blob: 48e54c7dc47a8280ac56002805d3afeb0e056dfd [file] [log] [blame]
/*
* Copyright (c) 2017 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/include/audio_device.h"
#include <algorithm>
#include <cstring>
#include <memory>
#include <numeric>
#include "absl/memory/memory.h"
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/scoped_refptr.h"
#include "api/task_queue/default_task_queue_factory.h"
#include "api/task_queue/task_queue_factory.h"
#include "modules/audio_device/audio_device_impl.h"
#include "modules/audio_device/include/mock_audio_transport.h"
#include "rtc_base/buffer.h"
#include "rtc_base/critical_section.h"
#include "rtc_base/event.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/race_checker.h"
#include "rtc_base/thread_annotations.h"
#include "rtc_base/thread_checker.h"
#include "rtc_base/time_utils.h"
#include "system_wrappers/include/sleep.h"
#include "test/gmock.h"
#include "test/gtest.h"
#ifdef WEBRTC_WIN
#include "modules/audio_device/include/audio_device_factory.h"
#include "modules/audio_device/win/core_audio_utility_win.h"
#endif
using ::testing::_;
using ::testing::AtLeast;
using ::testing::Ge;
using ::testing::Invoke;
using ::testing::Mock;
using ::testing::NiceMock;
using ::testing::NotNull;
namespace webrtc {
namespace {
// Using a #define for AUDIO_DEVICE since we will call *different* versions of
// the ADM functions, depending on the ID type.
#if defined(WEBRTC_WIN)
#define AUDIO_DEVICE_ID (AudioDeviceModule::WindowsDeviceType::kDefaultDevice)
#else
#define AUDIO_DEVICE_ID (0u)
#endif // defined(WEBRTC_WIN)
// #define ENABLE_DEBUG_PRINTF
#ifdef ENABLE_DEBUG_PRINTF
#define PRINTD(...) fprintf(stderr, __VA_ARGS__);
#else
#define PRINTD(...) ((void)0)
#endif
#define PRINT(...) fprintf(stderr, __VA_ARGS__);
// Don't run these tests if audio-related requirements are not met.
#define SKIP_TEST_IF_NOT(requirements_satisfied) \
do { \
if (!requirements_satisfied) { \
GTEST_SKIP() << "Skipped. No audio device found."; \
} \
} while (false)
// Number of callbacks (input or output) the tests waits for before we set
// an event indicating that the test was OK.
static constexpr size_t kNumCallbacks = 10;
// Max amount of time we wait for an event to be set while counting callbacks.
static constexpr size_t kTestTimeOutInMilliseconds = 10 * 1000;
// Average number of audio callbacks per second assuming 10ms packet size.
static constexpr size_t kNumCallbacksPerSecond = 100;
// Run the full-duplex test during this time (unit is in seconds).
static constexpr size_t kFullDuplexTimeInSec = 5;
// Length of round-trip latency measurements. Number of deteced impulses
// shall be kImpulseFrequencyInHz * kMeasureLatencyTimeInSec - 1 since the
// last transmitted pulse is not used.
static constexpr size_t kMeasureLatencyTimeInSec = 10;
// Sets the number of impulses per second in the latency test.
static constexpr size_t kImpulseFrequencyInHz = 1;
// Utilized in round-trip latency measurements to avoid capturing noise samples.
static constexpr int kImpulseThreshold = 1000;
enum class TransportType {
kInvalid,
kPlay,
kRecord,
kPlayAndRecord,
};
// Interface for processing the audio stream. Real implementations can e.g.
// run audio in loopback, read audio from a file or perform latency
// measurements.
class AudioStream {
public:
virtual void Write(rtc::ArrayView<const int16_t> source) = 0;
virtual void Read(rtc::ArrayView<int16_t> destination) = 0;
virtual ~AudioStream() = default;
};
// Converts index corresponding to position within a 10ms buffer into a
// delay value in milliseconds.
// Example: index=240, frames_per_10ms_buffer=480 => 5ms as output.
int IndexToMilliseconds(size_t index, size_t frames_per_10ms_buffer) {
return rtc::checked_cast<int>(
10.0 * (static_cast<double>(index) / frames_per_10ms_buffer) + 0.5);
}
} // namespace
// Simple first in first out (FIFO) class that wraps a list of 16-bit audio
// buffers of fixed size and allows Write and Read operations. The idea is to
// store recorded audio buffers (using Write) and then read (using Read) these
// stored buffers with as short delay as possible when the audio layer needs
// data to play out. The number of buffers in the FIFO will stabilize under
// normal conditions since there will be a balance between Write and Read calls.
// The container is a std::list container and access is protected with a lock
// since both sides (playout and recording) are driven by its own thread.
// Note that, we know by design that the size of the audio buffer will not
// change over time and that both sides will in most cases use the same size.
class FifoAudioStream : public AudioStream {
public:
void Write(rtc::ArrayView<const int16_t> source) override {
RTC_DCHECK_RUNS_SERIALIZED(&race_checker_);
const size_t size = [&] {
rtc::CritScope lock(&lock_);
fifo_.push_back(Buffer16(source.data(), source.size()));
return fifo_.size();
}();
if (size > max_size_) {
max_size_ = size;
}
// Add marker once per second to signal that audio is active.
if (write_count_++ % 100 == 0) {
PRINTD(".");
}
written_elements_ += size;
}
void Read(rtc::ArrayView<int16_t> destination) override {
rtc::CritScope lock(&lock_);
if (fifo_.empty()) {
std::fill(destination.begin(), destination.end(), 0);
} else {
const Buffer16& buffer = fifo_.front();
if (buffer.size() == destination.size()) {
// Default case where input and output uses same sample rate and
// channel configuration. No conversion is needed.
std::copy(buffer.begin(), buffer.end(), destination.begin());
} else if (destination.size() == 2 * buffer.size()) {
// Recorded input signal in |buffer| is in mono. Do channel upmix to
// match stereo output (1 -> 2).
