// 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 <stdint.h>
#include <memory>
#include "base/environment.h"
#include "base/functional/bind.h"
#include "base/memory/ptr_util.h"
#include "base/memory/raw_ptr.h"
#include "base/run_loop.h"
#include "base/task/single_thread_task_runner.h"
#include "base/test/task_environment.h"
#include "base/test/test_timeouts.h"
#include "base/threading/platform_thread.h"
#include "build/build_config.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_base.h"
#include "media/audio/audio_unittest_util.h"
#include "media/audio/apple/audio_low_latency_input.h"
#include "media/audio/test_audio_thread.h"
#include "media/base/audio_glitch_info.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::Ge;
using ::testing::NotNull;
namespace media {
ACTION_P4(CheckCountAndPostQuitTask, count, limit, task_runner, closure) {
if (++*count >= limit) {
task_runner->PostTask(FROM_HERE, closure);
}
}
class MockAudioInputCallback : public AudioInputStream::AudioInputCallback {
public:
MOCK_METHOD4(OnData,
void(const AudioBus* src,
base::TimeTicks capture_time,
double volume,
const AudioGlitchInfo& glitch_info));
MOCK_METHOD0(OnError, void());
};
// This audio sink implementation should be used for manual tests only since
// the recorded data is stored on a raw binary data file.
// The last test (WriteToFileAudioSink) - which is disabled by default -
// can use this audio sink to store the captured data on a file for offline
// analysis.
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 int kMaxBufferSize = 2 * 2 * 480 * 100 * 10;
explicit WriteToFileAudioSink(const char* file_name)
: buffer_(0, kMaxBufferSize),
file_(fopen(file_name, "wb")),
bytes_to_write_(0) {
}
~WriteToFileAudioSink() override {
int 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, file_);
buffer_.Seek(chunk_size);
bytes_written += chunk_size;
}
fclose(file_);
}
// AudioInputStream::AudioInputCallback implementation.
void OnData(const AudioBus* src,
base::TimeTicks capture_time,
double volume,
const AudioGlitchInfo& glitch_info) override {
const int num_samples = src->frames() * src->channels();
std::unique_ptr<int16_t> interleaved(new int16_t[num_samples]);
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 bytes_per_sample = sizeof(*interleaved);
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_;
raw_ptr<FILE> file_;
int bytes_to_write_;
};
class MacAudioInputTest : public testing::Test {
protected:
MacAudioInputTest()
: task_environment_(
base::test::SingleThreadTaskEnvironment::MainThreadType::UI),
audio_manager_(AudioManager::CreateForTesting(
std::make_unique<TestAudioThread>())) {
// Wait for the AudioManager to finish any initialization on the audio loop.
base::RunLoop().RunUntilIdle();
}
~MacAudioInputTest() override { audio_manager_->Shutdown(); }
bool InputDevicesAvailable() {
#if BUILDFLAG(IS_APPLE) && defined(ARCH_CPU_ARM64)
// TODO(crbug.com/40719640): macOS on ARM64 says it has devices, but won't
// let any of them be opened or listed.
return false;
#else
return AudioDeviceInfoAccessorForTests(audio_manager_.get())
.HasAudioInputDevices();
#endif
}
int HardwareSampleRateForDefaultInputDevice() {
// Determine the default input device's sample-rate.
AudioDeviceID input_device_id = kAudioObjectUnknown;
#if BUILDFLAG(IS_MAC)
AudioManagerMac::GetDefaultInputDevice(&input_device_id);
#endif
auto* manager = static_cast<AudioManagerApple*>(audio_manager_.get());
return manager->HardwareSampleRateForDevice(input_device_id);
}
// Convenience method which creates a default AudioInputStream object using
// a 10ms frame size and a sample rate which is set to the hardware sample
// rate.
