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 "audio/audio_state.h"

 #include <memory>
 #include <utility>
 #include <vector>

 #include "api/task_queue/test/mock_task_queue_base.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 "test/gtest.h"

 namespace webrtc {
 namespace test {
 namespace {

 using ::testing::_;
 using ::testing::Matcher;
 using ::testing::NiceMock;
 using ::testing::StrictMock;
 using ::testing::Values;

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

 struct FakeAsyncAudioProcessingHelper {
   class FakeTaskQueue : public StrictMock<MockTaskQueueBase> {
    public:
     FakeTaskQueue() = default;

     void Delete() override { delete this; }
     void PostTask(absl::AnyInvocable<void() &&> task) override {
       std::move(task)();
     }
   };

   class FakeTaskQueueFactory : public TaskQueueFactory {
    public:
     FakeTaskQueueFactory() = default;
     ~FakeTaskQueueFactory() override = default;
     std::unique_ptr<TaskQueueBase, TaskQueueDeleter> CreateTaskQueue(
         absl::string_view name,
         Priority priority) const override {
       return std::unique_ptr<webrtc::TaskQueueBase, webrtc::TaskQueueDeleter>(
           new FakeTaskQueue());
     }
   };

   class MockAudioFrameProcessor : public AudioFrameProcessor {
    public:
     ~MockAudioFrameProcessor() override = default;

     MOCK_METHOD(void, ProcessCalled, ());
     MOCK_METHOD(void, SinkSet, ());
     MOCK_METHOD(void, SinkCleared, ());

     void Process(std::unique_ptr<AudioFrame> frame) override {
       ProcessCalled();
       sink_callback_(std::move(frame));
     }

     void SetSink(OnAudioFrameCallback sink_callback) override {
       sink_callback_ = std::move(sink_callback);
       if (sink_callback_ == nullptr)
         SinkCleared();
       else
         SinkSet();
     }

    private:
     OnAudioFrameCallback sink_callback_;
   };

   NiceMock<MockAudioFrameProcessor> audio_frame_processor_;
   FakeTaskQueueFactory task_queue_factory_;

   rtc::scoped_refptr<AsyncAudioProcessing::Factory> CreateFactory() {
     return rtc::make_ref_counted<AsyncAudioProcessing::Factory>(
         audio_frame_processor_, task_queue_factory_);
   }
 };

 struct ConfigHelper {
   struct Params {
     bool use_null_audio_processing;
     bool use_async_audio_processing;
   };

   explicit ConfigHelper(const Params& params)
       : audio_mixer(AudioMixerImpl::Create()) {
     audio_state_config.audio_mixer = audio_mixer;
     audio_state_config.audio_processing =
         params.use_null_audio_processing
             ? nullptr
             : rtc::make_ref_counted<testing::NiceMock<MockAudioProcessing>>();
     audio_state_config.audio_device_module =
         rtc::make_ref_counted<NiceMock<MockAudioDeviceModule>>();
     if (params.use_async_audio_processing) {
       audio_state_config.async_audio_processing_factory =
           async_audio_processing_helper_.CreateFactory();
     }
   }
   AudioState::Config& config() { return audio_state_config; }
   rtc::scoped_refptr<AudioMixer> mixer() { return audio_mixer; }
   NiceMock<FakeAsyncAudioProcessingHelper::MockAudioFrameProcessor>&
   mock_audio_frame_processor() {
     return async_audio_processing_helper_.audio_frame_processor_;
   }

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

 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_METHOD(AudioFrameInfo,
               GetAudioFrameWithInfo,
               (int sample_rate_hz, AudioFrame*),
               (override));
 };

 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

 class AudioStateTest : public ::testing::TestWithParam<ConfigHelper::Params> {};

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

 TEST_P(AudioStateTest, ConstructDestruct) {
   ConfigHelper helper(GetParam());
   rtc::scoped_refptr<internal::AudioState> audio_state(
       rtc::make_ref_counted<internal::AudioState>(helper.config()));
 }

 TEST_P(AudioStateTest, RecordedAudioArrivesAtSingleStream) {
   ConfigHelper helper(GetParam());

   if (GetParam().use_async_audio_processing) {
     EXPECT_CALL(helper.mock_audio_frame_processor(), SinkSet);
     EXPECT_CALL(helper.mock_audio_frame_processor(), ProcessCalled);
     EXPECT_CALL(helper.mock_audio_frame_processor(), SinkCleared);
   }

   rtc::scoped_refptr<internal::AudioState> audio_state(
       rtc::make_ref_counted<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 =
       GetParam().use_null_audio_processing
           ? nullptr
           : static_cast<MockAudioProcessing*>(audio_state->audio_processing());
   if (ap) {
     EXPECT_CALL(*ap, set_stream_delay_ms(0));
     EXPECT_CALL(*ap, set_stream_key_pressed(false));
     EXPECT_CALL(*ap, ProcessStream(_, _, _, Matcher<int16_t*>(_)));
   }

