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a1ceae206b
This CL migrates unit tests to the new TaskQueueBase interface. Bug: chromium:1416199 Change-Id: Ic15c694b28eb67450ac99fdd56754de1246a4d95 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295621 Commit-Queue: Markus Handell <handellm@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39434}
368 lines
13 KiB
C++
368 lines
13 KiB
C++
/*
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* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "audio/audio_state.h"
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#include <memory>
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#include <utility>
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#include <vector>
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#include "api/task_queue/test/mock_task_queue_base.h"
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#include "call/test/mock_audio_send_stream.h"
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#include "modules/audio_device/include/mock_audio_device.h"
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#include "modules/audio_mixer/audio_mixer_impl.h"
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#include "modules/audio_processing/include/mock_audio_processing.h"
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#include "test/gtest.h"
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namespace webrtc {
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namespace test {
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namespace {
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using ::testing::_;
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using ::testing::Matcher;
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using ::testing::NiceMock;
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using ::testing::StrictMock;
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using ::testing::Values;
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constexpr int kSampleRate = 16000;
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constexpr int kNumberOfChannels = 1;
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struct FakeAsyncAudioProcessingHelper {
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class FakeTaskQueue : public StrictMock<MockTaskQueueBase> {
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public:
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FakeTaskQueue() = default;
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void Delete() override { delete this; }
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void PostTaskImpl(absl::AnyInvocable<void() &&> task,
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const PostTaskTraits& /*traits*/,
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const Location& /*location*/) override {
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std::move(task)();
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}
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};
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class FakeTaskQueueFactory : public TaskQueueFactory {
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public:
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FakeTaskQueueFactory() = default;
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~FakeTaskQueueFactory() override = default;
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std::unique_ptr<TaskQueueBase, TaskQueueDeleter> CreateTaskQueue(
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absl::string_view name,
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Priority priority) const override {
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return std::unique_ptr<webrtc::TaskQueueBase, webrtc::TaskQueueDeleter>(
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new FakeTaskQueue());
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}
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};
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class MockAudioFrameProcessor : public AudioFrameProcessor {
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public:
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~MockAudioFrameProcessor() override = default;
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MOCK_METHOD(void, ProcessCalled, ());
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MOCK_METHOD(void, SinkSet, ());
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MOCK_METHOD(void, SinkCleared, ());
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void Process(std::unique_ptr<AudioFrame> frame) override {
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ProcessCalled();
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sink_callback_(std::move(frame));
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}
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void SetSink(OnAudioFrameCallback sink_callback) override {
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sink_callback_ = std::move(sink_callback);
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if (sink_callback_ == nullptr)
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SinkCleared();
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else
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SinkSet();
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}
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private:
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OnAudioFrameCallback sink_callback_;
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};
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NiceMock<MockAudioFrameProcessor> audio_frame_processor_;
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FakeTaskQueueFactory task_queue_factory_;
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rtc::scoped_refptr<AsyncAudioProcessing::Factory> CreateFactory() {
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return rtc::make_ref_counted<AsyncAudioProcessing::Factory>(
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audio_frame_processor_, task_queue_factory_);
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}
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};
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struct ConfigHelper {
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struct Params {
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bool use_null_audio_processing;
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bool use_async_audio_processing;
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};
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explicit ConfigHelper(const Params& params)
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: audio_mixer(AudioMixerImpl::Create()) {
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audio_state_config.audio_mixer = audio_mixer;
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audio_state_config.audio_processing =
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params.use_null_audio_processing
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? nullptr
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: rtc::make_ref_counted<testing::NiceMock<MockAudioProcessing>>();
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audio_state_config.audio_device_module =
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rtc::make_ref_counted<NiceMock<MockAudioDeviceModule>>();
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if (params.use_async_audio_processing) {
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audio_state_config.async_audio_processing_factory =
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async_audio_processing_helper_.CreateFactory();
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}
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}
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AudioState::Config& config() { return audio_state_config; }
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rtc::scoped_refptr<AudioMixer> mixer() { return audio_mixer; }
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NiceMock<FakeAsyncAudioProcessingHelper::MockAudioFrameProcessor>&
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mock_audio_frame_processor() {
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return async_audio_processing_helper_.audio_frame_processor_;
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}
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private:
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AudioState::Config audio_state_config;
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rtc::scoped_refptr<AudioMixer> audio_mixer;
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FakeAsyncAudioProcessingHelper async_audio_processing_helper_;
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};
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class FakeAudioSource : public AudioMixer::Source {
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public:
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// TODO(aleloi): Valid overrides commented out, because the gmock
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// methods don't use any override declarations, and we want to avoid
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// warnings from -Winconsistent-missing-override. See
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// http://crbug.com/428099.
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int Ssrc() const /*override*/ { return 0; }
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int PreferredSampleRate() const /*override*/ { return kSampleRate; }
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MOCK_METHOD(AudioFrameInfo,
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GetAudioFrameWithInfo,
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(int sample_rate_hz, AudioFrame*),
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(override));
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};
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std::vector<int16_t> Create10msTestData(int sample_rate_hz,
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size_t num_channels) {
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const int samples_per_channel = sample_rate_hz / 100;
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std::vector<int16_t> audio_data(samples_per_channel * num_channels, 0);
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// Fill the first channel with a 1kHz sine wave.
