From 6bf0a5cb5034a7e684dcc3500e841785237ce2dd Mon Sep 17 00:00:00 2001 From: Daniel Baumann Date: Sun, 7 Apr 2024 19:32:43 +0200 Subject: Adding upstream version 1:115.7.0. Signed-off-by: Daniel Baumann --- .../libwebrtc/audio/audio_state_unittest.cc | 366 +++++++++++++++++++++ 1 file changed, 366 insertions(+) create mode 100644 third_party/libwebrtc/audio/audio_state_unittest.cc (limited to 'third_party/libwebrtc/audio/audio_state_unittest.cc') diff --git a/third_party/libwebrtc/audio/audio_state_unittest.cc b/third_party/libwebrtc/audio/audio_state_unittest.cc new file mode 100644 index 0000000000..4426a782d7 --- /dev/null +++ b/third_party/libwebrtc/audio/audio_state_unittest.cc @@ -0,0 +1,366 @@ +/* + * 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 +#include +#include + +#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 { + public: + FakeTaskQueue() = default; + + void Delete() override { delete this; } + void PostTask(absl::AnyInvocable task) override { + std::move(task)(); + } + }; + + class FakeTaskQueueFactory : public TaskQueueFactory { + public: + FakeTaskQueueFactory() = default; + ~FakeTaskQueueFactory() override = default; + std::unique_ptr CreateTaskQueue( + absl::string_view name, + Priority priority) const override { + return std::unique_ptr( + 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 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 audio_frame_processor_; + FakeTaskQueueFactory task_queue_factory_; + + rtc::scoped_refptr CreateFactory() { + return rtc::make_ref_counted( + 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>(); + audio_state_config.audio_device_module = + rtc::make_ref_counted>(); + 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 mixer() { return audio_mixer; } + NiceMock& + mock_audio_frame_processor() { + return async_audio_processing_helper_.audio_frame_processor_; + } + + private: + AudioState::Config audio_state_config; + rtc::scoped_refptr 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 Create10msTestData(int sample_rate_hz, + size_t num_channels) { + const int samples_per_channel = sample_rate_hz / 100; + std::vector 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(32767.f * std::sin(w)); + w += inc; + } + return audio_data; +} + +std::vector 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 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 {}; + +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 audio_state( + rtc::make_ref_counted(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 audio_state( + rtc::make_ref_counted(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(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(_))); + } + + 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 audio_state( + rtc::make_ref_counted(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(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(_))); + } + + 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 audio_state( + rtc::make_ref_counted(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 -- cgit v1.2.3