blob: 53b78a7d5f972de6d06597ff9b231bb04983c4dd [file] [log] [blame]
// Copyright 2013 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 "base/basictypes.h"
#include "base/file_util.h"
#include "base/memory/scoped_ptr.h"
#include "base/message_loop/message_loop.h"
#include "base/path_service.h"
#include "base/strings/stringprintf.h"
#include "base/synchronization/lock.h"
#include "base/synchronization/waitable_event.h"
#include "base/test/test_timeouts.h"
#include "base/time/time.h"
#include "build/build_config.h"
#include "media/audio/android/audio_manager_android.h"
#include "media/audio/audio_io.h"
#include "media/audio/audio_manager_base.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::AtLeast;
using ::testing::DoAll;
using ::testing::Invoke;
using ::testing::NotNull;
using ::testing::Return;
namespace media {
ACTION_P3(CheckCountAndPostQuitTask, count, limit, loop) {
if (++*count >= limit) {
loop->PostTask(FROM_HERE, base::MessageLoop::QuitClosure());
}
}
static const char kSpeechFile_16b_s_48k[] = "speech_16b_stereo_48kHz.raw";
static const char kSpeechFile_16b_m_48k[] = "speech_16b_mono_48kHz.raw";
static const char kSpeechFile_16b_s_44k[] = "speech_16b_stereo_44kHz.raw";
static const char kSpeechFile_16b_m_44k[] = "speech_16b_mono_44kHz.raw";
static const float kCallbackTestTimeMs = 2000.0;
static const int kBitsPerSample = 16;
static const int kBytesPerSample = kBitsPerSample / 8;
// Converts AudioParameters::Format enumerator to readable string.
static std::string FormatToString(AudioParameters::Format format) {
switch (format) {
case AudioParameters::AUDIO_PCM_LINEAR:
return std::string("AUDIO_PCM_LINEAR");
case AudioParameters::AUDIO_PCM_LOW_LATENCY:
return std::string("AUDIO_PCM_LOW_LATENCY");
case AudioParameters::AUDIO_FAKE:
return std::string("AUDIO_FAKE");
case AudioParameters::AUDIO_LAST_FORMAT:
return std::string("AUDIO_LAST_FORMAT");
default:
return std::string();
}
}
// Converts ChannelLayout enumerator to readable string. Does not include
// multi-channel cases since these layouts are not supported on Android.
static std::string LayoutToString(ChannelLayout channel_layout) {
switch (channel_layout) {
case CHANNEL_LAYOUT_NONE:
return std::string("CHANNEL_LAYOUT_NONE");
case CHANNEL_LAYOUT_MONO:
return std::string("CHANNEL_LAYOUT_MONO");
case CHANNEL_LAYOUT_STEREO:
return std::string("CHANNEL_LAYOUT_STEREO");
case CHANNEL_LAYOUT_UNSUPPORTED:
default:
return std::string("CHANNEL_LAYOUT_UNSUPPORTED");
}
}
static double ExpectedTimeBetweenCallbacks(AudioParameters params) {
return (base::TimeDelta::FromMicroseconds(
params.frames_per_buffer() * base::Time::kMicrosecondsPerSecond /
static_cast<double>(params.sample_rate()))).InMillisecondsF();
}
std::ostream& operator<<(std::ostream& os, const AudioParameters& params) {
using namespace std;
os << endl << "format: " << FormatToString(params.format()) << endl
<< "channel layout: " << LayoutToString(params.channel_layout()) << endl
<< "sample rate: " << params.sample_rate() << endl
<< "bits per sample: " << params.bits_per_sample() << endl
<< "frames per buffer: " << params.frames_per_buffer() << endl
<< "channels: " << params.channels() << endl
<< "bytes per buffer: " << params.GetBytesPerBuffer() << endl
<< "bytes per second: " << params.GetBytesPerSecond() << endl
<< "bytes per frame: " << params.GetBytesPerFrame() << endl
<< "frame size in ms: " << ExpectedTimeBetweenCallbacks(params);
return os;
}
// Gmock implementation of AudioInputStream::AudioInputCallback.
