| /* | |
| * 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 <cstddef> | |
| #include <cstdint> | |
| #include <cstring> | |
| #include <memory> | |
| #include <optional> | |
| #include <utility> | |
| #include <vector> | |
| #include "api/audio/audio_device_defines.h" | |
| #include "api/environment/environment.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/test_audio_device.h" | |
| #include "rtc_base/checks.h" | |
| #include "rtc_base/synchronization/mutex.h" | |
| #include "rtc_base/task_queue_for_test.h" | |
| #include "rtc_base/thread_annotations.h" | |
| #include "test/create_test_environment.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, | |
| std::optional<int64_t> /* estimated_capture_time_ns */) override { | |
| new_mic_level = 1; | |
| if (mode_ != Mode::kRecording) { | |
| EXPECT_TRUE(false) << "RecordedDataIsAvailable 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); | |
| const Environment env = CreateTestEnvironment({.time = &time_controller}); | |
| TestAudioTransport audio_transport(TestAudioTransport::Mode::kRecording); | |
| AudioDeviceBuffer audio_buffer(env); | |
| 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(env, 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)); | |
| } | |
| // Construct an AudioDeviceBuffer on one thread, use it and delete it on a | |
| // different thread. | |
| TEST(TestAudioDeviceTest, AudioDeviceBufferOnDifferentThread) { | |
| GlobalSimulatedTimeController time_controller(kStartTime); | |
| const Environment env = CreateTestEnvironment({.time = &time_controller}); | |
| auto audio_buffer = std::make_unique<AudioDeviceBuffer>(env); | |
| TaskQueueForTest queue; | |
| queue.SendTask([&] { | |
| TestAudioTransport audio_transport(TestAudioTransport::Mode::kRecording); | |
| ASSERT_EQ(audio_buffer->RegisterAudioCallback(&audio_transport), 0); | |
| audio_buffer = nullptr; | |
| }); | |
| } | |
| 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( | |
| CreateTestEnvironment({.time = &time_controller}), 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( | |
| CreateTestEnvironment({.time = &time_controller}), | |
| /*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); | |
| const Environment env = CreateTestEnvironment({.time = &time_controller}); | |
| TestAudioTransport audio_transport(TestAudioTransport::Mode::kPlaying); | |
| AudioDeviceBuffer audio_buffer(env); | |
| 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(env, | |
| /*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( | |
| CreateTestEnvironment({.time = &time_controller}), | |
| /*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( | |
| CreateTestEnvironment({.time = &time_controller}), | |
| /*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 |