for (size_t i = 0; i < buffer.size(); ++i) {
destination[2 * i] = buffer[i];
destination[2 * i + 1] = buffer[i];
}
} else if (buffer.size() == 2 * destination.size()) {
// Recorded input signal in |buffer| is in stereo. Do channel downmix
// to match mono output (2 -> 1).
for (size_t i = 0; i < destination.size(); ++i) {
destination[i] =
(static_cast<int32_t>(buffer[2 * i]) + buffer[2 * i + 1]) / 2;
}
} else {
RTC_NOTREACHED() << "Required conversion is not support";
}
fifo_.pop_front();
}
}
size_t size() const {
rtc::CritScope lock(&lock_);
return fifo_.size();
}
size_t max_size() const {
RTC_DCHECK_RUNS_SERIALIZED(&race_checker_);
return max_size_;
}
size_t average_size() const {
RTC_DCHECK_RUNS_SERIALIZED(&race_checker_);
return 0.5 + static_cast<float>(written_elements_ / write_count_);
}
using Buffer16 = rtc::BufferT<int16_t>;
rtc::CriticalSection lock_;
rtc::RaceChecker race_checker_;
std::list<Buffer16> fifo_ RTC_GUARDED_BY(lock_);
size_t write_count_ RTC_GUARDED_BY(race_checker_) = 0;
size_t max_size_ RTC_GUARDED_BY(race_checker_) = 0;
size_t written_elements_ RTC_GUARDED_BY(race_checker_) = 0;
};
// Inserts periodic impulses and measures the latency between the time of
// transmission and time of receiving the same impulse.
class LatencyAudioStream : public AudioStream {
public:
LatencyAudioStream() {
// Delay thread checkers from being initialized until first callback from
// respective thread.
read_thread_checker_.Detach();
write_thread_checker_.Detach();
}
// Insert periodic impulses in first two samples of |destination|.
void Read(rtc::ArrayView<int16_t> destination) override {
RTC_DCHECK_RUN_ON(&read_thread_checker_);
if (read_count_ == 0) {
PRINT("[");
}
read_count_++;
std::fill(destination.begin(), destination.end(), 0);
if (read_count_ % (kNumCallbacksPerSecond / kImpulseFrequencyInHz) == 0) {
PRINT(".");
{
rtc::CritScope lock(&lock_);
if (!pulse_time_) {
pulse_time_ = rtc::TimeMillis();
}
}
constexpr int16_t impulse = std::numeric_limits<int16_t>::max();
std::fill_n(destination.begin(), 2, impulse);
}
}
// Detect received impulses in |source|, derive time between transmission and
// detection and add the calculated delay to list of latencies.
void Write(rtc::ArrayView<const int16_t> source) override {
RTC_DCHECK_RUN_ON(&write_thread_checker_);
RTC_DCHECK_RUNS_SERIALIZED(&race_checker_);
rtc::CritScope lock(&lock_);
write_count_++;
if (!pulse_time_) {
// Avoid detection of new impulse response until a new impulse has
// been transmitted (sets |pulse_time_| to value larger than zero).
return;
}
// Find index (element position in vector) of the max element.
const size_t index_of_max =
std::max_element(source.begin(), source.end()) - source.begin();
// Derive time between transmitted pulse and received pulse if the level
// is high enough (removes noise).
const size_t max = source[index_of_max];
if (max > kImpulseThreshold) {
PRINTD("(%zu, %zu)", max, index_of_max);
int64_t now_time = rtc::TimeMillis();
int extra_delay = IndexToMilliseconds(index_of_max, source.size());
PRINTD("[%d]", rtc::checked_cast<int>(now_time - pulse_time_));
PRINTD("[%d]", extra_delay);
// Total latency is the difference between transmit time and detection
// tome plus the extra delay within the buffer in which we detected the
// received impulse. It is transmitted at sample 0 but can be received
// at sample N where N > 0. The term |extra_delay| accounts for N and it
// is a value between 0 and 10ms.
latencies_.push_back(now_time - *pulse_time_ + extra_delay);
pulse_time_.reset();
} else {
PRINTD("-");
}
}
size_t num_latency_values() const {
RTC_DCHECK_RUNS_SERIALIZED(&race_checker_);
return latencies_.size();
}
int min_latency() const {
RTC_DCHECK_RUNS_SERIALIZED(&race_checker_);
if (latencies_.empty())
return 0;
return *std::min_element(latencies_.begin(), latencies_.end());
}
int max_latency() const {
RTC_DCHECK_RUNS_SERIALIZED(&race_checker_);
if (latencies_.empty())
return 0;
return *std::max_element(latencies_.begin(), latencies_.end());
}
int average_latency() const {
RTC_DCHECK_RUNS_SERIALIZED(&race_checker_);
if (latencies_.empty())
return 0;
return 0.5 + static_cast<double>(
std::accumulate(latencies_.begin(), latencies_.end(), 0)) /
latencies_.size();
}
void PrintResults() const {
RTC_DCHECK_RUNS_SERIALIZED(&race_checker_);
PRINT("] ");
for (auto it = latencies_.begin(); it != latencies_.end(); ++it) {
PRINTD("%d ", *it);
}
PRINT("\n");
PRINT("[..........] [min, max, avg]=[%d, %d, %d] ms\n", min_latency(),
max_latency(), average_latency());
}
rtc::CriticalSection lock_;
rtc::RaceChecker race_checker_;
rtc::ThreadChecker read_thread_checker_;
rtc::ThreadChecker write_thread_checker_;
absl::optional<int64_t> pulse_time_ RTC_GUARDED_BY(lock_);
std::vector<int> latencies_ RTC_GUARDED_BY(race_checker_);
size_t read_count_ RTC_GUARDED_BY(read_thread_checker_) = 0;
size_t write_count_ RTC_GUARDED_BY(write_thread_checker_) = 0;
};
// Mocks the AudioTransport object and proxies actions for the two callbacks
// (RecordedDataIsAvailable and NeedMorePlayData) to different implementations
// of AudioStreamInterface.
class MockAudioTransport : public test::MockAudioTransport {
public:
explicit MockAudioTransport(TransportType type) : type_(type) {}
~MockAudioTransport() {}
// Set default actions of the mock object. We are delegating to fake
// implementation where the number of callbacks is counted and an event
// is set after a certain number of callbacks. Audio parameters are also
// checked.