AudioInputStream* CreateDefaultAudioInputStream() {
int fs = HardwareSampleRateForDefaultInputDevice();
int samples_per_packet = fs / 100;
#if BUILDFLAG(IS_MAC)
ChannelLayoutConfig channel_layout_config = ChannelLayoutConfig::Stereo();
#else
ChannelLayoutConfig channel_layout_config = ChannelLayoutConfig::Mono();
#endif
AudioInputStream* ais = audio_manager_->MakeAudioInputStream(
AudioParameters(AudioParameters::AUDIO_PCM_LOW_LATENCY,
channel_layout_config, fs, samples_per_packet),
AudioDeviceDescription::kDefaultDeviceId,
base::BindRepeating(&MacAudioInputTest::OnLogMessage,
base::Unretained(this)));
EXPECT_TRUE(ais);
return ais;
}
// Convenience method which creates an AudioInputStream object with a
// specified channel layout.
AudioInputStream* CreateAudioInputStream(
ChannelLayoutConfig channel_layout_config) {
int fs = HardwareSampleRateForDefaultInputDevice();
int samples_per_packet = fs / 100;
AudioInputStream* ais = audio_manager_->MakeAudioInputStream(
AudioParameters(AudioParameters::AUDIO_PCM_LOW_LATENCY,
channel_layout_config, fs, samples_per_packet),
AudioDeviceDescription::kDefaultDeviceId,
base::BindRepeating(&MacAudioInputTest::OnLogMessage,
base::Unretained(this)));
EXPECT_TRUE(ais);
return ais;
}
void OnLogMessage(const std::string& message) { log_message_ = message; }
base::test::SingleThreadTaskEnvironment task_environment_;
std::unique_ptr<AudioManager> audio_manager_;
std::string log_message_;
};
// Test Create(), Close().
TEST_F(MacAudioInputTest, AUAudioInputStreamCreateAndClose) {
ABORT_AUDIO_TEST_IF_NOT(InputDevicesAvailable());
AudioInputStream* ais = CreateDefaultAudioInputStream();
ais->Close();
}
// Test Open(), Close().
TEST_F(MacAudioInputTest, AUAudioInputStreamOpenAndClose) {
ABORT_AUDIO_TEST_IF_NOT(InputDevicesAvailable());
AudioInputStream* ais = CreateDefaultAudioInputStream();
EXPECT_EQ(ais->Open(), AudioInputStream::OpenOutcome::kSuccess);
ais->Close();
}
// Test Open(), Start(), Close().
TEST_F(MacAudioInputTest, AUAudioInputStreamOpenStartAndClose) {
ABORT_AUDIO_TEST_IF_NOT(InputDevicesAvailable());
AudioInputStream* ais = CreateDefaultAudioInputStream();
EXPECT_EQ(ais->Open(), AudioInputStream::OpenOutcome::kSuccess);
MockAudioInputCallback sink;
ais->Start(&sink);
ais->Close();
}
// Test Open(), Start(), Stop(), Close().
TEST_F(MacAudioInputTest, AUAudioInputStreamOpenStartStopAndClose) {
ABORT_AUDIO_TEST_IF_NOT(InputDevicesAvailable());
AudioInputStream* ais = CreateDefaultAudioInputStream();
EXPECT_EQ(ais->Open(), AudioInputStream::OpenOutcome::kSuccess);
MockAudioInputCallback sink;
ais->Start(&sink);
ais->Stop();
ais->Close();
}
// Verify that recording starts and stops correctly in mono using mocked sink.
TEST_F(MacAudioInputTest, AUAudioInputStreamVerifyMonoRecording) {
ABORT_AUDIO_TEST_IF_NOT(InputDevicesAvailable());
int count = 0;
// Create an audio input stream which records in mono.
AudioInputStream* ais = CreateAudioInputStream(ChannelLayoutConfig::Mono());
EXPECT_EQ(ais->Open(), AudioInputStream::OpenOutcome::kSuccess);
MockAudioInputCallback sink;
// 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, task_environment_.GetMainThreadTaskRunner(),
run_loop.QuitClosure()));
ais->Start(&sink);
run_loop.Run();
ais->Stop();
ais->Close();
EXPECT_FALSE(log_message_.empty());
}
// Verify that recording starts and stops correctly in mono using mocked sink.
TEST_F(MacAudioInputTest, AUAudioInputStreamVerifyStereoRecording) {
ABORT_AUDIO_TEST_IF_NOT(InputDevicesAvailable());
int count = 0;
// Create an audio input stream which records in stereo.
