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webrtc/modules/audio_device/test_audio_device_impl_test.cc
Florent Castelli f4673f97ed Move webrtc::AudioDeviceModule include to api/ folder 
Bug: webrtc:15874
Change-Id: I5bdb19d5e710838b41e6ca283d406c9f1f21286b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/348060
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42137}
2024-04-22 08:56:31 +00:00

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/*
* 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/audio/audio_device.h"
#include "api/audio/audio_device_defines.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/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) << "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);
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
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