// Copyright 2012 The Chromium Authors
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.
#ifdef UNSAFE_BUFFERS_BUILD
// TODO(crbug.com/40285824): Remove this and convert code to safer constructs.
#pragma allow_unsafe_buffers
#endif
#include "media/audio/win/audio_low_latency_output_win.h"
#include <windows.h>
#include <mmsystem.h>
#include <stddef.h>
#include <stdint.h>
#include <memory>
#include "base/environment.h"
#include "base/files/file_util.h"
#include "base/logging.h"
#include "base/memory/ptr_util.h"
#include "base/memory/raw_ptr.h"
#include "base/numerics/safe_conversions.h"
#include "base/path_service.h"
#include "base/run_loop.h"
#include "base/task/single_thread_task_runner.h"
#include "base/test/gmock_callback_support.h"
#include "base/test/task_environment.h"
#include "base/test/test_timeouts.h"
#include "base/time/time.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/mock_audio_source_callback.h"
#include "media/audio/test_audio_thread.h"
#include "media/audio/win/core_audio_util_win.h"
#include "media/base/decoder_buffer.h"
#include "media/base/seekable_buffer.h"
#include "media/base/test_data_util.h"
#include "testing/gmock/include/gmock/gmock.h"
#include "testing/gtest/include/gtest/gtest.h"
using ::testing::_;
using ::testing::DoAll;
using ::testing::NotNull;
using ::testing::Return;
namespace media {
static const int kBitsPerSample = 16;
static const size_t kMaxDeltaSamples = 1000;
static const char kDeltaTimeMsFileName[] = "delta_times_ms.txt";
MATCHER_P(HasValidDelay, value, "") {
// It is difficult to come up with a perfect test condition for the delay
// estimation. For now, verify that the produced output delay is always
// larger than the selected buffer size.
return arg >= value;
}
// This audio source implementation should be used for manual tests only since
// it takes about 20 seconds to play out a file.
class ReadFromFileAudioSource : public AudioOutputStream::AudioSourceCallback {
public:
explicit ReadFromFileAudioSource(const std::string& name)
: pos_(0),
previous_call_time_(base::TimeTicks::Now()),
text_file_(nullptr),
elements_to_write_(0) {
// Reads a test file from media/test/data directory.
file_ = ReadTestDataFile(name);
// Creates an array that will store delta times between callbacks.
// The content of this array will be written to a text file at
// destruction and can then be used for off-line analysis of the exact
// timing of callbacks. The text file will be stored in media/test/data.
delta_times_.reset(new int[kMaxDeltaSamples]);
}
~ReadFromFileAudioSource() override {
// Get complete file path to output file in directory containing
// media_unittests.exe.
base::FilePath file_name;
EXPECT_TRUE(base::PathService::Get(base::DIR_EXE, &file_name));
file_name = file_name.AppendASCII(kDeltaTimeMsFileName);
EXPECT_TRUE(!text_file_);
text_file_ = base::OpenFile(file_name, "wt");
DLOG_IF(ERROR, !text_file_) << "Failed to open log file.";
// Write the array which contains delta times to a text file.
size_t elements_written = 0;
while (elements_written < elements_to_write_) {
fprintf(text_file_.get(), "%d\n", delta_times_[elements_written]);
++elements_written;
}
base::CloseFile(text_file_);
}
// AudioOutputStream::AudioSourceCallback implementation.
int OnMoreData(base::TimeDelta /* delay */,
base::TimeTicks /* delay_timestamp */,
const AudioGlitchInfo& /* glitch_info */,
AudioBus* dest) override {
// Store time difference between two successive callbacks in an array.
