blob: 13cf3734ed2ac777833498cb42f7a068770bac2b [file] [log] [blame]
// Copyright (c) 2012 The Chromium Authors. All rights reserved.
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.
#include "media/audio/win/audio_low_latency_input_win.h"
#include <mmsystem.h>
#include <stddef.h>
#include <stdint.h>
#include <windows.h>
#include <memory>
#include "base/bind.h"
#include "base/environment.h"
#include "base/files/file_util.h"
#include "base/macros.h"
#include "base/message_loop/message_loop.h"
#include "base/path_service.h"
#include "base/run_loop.h"
#include "base/single_thread_task_runner.h"
#include "base/strings/stringprintf.h"
#include "base/test/test_timeouts.h"
#include "base/win/scoped_com_initializer.h"
#include "media/audio/audio_device_description.h"
#include "media/audio/audio_device_info_accessor_for_tests.h"
#include "media/audio/audio_io.h"
#include "media/audio/audio_manager.h"
#include "media/audio/audio_unittest_util.h"
#include "media/audio/test_audio_thread.h"
#include "media/audio/win/core_audio_util_win.h"
#include "media/base/seekable_buffer.h"
#include "testing/gmock/include/gmock/gmock.h"
#include "testing/gtest/include/gtest/gtest.h"
using ::testing::_;
using ::testing::AnyNumber;
using ::testing::AtLeast;
using ::testing::Gt;
using ::testing::NotNull;
namespace media {
namespace {
constexpr SampleFormat kSampleFormat = kSampleFormatS16;
void LogCallbackDummy(const std::string& /* message */) {}
} // namespace
ACTION_P4(CheckCountAndPostQuitTask, count, limit, task_runner, quit_closure) {
if (++*count >= limit)
task_runner->PostTask(FROM_HERE, quit_closure);
}
class MockAudioInputCallback : public AudioInputStream::AudioInputCallback {
public:
MOCK_METHOD3(OnData,
void(const AudioBus* src,
base::TimeTicks capture_time,
double volume));
MOCK_METHOD0(OnError, void());
};
class FakeAudioInputCallback : public AudioInputStream::AudioInputCallback {
public:
FakeAudioInputCallback()
: num_received_audio_frames_(0),
data_event_(base::WaitableEvent::ResetPolicy::AUTOMATIC,
base::WaitableEvent::InitialState::NOT_SIGNALED),
error_(false) {}
bool error() const { return error_; }
int num_received_audio_frames() const { return num_received_audio_frames_; }
// Waits until OnData() is called on another thread.
void WaitForData() { data_event_.Wait(); }
void OnData(const AudioBus* src,
base::TimeTicks capture_time,
double volume) override {
EXPECT_GE(capture_time, base::TimeTicks());
num_received_audio_frames_ += src->frames();
data_event_.Signal();
}
void OnError() override { error_ = true; }
private:
int num_received_audio_frames_;
base::WaitableEvent data_event_;
bool error_;
DISALLOW_COPY_AND_ASSIGN(FakeAudioInputCallback);
};
// This audio sink implementation should be used for manual tests only since
// the recorded data is stored on a raw binary data file.
class WriteToFileAudioSink : public AudioInputStream::AudioInputCallback {
public:
// Allocate space for ~10 seconds of data @ 48kHz in stereo:
// 2 bytes per sample, 2 channels, 10ms @ 48kHz, 10 seconds <=> 1920000 bytes.
static const size_t kMaxBufferSize = 2 * 2 * 480 * 100 * 10;
explicit WriteToFileAudioSink(const char* file_name)
: buffer_(0, kMaxBufferSize), bytes_to_write_(0) {
base::FilePath file_path;
EXPECT_TRUE(base::PathService::Get(base::DIR_EXE, &file_path));
file_path = file_path.AppendASCII(file_name);
binary_file_ = base::OpenFile(file_path, "wb");
DLOG_IF(ERROR, !binary_file_) << "Failed to open binary PCM data file.";
VLOG(0) << ">> Output file: " << file_path.value() << " has been created.";
}
~WriteToFileAudioSink() override {
size_t bytes_written = 0;
while (bytes_written < bytes_to_write_) {
const uint8_t* chunk;
int chunk_size;
// Stop writing if no more data is available.
if (!buffer_.GetCurrentChunk(&chunk, &chunk_size))
break;
// Write recorded data chunk to the file and prepare for next chunk.
fwrite(chunk, 1, chunk_size, binary_file_);
buffer_.Seek(chunk_size);
bytes_written += chunk_size;
}
base::CloseFile(binary_file_);
}
// AudioInputStream::AudioInputCallback implementation.
