blob: 60648d2c46f937cb31f567ca1b3afcf0dcfec75d [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_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/memory/ptr_util.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/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/gmock_mutant.h"
#include "testing/gtest/include/gtest/gtest.h"
using ::testing::_;
using ::testing::AnyNumber;
using ::testing::AtLeast;
using ::testing::Between;
using ::testing::CreateFunctor;
using ::testing::DoAll;
using ::testing::Gt;
using ::testing::InvokeWithoutArgs;
using ::testing::NotNull;
using ::testing::Return;
namespace media {
static const char kSpeechFile_16b_s_48k[] = "speech_16b_stereo_48kHz.raw";
static const char kSpeechFile_16b_s_44k[] = "speech_16b_stereo_44kHz.raw";
static const size_t kFileDurationMs = 20000;
static const size_t kNumFileSegments = 2;
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;
}
// Used to terminate a loop from a different thread than the loop belongs to.
// |task_runner| should be a SingleThreadTaskRunner.
ACTION_P(QuitLoop, task_runner) {
task_runner->PostTask(FROM_HERE,
base::RunLoop::QuitCurrentWhenIdleClosureDeprecated());
}
// 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_(NULL),
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_, "%d\n", delta_times_[elements_written]);
++elements_written;
}
base::CloseFile(text_file_);
}
// AudioOutputStream::AudioSourceCallback implementation.
int OnMoreData(base::TimeDelta /* delay */,
base::TimeTicks /* delay_timestamp */,
int /* prior_frames_skipped */,
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_ + static_cast<int>(max_size) > file_size())
max_size = file_size() - pos_;
int frames = max_size / (dest->channels() * kBitsPerSample / 8);
if (max_size) {
dest->FromInterleaved(file_->data() + pos_, frames, kBitsPerSample / 8);
pos_ += max_size;
}
return frames;
}
void OnError() override {}
int file_size() { return file_->data_size(); }
private:
scoped_refptr<DecoderBuffer> file_;
std::unique_ptr<int[]> delta_times_;
int pos_;
base::TimeTicks previous_call_time_;
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)));
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();
}
AudioParameters::Format format() const { return format_; }
int channels() const { return ChannelLayoutToChannelCount(channel_layout_); }
int sample_rate() const { return sample_rate_; }
int samples_per_packet() const { return samples_per_packet_; }
private:
AudioOutputStream* CreateOutputStream() {
AudioOutputStream* aos = audio_man_->MakeAudioOutputStream(
AudioParameters(format_, channel_layout_, sample_rate_,
samples_per_packet_),
std::string(), AudioManager::LogCallback());
EXPECT_TRUE(aos);
return aos;
}
AudioManager* audio_man_;
AudioParameters::Format format_;
ChannelLayout channel_layout_;
int sample_rate_;
int samples_per_packet_;
};
// 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::MessageLoopForUI message_loop_;
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());
// Derive the expected duration of each packet.
base::TimeDelta packet_duration = base::TimeDelta::FromSecondsD(
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), _, 0, NotNull()))
.WillOnce(DoAll(QuitLoop(message_loop_.task_runner()),
Return(aosw.samples_per_packet())))
.WillRepeatedly(Return(0));
aos->Start(&source);
message_loop_.task_runner()->PostDelayedTask(
FROM_HERE, base::RunLoop::QuitCurrentWhenIdleClosureDeprecated(),
TestTimeouts::action_timeout());
base::RunLoop().Run();
aos->Stop();
aos->Close();
}
// This test is intended for manual tests and should only be enabled
// when it is required to play out data from a local PCM 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.
// The test files are approximately 20 seconds long.
TEST_F(WASAPIAudioOutputStreamTest, DISABLED_ReadFromStereoFile) {
ABORT_AUDIO_TEST_IF_NOT(HasCoreAudioAndOutputDevices(audio_manager_.get()));
AudioOutputStreamWrapper aosw(audio_manager_.get());
AudioOutputStream* aos = aosw.Create();
EXPECT_TRUE(aos->Open());
std::string file_name;
if (aosw.sample_rate() == 48000) {
file_name = kSpeechFile_16b_s_48k;
} else if (aosw.sample_rate() == 44100) {
file_name = kSpeechFile_16b_s_44k;
} else if (aosw.sample_rate() == 96000) {
// Use 48kHz file at 96kHz as well. Will sound like Donald Duck.
file_name = kSpeechFile_16b_s_48k;
} else {
FAIL() << "This test supports 44.1, 48kHz and 96kHz only.";
return;
}
ReadFromFileAudioSource file_source(file_name);
DVLOG(0) << "File name : " << file_name.c_str();
DVLOG(0) << "Sample rate : " << aosw.sample_rate();
DVLOG(0) << "#channels : " << aosw.channels();
DVLOG(0) << "File size : " << file_source.file_size();
DVLOG(0) << "#file segments : " << kNumFileSegments;
DVLOG(0) << ">> Listen to the stereo file while playing...";
for (size_t i = 0; i < kNumFileSegments; i++) {
// Each segment will start with a short (~20ms) block of zeros, hence
// some short glitches might be heard in this test if kNumFileSegments
// is larger than one. The exact length of the silence period depends on
// the selected sample rate.
aos->Start(&file_source);
base::PlatformThread::Sleep(
base::TimeDelta::FromMilliseconds(kFileDurationMs / kNumFileSegments));
aos->Stop();
}
DVLOG(0) << ">> Stereo file playout has stopped.";
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());
// Derive the expected size in bytes of each packet.
base::TimeDelta packet_duration = base::TimeDelta::FromSecondsD(
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), _, 0, NotNull()))
.WillOnce(DoAll(QuitLoop(message_loop_.task_runner()),
Return(aosw.samples_per_packet())))
.WillRepeatedly(Return(aosw.samples_per_packet()));
aos->Start(&source);
message_loop_.task_runner()->PostDelayedTask(
FROM_HERE, base::RunLoop::QuitCurrentWhenIdleClosureDeprecated(),
TestTimeouts::action_timeout());
base::RunLoop().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());
// Derive the expected size in bytes of each packet.
base::TimeDelta packet_duration = base::TimeDelta::FromSecondsD(
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), _, 0, NotNull()))
.WillOnce(DoAll(QuitLoop(message_loop_.task_runner()),
Return(aosw.samples_per_packet())))
.WillRepeatedly(Return(aosw.samples_per_packet()));
aos->Start(&source);
message_loop_.task_runner()->PostDelayedTask(
FROM_HERE, base::RunLoop::QuitCurrentWhenIdleClosureDeprecated(),
TestTimeouts::action_timeout());
base::RunLoop().Run();
aos->Stop();
aos->Close();
}
} // namespace media