   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_P(AudioStateTest, RecordedAudioArrivesAtMultipleStreams) {
   ConfigHelper helper(GetParam());

   if (GetParam().use_async_audio_processing) {
     EXPECT_CALL(helper.mock_audio_frame_processor(), SinkSet);
     EXPECT_CALL(helper.mock_audio_frame_processor(), ProcessCalled);
     EXPECT_CALL(helper.mock_audio_frame_processor(), SinkCleared);
   }

   rtc::scoped_refptr<internal::AudioState> audio_state(
       rtc::make_ref_counted<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());
   if (ap) {
     EXPECT_CALL(*ap, set_stream_delay_ms(5));
     EXPECT_CALL(*ap, set_stream_key_pressed(true));
     EXPECT_CALL(*ap, ProcessStream(_, _, _, Matcher<int16_t*>(_)));
   }

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

   ConfigHelper helper(GetParam());

   if (GetParam().use_async_audio_processing) {
     EXPECT_CALL(helper.mock_audio_frame_processor(), SinkSet);
     EXPECT_CALL(helper.mock_audio_frame_processor(), ProcessCalled);
     EXPECT_CALL(helper.mock_audio_frame_processor(), SinkCleared);
   }

   rtc::scoped_refptr<internal::AudioState> audio_state(
       rtc::make_ref_counted<internal::AudioState>(helper.config()));

   audio_state->SetStereoChannelSwapping(true);

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

   EXPECT_CALL(stream, SendAudioDataForMock(_))
       .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_P(AudioStateTest,
        QueryingTransportForAudioShouldResultInGetAudioCallOnMixerSource) {
   ConfigHelper helper(GetParam());
   auto audio_state = AudioState::Create(helper.config());

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

   EXPECT_CALL(fake_source, GetAudioFrameWithInfo(_, _))
       .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);
 }

 INSTANTIATE_TEST_SUITE_P(AudioStateTest,
                          AudioStateTest,
                          Values(ConfigHelper::Params({false, false}),
                                 ConfigHelper::Params({true, false}),
                                 ConfigHelper::Params({false, true}),
                                 ConfigHelper::Params({true, true})));

 }  // 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 "audio/audio_state.h"

	#include <memory>
	#include <utility>
	#include <vector>

	#include "api/task_queue/test/mock_task_queue_base.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 "test/gtest.h"

	namespace webrtc {
	namespace test {
	namespace {

	using ::testing::_;
	using ::testing::Matcher;
	using ::testing::NiceMock;
	using ::testing::StrictMock;
	using ::testing::Values;

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

	struct FakeAsyncAudioProcessingHelper {
	class FakeTaskQueue : public StrictMock<MockTaskQueueBase> {
	public:
	FakeTaskQueue() = default;

	void Delete() override { delete this; }
	void PostTask(absl::AnyInvocable<void() &&> task) override {
	std::move(task)();
	}
	};

	class FakeTaskQueueFactory : public TaskQueueFactory {
	public:
	FakeTaskQueueFactory() = default;
	~FakeTaskQueueFactory() override = default;
	std::unique_ptr<TaskQueueBase, TaskQueueDeleter> CreateTaskQueue(
	absl::string_view name,
	Priority priority) const override {
	return std::unique_ptr<webrtc::TaskQueueBase, webrtc::TaskQueueDeleter>(
	new FakeTaskQueue());
	}
	};

	class MockAudioFrameProcessor : public AudioFrameProcessor {
	public:
	~MockAudioFrameProcessor() override = default;

	MOCK_METHOD(void, ProcessCalled, ());
	MOCK_METHOD(void, SinkSet, ());
	MOCK_METHOD(void, SinkCleared, ());

	void Process(std::unique_ptr<AudioFrame> frame) override {
	ProcessCalled();
	sink_callback_(std::move(frame));
	}

	void SetSink(OnAudioFrameCallback sink_callback) override {
	sink_callback_ = std::move(sink_callback);
	if (sink_callback_ == nullptr)
	SinkCleared();
	else
	SinkSet();
	}

	private:
	OnAudioFrameCallback sink_callback_;
	};

