0, stats.audio_level);
	EXPECT_THAT(stats.total_energy, testing::DoubleEq(0.0));
	EXPECT_THAT(stats.total_duration, testing::DoubleEq(0.01));
	}

	// Push 10 non-silent buffers -> Level stats should be non-zero.
	{
	auto audio_data = Create10msTestData(kSampleRate, kNumChannels);
	uint32_t new_mic_level = 667;
	for (int i = 0; i < 10; ++i) {
	audio_state->audio_transport()->RecordedDataIsAvailable(
	&audio_data[0], kSampleRate / 100, kNumChannels * 2, kNumChannels,
	kSampleRate, 0, 0, 0, false, new_mic_level);
	}
	auto stats = audio_state->GetAudioInputStats();
	EXPECT_EQ(32767, stats.audio_level);
	EXPECT_THAT(stats.total_energy, testing::DoubleEq(0.01));
	EXPECT_THAT(stats.total_duration, testing::DoubleEq(0.11));
	}
	}

	TEST(AudioStateTest,
	QueryingTransportForAudioShouldResultInGetAudioCallOnMixerSource) {
	ConfigHelper helper;
	auto audio_state = AudioState::Create(helper.config());

	FakeAudioSource fake_source;
	helper.mixer()->AddSource(&fake_source);

	EXPECT_CALL(fake_source, GetAudioFrameWithInfo(testing::_, testing::_))
	.WillOnce(
	testing::Invoke([](int sample_rate_hz, AudioFrame* audio_frame) {
	audio_frame->sample_rate_hz_ = sample_rate_hz;
	audio_frame->samples_per_channel_ = sample_rate_hz / 100;
	audio_frame->num_channels_ = kNumberOfChannels;
	return AudioMixer::Source::AudioFrameInfo::kNormal;
	}));

	int16_t audio_buffer[kSampleRate / 100 * kNumberOfChannels];
	size_t n_samples_out;
	int64_t elapsed_time_ms;
	int64_t ntp_time_ms;
	audio_state->audio_transport()->NeedMorePlayData(
	kSampleRate / 100, kNumberOfChannels * 2, kNumberOfChannels, kSampleRate,
	audio_buffer, n_samples_out, &elapsed_time_ms, &ntp_time_ms);
	}
	} // namespace test
	} // namespace webrtc