class MockAudioInputCallback : public AudioInputStream::AudioInputCallback {
public:
MOCK_METHOD5(OnData,
void(AudioInputStream* stream,
const uint8* src,
uint32 size,
uint32 hardware_delay_bytes,
double volume));
MOCK_METHOD1(OnClose, void(AudioInputStream* stream));
MOCK_METHOD1(OnError, void(AudioInputStream* stream));
};
// Gmock implementation of AudioOutputStream::AudioSourceCallback.
class MockAudioOutputCallback : public AudioOutputStream::AudioSourceCallback {
public:
MOCK_METHOD2(OnMoreData,
int(AudioBus* dest, AudioBuffersState buffers_state));
MOCK_METHOD3(OnMoreIOData,
int(AudioBus* source,
AudioBus* dest,
AudioBuffersState buffers_state));
MOCK_METHOD1(OnError, void(AudioOutputStream* stream));
// We clear the data bus to ensure that the test does not cause noise.
int RealOnMoreData(AudioBus* dest, AudioBuffersState buffers_state) {
dest->Zero();
return dest->frames();
}
};
// Implements AudioOutputStream::AudioSourceCallback and provides audio data
// by reading from a data file.
class FileAudioSource : public AudioOutputStream::AudioSourceCallback {
public:
explicit FileAudioSource(base::WaitableEvent* event, const std::string& name)
: event_(event), pos_(0) {
// Reads a test file from media/test/data directory and stores it in
// a DecoderBuffer.
file_ = ReadTestDataFile(name);
// Log the name of the file which is used as input for this test.
base::FilePath file_path = GetTestDataFilePath(name);
VLOG(0) << "Reading from file: " << file_path.value().c_str();
}
virtual ~FileAudioSource() {}
// AudioOutputStream::AudioSourceCallback implementation.
// Use samples read from a data file and fill up the audio buffer
// provided to us in the callback.
virtual int OnMoreData(AudioBus* audio_bus,
AudioBuffersState buffers_state) OVERRIDE {
bool stop_playing = false;
int max_size =
audio_bus->frames() * audio_bus->channels() * kBytesPerSample;
// Adjust data size and prepare for end signal if file has ended.
if (pos_ + max_size > file_size()) {
stop_playing = true;
max_size = file_size() - pos_;
}
// File data is stored as interleaved 16-bit values. Copy data samples from
// the file and deinterleave to match the audio bus format.
// FromInterleaved() will zero out any unfilled frames when there is not
// sufficient data remaining in the file to fill up the complete frame.
int frames = max_size / (audio_bus->channels() * kBytesPerSample);
if (max_size) {
audio_bus->FromInterleaved(file_->data() + pos_, frames, kBytesPerSample);
pos_ += max_size;
}
// Set event to ensure that the test can stop when the file has ended.
if (stop_playing)
event_->Signal();
return frames;
}
virtual int OnMoreIOData(AudioBus* source,
AudioBus* dest,
AudioBuffersState buffers_state) OVERRIDE {
NOTREACHED();
return 0;
}
virtual void OnError(AudioOutputStream* stream) OVERRIDE {}
int file_size() { return file_->data_size(); }
private:
base::WaitableEvent* event_;
int pos_;
scoped_refptr<DecoderBuffer> file_;
DISALLOW_COPY_AND_ASSIGN(FileAudioSource);
};
// Implements AudioInputStream::AudioInputCallback and writes the recorded
// audio data to a local output file. Note that this implementation should
// only be used for manually invoked and evaluated tests, hence the created
// file will not be destroyed after the test is done since the intention is
// that it shall be available for off-line analysis.
class FileAudioSink : public AudioInputStream::AudioInputCallback {
public:
explicit FileAudioSink(base::WaitableEvent* event,
const AudioParameters& params,
const std::string& file_name)
: event_(event), params_(params) {
// Allocate space for ~10 seconds of data.
const int kMaxBufferSize = 10 * params.GetBytesPerSecond();
buffer_.reset(new media::SeekableBuffer(0, kMaxBufferSize));
// Open up the binary file which will be written to in the destructor.
base::FilePath file_path;
EXPECT_TRUE(PathService::Get(base::DIR_SOURCE_ROOT, &file_path));
file_path = file_path.AppendASCII(file_name.c_str());
binary_file_ = file_util::OpenFile(file_path, "wb");
DLOG_IF(ERROR, !binary_file_) << "Failed to open binary PCM data file.";
VLOG(0) << "Writing to file: " << file_path.value().c_str();
}
virtual ~FileAudioSink() {
int bytes_written = 0;
while (bytes_written < buffer_->forward_capacity()) {
const uint8* 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.
