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// 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/command_line.h"
#include "base/file_util.h"
#include "base/files/file_path.h"
#include "base/logging.h"
#include "base/path_service.h"
#include "base/time/time.h"
#include "content/public/common/content_switches.h"
#include "content/renderer/media/media_stream_audio_processor.h"
#include "content/renderer/media/rtc_media_constraints.h"
#include "media/audio/audio_parameters.h"
#include "media/base/audio_bus.h"
#include "testing/gmock/include/gmock/gmock.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
using ::testing::_;
using ::testing::AnyNumber;
using ::testing::AtLeast;
using ::testing::Return;
namespace content {
namespace {
#if defined(ANDROID)
const int kAudioProcessingSampleRate = 16000;
#else
const int kAudioProcessingSampleRate = 32000;
#endif
const int kAudioProcessingNumberOfChannel = 1;
// The number of packers used for testing.
const int kNumberOfPacketsForTest = 100;
void ReadDataFromSpeechFile(char* data, int length) {
base::FilePath file;
CHECK(PathService::Get(base::DIR_SOURCE_ROOT, &file));
file = file.Append(FILE_PATH_LITERAL("media"))
.Append(FILE_PATH_LITERAL("test"))
.Append(FILE_PATH_LITERAL("data"))
.Append(FILE_PATH_LITERAL("speech_16b_stereo_48kHz.raw"));
DCHECK(base::PathExists(file));
int64 data_file_size64 = 0;
DCHECK(base::GetFileSize(file, &data_file_size64));
EXPECT_EQ(length, file_util::ReadFile(file, data, length));
DCHECK(data_file_size64 > length);
}
void ApplyFixedAudioConstraints(RTCMediaConstraints* constraints) {
// Constant constraint keys which enables default audio constraints on
// mediastreams with audio.
struct {
const char* key;
const char* value;
} static const kDefaultAudioConstraints[] = {
{ webrtc::MediaConstraintsInterface::kEchoCancellation,
webrtc::MediaConstraintsInterface::kValueTrue },
#if defined(OS_CHROMEOS) || defined(OS_MACOSX)
// Enable the extended filter mode AEC on platforms with known echo issues.
{ webrtc::MediaConstraintsInterface::kExperimentalEchoCancellation,
webrtc::MediaConstraintsInterface::kValueTrue },
#endif
{ webrtc::MediaConstraintsInterface::kAutoGainControl,
webrtc::MediaConstraintsInterface::kValueTrue },
{ webrtc::MediaConstraintsInterface::kExperimentalAutoGainControl,
webrtc::MediaConstraintsInterface::kValueTrue },
{ webrtc::MediaConstraintsInterface::kNoiseSuppression,
webrtc::MediaConstraintsInterface::kValueTrue },
{ webrtc::MediaConstraintsInterface::kHighpassFilter,
webrtc::MediaConstraintsInterface::kValueTrue },
};
for (size_t i = 0; i < ARRAYSIZE_UNSAFE(kDefaultAudioConstraints); ++i) {
constraints->AddMandatory(kDefaultAudioConstraints[i].key,
kDefaultAudioConstraints[i].value, false);
}
}
} // namespace
class MediaStreamAudioProcessorTest : public ::testing::Test {
public:
MediaStreamAudioProcessorTest()
: params_(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
media::CHANNEL_LAYOUT_STEREO, 48000, 16, 512) {
CommandLine::ForCurrentProcess()->AppendSwitch(
switches::kEnableAudioTrackProcessing);
}
protected:
// Helper method to save duplicated code.
void ProcessDataAndVerifyFormat(MediaStreamAudioProcessor* audio_processor,
int expected_output_sample_rate,
int expected_output_channels,
int expected_output_buffer_size) {
// Read the audio data from a file.
const int packet_size =
params_.frames_per_buffer() * 2 * params_.channels();
const size_t length = packet_size * kNumberOfPacketsForTest;
scoped_ptr<char[]> capture_data(new char[length]);
ReadDataFromSpeechFile(capture_data.get(), length);
const int16* data_ptr = reinterpret_cast<const int16*>(capture_data.get());
scoped_ptr<media::AudioBus> data_bus = media::AudioBus::Create(
params_.channels(), params_.frames_per_buffer());
for (int i = 0; i < kNumberOfPacketsForTest; ++i) {
data_bus->FromInterleaved(data_ptr, data_bus->frames(), 2);
audio_processor->PushCaptureData(data_bus.get());
// |audio_processor| does nothing when the audio processing is off in
// the processor.
audio_processor->PushRenderData(
data_ptr,
params_.sample_rate(), params_.channels(),
params_.frames_per_buffer(), base::TimeDelta::FromMilliseconds(10));
int16* output = NULL;
while(audio_processor->ProcessAndConsumeData(
base::TimeDelta::FromMilliseconds(10), 255, false, &output)) {
EXPECT_TRUE(output != NULL);
EXPECT_EQ(audio_processor->OutputFormat().sample_rate(),
expected_output_sample_rate);
EXPECT_EQ(audio_processor->OutputFormat().channels(),
expected_output_channels);
EXPECT_EQ(audio_processor->OutputFormat().frames_per_buffer(),
expected_output_buffer_size);
}
data_ptr += params_.frames_per_buffer() * params_.channels();
}
}
media::AudioParameters params_;
};
TEST_F(MediaStreamAudioProcessorTest, WithoutAudioProcessing) {
// Setup the audio processor with empty constraint.
RTCMediaConstraints constraints;
MediaStreamAudioProcessor audio_processor(&constraints);
audio_processor.SetCaptureFormat(params_);
EXPECT_FALSE(audio_processor.has_audio_processing());
ProcessDataAndVerifyFormat(&audio_processor,
params_.sample_rate(),
params_.channels(),
params_.sample_rate() / 100);
}
TEST_F(MediaStreamAudioProcessorTest, WithAudioProcessing) {
// Setup the audio processor with default constraint.
RTCMediaConstraints constraints;
ApplyFixedAudioConstraints(&constraints);
MediaStreamAudioProcessor audio_processor(&constraints);
audio_processor.SetCaptureFormat(params_);
EXPECT_TRUE(audio_processor.has_audio_processing());
ProcessDataAndVerifyFormat(&audio_processor,
kAudioProcessingSampleRate,
kAudioProcessingNumberOfChannel,
kAudioProcessingSampleRate / 100);
}
} // namespace content
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