void HandleCallbacks(rtc::Event* event,
AudioStream* audio_stream,
int num_callbacks) {
event_ = event;
audio_stream_ = audio_stream;
num_callbacks_ = num_callbacks;
if (play_mode()) {
ON_CALL(*this, NeedMorePlayData(_, _, _, _, _, _, _, _))
.WillByDefault(
Invoke(this, &MockAudioTransport::RealNeedMorePlayData));
}
if (rec_mode()) {
ON_CALL(*this, RecordedDataIsAvailable(_, _, _, _, _, _, _, _, _, _))
.WillByDefault(
Invoke(this, &MockAudioTransport::RealRecordedDataIsAvailable));
}
}
// Special constructor used in manual tests where the user wants to run audio
// until e.g. a keyboard key is pressed. The event flag is set to nullptr by
// default since it is up to the user to stop the test. See e.g.
// DISABLED_RunPlayoutAndRecordingInFullDuplexAndWaitForEnterKey().
void HandleCallbacks(AudioStream* audio_stream) {
HandleCallbacks(nullptr, audio_stream, 0);
}
int32_t RealRecordedDataIsAvailable(const void* audio_buffer,
const size_t samples_per_channel,
const size_t bytes_per_frame,
const size_t channels,
const uint32_t sample_rate,
const uint32_t total_delay_ms,
const int32_t clock_drift,
const uint32_t current_mic_level,
const bool typing_status,
uint32_t& new_mic_level) {
EXPECT_TRUE(rec_mode()) << "No test is expecting these callbacks.";
// Store audio parameters once in the first callback. For all other
// callbacks, verify that the provided audio parameters are maintained and
// that each callback corresponds to 10ms for any given sample rate.
if (!record_parameters_.is_complete()) {
record_parameters_.reset(sample_rate, channels, samples_per_channel);
} else {
EXPECT_EQ(samples_per_channel, record_parameters_.frames_per_buffer());
EXPECT_EQ(bytes_per_frame, record_parameters_.GetBytesPerFrame());
EXPECT_EQ(channels, record_parameters_.channels());
EXPECT_EQ(static_cast<int>(sample_rate),
record_parameters_.sample_rate());
EXPECT_EQ(samples_per_channel,
record_parameters_.frames_per_10ms_buffer());
}
{
rtc::CritScope lock(&lock_);
rec_count_++;
}
// Write audio data to audio stream object if one has been injected.
if (audio_stream_) {
audio_stream_->Write(
rtc::MakeArrayView(static_cast<const int16_t*>(audio_buffer),
samples_per_channel * channels));
}
// Signal the event after given amount of callbacks.
if (event_ && ReceivedEnoughCallbacks()) {
event_->Set();
}
return 0;
}
int32_t RealNeedMorePlayData(const size_t samples_per_channel,
const size_t bytes_per_frame,
const size_t channels,
const uint32_t sample_rate,
void* audio_buffer,
size_t& samples_out,
int64_t* elapsed_time_ms,
int64_t* ntp_time_ms) {
EXPECT_TRUE(play_mode()) << "No test is expecting these callbacks.";
// Store audio parameters once in the first callback. For all other
// callbacks, verify that the provided audio parameters are maintained and
// that each callback corresponds to 10ms for any given sample rate.
if (!playout_parameters_.is_complete()) {
playout_parameters_.reset(sample_rate, channels, samples_per_channel);
} else {
EXPECT_EQ(samples_per_channel, playout_parameters_.frames_per_buffer());
EXPECT_EQ(bytes_per_frame, playout_parameters_.GetBytesPerFrame());
EXPECT_EQ(channels, playout_parameters_.channels());
EXPECT_EQ(static_cast<int>(sample_rate),
playout_parameters_.sample_rate());
EXPECT_EQ(samples_per_channel,
playout_parameters_.frames_per_10ms_buffer());
}
{
rtc::CritScope lock(&lock_);
play_count_++;
}
samples_out = samples_per_channel * channels;
// Read audio data from audio stream object if one has been injected.
if (audio_stream_) {
audio_stream_->Read(rtc::MakeArrayView(
static_cast<int16_t*>(audio_buffer), samples_per_channel * channels));
} else {
// Fill the audio buffer with zeros to avoid disturbing audio.
const size_t num_bytes = samples_per_channel * bytes_per_frame;
std::memset(audio_buffer, 0, num_bytes);
}
// Signal the event after given amount of callbacks.
if (event_ && ReceivedEnoughCallbacks()) {
event_->Set();
}
return 0;
}
bool ReceivedEnoughCallbacks() {
bool recording_done = false;
if (rec_mode()) {
rtc::CritScope lock(&lock_);
recording_done = rec_count_ >= num_callbacks_;
} else {
recording_done = true;
}
bool playout_done = false;
if (play_mode()) {
rtc::CritScope lock(&lock_);
playout_done = play_count_ >= num_callbacks_;
} else {
playout_done = true;
}
return recording_done && playout_done;
}
bool play_mode() const {
return type_ == TransportType::kPlay ||
type_ == TransportType::kPlayAndRecord;
}
bool rec_mode() const {
return type_ == TransportType::kRecord ||
type_ == TransportType::kPlayAndRecord;
}
void ResetCallbackCounters() {
rtc::CritScope lock(&lock_);
if (play_mode()) {
play_count_ = 0;
}
if (rec_mode()) {
rec_count_ = 0;
}
}
private:
rtc::CriticalSection lock_;
TransportType type_ = TransportType::kInvalid;
rtc::Event* event_ = nullptr;
AudioStream* audio_stream_ = nullptr;
size_t num_callbacks_ = 0;
size_t play_count_ RTC_GUARDED_BY(lock_) = 0;
size_t rec_count_ RTC_GUARDED_BY(lock_) = 0;
AudioParameters playout_parameters_;
AudioParameters record_parameters_;
};
// AudioDeviceTest test fixture.
// Don't run these tests in combination with sanitizers.