// TODO(henrika): use file_util:: instead.
fwrite(chunk, 1, chunk_size, binary_file_);
buffer_->Seek(chunk_size);
bytes_written += chunk_size;
}
file_util::CloseFile(binary_file_);
}
// AudioInputStream::AudioInputCallback implementation.
virtual void OnData(AudioInputStream* stream,
const uint8* src,
uint32 size,
uint32 hardware_delay_bytes,
double volume) OVERRIDE {
// 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.
if (!buffer_->Append(src, size))
event_->Signal();
}
virtual void OnClose(AudioInputStream* stream) OVERRIDE {}
virtual void OnError(AudioInputStream* stream) OVERRIDE {}
private:
base::WaitableEvent* event_;
AudioParameters params_;
scoped_ptr<media::SeekableBuffer> buffer_;
FILE* binary_file_;
DISALLOW_COPY_AND_ASSIGN(FileAudioSink);
};
// Implements AudioInputCallback and AudioSourceCallback to support full
// duplex audio where captured samples are played out in loopback after
// reading from a temporary FIFO storage.
class FullDuplexAudioSinkSource
: public AudioInputStream::AudioInputCallback,
public AudioOutputStream::AudioSourceCallback {
public:
explicit FullDuplexAudioSinkSource(const AudioParameters& params)
: params_(params),
previous_time_(base::TimeTicks::Now()),
started_(false) {
// Start with a reasonably small FIFO size. It will be increased
// dynamically during the test if required.
fifo_.reset(new media::SeekableBuffer(0, 2 * params.GetBytesPerBuffer()));
buffer_.reset(new uint8[params_.GetBytesPerBuffer()]);
}
virtual ~FullDuplexAudioSinkSource() {}
// AudioInputStream::AudioInputCallback implementation
virtual void OnData(AudioInputStream* stream,
const uint8* src,
uint32 size,
uint32 hardware_delay_bytes,
double volume) OVERRIDE {
const base::TimeTicks now_time = base::TimeTicks::Now();
const int diff = (now_time - previous_time_).InMilliseconds();
base::AutoLock lock(lock_);
if (diff > 1000) {
started_ = true;
previous_time_ = now_time;
// Log out the extra delay added by the FIFO. This is a best effort
// estimate. We might be +- 10ms off here.
int extra_fifo_delay =
static_cast<int>(BytesToMilliseconds(fifo_->forward_bytes() + size));
DVLOG(1) << extra_fifo_delay;
}
// We add an initial delay of ~1 second before loopback starts to ensure
// a stable callback sequence and to avoid initial bursts which might add
// to the extra FIFO delay.
if (!started_)
return;
// Append new data to the FIFO and extend the size if the max capacity
// was exceeded. Flush the FIFO when extended just in case.
if (!fifo_->Append(src, size)) {
fifo_->set_forward_capacity(2 * fifo_->forward_capacity());
fifo_->Clear();
}
}
virtual void OnClose(AudioInputStream* stream) OVERRIDE {}
virtual void OnError(AudioInputStream* stream) OVERRIDE {}
// AudioOutputStream::AudioSourceCallback implementation
virtual int OnMoreData(AudioBus* dest,
AudioBuffersState buffers_state) OVERRIDE {
const int size_in_bytes =
(params_.bits_per_sample() / 8) * dest->frames() * dest->channels();
EXPECT_EQ(size_in_bytes, params_.GetBytesPerBuffer());
base::AutoLock lock(lock_);
// We add an initial delay of ~1 second before loopback starts to ensure
// a stable callback sequences and to avoid initial bursts which might add
// to the extra FIFO delay.
if (!started_) {
dest->Zero();
return dest->frames();
}
// Fill up destination with zeros if the FIFO does not contain enough
// data to fulfill the request.