	NiceMock<MockAudioFrameProcessor> audio_frame_processor_;
	FakeTaskQueueFactory task_queue_factory_;

	rtc::scoped_refptr<AsyncAudioProcessing::Factory> CreateFactory() {
	return rtc::make_ref_counted<AsyncAudioProcessing::Factory>(
	audio_frame_processor_, task_queue_factory_);
	}
	};

	struct ConfigHelper {
	struct Params {
	bool use_null_audio_processing;
	bool use_async_audio_processing;
	};

	explicit ConfigHelper(const Params& params)
	: audio_mixer(AudioMixerImpl::Create()) {
	audio_state_config.audio_mixer = audio_mixer;
	audio_state_config.audio_processing =
	params.use_null_audio_processing
	? nullptr
	: rtc::make_ref_counted<testing::NiceMock<MockAudioProcessing>>();
	audio_state_config.audio_device_module =
	rtc::make_ref_counted<NiceMock<MockAudioDeviceModule>>();
	if (params.use_async_audio_processing) {
	audio_state_config.async_audio_processing_factory =
	async_audio_processing_helper_.CreateFactory();
	}
	}
	AudioState::Config& config() { return audio_state_config; }
	rtc::scoped_refptr<AudioMixer> mixer() { return audio_mixer; }
	NiceMock<FakeAsyncAudioProcessingHelper::MockAudioFrameProcessor>&
	mock_audio_frame_processor() {
	return async_audio_processing_helper_.audio_frame_processor_;
	}

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

	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_METHOD(AudioFrameInfo,
	GetAudioFrameWithInfo,
	(int sample_rate_hz, AudioFrame*),
	(override));
	};

	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

	class AudioStateTest : public ::testing::TestWithParam<ConfigHelper::Params> {};

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

	TEST_P(AudioStateTest, ConstructDestruct) {
	ConfigHelper helper(GetParam());
	rtc::scoped_refptr<internal::AudioState> audio_state(
	rtc::make_ref_counted<internal::AudioState>(helper.config()));
	}

	TEST_P(AudioStateTest, RecordedAudioArrivesAtSingleStream) {
	ConfigHelper helper(GetParam());

	if (GetParam().use_async_audio_processing) {
	EXPECT_CALL(helper.mock_audio_frame_processor(), SinkSet);
	EXPECT_CALL(helper.mock_audio_frame_processor(), ProcessCalled);
	EXPECT_CALL(helper.mock_audio_frame_processor(), SinkCleared);
	}

	rtc::scoped_refptr<internal::AudioState> audio_state(
	rtc::make_ref_counted<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 =
	GetParam().use_null_audio_processing
	? nullptr
	: static_cast<MockAudioProcessing*>(audio_state->audio_processing());
	if (ap) {
	EXPECT_CALL(*ap, set_stream_delay_ms(0));
	EXPECT_CALL(*ap, set_stream_key_pressed(false));
	EXPECT_CALL(ap, ProcessStream(_, _, _, Matcher<int16_t>(_)));
	}

	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_P(AudioStateTest, RecordedAudioArrivesAtMultipleStreams) {
	ConfigHelper helper(GetParam());

	if (GetParam().use_async_audio_processing) {
	EXPECT_CALL(helper.mock_audio_frame_processor(), SinkSet);
	EXPECT_CALL(helper.mock_audio_frame_processor(), ProcessCalled);
	EXPECT_CALL(helper.mock_audio_frame_processor(), SinkCleared);
	}

	rtc::scoped_refptr<internal::AudioState> audio_state(
	rtc::make_ref_counted<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());
	if (ap) {
	EXPECT_CALL(*ap, set_stream_delay_ms(5));
	EXPECT_CALL(*ap, set_stream_key_pressed(true));
	EXPECT_CALL(ap, ProcessStream(_, _, _, Matcher<int16_t>(_)));
	}

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

	ConfigHelper helper(GetParam());

	if (GetParam().use_async_audio_processing) {
	EXPECT_CALL(helper.mock_audio_frame_processor(), SinkSet);
	EXPECT_CALL(helper.mock_audio_frame_processor(), ProcessCalled);
	EXPECT_CALL(helper.mock_audio_frame_processor(), SinkCleared);
	}

	rtc::scoped_refptr<internal::AudioState> audio_state(
	rtc::make_ref_counted<internal::AudioState>(helper.config()));

	audio_state->SetStereoChannelSwapping(true);

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

	EXPECT_CALL(stream, SendAudioDataForMock(_))
	.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_P(AudioStateTest,
	QueryingTransportForAudioShouldResultInGetAudioCallOnMixerSource) {
	ConfigHelper helper(GetParam());
	auto audio_state = AudioState::Create(helper.config());

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

	EXPECT_CALL(fake_source, GetAudioFrameWithInfo(_, _))
	.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);
	}

	INSTANTIATE_TEST_SUITE_P(AudioStateTest,
	AudioStateTest,
	Values(ConfigHelper::Params({false, false}),
	ConfigHelper::Params({true, false}),
	ConfigHelper::Params({false, true}),
	ConfigHelper::Params({true, true})));

	} // namespace test
	} // namespace webrtc