// These values will be written to a file in the destructor.
const base::TimeTicks now_time = base::TimeTicks::Now();
const int diff = (now_time - previous_call_time_).InMilliseconds();
previous_call_time_ = now_time;
if (elements_to_write_ < kMaxDeltaSamples) {
delta_times_[elements_to_write_] = diff;
++elements_to_write_;
}
int max_size = dest->frames() * dest->channels() * kBitsPerSample / 8;
// Use samples read from a data file and fill up the audio buffer
// provided to us in the callback.
if (pos_ + max_size > file_size())
max_size = file_size() - pos_;
int frames = max_size / (dest->channels() * kBitsPerSample / 8);
if (max_size) {
static_assert(kBitsPerSample == 16, "FromInterleaved expects 2 bytes.");
dest->FromInterleaved<SignedInt16SampleTypeTraits>(
reinterpret_cast<const int16_t*>(file_->data() + pos_), frames);
pos_ += max_size;
}
return frames;
}
void OnError(ErrorType type) override {}
int file_size() { return base::checked_cast<int>(file_->size()); }
private:
scoped_refptr<DecoderBuffer> file_;
std::unique_ptr<int[]> delta_times_;
int pos_;
base::TimeTicks previous_call_time_;
raw_ptr<FILE> text_file_;
size_t elements_to_write_;
};
static bool ExclusiveModeIsEnabled() {
return (WASAPIAudioOutputStream::GetShareMode() ==
AUDCLNT_SHAREMODE_EXCLUSIVE);
}
static bool HasCoreAudioAndOutputDevices(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 output devices.
return CoreAudioUtil::IsSupported() &&
AudioDeviceInfoAccessorForTests(audio_man).HasAudioOutputDevices();
}
// Convenience method which creates a default AudioOutputStream object but
// also allows the user to modify the default settings.
class AudioOutputStreamWrapper {
public:
explicit AudioOutputStreamWrapper(AudioManager* audio_manager)
: audio_man_(audio_manager),
format_(AudioParameters::AUDIO_PCM_LOW_LATENCY) {
AudioParameters preferred_params;
EXPECT_TRUE(SUCCEEDED(CoreAudioUtil::GetPreferredAudioParameters(
AudioDeviceDescription::kDefaultDeviceId, true, &preferred_params)));
channels_ = preferred_params.channels();
channel_layout_ = preferred_params.channel_layout();
sample_rate_ = preferred_params.sample_rate();
samples_per_packet_ = preferred_params.frames_per_buffer();
}
~AudioOutputStreamWrapper() {}
// Creates AudioOutputStream object using default parameters.
AudioOutputStream* Create() { return CreateOutputStream(); }
// Creates AudioOutputStream object using non-default parameters where the
// frame size is modified.
AudioOutputStream* Create(int samples_per_packet) {
samples_per_packet_ = samples_per_packet;
return CreateOutputStream();
}
// Creates AudioOutputStream object using non-default parameters where the
// sample rate and frame size are modified.
AudioOutputStream* Create(int sample_rate, int samples_per_packet) {
sample_rate_ = sample_rate;
samples_per_packet_ = samples_per_packet;
return CreateOutputStream();
}
// Creates AudioOutputStream object using non-default parameters where the
// sample rate, frame size and audio offload are modified.
AudioOutputStream* Create(int sample_rate,
int samples_per_packet,
bool audio_offload) {
sample_rate_ = sample_rate;
samples_per_packet_ = samples_per_packet;
AudioParameters::HardwareCapabilities hardware_cap(0, true);
hardware_cap.require_audio_offload = true;
hardware_capabilities_ = hardware_cap;
return CreateOutputStream();
}
AudioParameters::Format format() const { return format_; }
int channels() const { return channels_; }
int sample_rate() const { return sample_rate_; }
int samples_per_packet() const { return samples_per_packet_; }
private:
AudioOutputStream* CreateOutputStream() {
AudioParameters params(format_, {channel_layout_, channels_}, sample_rate_,
samples_per_packet_);
if (hardware_capabilities_) {
params.set_hardware_capabilities(hardware_capabilities_.value());
}
DVLOG(1) << params.AsHumanReadableString();
AudioOutputStream* aos = audio_man_->MakeAudioOutputStream(
params, std::string(), AudioManager::LogCallback());
EXPECT_TRUE(aos);
return aos;
}
raw_ptr<AudioManager> audio_man_;
AudioParameters::Format format_;
int channels_;
ChannelLayout channel_layout_;
int sample_rate_;
int samples_per_packet_;
std::optional<AudioParameters::HardwareCapabilities> hardware_capabilities_;
};
// Convenience method which creates a default AudioOutputStream object.
