modules/audio_device/test_audio_device_impl_test.cc - src - Git at Google

 /*
  *  Copyright (c) 2023 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 "modules/audio_device/test_audio_device_impl.h"

 #include <memory>
 #include <utility>

 #include "absl/types/optional.h"
 #include "api/task_queue/task_queue_factory.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
 #include "modules/audio_device/audio_device_buffer.h"
 #include "modules/audio_device/audio_device_generic.h"
 #include "modules/audio_device/include/audio_device.h"
 #include "modules/audio_device/include/audio_device_defines.h"
 #include "modules/audio_device/include/test_audio_device.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/synchronization/mutex.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
 #include "test/time_controller/simulated_time_controller.h"

 namespace webrtc {
 namespace {

 using ::testing::ElementsAre;

 constexpr Timestamp kStartTime = Timestamp::Millis(10000);

 class TestAudioTransport : public AudioTransport {
  public:
   enum class Mode { kPlaying, kRecording };

   explicit TestAudioTransport(Mode mode) : mode_(mode) {}
   ~TestAudioTransport() override = default;

   int32_t RecordedDataIsAvailable(
       const void* audioSamples,
       size_t samples_per_channel,
       size_t bytes_per_sample,
       size_t number_of_channels,
       uint32_t samples_per_second,
       uint32_t total_delay_ms,
       int32_t clock_drift,
       uint32_t current_mic_level,
       bool key_pressed,
       uint32_t& new_mic_level,
       absl::optional<int64_t> estimated_capture_time_ns) override {
     new_mic_level = 1;

     if (mode_ != Mode::kRecording) {
       EXPECT_TRUE(false)
           << "NeedMorePlayData mustn't be called when mode isn't kRecording";
       return -1;
     }

     MutexLock lock(&mutex_);
     samples_per_channel_.push_back(samples_per_channel);
     number_of_channels_.push_back(number_of_channels);
     bytes_per_sample_.push_back(bytes_per_sample);
     samples_per_second_.push_back(samples_per_second);
     return 0;
   }

   int32_t NeedMorePlayData(size_t samples_per_channel,
                            size_t bytes_per_sample,
                            size_t number_of_channels,
                            uint32_t samples_per_second,
                            void* audio_samples,
                            size_t& samples_out,
                            int64_t* elapsed_time_ms,
                            int64_t* ntp_time_ms) override {
     const size_t num_bytes = samples_per_channel * number_of_channels;
     std::memset(audio_samples, 1, num_bytes);
     samples_out = samples_per_channel * number_of_channels;
     *elapsed_time_ms = 0;
     *ntp_time_ms = 0;

     if (mode_ != Mode::kPlaying) {
       EXPECT_TRUE(false)
           << "NeedMorePlayData mustn't be called when mode isn't kPlaying";
       return -1;
     }

     MutexLock lock(&mutex_);
     samples_per_channel_.push_back(samples_per_channel);
     number_of_channels_.push_back(number_of_channels);
     bytes_per_sample_.push_back(bytes_per_sample);
     samples_per_second_.push_back(samples_per_second);
     return 0;
   }

   int32_t RecordedDataIsAvailable(const void* audio_samples,
                                   size_t samples_per_channel,
                                   size_t bytes_per_sample,
                                   size_t number_of_channels,
                                   uint32_t samples_per_second,
                                   uint32_t total_delay_ms,
                                   int32_t clockDrift,
                                   uint32_t current_mic_level,
                                   bool key_pressed,
                                   uint32_t& new_mic_level) override {
     RTC_CHECK(false) << "This methods should be never executed";
   }

   void PullRenderData(int bits_per_sample,
                       int sample_rate,
                       size_t number_of_channels,
                       size_t number_of_frames,
                       void* audio_data,
                       int64_t* elapsed_time_ms,
                       int64_t* ntp_time_ms) override {
     RTC_CHECK(false) << "This methods should be never executed";
   }