if (fifo_->forward_bytes() < size_in_bytes) {
dest->Zero();
} else {
fifo_->Read(buffer_.get(), size_in_bytes);
dest->FromInterleaved(
buffer_.get(), dest->frames(), params_.bits_per_sample() / 8);
}
return dest->frames();
}
virtual int OnMoreIOData(AudioBus* source,
AudioBus* dest,
AudioBuffersState buffers_state) OVERRIDE {
NOTREACHED();
return 0;
}
virtual void OnError(AudioOutputStream* stream) OVERRIDE {}
private:
// Converts from bytes to milliseconds given number of bytes and existing
// audio parameters.
double BytesToMilliseconds(int bytes) const {
const int frames = bytes / params_.GetBytesPerFrame();
return (base::TimeDelta::FromMicroseconds(
frames * base::Time::kMicrosecondsPerSecond /
static_cast<double>(params_.sample_rate()))).InMillisecondsF();
}
AudioParameters params_;
base::TimeTicks previous_time_;
base::Lock lock_;
scoped_ptr<media::SeekableBuffer> fifo_;
scoped_ptr<uint8[]> buffer_;
bool started_;
DISALLOW_COPY_AND_ASSIGN(FullDuplexAudioSinkSource);
};
// Test fixture class.
class AudioAndroidTest : public testing::Test {
public:
AudioAndroidTest() {}
protected:
virtual void SetUp() {
audio_manager_.reset(AudioManager::Create());
loop_.reset(new base::MessageLoopForUI());
}
virtual void TearDown() {}
AudioManager* audio_manager() { return audio_manager_.get(); }
base::MessageLoopForUI* loop() { return loop_.get(); }
AudioParameters GetDefaultInputStreamParameters() {
return audio_manager()->GetInputStreamParameters(
AudioManagerBase::kDefaultDeviceId);
}
AudioParameters GetDefaultOutputStreamParameters() {
return audio_manager()->GetDefaultOutputStreamParameters();
}
double AverageTimeBetweenCallbacks(int num_callbacks) const {
return ((end_time_ - start_time_) / static_cast<double>(num_callbacks - 1))
.InMillisecondsF();
}
void StartInputStreamCallbacks(const AudioParameters& params) {
double expected_time_between_callbacks_ms =
ExpectedTimeBetweenCallbacks(params);
const int num_callbacks =
(kCallbackTestTimeMs / expected_time_between_callbacks_ms);
AudioInputStream* stream = audio_manager()->MakeAudioInputStream(
params, AudioManagerBase::kDefaultDeviceId);
EXPECT_TRUE(stream);
int count = 0;
MockAudioInputCallback sink;
EXPECT_CALL(sink,
OnData(stream, NotNull(), params.GetBytesPerBuffer(), _, _))
.Times(AtLeast(num_callbacks))
.WillRepeatedly(
CheckCountAndPostQuitTask(&count, num_callbacks, loop()));
EXPECT_CALL(sink, OnError(stream)).Times(0);
EXPECT_CALL(sink, OnClose(stream)).Times(1);
EXPECT_TRUE(stream->Open());
stream->Start(&sink);
start_time_ = base::TimeTicks::Now();
loop()->Run();
end_time_ = base::TimeTicks::Now();
stream->Stop();
stream->Close();
double average_time_between_callbacks_ms =
AverageTimeBetweenCallbacks(num_callbacks);
VLOG(0) << "expected time between callbacks: "
<< expected_time_between_callbacks_ms << " ms";
VLOG(0) << "average time between callbacks: "
<< average_time_between_callbacks_ms << " ms";
EXPECT_GE(average_time_between_callbacks_ms,
0.70 * expected_time_between_callbacks_ms);
EXPECT_LE(average_time_between_callbacks_ms,
1.30 * expected_time_between_callbacks_ms);
}
void StartOutputStreamCallbacks(const AudioParameters& params) {
double expected_time_between_callbacks_ms =
ExpectedTimeBetweenCallbacks(params);
const int num_callbacks =
(kCallbackTestTimeMs / expected_time_between_callbacks_ms);
AudioOutputStream* stream = audio_manager()->MakeAudioOutputStream(
params, std::string(), std::string());
EXPECT_TRUE(stream);
int count = 0;
MockAudioOutputCallback source;
EXPECT_CALL(source, OnMoreData(NotNull(), _))