// TODO(webrtc:9778): Re-enable on THREAD_SANITIZER?
#if defined(ADDRESS_SANITIZER) || defined(MEMORY_SANITIZER) || \
defined(THREAD_SANITIZER)
#define MAYBE_AudioDeviceTest DISABLED_AudioDeviceTest
#else
#define MAYBE_AudioDeviceTest AudioDeviceTest
#endif
class MAYBE_AudioDeviceTest
: public ::testing::TestWithParam<webrtc::AudioDeviceModule::AudioLayer> {
protected:
MAYBE_AudioDeviceTest()
: audio_layer_(GetParam()),
task_queue_factory_(CreateDefaultTaskQueueFactory()) {
rtc::LogMessage::LogToDebug(rtc::LS_INFO);
// Add extra logging fields here if needed for debugging.
rtc::LogMessage::LogTimestamps();
rtc::LogMessage::LogThreads();
audio_device_ = CreateAudioDevice();
EXPECT_NE(audio_device_.get(), nullptr);
AudioDeviceModule::AudioLayer audio_layer;
int got_platform_audio_layer =
audio_device_->ActiveAudioLayer(&audio_layer);
// First, ensure that a valid audio layer can be activated.
if (got_platform_audio_layer != 0) {
requirements_satisfied_ = false;
}
// Next, verify that the ADM can be initialized.
if (requirements_satisfied_) {
requirements_satisfied_ = (audio_device_->Init() == 0);
}
// Finally, ensure that at least one valid device exists in each direction.
if (requirements_satisfied_) {
const int16_t num_playout_devices = audio_device_->PlayoutDevices();
const int16_t num_record_devices = audio_device_->RecordingDevices();
requirements_satisfied_ =
num_playout_devices > 0 && num_record_devices > 0;
}
if (requirements_satisfied_) {
EXPECT_EQ(0, audio_device_->SetPlayoutDevice(AUDIO_DEVICE_ID));
EXPECT_EQ(0, audio_device_->InitSpeaker());
EXPECT_EQ(0, audio_device_->StereoPlayoutIsAvailable(&stereo_playout_));
EXPECT_EQ(0, audio_device_->SetStereoPlayout(stereo_playout_));
EXPECT_EQ(0, audio_device_->SetRecordingDevice(AUDIO_DEVICE_ID));
EXPECT_EQ(0, audio_device_->InitMicrophone());
// Avoid asking for input stereo support and always record in mono
// since asking can cause issues in combination with remote desktop.
// See https://bugs.chromium.org/p/webrtc/issues/detail?id=7397 for
// details.
EXPECT_EQ(0, audio_device_->SetStereoRecording(false));
}
}
virtual ~MAYBE_AudioDeviceTest() {
if (audio_device_) {
EXPECT_EQ(0, audio_device_->Terminate());
}
}
bool requirements_satisfied() const { return requirements_satisfied_; }
rtc::Event* event() { return &event_; }
AudioDeviceModule::AudioLayer audio_layer() const { return audio_layer_; }
// AudioDeviceModuleForTest extends the default ADM interface with some extra
// test methods. Intended for usage in tests only and requires a unique
// factory method. See CreateAudioDevice() for details.
const rtc::scoped_refptr<AudioDeviceModuleForTest>& audio_device() const {
return audio_device_;
}
rtc::scoped_refptr<AudioDeviceModuleForTest> CreateAudioDevice() {
// Use the default factory for kPlatformDefaultAudio and a special factory
// CreateWindowsCoreAudioAudioDeviceModuleForTest() for kWindowsCoreAudio2.
// The value of |audio_layer_| is set at construction by GetParam() and two
// different layers are tested on Windows only.
if (audio_layer_ == AudioDeviceModule::kPlatformDefaultAudio) {
return AudioDeviceModule::CreateForTest(audio_layer_,
task_queue_factory_.get());
} else if (audio_layer_ == AudioDeviceModule::kWindowsCoreAudio2) {
#ifdef WEBRTC_WIN
// We must initialize the COM library on a thread before we calling any of
// the library functions. All COM functions in the ADM will return
// CO_E_NOTINITIALIZED otherwise.
com_initializer_ = absl::make_unique<webrtc_win::ScopedCOMInitializer>(
webrtc_win::ScopedCOMInitializer::kMTA);
EXPECT_TRUE(com_initializer_->Succeeded());
EXPECT_TRUE(webrtc_win::core_audio_utility::IsSupported());
EXPECT_TRUE(webrtc_win::core_audio_utility::IsMMCSSSupported());
return CreateWindowsCoreAudioAudioDeviceModuleForTest(
task_queue_factory_.get(), true);
#else
return nullptr;
#endif
} else {
return nullptr;
}
}
void StartPlayout() {
EXPECT_FALSE(audio_device()->Playing());
EXPECT_EQ(0, audio_device()->InitPlayout());
EXPECT_TRUE(audio_device()->PlayoutIsInitialized());
EXPECT_EQ(0, audio_device()->StartPlayout());
EXPECT_TRUE(audio_device()->Playing());
}
void StopPlayout() {
EXPECT_EQ(0, audio_device()->StopPlayout());
EXPECT_FALSE(audio_device()->Playing());
EXPECT_FALSE(audio_device()->PlayoutIsInitialized());
}
void StartRecording() {
EXPECT_FALSE(audio_device()->Recording());
EXPECT_EQ(0, audio_device()->InitRecording());
EXPECT_TRUE(audio_device()->RecordingIsInitialized());
EXPECT_EQ(0, audio_device()->StartRecording());
EXPECT_TRUE(audio_device()->Recording());
}
void StopRecording() {
EXPECT_EQ(0, audio_device()->StopRecording());
EXPECT_FALSE(audio_device()->Recording());
EXPECT_FALSE(audio_device()->RecordingIsInitialized());
}
bool NewWindowsAudioDeviceModuleIsUsed() {
#ifdef WEBRTC_WIN
AudioDeviceModule::AudioLayer audio_layer;
EXPECT_EQ(0, audio_device()->ActiveAudioLayer(&audio_layer));
if (audio_layer == AudioDeviceModule::kWindowsCoreAudio2) {
// Default device is always added as first element in the list and the
// default communication device as the second element. Hence, the list
// contains two extra elements in this case.
return true;
}
#endif
return false;
}
private:
#ifdef WEBRTC_WIN
// Windows Core Audio based ADM needs to run on a COM initialized thread.
std::unique_ptr<webrtc_win::ScopedCOMInitializer> com_initializer_;
#endif
AudioDeviceModule::AudioLayer audio_layer_;
std::unique_ptr<TaskQueueFactory> task_queue_factory_;
bool requirements_satisfied_ = true;
rtc::Event event_;
rtc::scoped_refptr<AudioDeviceModuleForTest> audio_device_;
bool stereo_playout_ = false;
};
// Instead of using the test fixture, verify that the different factory methods
// work as intended.