   std::vector<size_t> samples_per_channel() const {
     MutexLock lock(&mutex_);
     return samples_per_channel_;
   }
   std::vector<size_t> number_of_channels() const {
     MutexLock lock(&mutex_);
     return number_of_channels_;
   }
   std::vector<size_t> bytes_per_sample() const {
     MutexLock lock(&mutex_);
     return bytes_per_sample_;
   }
   std::vector<size_t> samples_per_second() const {
     MutexLock lock(&mutex_);
     return samples_per_second_;
   }

  private:
   const Mode mode_;

   mutable Mutex mutex_;
   std::vector<size_t> samples_per_channel_ RTC_GUARDED_BY(mutex_);
   std::vector<size_t> number_of_channels_ RTC_GUARDED_BY(mutex_);
   std::vector<size_t> bytes_per_sample_ RTC_GUARDED_BY(mutex_);
   std::vector<size_t> samples_per_second_ RTC_GUARDED_BY(mutex_);
 };

 TEST(TestAudioDeviceTest, EnablingRecordingProducesAudio) {
   GlobalSimulatedTimeController time_controller(kStartTime);
   TestAudioTransport audio_transport(TestAudioTransport::Mode::kRecording);
   AudioDeviceBuffer audio_buffer(time_controller.GetTaskQueueFactory());
   ASSERT_EQ(audio_buffer.RegisterAudioCallback(&audio_transport), 0);
   std::unique_ptr<TestAudioDeviceModule::PulsedNoiseCapturer> capturer =
       TestAudioDeviceModule::CreatePulsedNoiseCapturer(
           /*max_amplitude=*/1000,
           /*sampling_frequency_in_hz=*/48000, /*num_channels=*/2);

   TestAudioDevice audio_device(time_controller.GetTaskQueueFactory(),
                                std::move(capturer),
                                /*renderer=*/nullptr);
   ASSERT_EQ(audio_device.Init(), AudioDeviceGeneric::InitStatus::OK);
   audio_device.AttachAudioBuffer(&audio_buffer);

   EXPECT_FALSE(audio_device.RecordingIsInitialized());
   ASSERT_EQ(audio_device.InitRecording(), 0);
   EXPECT_TRUE(audio_device.RecordingIsInitialized());
   audio_buffer.StartRecording();
   ASSERT_EQ(audio_device.StartRecording(), 0);
   time_controller.AdvanceTime(TimeDelta::Millis(10));
   ASSERT_TRUE(audio_device.Recording());
   time_controller.AdvanceTime(TimeDelta::Millis(10));
   ASSERT_EQ(audio_device.StopRecording(), 0);
   audio_buffer.StopRecording();

   EXPECT_THAT(audio_transport.samples_per_channel(),
               ElementsAre(480, 480, 480));
   EXPECT_THAT(audio_transport.number_of_channels(), ElementsAre(2, 2, 2));
   EXPECT_THAT(audio_transport.bytes_per_sample(), ElementsAre(4, 4, 4));
   EXPECT_THAT(audio_transport.samples_per_second(),
               ElementsAre(48000, 48000, 48000));
 }

 TEST(TestAudioDeviceTest, RecordingIsAvailableWhenCapturerIsSet) {
   GlobalSimulatedTimeController time_controller(kStartTime);
   std::unique_ptr<TestAudioDeviceModule::PulsedNoiseCapturer> capturer =
       TestAudioDeviceModule::CreatePulsedNoiseCapturer(
           /*max_amplitude=*/1000,
           /*sampling_frequency_in_hz=*/48000, /*num_channels=*/2);

   TestAudioDevice audio_device(time_controller.GetTaskQueueFactory(),
                                std::move(capturer),
                                /*renderer=*/nullptr);
   ASSERT_EQ(audio_device.Init(), AudioDeviceGeneric::InitStatus::OK);

   bool available;
   EXPECT_EQ(audio_device.RecordingIsAvailable(available), 0);
   EXPECT_TRUE(available);
 }