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const float inc = (2 * 3.14159265f * 1000) / sample_rate_hz;
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float w = 0.f;
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for (int i = 0; i < samples_per_channel; ++i) {
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audio_data[i * num_channels] = static_cast<int16_t>(32767.f * std::sin(w));
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w += inc;
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}
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return audio_data;
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}
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std::vector<uint32_t> ComputeChannelLevels(AudioFrame* audio_frame) {
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const size_t num_channels = audio_frame->num_channels_;
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const size_t samples_per_channel = audio_frame->samples_per_channel_;
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std::vector<uint32_t> levels(num_channels, 0);
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for (size_t i = 0; i < samples_per_channel; ++i) {
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for (size_t j = 0; j < num_channels; ++j) {
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levels[j] += std::abs(audio_frame->data()[i * num_channels + j]);
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}
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}
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return levels;
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}
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} // namespace
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class AudioStateTest : public ::testing::TestWithParam<ConfigHelper::Params> {};
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TEST_P(AudioStateTest, Create) {
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ConfigHelper helper(GetParam());
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auto audio_state = AudioState::Create(helper.config());
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EXPECT_TRUE(audio_state.get());
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}
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TEST_P(AudioStateTest, ConstructDestruct) {
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ConfigHelper helper(GetParam());
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rtc::scoped_refptr<internal::AudioState> audio_state(
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rtc::make_ref_counted<internal::AudioState>(helper.config()));
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}
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TEST_P(AudioStateTest, RecordedAudioArrivesAtSingleStream) {
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ConfigHelper helper(GetParam());
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if (GetParam().use_async_audio_processing) {
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EXPECT_CALL(helper.mock_audio_frame_processor(), SinkSet);
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EXPECT_CALL(helper.mock_audio_frame_processor(), ProcessCalled);
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EXPECT_CALL(helper.mock_audio_frame_processor(), SinkCleared);
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}
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rtc::scoped_refptr<internal::AudioState> audio_state(
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rtc::make_ref_counted<internal::AudioState>(helper.config()));
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MockAudioSendStream stream;
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audio_state->AddSendingStream(&stream, 8000, 2);
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EXPECT_CALL(
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stream,
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SendAudioDataForMock(::testing::AllOf(
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::testing::Field(&AudioFrame::sample_rate_hz_, ::testing::Eq(8000)),
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::testing::Field(&AudioFrame::num_channels_, ::testing::Eq(2u)))))
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.WillOnce(
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// Verify that channels are not swapped by default.
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::testing::Invoke([](AudioFrame* audio_frame) {
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auto levels = ComputeChannelLevels(audio_frame);
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EXPECT_LT(0u, levels[0]);
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EXPECT_EQ(0u, levels[1]);
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}));
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MockAudioProcessing* ap =
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GetParam().use_null_audio_processing
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? nullptr
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: static_cast<MockAudioProcessing*>(audio_state->audio_processing());
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if (ap) {
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EXPECT_CALL(*ap, set_stream_delay_ms(0));
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EXPECT_CALL(*ap, set_stream_key_pressed(false));
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EXPECT_CALL(*ap, ProcessStream(_, _, _, Matcher<int16_t*>(_)));
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}
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constexpr int kSampleRate = 16000;
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constexpr size_t kNumChannels = 2;
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auto audio_data = Create10msTestData(kSampleRate, kNumChannels);
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uint32_t new_mic_level = 667;
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audio_state->audio_transport()->RecordedDataIsAvailable(
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&audio_data[0], kSampleRate / 100, kNumChannels * 2, kNumChannels,
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kSampleRate, 0, 0, 0, false, new_mic_level);
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EXPECT_EQ(667u, new_mic_level);
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audio_state->RemoveSendingStream(&stream);
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}
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TEST_P(AudioStateTest, RecordedAudioArrivesAtMultipleStreams) {
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ConfigHelper helper(GetParam());
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if (GetParam().use_async_audio_processing) {
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EXPECT_CALL(helper.mock_audio_frame_processor(), SinkSet);
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EXPECT_CALL(helper.mock_audio_frame_processor(), ProcessCalled);
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EXPECT_CALL(helper.mock_audio_frame_processor(), SinkCleared);
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}
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rtc::scoped_refptr<internal::AudioState> audio_state(
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rtc::make_ref_counted<internal::AudioState>(helper.config()));
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MockAudioSendStream stream_1;
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MockAudioSendStream stream_2;
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audio_state->AddSendingStream(&stream_1, 8001, 2);
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audio_state->AddSendingStream(&stream_2, 32000, 1);
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EXPECT_CALL(
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stream_1,
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SendAudioDataForMock(::testing::AllOf(
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::testing::Field(&AudioFrame::sample_rate_hz_, ::testing::Eq(16000)),
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::testing::Field(&AudioFrame::num_channels_, ::testing::Eq(1u)))))
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.WillOnce(
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// Verify that there is output signal.