TEST(AudioDeviceTestWin, ConstructDestructWithFactory) {
std::unique_ptr<TaskQueueFactory> task_queue_factory =
CreateDefaultTaskQueueFactory();
rtc::scoped_refptr<AudioDeviceModule> audio_device;
// The default factory should work for all platforms when a default ADM is
// requested.
audio_device = AudioDeviceModule::Create(
AudioDeviceModule::kPlatformDefaultAudio, task_queue_factory.get());
EXPECT_TRUE(audio_device);
audio_device = nullptr;
#ifdef WEBRTC_WIN
// For Windows, the old factory method creates an ADM where the platform-
// specific parts are implemented by an AudioDeviceGeneric object. Verify
// that the old factory can't be used in combination with the latest audio
// layer AudioDeviceModule::kWindowsCoreAudio2.
audio_device = AudioDeviceModule::Create(
AudioDeviceModule::kWindowsCoreAudio2, task_queue_factory.get());
EXPECT_FALSE(audio_device);
audio_device = nullptr;
// Instead, ensure that the new dedicated factory method called
// CreateWindowsCoreAudioAudioDeviceModule() can be used on Windows and that
// it sets the audio layer to kWindowsCoreAudio2 implicitly. Note that, the
// new ADM for Windows must be created on a COM thread.
webrtc_win::ScopedCOMInitializer com_initializer(
webrtc_win::ScopedCOMInitializer::kMTA);
EXPECT_TRUE(com_initializer.Succeeded());
audio_device =
CreateWindowsCoreAudioAudioDeviceModule(task_queue_factory.get());
EXPECT_TRUE(audio_device);
AudioDeviceModule::AudioLayer audio_layer;
EXPECT_EQ(0, audio_device->ActiveAudioLayer(&audio_layer));
EXPECT_EQ(audio_layer, AudioDeviceModule::kWindowsCoreAudio2);
#endif
}
// Uses the test fixture to create, initialize and destruct the ADM.
TEST_P(MAYBE_AudioDeviceTest, ConstructDestructDefault) {}
TEST_P(MAYBE_AudioDeviceTest, InitTerminate) {
SKIP_TEST_IF_NOT(requirements_satisfied());
// Initialization is part of the test fixture.
EXPECT_TRUE(audio_device()->Initialized());
EXPECT_EQ(0, audio_device()->Terminate());
EXPECT_FALSE(audio_device()->Initialized());
}
// Enumerate all available and active output devices.
TEST_P(MAYBE_AudioDeviceTest, PlayoutDeviceNames) {
SKIP_TEST_IF_NOT(requirements_satisfied());
char device_name[kAdmMaxDeviceNameSize];
char unique_id[kAdmMaxGuidSize];
int num_devices = audio_device()->PlayoutDevices();
if (NewWindowsAudioDeviceModuleIsUsed()) {
num_devices += 2;
}
EXPECT_GT(num_devices, 0);
for (int i = 0; i < num_devices; ++i) {
EXPECT_EQ(0, audio_device()->PlayoutDeviceName(i, device_name, unique_id));
}
EXPECT_EQ(-1, audio_device()->PlayoutDeviceName(num_devices, device_name,
unique_id));
}
// Enumerate all available and active input devices.
TEST_P(MAYBE_AudioDeviceTest, RecordingDeviceNames) {
SKIP_TEST_IF_NOT(requirements_satisfied());
char device_name[kAdmMaxDeviceNameSize];
char unique_id[kAdmMaxGuidSize];
int num_devices = audio_device()->RecordingDevices();
if (NewWindowsAudioDeviceModuleIsUsed()) {
num_devices += 2;
}
EXPECT_GT(num_devices, 0);
for (int i = 0; i < num_devices; ++i) {
EXPECT_EQ(0,
audio_device()->RecordingDeviceName(i, device_name, unique_id));
}
EXPECT_EQ(-1, audio_device()->RecordingDeviceName(num_devices, device_name,
unique_id));
}
// Counts number of active output devices and ensure that all can be selected.
TEST_P(MAYBE_AudioDeviceTest, SetPlayoutDevice) {
SKIP_TEST_IF_NOT(requirements_satisfied());
int num_devices = audio_device()->PlayoutDevices();
if (NewWindowsAudioDeviceModuleIsUsed()) {
num_devices += 2;
}
EXPECT_GT(num_devices, 0);
// Verify that all available playout devices can be set (not enabled yet).
for (int i = 0; i < num_devices; ++i) {
EXPECT_EQ(0, audio_device()->SetPlayoutDevice(i));
}
EXPECT_EQ(-1, audio_device()->SetPlayoutDevice(num_devices));
#ifdef WEBRTC_WIN
// On Windows, verify the alternative method where the user can select device
// by role.
EXPECT_EQ(
0, audio_device()->SetPlayoutDevice(AudioDeviceModule::kDefaultDevice));
EXPECT_EQ(0, audio_device()->SetPlayoutDevice(
AudioDeviceModule::kDefaultCommunicationDevice));
#endif
}
// Counts number of active input devices and ensure that all can be selected.