static AudioOutputStream* CreateDefaultAudioOutputStream(
AudioManager* audio_manager) {
AudioOutputStreamWrapper aosw(audio_manager);
AudioOutputStream* aos = aosw.Create();
return aos;
}
class WASAPIAudioOutputStreamTest : public ::testing::Test {
public:
WASAPIAudioOutputStreamTest() {
audio_manager_ =
AudioManager::CreateForTesting(std::make_unique<TestAudioThread>());
base::RunLoop().RunUntilIdle();
}
~WASAPIAudioOutputStreamTest() override { audio_manager_->Shutdown(); }
protected:
base::test::SingleThreadTaskEnvironment task_environment_{
base::test::SingleThreadTaskEnvironment::MainThreadType::UI};
std::unique_ptr<AudioManager> audio_manager_;
};
// Test Create(), Close() calling sequence.
TEST_F(WASAPIAudioOutputStreamTest, CreateAndClose) {
ABORT_AUDIO_TEST_IF_NOT(HasCoreAudioAndOutputDevices(audio_manager_.get()));
AudioOutputStream* aos = CreateDefaultAudioOutputStream(audio_manager_.get());
aos->Close();
}
// Test Open(), Close() calling sequence.
TEST_F(WASAPIAudioOutputStreamTest, OpenAndClose) {
ABORT_AUDIO_TEST_IF_NOT(HasCoreAudioAndOutputDevices(audio_manager_.get()));
AudioOutputStream* aos = CreateDefaultAudioOutputStream(audio_manager_.get());
EXPECT_TRUE(aos->Open());
aos->Close();
}
// Test Open(), Start(), Close() calling sequence.
TEST_F(WASAPIAudioOutputStreamTest, OpenStartAndClose) {
ABORT_AUDIO_TEST_IF_NOT(HasCoreAudioAndOutputDevices(audio_manager_.get()));
AudioOutputStream* aos = CreateDefaultAudioOutputStream(audio_manager_.get());
EXPECT_TRUE(aos->Open());
MockAudioSourceCallback source;
EXPECT_CALL(source, OnError(_)).Times(0);
aos->Start(&source);
aos->Close();
}
// Test Open(), Start(), Stop(), Close() calling sequence.
TEST_F(WASAPIAudioOutputStreamTest, OpenStartStopAndClose) {
ABORT_AUDIO_TEST_IF_NOT(HasCoreAudioAndOutputDevices(audio_manager_.get()));
AudioOutputStream* aos = CreateDefaultAudioOutputStream(audio_manager_.get());
EXPECT_TRUE(aos->Open());
MockAudioSourceCallback source;
EXPECT_CALL(source, OnError(_)).Times(0);
aos->Start(&source);
aos->Stop();
aos->Close();
}
// Test SetVolume(), GetVolume()
TEST_F(WASAPIAudioOutputStreamTest, Volume) {
ABORT_AUDIO_TEST_IF_NOT(HasCoreAudioAndOutputDevices(audio_manager_.get()));
AudioOutputStream* aos = CreateDefaultAudioOutputStream(audio_manager_.get());
// Initial volume should be full volume (1.0).
double volume = 0.0;
aos->GetVolume(&volume);
EXPECT_EQ(1.0, volume);
// Verify some valid volume settings.
aos->SetVolume(0.0);
aos->GetVolume(&volume);
EXPECT_EQ(0.0, volume);
aos->SetVolume(0.5);
aos->GetVolume(&volume);
EXPECT_EQ(0.5, volume);
aos->SetVolume(1.0);
aos->GetVolume(&volume);
EXPECT_EQ(1.0, volume);
// Ensure that invalid volume setting have no effect.
aos->SetVolume(1.5);
aos->GetVolume(&volume);
EXPECT_EQ(1.0, volume);
aos->SetVolume(-0.5);
aos->GetVolume(&volume);
EXPECT_EQ(1.0, volume);
aos->Close();
}
// Test some additional calling sequences.