void OnData(const AudioBus* src,
base::TimeTicks capture_time,
double volume) override {
const int num_samples = src->frames() * src->channels();
auto interleaved = std::make_unique<int16_t[]>(num_samples);
const int bytes_per_sample = sizeof(interleaved[0]);
src->ToInterleaved<SignedInt16SampleTypeTraits>(src->frames(),
interleaved.get());
// Store data data in a temporary buffer to avoid making blocking
// fwrite() calls in the audio callback. The complete buffer will be
// written to file in the destructor.
const int size = bytes_per_sample * num_samples;
if (buffer_.Append((const uint8_t*)interleaved.get(), size)) {
bytes_to_write_ += size;
}
}
void OnError() override {}
private:
media::SeekableBuffer buffer_;
FILE* binary_file_;
size_t bytes_to_write_;
};
static bool HasCoreAudioAndInputDevices(AudioManager* audio_man) {
// The low-latency (WASAPI-based) version requires Windows Vista or higher.
// TODO(henrika): note that we use Wave today to query the number of
// existing input devices.
return CoreAudioUtil::IsSupported() &&
AudioDeviceInfoAccessorForTests(audio_man).HasAudioInputDevices();
}
// Convenience method which creates a default AudioInputStream object but
// also allows the user to modify the default settings.
class AudioInputStreamWrapper {
public:
explicit AudioInputStreamWrapper(AudioManager* audio_manager)
: audio_man_(audio_manager) {
EXPECT_TRUE(SUCCEEDED(CoreAudioUtil::GetPreferredAudioParameters(
AudioDeviceDescription::kDefaultDeviceId, false, &default_params_)));
EXPECT_EQ(format(), AudioParameters::AUDIO_PCM_LOW_LATENCY);
frames_per_buffer_ = default_params_.frames_per_buffer();
}
AudioInputStreamWrapper(AudioManager* audio_manager,
const AudioParameters& default_params)
: audio_man_(audio_manager), default_params_(default_params) {
EXPECT_EQ(format(), AudioParameters::AUDIO_PCM_LOW_LATENCY);
frames_per_buffer_ = default_params_.frames_per_buffer();
}
~AudioInputStreamWrapper() {}
// Creates AudioInputStream object using default parameters.
AudioInputStream* Create() { return CreateInputStream(); }
// Creates AudioInputStream object using non-default parameters where the
// frame size is modified.
AudioInputStream* Create(int frames_per_buffer) {
frames_per_buffer_ = frames_per_buffer;
return CreateInputStream();
}
AudioParameters::Format format() const { return default_params_.format(); }
int channels() const {
return ChannelLayoutToChannelCount(default_params_.channel_layout());
}
int sample_rate() const { return default_params_.sample_rate(); }
int frames_per_buffer() const { return frames_per_buffer_; }
private:
AudioInputStream* CreateInputStream() {
AudioParameters params = default_params_;
params.set_frames_per_buffer(frames_per_buffer_);
AudioInputStream* ais = audio_man_->MakeAudioInputStream(
params, AudioDeviceDescription::kDefaultDeviceId,
base::BindRepeating(&LogCallbackDummy));
EXPECT_TRUE(ais);
return ais;
}
AudioManager* audio_man_;
AudioParameters default_params_;
int frames_per_buffer_;
};
// Convenience method which creates a default AudioInputStream object.
static AudioInputStream* CreateDefaultAudioInputStream(
AudioManager* audio_manager) {
AudioInputStreamWrapper aisw(audio_manager);
AudioInputStream* ais = aisw.Create();
return ais;
}
class ScopedAudioInputStream {
public:
explicit ScopedAudioInputStream(AudioInputStream* stream) : stream_(stream) {}
~ScopedAudioInputStream() {
if (stream_)
stream_->Close();
}
void Close() {
if (stream_)
stream_->Close();
stream_ = nullptr;
}
AudioInputStream* operator->() { return stream_; }
AudioInputStream* get() const { return stream_; }
void Reset(AudioInputStream* new_stream) {
Close();
stream_ = new_stream;
}
private:
AudioInputStream* stream_;
DISALLOW_COPY_AND_ASSIGN(ScopedAudioInputStream);
};
class WinAudioInputTest : public ::testing::Test {
public:
WinAudioInputTest() {
audio_manager_ =
AudioManager::CreateForTesting(std::make_unique<TestAudioThread>());
base::RunLoop().RunUntilIdle();
}
~WinAudioInputTest() override { audio_manager_->Shutdown(); }
protected:
base::MessageLoop message_loop_;
std::unique_ptr<AudioManager> audio_manager_;
};
// Verify that we can retrieve the current hardware/mixing sample rate
// for all available input devices.