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::testing::Invoke([](AudioFrame* audio_frame) {
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auto levels = ComputeChannelLevels(audio_frame);
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EXPECT_LT(0u, levels[0]);
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}));
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EXPECT_CALL(
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stream_2,
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SendAudioDataForMock(::testing::AllOf(
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::testing::Field(&AudioFrame::sample_rate_hz_, ::testing::Eq(16000)),
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::testing::Field(&AudioFrame::num_channels_, ::testing::Eq(1u)))))
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.WillOnce(
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// Verify that there is output signal.
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::testing::Invoke([](AudioFrame* audio_frame) {
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auto levels = ComputeChannelLevels(audio_frame);
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EXPECT_LT(0u, levels[0]);
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}));
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MockAudioProcessing* ap =
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static_cast<MockAudioProcessing*>(audio_state->audio_processing());
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if (ap) {
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EXPECT_CALL(*ap, set_stream_delay_ms(5));
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EXPECT_CALL(*ap, set_stream_key_pressed(true));
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EXPECT_CALL(*ap, ProcessStream(_, _, _, Matcher<int16_t*>(_)));
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}
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constexpr int kSampleRate = 16000;
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constexpr size_t kNumChannels = 1;
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auto audio_data = Create10msTestData(kSampleRate, kNumChannels);
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uint32_t new_mic_level = 667;
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audio_state->audio_transport()->RecordedDataIsAvailable(
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&audio_data[0], kSampleRate / 100, kNumChannels * 2, kNumChannels,
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kSampleRate, 5, 0, 0, true, new_mic_level);
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EXPECT_EQ(667u, new_mic_level);
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audio_state->RemoveSendingStream(&stream_1);
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audio_state->RemoveSendingStream(&stream_2);
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}
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TEST_P(AudioStateTest, EnableChannelSwap) {
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constexpr int kSampleRate = 16000;
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constexpr size_t kNumChannels = 2;
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ConfigHelper helper(GetParam());
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if (GetParam().use_async_audio_processing) {
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EXPECT_CALL(helper.mock_audio_frame_processor(), SinkSet);
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EXPECT_CALL(helper.mock_audio_frame_processor(), ProcessCalled);
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EXPECT_CALL(helper.mock_audio_frame_processor(), SinkCleared);
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}
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rtc::scoped_refptr<internal::AudioState> audio_state(
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rtc::make_ref_counted<internal::AudioState>(helper.config()));
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audio_state->SetStereoChannelSwapping(true);
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MockAudioSendStream stream;
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audio_state->AddSendingStream(&stream, kSampleRate, kNumChannels);
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EXPECT_CALL(stream, SendAudioDataForMock(_))
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.WillOnce(
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// Verify that channels are swapped.
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::testing::Invoke([](AudioFrame* audio_frame) {
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auto levels = ComputeChannelLevels(audio_frame);
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EXPECT_EQ(0u, levels[0]);
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EXPECT_LT(0u, levels[1]);
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}));
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auto audio_data = Create10msTestData(kSampleRate, kNumChannels);
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uint32_t new_mic_level = 667;
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audio_state->audio_transport()->RecordedDataIsAvailable(
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&audio_data[0], kSampleRate / 100, kNumChannels * 2, kNumChannels,
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kSampleRate, 0, 0, 0, false, new_mic_level);
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EXPECT_EQ(667u, new_mic_level);
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audio_state->RemoveSendingStream(&stream);
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}
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TEST_P(AudioStateTest,
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QueryingTransportForAudioShouldResultInGetAudioCallOnMixerSource) {
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ConfigHelper helper(GetParam());
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auto audio_state = AudioState::Create(helper.config());
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FakeAudioSource fake_source;
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helper.mixer()->AddSource(&fake_source);
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EXPECT_CALL(fake_source, GetAudioFrameWithInfo(_, _))
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.WillOnce(
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::testing::Invoke([](int sample_rate_hz, AudioFrame* audio_frame) {
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audio_frame->sample_rate_hz_ = sample_rate_hz;
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audio_frame->samples_per_channel_ = sample_rate_hz / 100;
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audio_frame->num_channels_ = kNumberOfChannels;
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return AudioMixer::Source::AudioFrameInfo::kNormal;
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}));
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int16_t audio_buffer[kSampleRate / 100 * kNumberOfChannels];
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size_t n_samples_out;
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int64_t elapsed_time_ms;
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int64_t ntp_time_ms;
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audio_state->audio_transport()->NeedMorePlayData(
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kSampleRate / 100, kNumberOfChannels * 2, kNumberOfChannels, kSampleRate,
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audio_buffer, n_samples_out, &elapsed_time_ms, &ntp_time_ms);
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}
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INSTANTIATE_TEST_SUITE_P(AudioStateTest,
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AudioStateTest,
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Values(ConfigHelper::Params({false, false}),
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ConfigHelper::Params({true, false}),
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ConfigHelper::Params({false, true}),
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ConfigHelper::Params({true, true})));
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} // namespace test
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} // namespace webrtc
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