TEST_P(MAYBE_AudioDeviceTest, SetRecordingDevice) {
SKIP_TEST_IF_NOT(requirements_satisfied());
int num_devices = audio_device()->RecordingDevices();
if (NewWindowsAudioDeviceModuleIsUsed()) {
num_devices += 2;
}
EXPECT_GT(num_devices, 0);
// Verify that all available recording devices can be set (not enabled yet).
for (int i = 0; i < num_devices; ++i) {
EXPECT_EQ(0, audio_device()->SetRecordingDevice(i));
}
EXPECT_EQ(-1, audio_device()->SetRecordingDevice(num_devices));
#ifdef WEBRTC_WIN
// On Windows, verify the alternative method where the user can select device
// by role.
EXPECT_EQ(
0, audio_device()->SetRecordingDevice(AudioDeviceModule::kDefaultDevice));
EXPECT_EQ(0, audio_device()->SetRecordingDevice(
AudioDeviceModule::kDefaultCommunicationDevice));
#endif
}
// Tests Start/Stop playout without any registered audio callback.
TEST_P(MAYBE_AudioDeviceTest, StartStopPlayout) {
SKIP_TEST_IF_NOT(requirements_satisfied());
StartPlayout();
StopPlayout();
}
// Tests Start/Stop recording without any registered audio callback.
TEST_P(MAYBE_AudioDeviceTest, StartStopRecording) {
SKIP_TEST_IF_NOT(requirements_satisfied());
StartRecording();
StopRecording();
}
// Tests Init/Stop/Init recording without any registered audio callback.
// See https://bugs.chromium.org/p/webrtc/issues/detail?id=8041 for details
// on why this test is useful.
TEST_P(MAYBE_AudioDeviceTest, InitStopInitRecording) {
SKIP_TEST_IF_NOT(requirements_satisfied());
EXPECT_EQ(0, audio_device()->InitRecording());
EXPECT_TRUE(audio_device()->RecordingIsInitialized());
StopRecording();
EXPECT_EQ(0, audio_device()->InitRecording());
StopRecording();
}
// Verify that additional attempts to initialize or start recording while
// already being active works. Additional calls should just be ignored.
TEST_P(MAYBE_AudioDeviceTest, StartInitRecording) {
SKIP_TEST_IF_NOT(requirements_satisfied());
StartRecording();
// An additional attempt to initialize at this stage should be ignored.
EXPECT_EQ(0, audio_device()->InitRecording());
// Same for additional request to start recording while already active.
EXPECT_EQ(0, audio_device()->StartRecording());
StopRecording();
}
// Verify that additional attempts to initialize or start playou while
// already being active works. Additional calls should just be ignored.
TEST_P(MAYBE_AudioDeviceTest, StartInitPlayout) {
SKIP_TEST_IF_NOT(requirements_satisfied());
StartPlayout();
// An additional attempt to initialize at this stage should be ignored.
EXPECT_EQ(0, audio_device()->InitPlayout());
// Same for additional request to start playout while already active.
EXPECT_EQ(0, audio_device()->StartPlayout());
StopPlayout();
}
// Tests Init/Stop/Init recording while playout is active.
TEST_P(MAYBE_AudioDeviceTest, InitStopInitRecordingWhilePlaying) {
SKIP_TEST_IF_NOT(requirements_satisfied());
StartPlayout();
EXPECT_EQ(0, audio_device()->InitRecording());
EXPECT_TRUE(audio_device()->RecordingIsInitialized());
StopRecording();
EXPECT_EQ(0, audio_device()->InitRecording());
StopRecording();
StopPlayout();
}
// Tests Init/Stop/Init playout without any registered audio callback.
TEST_P(MAYBE_AudioDeviceTest, InitStopInitPlayout) {
SKIP_TEST_IF_NOT(requirements_satisfied());
EXPECT_EQ(0, audio_device()->InitPlayout());
EXPECT_TRUE(audio_device()->PlayoutIsInitialized());
StopPlayout();
EXPECT_EQ(0, audio_device()->InitPlayout());
StopPlayout();
}
// Tests Init/Stop/Init playout while recording is active.
TEST_P(MAYBE_AudioDeviceTest, InitStopInitPlayoutWhileRecording) {
SKIP_TEST_IF_NOT(requirements_satisfied());
StartRecording();
EXPECT_EQ(0, audio_device()->InitPlayout());
EXPECT_TRUE(audio_device()->PlayoutIsInitialized());
StopPlayout();
EXPECT_EQ(0, audio_device()->InitPlayout());
StopPlayout();
StopRecording();
}
// TODO(henrika): restart without intermediate destruction is currently only
// supported on Windows.
#ifdef WEBRTC_WIN
// Tests Start/Stop playout followed by a second session (emulates a restart
// triggered by a user using public APIs).
TEST_P(MAYBE_AudioDeviceTest, StartStopPlayoutWithExternalRestart) {
SKIP_TEST_IF_NOT(requirements_satisfied());
StartPlayout();
StopPlayout();
// Restart playout without destroying the ADM in between. Ensures that we
// support: Init(), Start(), Stop(), Init(), Start(), Stop().
StartPlayout();
StopPlayout();
}
// Tests Start/Stop recording followed by a second session (emulates a restart
// triggered by a user using public APIs).
TEST_P(MAYBE_AudioDeviceTest, StartStopRecordingWithExternalRestart) {
SKIP_TEST_IF_NOT(requirements_satisfied());
StartRecording();
StopRecording();
// Restart recording without destroying the ADM in between. Ensures that we
// support: Init(), Start(), Stop(), Init(), Start(), Stop().
StartRecording();
StopRecording();
}
// Tests Start/Stop playout followed by a second session (emulates a restart
// triggered by an internal callback e.g. corresponding to a device switch).
// Note that, internal restart is only supported in combination with the latest
// Windows ADM.
TEST_P(MAYBE_AudioDeviceTest, StartStopPlayoutWithInternalRestart) {
SKIP_TEST_IF_NOT(requirements_satisfied());
if (audio_layer() != AudioDeviceModule::kWindowsCoreAudio2) {
return;
}
MockAudioTransport mock(TransportType::kPlay);
mock.HandleCallbacks(event(), nullptr, kNumCallbacks);
EXPECT_CALL(mock, NeedMorePlayData(_, _, _, _, NotNull(), _, _, _))
.Times(AtLeast(kNumCallbacks));
EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
StartPlayout();
event()->Wait(kTestTimeOutInMilliseconds);
EXPECT_TRUE(audio_device()->Playing());
// Restart playout but without stopping the internal audio thread.