.Times(AtLeast(num_callbacks))
.WillRepeatedly(
DoAll(CheckCountAndPostQuitTask(&count, num_callbacks, loop()),
Invoke(&source, &MockAudioOutputCallback::RealOnMoreData)));
EXPECT_CALL(source, OnError(stream)).Times(0);
EXPECT_CALL(source, OnMoreIOData(_, _, _)).Times(0);
EXPECT_TRUE(stream->Open());
stream->Start(&source);
start_time_ = base::TimeTicks::Now();
loop()->Run();
end_time_ = base::TimeTicks::Now();
stream->Stop();
stream->Close();
double average_time_between_callbacks_ms =
AverageTimeBetweenCallbacks(num_callbacks);
VLOG(0) << "expected time between callbacks: "
<< expected_time_between_callbacks_ms << " ms";
VLOG(0) << "average time between callbacks: "
<< average_time_between_callbacks_ms << " ms";
EXPECT_GE(average_time_between_callbacks_ms,
0.70 * expected_time_between_callbacks_ms);
EXPECT_LE(average_time_between_callbacks_ms,
1.30 * expected_time_between_callbacks_ms);
}
scoped_ptr<base::MessageLoopForUI> loop_;
scoped_ptr<AudioManager> audio_manager_;
base::TimeTicks start_time_;
base::TimeTicks end_time_;
DISALLOW_COPY_AND_ASSIGN(AudioAndroidTest);
};
// Get the default audio input parameters and log the result.
TEST_F(AudioAndroidTest, GetInputStreamParameters) {
AudioParameters params = GetDefaultInputStreamParameters();
EXPECT_TRUE(params.IsValid());
VLOG(1) << params;
}
// Get the default audio output parameters and log the result.
TEST_F(AudioAndroidTest, GetDefaultOutputStreamParameters) {
AudioParameters params = GetDefaultOutputStreamParameters();
EXPECT_TRUE(params.IsValid());
VLOG(1) << params;
}
// Check if low-latency output is supported and log the result as output.
TEST_F(AudioAndroidTest, IsAudioLowLatencySupported) {
AudioManagerAndroid* manager =
static_cast<AudioManagerAndroid*>(audio_manager());
bool low_latency = manager->IsAudioLowLatencySupported();
low_latency ? VLOG(0) << "Low latency output is supported"
: VLOG(0) << "Low latency output is *not* supported";
}
// Ensure that a default input stream can be created and closed.
TEST_F(AudioAndroidTest, CreateAndCloseInputStream) {
AudioParameters params = GetDefaultInputStreamParameters();
AudioInputStream* ais = audio_manager()->MakeAudioInputStream(
params, AudioManagerBase::kDefaultDeviceId);
EXPECT_TRUE(ais);
ais->Close();
}
// Ensure that a default output stream can be created and closed.
// TODO(henrika): should we also verify that this API changes the audio mode
// to communication mode, and calls RegisterHeadsetReceiver, the first time
// it is called?
TEST_F(AudioAndroidTest, CreateAndCloseOutputStream) {
AudioParameters params = GetDefaultOutputStreamParameters();
AudioOutputStream* aos = audio_manager()->MakeAudioOutputStream(
params, std::string(), std::string());
EXPECT_TRUE(aos);
aos->Close();
}
// Ensure that a default input stream can be opened and closed.
TEST_F(AudioAndroidTest, OpenAndCloseInputStream) {
AudioParameters params = GetDefaultInputStreamParameters();
AudioInputStream* ais = audio_manager()->MakeAudioInputStream(
params, AudioManagerBase::kDefaultDeviceId);
EXPECT_TRUE(ais);
EXPECT_TRUE(ais->Open());
ais->Close();
}
// Ensure that a default output stream can be opened and closed.
TEST_F(AudioAndroidTest, OpenAndCloseOutputStream) {
AudioParameters params = GetDefaultOutputStreamParameters();
AudioOutputStream* aos = audio_manager()->MakeAudioOutputStream(
params, std::string(), std::string());
EXPECT_TRUE(aos);
EXPECT_TRUE(aos->Open());
aos->Close();
}
// Start input streaming using default input parameters and ensure that the
// callback sequence is sane.