TEST_F(WinAudioInputTest, WASAPIAudioInputStreamHardwareSampleRate) {
ABORT_AUDIO_TEST_IF_NOT(HasCoreAudioAndInputDevices(audio_manager_.get()));
// Retrieve a list of all available input devices.
media::AudioDeviceDescriptions device_descriptions;
AudioDeviceInfoAccessorForTests(audio_manager_.get())
.GetAudioInputDeviceDescriptions(&device_descriptions);
// Scan all available input devices and repeat the same test for all of them.
for (const auto& device : device_descriptions) {
// Retrieve the hardware sample rate given a specified audio input device.
AudioParameters params;
ASSERT_TRUE(SUCCEEDED(CoreAudioUtil::GetPreferredAudioParameters(
device.unique_id, false, &params)));
EXPECT_GE(params.sample_rate(), 0);
}
}
// Test effects.
TEST_F(WinAudioInputTest, WASAPIAudioInputStreamEffects) {
AudioDeviceInfoAccessorForTests device_info_accessor(audio_manager_.get());
ABORT_AUDIO_TEST_IF_NOT(device_info_accessor.HasAudioInputDevices() &&
device_info_accessor.HasAudioOutputDevices() &&
CoreAudioUtil::IsSupported());
// Retrieve a list of all available input devices.
media::AudioDeviceDescriptions device_descriptions;
device_info_accessor.GetAudioInputDeviceDescriptions(&device_descriptions);
// No device should have any effects.
for (const auto& device : device_descriptions) {
AudioParameters params =
device_info_accessor.GetInputStreamParameters(device.unique_id);
EXPECT_EQ(params.effects(), AudioParameters::NO_EFFECTS);
}
// The two loopback devices are not included in the device description list
// above. They should also have no effects.
AudioParameters params = device_info_accessor.GetInputStreamParameters(
AudioDeviceDescription::kLoopbackInputDeviceId);
EXPECT_EQ(params.effects(), AudioParameters::NO_EFFECTS);
params = device_info_accessor.GetInputStreamParameters(
AudioDeviceDescription::kLoopbackWithMuteDeviceId);
EXPECT_EQ(params.effects(), AudioParameters::NO_EFFECTS);
}
// Test Create(), Close() calling sequence.
TEST_F(WinAudioInputTest, WASAPIAudioInputStreamCreateAndClose) {
ABORT_AUDIO_TEST_IF_NOT(HasCoreAudioAndInputDevices(audio_manager_.get()));
ScopedAudioInputStream ais(
CreateDefaultAudioInputStream(audio_manager_.get()));
ais.Close();
}
// Test Open(), Close() calling sequence.
TEST_F(WinAudioInputTest, WASAPIAudioInputStreamOpenAndClose) {
ABORT_AUDIO_TEST_IF_NOT(HasCoreAudioAndInputDevices(audio_manager_.get()));
ScopedAudioInputStream ais(
CreateDefaultAudioInputStream(audio_manager_.get()));
EXPECT_TRUE(ais->Open());
ais.Close();
}
// Test Open(), Start(), Close() calling sequence.
TEST_F(WinAudioInputTest, WASAPIAudioInputStreamOpenStartAndClose) {
ABORT_AUDIO_TEST_IF_NOT(HasCoreAudioAndInputDevices(audio_manager_.get()));
ScopedAudioInputStream ais(
CreateDefaultAudioInputStream(audio_manager_.get()));
EXPECT_TRUE(ais->Open());
MockAudioInputCallback sink;
ais->Start(&sink);
ais.Close();
}
// Test Open(), Start(), Stop(), Close() calling sequence.
TEST_F(WinAudioInputTest, WASAPIAudioInputStreamOpenStartStopAndClose) {
ABORT_AUDIO_TEST_IF_NOT(HasCoreAudioAndInputDevices(audio_manager_.get()));
ScopedAudioInputStream ais(
CreateDefaultAudioInputStream(audio_manager_.get()));
EXPECT_TRUE(ais->Open());
MockAudioInputCallback sink;
ais->Start(&sink);
ais->Stop();
ais.Close();
}
// Test some additional calling sequences.