// This procedure uses a non-public test API and it emulates what happens
// inside the ADM when e.g. a device is removed.
EXPECT_EQ(0, audio_device()->RestartPlayoutInternally());
// Run basic tests of public APIs while a restart attempt is active.
// These calls should now be very thin and not trigger any new actions.
EXPECT_EQ(-1, audio_device()->StopPlayout());
EXPECT_TRUE(audio_device()->Playing());
EXPECT_TRUE(audio_device()->PlayoutIsInitialized());
EXPECT_EQ(0, audio_device()->InitPlayout());
EXPECT_EQ(0, audio_device()->StartPlayout());
// Wait until audio has restarted and a new sequence of audio callbacks
// becomes active.
// TODO(henrika): is it possible to verify that the internal state transition
// is Stop->Init->Start?
ASSERT_TRUE(Mock::VerifyAndClearExpectations(&mock));
mock.ResetCallbackCounters();
EXPECT_CALL(mock, NeedMorePlayData(_, _, _, _, NotNull(), _, _, _))
.Times(AtLeast(kNumCallbacks));
event()->Wait(kTestTimeOutInMilliseconds);
EXPECT_TRUE(audio_device()->Playing());
// Stop playout and the audio thread after successful internal restart.
StopPlayout();
}
// Tests Start/Stop recording followed by a second session (emulates a restart
// triggered by an internal callback e.g. corresponding to a device switch).
// Note that, internal restart is only supported in combination with the latest
// Windows ADM.
TEST_P(MAYBE_AudioDeviceTest, StartStopRecordingWithInternalRestart) {
SKIP_TEST_IF_NOT(requirements_satisfied());
if (audio_layer() != AudioDeviceModule::kWindowsCoreAudio2) {
return;
}
MockAudioTransport mock(TransportType::kRecord);
mock.HandleCallbacks(event(), nullptr, kNumCallbacks);
EXPECT_CALL(mock, RecordedDataIsAvailable(NotNull(), _, _, _, _, Ge(0u), 0, _,
false, _))
.Times(AtLeast(kNumCallbacks));
EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
StartRecording();
event()->Wait(kTestTimeOutInMilliseconds);
EXPECT_TRUE(audio_device()->Recording());
// Restart recording but without stopping the internal audio thread.
// This procedure uses a non-public test API and it emulates what happens
// inside the ADM when e.g. a device is removed.
EXPECT_EQ(0, audio_device()->RestartRecordingInternally());
// Run basic tests of public APIs while a restart attempt is active.
// These calls should now be very thin and not trigger any new actions.
EXPECT_EQ(-1, audio_device()->StopRecording());
EXPECT_TRUE(audio_device()->Recording());
EXPECT_TRUE(audio_device()->RecordingIsInitialized());
EXPECT_EQ(0, audio_device()->InitRecording());
EXPECT_EQ(0, audio_device()->StartRecording());
// Wait until audio has restarted and a new sequence of audio callbacks
// becomes active.
// TODO(henrika): is it possible to verify that the internal state transition
// is Stop->Init->Start?
ASSERT_TRUE(Mock::VerifyAndClearExpectations(&mock));
mock.ResetCallbackCounters();
EXPECT_CALL(mock, RecordedDataIsAvailable(NotNull(), _, _, _, _, Ge(0u), 0, _,
false, _))
.Times(AtLeast(kNumCallbacks));
event()->Wait(kTestTimeOutInMilliseconds);
EXPECT_TRUE(audio_device()->Recording());
// Stop recording and the audio thread after successful internal restart.
StopRecording();
}
#endif // #ifdef WEBRTC_WIN
// Start playout and verify that the native audio layer starts asking for real
// audio samples to play out using the NeedMorePlayData() callback.
// Note that we can't add expectations on audio parameters in EXPECT_CALL
// since parameter are not provided in the each callback. We therefore test and
// verify the parameters in the fake audio transport implementation instead.
TEST_P(MAYBE_AudioDeviceTest, StartPlayoutVerifyCallbacks) {
SKIP_TEST_IF_NOT(requirements_satisfied());
MockAudioTransport mock(TransportType::kPlay);
mock.HandleCallbacks(event(), nullptr, kNumCallbacks);
EXPECT_CALL(mock, NeedMorePlayData(_, _, _, _, NotNull(), _, _, _))
.Times(AtLeast(kNumCallbacks));
EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
StartPlayout();
event()->Wait(kTestTimeOutInMilliseconds);
StopPlayout();
}
// Start recording and verify that the native audio layer starts providing real
// audio samples using the RecordedDataIsAvailable() callback.
TEST_P(MAYBE_AudioDeviceTest, StartRecordingVerifyCallbacks) {
SKIP_TEST_IF_NOT(requirements_satisfied());
MockAudioTransport mock(TransportType::kRecord);
mock.HandleCallbacks(event(), nullptr, kNumCallbacks);
EXPECT_CALL(mock, RecordedDataIsAvailable(NotNull(), _, _, _, _, Ge(0u), 0, _,
false, _))
.Times(AtLeast(kNumCallbacks));
EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
StartRecording();
event()->Wait(kTestTimeOutInMilliseconds);
StopRecording();
}
// Start playout and recording (full-duplex audio) and verify that audio is
// active in both directions.