TEST_F(AudioAndroidTest, StartInputStreamCallbacks) {
AudioParameters params = GetDefaultInputStreamParameters();
StartInputStreamCallbacks(params);
}
// Start input streaming using non default input parameters and ensure that the
// callback sequence is sane. The only change we make in this test is to select
// a 10ms buffer size instead of the default size.
// TODO(henrika): possibly add support for more variations.
TEST_F(AudioAndroidTest, StartInputStreamCallbacksNonDefaultParameters) {
AudioParameters native_params = GetDefaultInputStreamParameters();
AudioParameters params(native_params.format(),
native_params.channel_layout(),
native_params.sample_rate(),
native_params.bits_per_sample(),
native_params.sample_rate() / 100);
StartInputStreamCallbacks(params);
}
// Start output streaming using default output parameters and ensure that the
// callback sequence is sane.
TEST_F(AudioAndroidTest, StartOutputStreamCallbacks) {
AudioParameters params = GetDefaultOutputStreamParameters();
StartOutputStreamCallbacks(params);
}
// Start output streaming using non default output parameters and ensure that
// the callback sequence is sane. The only change we make in this test is to
// select a 10ms buffer size instead of the default size and to open up the
// device in mono.
// TODO(henrika): possibly add support for more variations.
TEST_F(AudioAndroidTest, StartOutputStreamCallbacksNonDefaultParameters) {
AudioParameters native_params = GetDefaultOutputStreamParameters();
AudioParameters params(native_params.format(),
CHANNEL_LAYOUT_MONO,
native_params.sample_rate(),
native_params.bits_per_sample(),
native_params.sample_rate() / 100);
StartOutputStreamCallbacks(params);
}
// Play out a PCM file segment in real time and allow the user to verify that
// the rendered audio sounds OK.
// NOTE: this test requires user interaction and is not designed to run as an
// automatized test on bots.
TEST_F(AudioAndroidTest, DISABLED_RunOutputStreamWithFileAsSource) {
AudioParameters params = GetDefaultOutputStreamParameters();
VLOG(1) << params;
AudioOutputStream* aos = audio_manager()->MakeAudioOutputStream(
params, std::string(), std::string());
EXPECT_TRUE(aos);
std::string file_name;
if (params.sample_rate() == 48000 && params.channels() == 2) {
file_name = kSpeechFile_16b_s_48k;
} else if (params.sample_rate() == 48000 && params.channels() == 1) {
file_name = kSpeechFile_16b_m_48k;
} else if (params.sample_rate() == 44100 && params.channels() == 2) {
file_name = kSpeechFile_16b_s_44k;
} else if (params.sample_rate() == 44100 && params.channels() == 1) {
file_name = kSpeechFile_16b_m_44k;
} else {
FAIL() << "This test supports 44.1kHz and 48kHz mono/stereo only.";
return;
}
base::WaitableEvent event(false, false);
FileAudioSource source(&event, file_name);
EXPECT_TRUE(aos->Open());
aos->SetVolume(1.0);
aos->Start(&source);
VLOG(0) << ">> Verify that the file is played out correctly...";
EXPECT_TRUE(event.TimedWait(TestTimeouts::action_max_timeout()));
aos->Stop();
aos->Close();
}
// Start input streaming and run it for ten seconds while recording to a
// local audio file.
// NOTE: this test requires user interaction and is not designed to run as an
// automatized test on bots.
TEST_F(AudioAndroidTest, DISABLED_RunSimplexInputStreamWithFileAsSink) {
AudioParameters params = GetDefaultInputStreamParameters();
VLOG(1) << params;
AudioInputStream* ais = audio_manager()->MakeAudioInputStream(
params, AudioManagerBase::kDefaultDeviceId);
EXPECT_TRUE(ais);
std::string file_name = base::StringPrintf("out_simplex_%d_%d_%d.pcm",
params.sample_rate(),
params.frames_per_buffer(),
params.channels());
base::WaitableEvent event(false, false);
FileAudioSink sink(&event, params, file_name);
EXPECT_TRUE(ais->Open());
ais->Start(&sink);
VLOG(0) << ">> Speak into the microphone to record audio...";
EXPECT_TRUE(event.TimedWait(TestTimeouts::action_max_timeout()));
ais->Stop();
ais->Close();
}
// Same test as RunSimplexInputStreamWithFileAsSink but this time output
// streaming is active as well (reads zeros only).