TEST_F(WASAPIAudioOutputStreamTest, MiscCallingSequences) {
ABORT_AUDIO_TEST_IF_NOT(HasCoreAudioAndOutputDevices(audio_manager_.get()));
AudioOutputStream* aos = CreateDefaultAudioOutputStream(audio_manager_.get());
WASAPIAudioOutputStream* waos = static_cast<WASAPIAudioOutputStream*>(aos);
// Open(), Open() is a valid calling sequence (second call does nothing).
EXPECT_TRUE(aos->Open());
EXPECT_TRUE(aos->Open());
MockAudioSourceCallback source;
// Start(), Start() is a valid calling sequence (second call does nothing).
aos->Start(&source);
EXPECT_TRUE(waos->started());
aos->Start(&source);
EXPECT_TRUE(waos->started());
// Stop(), Stop() is a valid calling sequence (second call does nothing).
aos->Stop();
EXPECT_FALSE(waos->started());
aos->Stop();
EXPECT_FALSE(waos->started());
// Start(), Stop(), Start(), Stop().
aos->Start(&source);
EXPECT_TRUE(waos->started());
aos->Stop();
EXPECT_FALSE(waos->started());
aos->Start(&source);
EXPECT_TRUE(waos->started());
aos->Stop();
EXPECT_FALSE(waos->started());
aos->Close();
}
// Use preferred packet size and verify that rendering starts.
TEST_F(WASAPIAudioOutputStreamTest, ValidPacketSize) {
ABORT_AUDIO_TEST_IF_NOT(HasCoreAudioAndOutputDevices(audio_manager_.get()));
MockAudioSourceCallback source;
// Create default WASAPI output stream which plays out in stereo using
// the shared mixing rate. The default buffer size is 10ms.
AudioOutputStreamWrapper aosw(audio_manager_.get());
AudioOutputStream* aos = aosw.Create();
EXPECT_TRUE(aos->Open());
base::RunLoop loop;
// Derive the expected duration of each packet.
base::TimeDelta packet_duration = base::Seconds(
static_cast<double>(aosw.samples_per_packet()) / aosw.sample_rate());
// Wait for the first callback and verify its parameters. Ignore any
// subsequent callbacks that might arrive.
EXPECT_CALL(source, OnMoreData(HasValidDelay(packet_duration), _,
AudioGlitchInfo(), NotNull()))
.WillOnce(DoAll(base::test::RunClosure(loop.QuitWhenIdleClosure()),
Return(aosw.samples_per_packet())))
.WillRepeatedly(Return(0));
aos->Start(&source);
base::SingleThreadTaskRunner::GetCurrentDefault()->PostDelayedTask(
FROM_HERE, loop.QuitWhenIdleClosure(), TestTimeouts::action_timeout());
loop.Run();
aos->Stop();
aos->Close();
}
// Verify that we are not allowed to open the output stream with audio offload
// enabled in exclusive mode.
TEST_F(WASAPIAudioOutputStreamTest, ExclusiveModeWithAudioOffload) {
ABORT_AUDIO_TEST_IF_NOT(HasCoreAudioAndOutputDevices(audio_manager_.get()) &&
ExclusiveModeIsEnabled());
// Create exclusive-mode WASAPI output stream which plays out in stereo
// using the minimum buffer size at 48kHz sample rate.
AudioOutputStreamWrapper aosw(audio_manager_.get());
// Open should fail with offload stream in exclusive mode.