TEST_P(MAYBE_AudioDeviceTest, StartPlayoutAndRecordingVerifyCallbacks) {
SKIP_TEST_IF_NOT(requirements_satisfied());
MockAudioTransport mock(TransportType::kPlayAndRecord);
mock.HandleCallbacks(event(), nullptr, kNumCallbacks);
EXPECT_CALL(mock, NeedMorePlayData(_, _, _, _, NotNull(), _, _, _))
.Times(AtLeast(kNumCallbacks));
EXPECT_CALL(mock, RecordedDataIsAvailable(NotNull(), _, _, _, _, Ge(0u), 0, _,
false, _))
.Times(AtLeast(kNumCallbacks));
EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
StartPlayout();
StartRecording();
event()->Wait(kTestTimeOutInMilliseconds);
StopRecording();
StopPlayout();
}
// Start playout and recording and store recorded data in an intermediate FIFO
// buffer from which the playout side then reads its samples in the same order
// as they were stored. Under ideal circumstances, a callback sequence would
// look like: ...+-+-+-+-+-+-+-..., where '+' means 'packet recorded' and '-'
// means 'packet played'. Under such conditions, the FIFO would contain max 1,
// with an average somewhere in (0,1) depending on how long the packets are
// buffered. However, under more realistic conditions, the size
// of the FIFO will vary more due to an unbalance between the two sides.
// This test tries to verify that the device maintains a balanced callback-
// sequence by running in loopback for a few seconds while measuring the size
// (max and average) of the FIFO. The size of the FIFO is increased by the
// recording side and decreased by the playout side.
TEST_P(MAYBE_AudioDeviceTest, RunPlayoutAndRecordingInFullDuplex) {
SKIP_TEST_IF_NOT(requirements_satisfied());
NiceMock<MockAudioTransport> mock(TransportType::kPlayAndRecord);
FifoAudioStream audio_stream;
mock.HandleCallbacks(event(), &audio_stream,
kFullDuplexTimeInSec * kNumCallbacksPerSecond);
EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
// Run both sides using the same channel configuration to avoid conversions
// between mono/stereo while running in full duplex mode. Also, some devices
// (mainly on Windows) do not support mono.
EXPECT_EQ(0, audio_device()->SetStereoPlayout(true));
EXPECT_EQ(0, audio_device()->SetStereoRecording(true));
// Mute speakers to prevent howling.
EXPECT_EQ(0, audio_device()->SetSpeakerVolume(0));
StartPlayout();
StartRecording();
event()->Wait(static_cast<int>(
std::max(kTestTimeOutInMilliseconds, 1000 * kFullDuplexTimeInSec)));
StopRecording();
StopPlayout();
// This thresholds is set rather high to accommodate differences in hardware
// in several devices. The main idea is to capture cases where a very large
// latency is built up. See http://bugs.webrtc.org/7744 for examples on
// bots where relatively large average latencies can happen.
EXPECT_LE(audio_stream.average_size(), 25u);
PRINT("\n");
}
// Runs audio in full duplex until user hits Enter. Intended as a manual test
// to ensure that the audio quality is good and that real device switches works
// as intended.
TEST_P(MAYBE_AudioDeviceTest,
DISABLED_RunPlayoutAndRecordingInFullDuplexAndWaitForEnterKey) {
SKIP_TEST_IF_NOT(requirements_satisfied());
if (audio_layer() != AudioDeviceModule::kWindowsCoreAudio2) {
return;
}
NiceMock<MockAudioTransport> mock(TransportType::kPlayAndRecord);
FifoAudioStream audio_stream;
mock.HandleCallbacks(&audio_stream);
EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
EXPECT_EQ(0, audio_device()->SetStereoPlayout(true));
EXPECT_EQ(0, audio_device()->SetStereoRecording(true));
// Ensure that the sample rate for both directions are identical so that we
// always can listen to our own voice. Will lead to rate conversion (and
// higher latency) if the native sample rate is not 48kHz.
EXPECT_EQ(0, audio_device()->SetPlayoutSampleRate(48000));
EXPECT_EQ(0, audio_device()->SetRecordingSampleRate(48000));
StartPlayout();
StartRecording();
do {
PRINT("Loopback audio is active at 48kHz. Press Enter to stop.\n");
} while (getchar() != '\n');
StopRecording();
StopPlayout();
}
// Measures loopback latency and reports the min, max and average values for
// a full duplex audio session.
// The latency is measured like so:
// - Insert impulses periodically on the output side.
// - Detect the impulses on the input side.
// - Measure the time difference between the transmit time and receive time.
// - Store time differences in a vector and calculate min, max and average.
// This test needs the '--gtest_also_run_disabled_tests' flag to run and also
// some sort of audio feedback loop. E.g. a headset where the mic is placed
// close to the speaker to ensure highest possible echo. It is also recommended
// to run the test at highest possible output volume.
TEST_P(MAYBE_AudioDeviceTest, DISABLED_MeasureLoopbackLatency) {
SKIP_TEST_IF_NOT(requirements_satisfied());
NiceMock<MockAudioTransport> mock(TransportType::kPlayAndRecord);
LatencyAudioStream audio_stream;
mock.HandleCallbacks(event(), &audio_stream,
kMeasureLatencyTimeInSec * kNumCallbacksPerSecond);
EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
EXPECT_EQ(0, audio_device()->SetStereoPlayout(true));
EXPECT_EQ(0, audio_device()->SetStereoRecording(true));
StartPlayout();
StartRecording();
event()->Wait(static_cast<int>(
std::max(kTestTimeOutInMilliseconds, 1000 * kMeasureLatencyTimeInSec)));
StopRecording();
StopPlayout();
// Verify that a sufficient number of transmitted impulses are detected.
EXPECT_GE(audio_stream.num_latency_values(),
static_cast<size_t>(
kImpulseFrequencyInHz * kMeasureLatencyTimeInSec - 2));
// Print out min, max and average delay values for debugging purposes.
audio_stream.PrintResults();
}
#ifdef WEBRTC_WIN
// Test two different audio layers (or rather two different Core Audio
// implementations) for Windows.
INSTANTIATE_TEST_SUITE_P(
AudioLayerWin,
MAYBE_AudioDeviceTest,
::testing::Values(AudioDeviceModule::kPlatformDefaultAudio,
AudioDeviceModule::kWindowsCoreAudio2));
#else
// For all platforms but Windows, only test the default audio layer.
INSTANTIATE_TEST_SUITE_P(
AudioLayer,
MAYBE_AudioDeviceTest,
::testing::Values(AudioDeviceModule::kPlatformDefaultAudio));
#endif
} // namespace webrtc