TEST_F(WinAudioInputTest, WASAPIAudioInputStreamMiscCallingSequences) {
ABORT_AUDIO_TEST_IF_NOT(HasCoreAudioAndInputDevices(audio_manager_.get()));
ScopedAudioInputStream ais(
CreateDefaultAudioInputStream(audio_manager_.get()));
// Open(), Open() should fail the second time.
EXPECT_TRUE(ais->Open());
EXPECT_FALSE(ais->Open());
FakeAudioInputCallback sink;
// Start(), Start() is a valid calling sequence (second call does nothing).
ais->Start(&sink);
sink.WaitForData();
ais->Start(&sink);
// Ensure the stream is still started.
sink.WaitForData();
sink.WaitForData();
// Stop(), Stop() is a valid calling sequence (second call does nothing).
ais->Stop();
ais->Stop();
ais.Close();
}
TEST_F(WinAudioInputTest, WASAPIAudioInputStreamTestPacketSizes) {
ABORT_AUDIO_TEST_IF_NOT(HasCoreAudioAndInputDevices(audio_manager_.get()));
int count = 0;
// 10 ms packet size.
// Create default WASAPI input stream which records in stereo using
// the shared mixing rate. The default buffer size is 10ms.
AudioInputStreamWrapper aisw(audio_manager_.get());
ScopedAudioInputStream ais(aisw.Create());
EXPECT_TRUE(ais->Open());
MockAudioInputCallback sink;
// Derive the expected size in bytes of each recorded packet.
uint32_t bytes_per_packet = aisw.channels() * aisw.frames_per_buffer() *
SampleFormatToBytesPerChannel(kSampleFormat);
{
// We use 10ms packets and will run the test until ten packets are received.
// All should contain valid packets of the same size and a valid delay
// estimate.
base::RunLoop run_loop;
EXPECT_CALL(sink, OnData(NotNull(), _, _))
.Times(AtLeast(10))
.WillRepeatedly(
CheckCountAndPostQuitTask(&count, 10, message_loop_.task_runner(),
run_loop.QuitWhenIdleClosure()));
ais->Start(&sink);
run_loop.Run();
ais->Stop();
}
// Store current packet size (to be used in the subsequent tests).
int frames_per_buffer_10ms = aisw.frames_per_buffer();
ais.Close();
// 20 ms packet size.
count = 0;
ais.Reset(aisw.Create(2 * frames_per_buffer_10ms));
EXPECT_TRUE(ais->Open());
bytes_per_packet = aisw.channels() * aisw.frames_per_buffer() *
SampleFormatToBytesPerChannel(kSampleFormat);
{
base::RunLoop run_loop;
EXPECT_CALL(sink, OnData(NotNull(), _, _))
.Times(AtLeast(10))
.WillRepeatedly(
CheckCountAndPostQuitTask(&count, 10, message_loop_.task_runner(),
run_loop.QuitWhenIdleClosure()));
ais->Start(&sink);
run_loop.Run();
ais->Stop();
ais.Close();
}
// 5 ms packet size.
count = 0;
ais.Reset(aisw.Create(frames_per_buffer_10ms / 2));
EXPECT_TRUE(ais->Open());
bytes_per_packet = aisw.channels() * aisw.frames_per_buffer() *
SampleFormatToBytesPerChannel(kSampleFormat);
{
base::RunLoop run_loop;
EXPECT_CALL(sink, OnData(NotNull(), _, _))
.Times(AtLeast(10))
.WillRepeatedly(
CheckCountAndPostQuitTask(&count, 10, message_loop_.task_runner(),
run_loop.QuitWhenIdleClosure()));
ais->Start(&sink);
run_loop.Run();
ais->Stop();
ais.Close();
}
}
// Test that we can capture a stream in loopback.
TEST_F(WinAudioInputTest, WASAPIAudioInputStreamLoopback) {
AudioDeviceInfoAccessorForTests device_info_accessor(audio_manager_.get());
ABORT_AUDIO_TEST_IF_NOT(device_info_accessor.HasAudioOutputDevices() &&
CoreAudioUtil::IsSupported());
AudioParameters params = device_info_accessor.GetInputStreamParameters(
AudioDeviceDescription::kLoopbackInputDeviceId);
EXPECT_EQ(params.effects(), AudioParameters::NO_EFFECTS);
AudioParameters output_params =
device_info_accessor.GetOutputStreamParameters(std::string());
EXPECT_EQ(params.sample_rate(), output_params.sample_rate());
EXPECT_EQ(params.channel_layout(), output_params.channel_layout());
ScopedAudioInputStream stream(audio_manager_->MakeAudioInputStream(
params, AudioDeviceDescription::kLoopbackInputDeviceId,
base::BindRepeating(&LogCallbackDummy)));
ASSERT_TRUE(stream->Open());
FakeAudioInputCallback sink;
stream->Start(&sink);
ASSERT_FALSE(sink.error());
sink.WaitForData();
stream.Close();
EXPECT_GT(sink.num_received_audio_frames(), 0);
EXPECT_FALSE(sink.error());
}
// This test is intended for manual tests and should only be enabled
// when it is required to store the captured data on a local file.