 TEST(TestAudioDeviceTest, RecordingIsNotAvailableWhenCapturerIsNotSet) {
   GlobalSimulatedTimeController time_controller(kStartTime);
   TestAudioDevice audio_device(time_controller.GetTaskQueueFactory(),
                                /*capturer=*/nullptr,
                                /*renderer=*/nullptr);
   ASSERT_EQ(audio_device.Init(), AudioDeviceGeneric::InitStatus::OK);

   bool available;
   EXPECT_EQ(audio_device.RecordingIsAvailable(available), 0);
   EXPECT_FALSE(available);
 }

 TEST(TestAudioDeviceTest, EnablingPlayoutProducesAudio) {
   GlobalSimulatedTimeController time_controller(kStartTime);
   TestAudioTransport audio_transport(TestAudioTransport::Mode::kPlaying);
   AudioDeviceBuffer audio_buffer(time_controller.GetTaskQueueFactory());
   ASSERT_EQ(audio_buffer.RegisterAudioCallback(&audio_transport), 0);
   std::unique_ptr<TestAudioDeviceModule::Renderer> renderer =
       TestAudioDeviceModule::CreateDiscardRenderer(
           /*sampling_frequency_in_hz=*/48000, /*num_channels=*/2);

   TestAudioDevice audio_device(time_controller.GetTaskQueueFactory(),
                                /*capturer=*/nullptr, std::move(renderer));
   ASSERT_EQ(audio_device.Init(), AudioDeviceGeneric::InitStatus::OK);
   audio_device.AttachAudioBuffer(&audio_buffer);

   EXPECT_FALSE(audio_device.PlayoutIsInitialized());
   ASSERT_EQ(audio_device.InitPlayout(), 0);
   EXPECT_TRUE(audio_device.PlayoutIsInitialized());
   audio_buffer.StartPlayout();
   ASSERT_EQ(audio_device.StartPlayout(), 0);
   time_controller.AdvanceTime(TimeDelta::Millis(10));
   ASSERT_TRUE(audio_device.Playing());
   time_controller.AdvanceTime(TimeDelta::Millis(10));
   ASSERT_EQ(audio_device.StopPlayout(), 0);
   audio_buffer.StopPlayout();

   EXPECT_THAT(audio_transport.samples_per_channel(),
               ElementsAre(480, 480, 480));
   EXPECT_THAT(audio_transport.number_of_channels(), ElementsAre(2, 2, 2));
   EXPECT_THAT(audio_transport.bytes_per_sample(), ElementsAre(4, 4, 4));
   EXPECT_THAT(audio_transport.samples_per_second(),
               ElementsAre(48000, 48000, 48000));
 }

 TEST(TestAudioDeviceTest, PlayoutIsAvailableWhenRendererIsSet) {
   GlobalSimulatedTimeController time_controller(kStartTime);
   std::unique_ptr<TestAudioDeviceModule::Renderer> renderer =
       TestAudioDeviceModule::CreateDiscardRenderer(
           /*sampling_frequency_in_hz=*/48000, /*num_channels=*/2);

   TestAudioDevice audio_device(time_controller.GetTaskQueueFactory(),
                                /*capturer=*/nullptr, std::move(renderer));
   ASSERT_EQ(audio_device.Init(), AudioDeviceGeneric::InitStatus::OK);

   bool available;
   EXPECT_EQ(audio_device.PlayoutIsAvailable(available), 0);
   EXPECT_TRUE(available);
 }

 TEST(TestAudioDeviceTest, PlayoutIsNotAvailableWhenRendererIsNotSet) {
   GlobalSimulatedTimeController time_controller(kStartTime);
   TestAudioDevice audio_device(time_controller.GetTaskQueueFactory(),
                                /*capturer=*/nullptr,
                                /*renderer=*/nullptr);
   ASSERT_EQ(audio_device.Init(), AudioDeviceGeneric::InitStatus::OK);

   bool available;
   EXPECT_EQ(audio_device.PlayoutIsAvailable(available), 0);
   EXPECT_FALSE(available);
 }