AudioOutputStream* aos = aosw.Create(48000, 160, true);
EXPECT_FALSE(aos->Open());
aos->Close();
}
// Verify that we can open the output stream in exclusive mode using a
// certain set of audio parameters and a sample rate of 48kHz.
// The expected outcomes of each setting in this test has been derived
// manually using log outputs (--v=1).
// It's disabled by default because a flag is required to enable exclusive mode.
TEST_F(WASAPIAudioOutputStreamTest, DISABLED_ExclusiveModeBufferSizesAt48kHz) {
ABORT_AUDIO_TEST_IF_NOT(HasCoreAudioAndOutputDevices(audio_manager_.get()) &&
ExclusiveModeIsEnabled());
AudioOutputStreamWrapper aosw(audio_manager_.get());
// 10ms @ 48kHz shall work.
// Note that, this is the same size as we can use for shared-mode streaming
// but here the endpoint buffer delay is only 10ms instead of 20ms.
AudioOutputStream* aos = aosw.Create(48000, 480);
EXPECT_TRUE(aos->Open());
aos->Close();
// 5ms @ 48kHz does not work due to misalignment.
// This test will propose an aligned buffer size of 5.3333ms.
// Note that we must call Close() even is Open() fails since Close() also
// deletes the object and we want to create a new object in the next test.
aos = aosw.Create(48000, 240);
EXPECT_FALSE(aos->Open());
aos->Close();
// 5.3333ms @ 48kHz should work (see test above).
aos = aosw.Create(48000, 256);
EXPECT_TRUE(aos->Open());
aos->Close();
// 2.6667ms is smaller than the minimum supported size (=3ms).
aos = aosw.Create(48000, 128);
EXPECT_FALSE(aos->Open());
aos->Close();
// 3ms does not correspond to an aligned buffer size.
// This test will propose an aligned buffer size of 3.3333ms.
aos = aosw.Create(48000, 144);
EXPECT_FALSE(aos->Open());
aos->Close();
// 3.3333ms @ 48kHz <=> smallest possible buffer size we can use.
aos = aosw.Create(48000, 160);
EXPECT_TRUE(aos->Open());
aos->Close();
}
// Verify that we can open the output stream in exclusive mode using a
// certain set of audio parameters and a sample rate of 44.1kHz.
// The expected outcomes of each setting in this test has been derived
// manually using log outputs (--v=1).
// It's disabled by default because a flag is required to enable exclusive mode.
TEST_F(WASAPIAudioOutputStreamTest, DISABLED_ExclusiveModeBufferSizesAt44kHz) {
ABORT_AUDIO_TEST_IF_NOT(HasCoreAudioAndOutputDevices(audio_manager_.get()) &&
ExclusiveModeIsEnabled());
AudioOutputStreamWrapper aosw(audio_manager_.get());
// 10ms @ 44.1kHz does not work due to misalignment.
// This test will propose an aligned buffer size of 10.1587ms.
AudioOutputStream* aos = aosw.Create(44100, 441);
EXPECT_FALSE(aos->Open());
aos->Close();
// 10.1587ms @ 44.1kHz shall work (see test above).
aos = aosw.Create(44100, 448);
EXPECT_TRUE(aos->Open());
aos->Close();
// 5.8050ms @ 44.1 should work.
aos = aosw.Create(44100, 256);
EXPECT_TRUE(aos->Open());
aos->Close();
// 4.9887ms @ 44.1kHz does not work to misalignment.
// This test will propose an aligned buffer size of 5.0794ms.
// Note that we must call Close() even is Open() fails since Close() also
// deletes the object and we want to create a new object in the next test.
aos = aosw.Create(44100, 220);
EXPECT_FALSE(aos->Open());
aos->Close();
// 5.0794ms @ 44.1kHz shall work (see test above).
aos = aosw.Create(44100, 224);
EXPECT_TRUE(aos->Open());
aos->Close();
// 2.9025ms is smaller than the minimum supported size (=3ms).
aos = aosw.Create(44100, 132);
EXPECT_FALSE(aos->Open());
aos->Close();
// 3.01587ms is larger than the minimum size but is not aligned.