// By default, GTest will print out YOU HAVE 1 DISABLED TEST.
// To include disabled tests in test execution, just invoke the test program
// with --gtest_also_run_disabled_tests or set the GTEST_ALSO_RUN_DISABLED_TESTS
// environment variable to a value greater than 0.
TEST_F(WinAudioInputTest, DISABLED_WASAPIAudioInputStreamRecordToFile) {
ABORT_AUDIO_TEST_IF_NOT(HasCoreAudioAndInputDevices(audio_manager_.get()));
// Name of the output PCM file containing captured data. The output file
// will be stored in the directory containing 'media_unittests.exe'.
// Example of full name: \src\build\Debug\out_stereo_10sec.pcm.
const char* file_name = "out_10sec.pcm";
AudioInputStreamWrapper aisw(audio_manager_.get());
ScopedAudioInputStream ais(aisw.Create());
ASSERT_TRUE(ais->Open());
VLOG(0) << ">> Sample rate: " << aisw.sample_rate() << " [Hz]";
WriteToFileAudioSink file_sink(file_name);
VLOG(0) << ">> Speak into the default microphone while recording.";
ais->Start(&file_sink);
base::PlatformThread::Sleep(TestTimeouts::action_timeout());
ais->Stop();
VLOG(0) << ">> Recording has stopped.";
ais.Close();
}
TEST_F(WinAudioInputTest, DISABLED_WASAPIAudioInputStreamResampleToFile) {
ABORT_AUDIO_TEST_IF_NOT(HasCoreAudioAndInputDevices(audio_manager_.get()));
// This is basically the same test as WASAPIAudioInputStreamRecordToFile
// except it forces use of a different sample rate than is preferred by
// the hardware. This functionality is offered while we still have code
// that doesn't ask the lower levels for what the preferred audio parameters
// are (and previously depended on the old Wave API to do this automatically).
struct TestData {
const int rate;
const int frames;
ChannelLayout layout;
} tests[] = {
{8000, 80, CHANNEL_LAYOUT_MONO},
{8000, 80, CHANNEL_LAYOUT_STEREO},
{44100, 441, CHANNEL_LAYOUT_MONO},
{44100, 1024, CHANNEL_LAYOUT_STEREO},
};
for (const auto& test : tests) {
AudioParameters params;
ASSERT_TRUE(SUCCEEDED(CoreAudioUtil::GetPreferredAudioParameters(
AudioDeviceDescription::kDefaultDeviceId, false, &params)));
VLOG(0) << ">> Hardware sample rate: " << params.sample_rate() << " [Hz]";
VLOG(0) << ">> Hardware channel layout: "
<< ChannelLayoutToString(params.channel_layout());
// Pick a somewhat difficult sample rate to convert too.
// If the sample rate is 8kHz, 16kHz, 32kHz, 48kHz etc, we convert to
// 44.1kHz.
// Otherwise (e.g. 44.1kHz, 22.05kHz etc) we convert to 48kHz.
const int hw_sample_rate = params.sample_rate();
params.Reset(params.format(), test.layout, test.rate, test.frames);
std::string file_name(base::StringPrintf(
"resampled_10sec_%i_to_%i_%s.pcm", hw_sample_rate, params.sample_rate(),
ChannelLayoutToString(params.channel_layout())));
AudioInputStreamWrapper aisw(audio_manager_.get(), params);
ScopedAudioInputStream ais(aisw.Create());
ASSERT_TRUE(ais->Open());
VLOG(0) << ">> Resampled rate will be: " << aisw.sample_rate() << " [Hz]";
VLOG(0) << ">> New layout will be: "
<< ChannelLayoutToString(params.channel_layout());
WriteToFileAudioSink file_sink(file_name.c_str());
VLOG(0) << ">> Speak into the default microphone while recording.";
ais->Start(&file_sink);
base::PlatformThread::Sleep(TestTimeouts::action_timeout());
// base::PlatformThread::Sleep(base::TimeDelta::FromMinutes(10));
ais->Stop();
VLOG(0) << ">> Recording has stopped.";
ais.Close();
}
}
} // namespace media