 }  // namespace
 }  // namespace webrtc
	/*
	* Copyright (c) 2023 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 "modules/audio_device/test_audio_device_impl.h"

	#include <memory>
	#include <utility>

	#include "absl/types/optional.h"
	#include "api/task_queue/task_queue_factory.h"
	#include "api/units/time_delta.h"
	#include "api/units/timestamp.h"
	#include "modules/audio_device/audio_device_buffer.h"
	#include "modules/audio_device/audio_device_generic.h"
	#include "modules/audio_device/include/audio_device.h"
	#include "modules/audio_device/include/audio_device_defines.h"
	#include "modules/audio_device/include/test_audio_device.h"
	#include "rtc_base/checks.h"
	#include "rtc_base/synchronization/mutex.h"
	#include "test/gmock.h"
	#include "test/gtest.h"
	#include "test/time_controller/simulated_time_controller.h"

	namespace webrtc {
	namespace {

	using ::testing::ElementsAre;

	constexpr Timestamp kStartTime = Timestamp::Millis(10000);

	class TestAudioTransport : public AudioTransport {
	public:
	enum class Mode { kPlaying, kRecording };

	explicit TestAudioTransport(Mode mode) : mode_(mode) {}
	~TestAudioTransport() override = default;

	int32_t RecordedDataIsAvailable(
	const void* audioSamples,
	size_t samples_per_channel,
	size_t bytes_per_sample,
	size_t number_of_channels,
	uint32_t samples_per_second,
	uint32_t total_delay_ms,
	int32_t clock_drift,
	uint32_t current_mic_level,
	bool key_pressed,
	uint32_t& new_mic_level,
	absl::optional<int64_t> estimated_capture_time_ns) override {
	new_mic_level = 1;

	if (mode_ != Mode::kRecording) {
	EXPECT_TRUE(false)
	<< "NeedMorePlayData mustn't be called when mode isn't kRecording";
	return -1;
	}

	MutexLock lock(&mutex_);
	samples_per_channel_.push_back(samples_per_channel);
	number_of_channels_.push_back(number_of_channels);
	bytes_per_sample_.push_back(bytes_per_sample);
	samples_per_second_.push_back(samples_per_second);
	return 0;
	}

	int32_t NeedMorePlayData(size_t samples_per_channel,
	size_t bytes_per_sample,
	size_t number_of_channels,
	uint32_t samples_per_second,
	void* audio_samples,
	size_t& samples_out,
	int64_t* elapsed_time_ms,
	int64_t* ntp_time_ms) override {
	const size_t num_bytes = samples_per_channel * number_of_channels;
	std::memset(audio_samples, 1, num_bytes);
	samples_out = samples_per_channel * number_of_channels;
	*elapsed_time_ms = 0;
	*ntp_time_ms = 0;

	if (mode_ != Mode::kPlaying) {
	EXPECT_TRUE(false)
	<< "NeedMorePlayData mustn't be called when mode isn't kPlaying";
	return -1;
	}

	MutexLock lock(&mutex_);
	samples_per_channel_.push_back(samples_per_channel);
	number_of_channels_.push_back(number_of_channels);
	bytes_per_sample_.push_back(bytes_per_sample);
	samples_per_second_.push_back(samples_per_second);
	return 0;
	}

	int32_t RecordedDataIsAvailable(const void* audio_samples,
	size_t samples_per_channel,
	size_t bytes_per_sample,
	size_t number_of_channels,
	uint32_t samples_per_second,
	uint32_t total_delay_ms,
	int32_t clockDrift,
	uint32_t current_mic_level,
	bool key_pressed,
	uint32_t& new_mic_level) override {
	RTC_CHECK(false) << "This methods should be never executed";
	}

	void PullRenderData(int bits_per_sample,
	int sample_rate,
	size_t number_of_channels,
	size_t number_of_frames,
	void* audio_data,
	int64_t* elapsed_time_ms,
	int64_t* ntp_time_ms) override {
	RTC_CHECK(false) << "This methods should be never executed";
	}