// NOTE: this test requires user interaction and is not designed to run as an
// automatized test on bots.
TEST_F(AudioAndroidTest, DISABLED_RunDuplexInputStreamWithFileAsSink) {
AudioParameters in_params = GetDefaultInputStreamParameters();
AudioInputStream* ais = audio_manager()->MakeAudioInputStream(
in_params, AudioManagerBase::kDefaultDeviceId);
EXPECT_TRUE(ais);
AudioParameters out_params =
audio_manager()->GetDefaultOutputStreamParameters();
AudioOutputStream* aos = audio_manager()->MakeAudioOutputStream(
out_params, std::string(), std::string());
EXPECT_TRUE(aos);
std::string file_name = base::StringPrintf("out_duplex_%d_%d_%d.pcm",
in_params.sample_rate(),
in_params.frames_per_buffer(),
in_params.channels());
base::WaitableEvent event(false, false);
FileAudioSink sink(&event, in_params, file_name);
MockAudioOutputCallback source;
EXPECT_CALL(source, OnMoreData(NotNull(), _)).WillRepeatedly(
Invoke(&source, &MockAudioOutputCallback::RealOnMoreData));
EXPECT_CALL(source, OnError(aos)).Times(0);
EXPECT_CALL(source, OnMoreIOData(_, _, _)).Times(0);
EXPECT_TRUE(ais->Open());
EXPECT_TRUE(aos->Open());
ais->Start(&sink);
aos->Start(&source);
VLOG(0) << ">> Speak into the microphone to record audio";
EXPECT_TRUE(event.TimedWait(TestTimeouts::action_max_timeout()));
aos->Stop();
ais->Stop();
aos->Close();
ais->Close();
}
// Start audio in both directions while feeding captured data into a FIFO so
// it can be read directly (in loopback) by the render side. A small extra
// delay will be added by the FIFO and an estimate of this delay will be
// printed out during the test.
// NOTE: this test requires user interaction and is not designed to run as an
// automatized test on bots.
TEST_F(AudioAndroidTest,
DISABLED_RunSymmetricInputAndOutputStreamsInFullDuplex) {
// Get native audio parameters for the input side.
AudioParameters default_input_params = GetDefaultInputStreamParameters();
// Modify the parameters so that both input and output can use the same
// parameters by selecting 10ms as buffer size. This will also ensure that
// the output stream will be a mono stream since mono is default for input
// audio on Android.
AudioParameters io_params(default_input_params.format(),
default_input_params.channel_layout(),
default_input_params.sample_rate(),
default_input_params.bits_per_sample(),
default_input_params.sample_rate() / 100);
VLOG(1) << io_params;
// Create input and output streams using the common audio parameters.
AudioInputStream* ais = audio_manager()->MakeAudioInputStream(
io_params, AudioManagerBase::kDefaultDeviceId);
EXPECT_TRUE(ais);
AudioOutputStream* aos = audio_manager()->MakeAudioOutputStream(
io_params, std::string(), std::string());
EXPECT_TRUE(aos);
FullDuplexAudioSinkSource full_duplex(io_params);
// Start a full duplex audio session and print out estimates of the extra
// delay we should expect from the FIFO. If real-time delay measurements are
// performed, the result should be reduced by this extra delay since it is
// something that has been added by the test.
EXPECT_TRUE(ais->Open());
EXPECT_TRUE(aos->Open());
ais->Start(&full_duplex);
aos->Start(&full_duplex);
VLOG(1) << "HINT: an estimate of the extra FIFO delay will be updated "
<< "once per second during this test.";
VLOG(0) << ">> Speak into the mic and listen to the audio in loopback...";
fflush(stdout);
base::PlatformThread::Sleep(base::TimeDelta::FromSeconds(20));
printf("\n");
aos->Stop();
ais->Stop();
aos->Close();
ais->Close();
}
} // namespace media