AudioInputStream* ais = CreateAudioInputStream(ChannelLayoutConfig::Stereo());
EXPECT_EQ(ais->Open(), AudioInputStream::OpenOutcome::kSuccess);
MockAudioInputCallback sink;
// 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.
// TODO(henrika): http://crbug.com/154352 forced us to run the capture side
// using a native buffer size of 128 audio frames and combine it with a FIFO
// to match the requested size by the client. This change might also have
// modified the delay estimates since the existing Ge(bytes_per_packet) for
// parameter #4 does no longer pass. I am removing this restriction here to
// ensure that we can land the patch but will revisit this test again when
// more analysis of the delay estimates are done.
base::RunLoop run_loop;
EXPECT_CALL(sink, OnData(NotNull(), _, _, _))
.Times(AtLeast(10))
.WillRepeatedly(CheckCountAndPostQuitTask(
&count, 10, task_environment_.GetMainThreadTaskRunner(),
run_loop.QuitClosure()));
ais->Start(&sink);
run_loop.Run();
ais->Stop();
ais->Close();
EXPECT_FALSE(log_message_.empty());
}
// 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(MacAudioInputTest, DISABLED_AUAudioInputStreamRecordToFile) {
ABORT_AUDIO_TEST_IF_NOT(InputDevicesAvailable());
const char* file_name = "out_stereo_10sec.pcm";
int fs = HardwareSampleRateForDefaultInputDevice();
AudioInputStream* ais = CreateDefaultAudioInputStream();
EXPECT_EQ(ais->Open(), AudioInputStream::OpenOutcome::kSuccess);
fprintf(stderr, " File name : %s\n", file_name);
fprintf(stderr, " Sample rate: %d\n", fs);
WriteToFileAudioSink file_sink(file_name);
fprintf(stderr, " >> Speak into the mic while recording...\n");
ais->Start(&file_sink);
base::PlatformThread::Sleep(TestTimeouts::action_timeout());
ais->Stop();
fprintf(stderr, " >> Recording has stopped.\n");
ais->Close();
}
TEST(MacAudioInputUpmixerTest, Upmix16bit) {
constexpr int kNumFrames = 512;
constexpr int kBytesPerSample = sizeof(int16_t);
int16_t mono[kNumFrames];
int16_t stereo[kNumFrames * 2];
// Fill the mono buffer and the first half of the stereo buffer with data
for (int i = 0; i != kNumFrames; ++i) {
mono[i] = i;
stereo[i] = i;
}
AudioBuffer audio_buffer;
audio_buffer.mNumberChannels = 2;
audio_buffer.mDataByteSize = kNumFrames * kBytesPerSample * 2;
audio_buffer.mData = stereo;
AUAudioInputStream::UpmixMonoToStereoInPlace(&audio_buffer, kBytesPerSample);
// Assert that the samples have been distributed properly
for (int i = 0; i != kNumFrames; ++i) {
ASSERT_EQ(mono[i], stereo[i * 2]);
ASSERT_EQ(mono[i], stereo[i * 2 + 1]);
}
}
TEST(MacAudioInputUpmixerTest, Upmix32bit) {
constexpr int kNumFrames = 512;
constexpr int kBytesPerSample = sizeof(int32_t);
int32_t mono[kNumFrames];
int32_t stereo[kNumFrames * 2];
// Fill the mono buffer and the first half of the stereo buffer with data
for (int i = 0; i != kNumFrames; ++i) {
mono[i] = i;
stereo[i] = i;
}
AudioBuffer audio_buffer;
audio_buffer.mNumberChannels = 2;
audio_buffer.mDataByteSize = kNumFrames * kBytesPerSample * 2;
audio_buffer.mData = stereo;
AUAudioInputStream::UpmixMonoToStereoInPlace(&audio_buffer, kBytesPerSample);
// Assert that the samples have been distributed properly
for (int i = 0; i != kNumFrames; ++i) {
ASSERT_EQ(mono[i], stereo[i * 2]);
ASSERT_EQ(mono[i], stereo[i * 2 + 1]);
}
}
} // namespace media
Codebase Browser
chromium