// This test will propose an aligned buffer size of 3.6281ms.
aos = aosw.Create(44100, 133);
EXPECT_FALSE(aos->Open());
aos->Close();
// 3.6281ms @ 44.1kHz <=> smallest possible buffer size we can use.
aos = aosw.Create(44100, 160);
EXPECT_TRUE(aos->Open());
aos->Close();
}
// Verify that we can open and start the output stream in exclusive mode at
// the lowest possible delay at 48kHz.
// It's disabled by default because a flag is required to enable exclusive mode.
TEST_F(WASAPIAudioOutputStreamTest,
DISABLED_ExclusiveModeMinBufferSizeAt48kHz) {
ABORT_AUDIO_TEST_IF_NOT(HasCoreAudioAndOutputDevices(audio_manager_.get()) &&
ExclusiveModeIsEnabled());
MockAudioSourceCallback source;
// Create exclusive-mode WASAPI output stream which plays out in stereo
// using the minimum buffer size at 48kHz sample rate.
AudioOutputStreamWrapper aosw(audio_manager_.get());
AudioOutputStream* aos = aosw.Create(48000, 160);
EXPECT_TRUE(aos->Open());
base::RunLoop loop;
// Derive the expected size in bytes of each packet.
base::TimeDelta packet_duration = base::Seconds(
static_cast<double>(aosw.samples_per_packet()) / aosw.sample_rate());
// Wait for the first callback and verify its parameters.
EXPECT_CALL(source, OnMoreData(HasValidDelay(packet_duration), _,
AudioGlitchInfo(), NotNull()))
.WillOnce(DoAll(base::test::RunClosure(loop.QuitWhenIdleClosure()),
Return(aosw.samples_per_packet())))
.WillRepeatedly(Return(aosw.samples_per_packet()));
aos->Start(&source);
base::SingleThreadTaskRunner::GetCurrentDefault()->PostDelayedTask(
FROM_HERE, loop.QuitWhenIdleClosure(), TestTimeouts::action_timeout());
loop.Run();
aos->Stop();
aos->Close();
}
// Verify that we can open and start the output stream in exclusive mode at
// the lowest possible delay at 44.1kHz.
// It's disabled by default because a flag is required to enable exclusive mode.
TEST_F(WASAPIAudioOutputStreamTest,
DISABLED_ExclusiveModeMinBufferSizeAt44kHz) {
ABORT_AUDIO_TEST_IF_NOT(ExclusiveModeIsEnabled());
MockAudioSourceCallback source;
// Create exclusive-mode WASAPI output stream which plays out in stereo
// using the minimum buffer size at 44.1kHz sample rate.
AudioOutputStreamWrapper aosw(audio_manager_.get());
AudioOutputStream* aos = aosw.Create(44100, 160);
EXPECT_TRUE(aos->Open());
base::RunLoop loop;
// Derive the expected size in bytes of each packet.
base::TimeDelta packet_duration = base::Seconds(
static_cast<double>(aosw.samples_per_packet()) / aosw.sample_rate());
// Wait for the first callback and verify its parameters.
EXPECT_CALL(source, OnMoreData(HasValidDelay(packet_duration), _,
AudioGlitchInfo(), NotNull()))
.WillOnce(DoAll(base::test::RunClosure(loop.QuitWhenIdleClosure()),
Return(aosw.samples_per_packet())))
.WillRepeatedly(Return(aosw.samples_per_packet()));
aos->Start(&source);
base::SingleThreadTaskRunner::GetCurrentDefault()->PostDelayedTask(
FROM_HERE, loop.QuitWhenIdleClosure(), TestTimeouts::action_timeout());
loop.Run();
aos->Stop();
aos->Close();
}
} // namespace media
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