	std::vector<size_t> samples_per_channel() const {
	MutexLock lock(&mutex_);
	return samples_per_channel_;
	}
	std::vector<size_t> number_of_channels() const {
	MutexLock lock(&mutex_);
	return number_of_channels_;
	}
	std::vector<size_t> bytes_per_sample() const {
	MutexLock lock(&mutex_);
	return bytes_per_sample_;
	}
	std::vector<size_t> samples_per_second() const {
	MutexLock lock(&mutex_);
	return samples_per_second_;
	}

	private:
	const Mode mode_;

	mutable Mutex mutex_;
	std::vector<size_t> samples_per_channel_ RTC_GUARDED_BY(mutex_);
	std::vector<size_t> number_of_channels_ RTC_GUARDED_BY(mutex_);
	std::vector<size_t> bytes_per_sample_ RTC_GUARDED_BY(mutex_);
	std::vector<size_t> samples_per_second_ RTC_GUARDED_BY(mutex_);
	};

	TEST(TestAudioDeviceTest, EnablingRecordingProducesAudio) {
	GlobalSimulatedTimeController time_controller(kStartTime);
	TestAudioTransport audio_transport(TestAudioTransport::Mode::kRecording);
	AudioDeviceBuffer audio_buffer(time_controller.GetTaskQueueFactory());
	ASSERT_EQ(audio_buffer.RegisterAudioCallback(&audio_transport), 0);
	std::unique_ptr<TestAudioDeviceModule::PulsedNoiseCapturer> capturer =
	TestAudioDeviceModule::CreatePulsedNoiseCapturer(
	/max_amplitude=/1000,
	/sampling_frequency_in_hz=/48000, /num_channels=/2);

	TestAudioDevice audio_device(time_controller.GetTaskQueueFactory(),
	std::move(capturer),
	/renderer=/nullptr);
	ASSERT_EQ(audio_device.Init(), AudioDeviceGeneric::InitStatus::OK);
	audio_device.AttachAudioBuffer(&audio_buffer);

	EXPECT_FALSE(audio_device.RecordingIsInitialized());
	ASSERT_EQ(audio_device.InitRecording(), 0);
	EXPECT_TRUE(audio_device.RecordingIsInitialized());
	audio_buffer.StartRecording();
	ASSERT_EQ(audio_device.StartRecording(), 0);
	time_controller.AdvanceTime(TimeDelta::Millis(10));
	ASSERT_TRUE(audio_device.Recording());
	time_controller.AdvanceTime(TimeDelta::Millis(10));
	ASSERT_EQ(audio_device.StopRecording(), 0);
	audio_buffer.StopRecording();

	EXPECT_THAT(audio_transport.samples_per_channel(),
	ElementsAre(480, 480, 480));
	EXPECT_THAT(audio_transport.number_of_channels(), ElementsAre(2, 2, 2));
	EXPECT_THAT(audio_transport.bytes_per_sample(), ElementsAre(4, 4, 4));
	EXPECT_THAT(audio_transport.samples_per_second(),
	ElementsAre(48000, 48000, 48000));
	}

	TEST(TestAudioDeviceTest, RecordingIsAvailableWhenCapturerIsSet) {
	GlobalSimulatedTimeController time_controller(kStartTime);
	std::unique_ptr<TestAudioDeviceModule::PulsedNoiseCapturer> capturer =
	TestAudioDeviceModule::CreatePulsedNoiseCapturer(
	/max_amplitude=/1000,
	/sampling_frequency_in_hz=/48000, /num_channels=/2);

	TestAudioDevice audio_device(time_controller.GetTaskQueueFactory(),
	std::move(capturer),
	/renderer=/nullptr);
	ASSERT_EQ(audio_device.Init(), AudioDeviceGeneric::InitStatus::OK);

	bool available;
	EXPECT_EQ(audio_device.RecordingIsAvailable(available), 0);
	EXPECT_TRUE(available);
	}

	TEST(TestAudioDeviceTest, RecordingIsNotAvailableWhenCapturerIsNotSet) {
	GlobalSimulatedTimeController time_controller(kStartTime);
	TestAudioDevice audio_device(time_controller.GetTaskQueueFactory(),
	/capturer=/nullptr,
	/renderer=/nullptr);
	ASSERT_EQ(audio_device.Init(), AudioDeviceGeneric::InitStatus::OK);

	bool available;
	EXPECT_EQ(audio_device.RecordingIsAvailable(available), 0);
	EXPECT_FALSE(available);
	}

	TEST(TestAudioDeviceTest, EnablingPlayoutProducesAudio) {
	GlobalSimulatedTimeController time_controller(kStartTime);
	TestAudioTransport audio_transport(TestAudioTransport::Mode::kPlaying);
	AudioDeviceBuffer audio_buffer(time_controller.GetTaskQueueFactory());
	ASSERT_EQ(audio_buffer.RegisterAudioCallback(&audio_transport), 0);
	std::unique_ptr<TestAudioDeviceModule::Renderer> renderer =
	TestAudioDeviceModule::CreateDiscardRenderer(
	/sampling_frequency_in_hz=/48000, /num_channels=/2);

	TestAudioDevice audio_device(time_controller.GetTaskQueueFactory(),
	/capturer=/nullptr, std::move(renderer));
	ASSERT_EQ(audio_device.Init(), AudioDeviceGeneric::InitStatus::OK);
	audio_device.AttachAudioBuffer(&audio_buffer);

	EXPECT_FALSE(audio_device.PlayoutIsInitialized());
	ASSERT_EQ(audio_device.InitPlayout(), 0);
	EXPECT_TRUE(audio_device.PlayoutIsInitialized());
	audio_buffer.StartPlayout();
	ASSERT_EQ(audio_device.StartPlayout(), 0);
	time_controller.AdvanceTime(TimeDelta::Millis(10));
	ASSERT_TRUE(audio_device.Playing());
	time_controller.AdvanceTime(TimeDelta::Millis(10));
	ASSERT_EQ(audio_device.StopPlayout(), 0);
	audio_buffer.StopPlayout();

	EXPECT_THAT(audio_transport.samples_per_channel(),
	ElementsAre(480, 480, 480));
	EXPECT_THAT(audio_transport.number_of_channels(), ElementsAre(2, 2, 2));
	EXPECT_THAT(audio_transport.bytes_per_sample(), ElementsAre(4, 4, 4));
	EXPECT_THAT(audio_transport.samples_per_second(),
	ElementsAre(48000, 48000, 48000));
	}

	TEST(TestAudioDeviceTest, PlayoutIsAvailableWhenRendererIsSet) {
	GlobalSimulatedTimeController time_controller(kStartTime);
	std::unique_ptr<TestAudioDeviceModule::Renderer> renderer =
	TestAudioDeviceModule::CreateDiscardRenderer(
	/sampling_frequency_in_hz=/48000, /num_channels=/2);

	TestAudioDevice audio_device(time_controller.GetTaskQueueFactory(),
	/capturer=/nullptr, std::move(renderer));
	ASSERT_EQ(audio_device.Init(), AudioDeviceGeneric::InitStatus::OK);

	bool available;
	EXPECT_EQ(audio_device.PlayoutIsAvailable(available), 0);
	EXPECT_TRUE(available);
	}

	TEST(TestAudioDeviceTest, PlayoutIsNotAvailableWhenRendererIsNotSet) {
	GlobalSimulatedTimeController time_controller(kStartTime);
	TestAudioDevice audio_device(time_controller.GetTaskQueueFactory(),
	/capturer=/nullptr,
	/renderer=/nullptr);
	ASSERT_EQ(audio_device.Init(), AudioDeviceGeneric::InitStatus::OK);

	bool available;
	EXPECT_EQ(audio_device.PlayoutIsAvailable(available), 0);
	EXPECT_FALSE(available);
	}

	} // namespace
	} // namespace webrtc