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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/public/common/media_stream_request.h"
#include "content/renderer/media/media_stream_audio_processor.h"
#include "content/renderer/media/mock_media_constraint_factory.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/WebKit/public/platform/WebMediaConstraints.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, base::ReadFile(file, data, length));
DCHECK(data_file_size64 > length);
}
} // namespace
class MediaStreamAudioProcessorTest : public ::testing::Test {
public:
MediaStreamAudioProcessorTest()
: params_(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
media::CHANNEL_LAYOUT_STEREO, 48000, 16, 512) {
}
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.
webrtc::AudioProcessing* ap = audio_processor->audio_processing_.get();
#if defined(OS_ANDROID) || defined(OS_IOS)
const bool is_aec_enabled = ap && ap->echo_control_mobile()->is_enabled();
// AEC should be turned off for mobiles.
DCHECK(!ap || !ap->echo_cancellation()->is_enabled());
#else
const bool is_aec_enabled = ap && ap->echo_cancellation()->is_enabled();
#endif
if (is_aec_enabled) {
audio_processor->OnPlayoutData(data_bus.get(), params_.sample_rate(),
10);
}
int16* output = NULL;
int new_volume = 0;
while(audio_processor->ProcessAndConsumeData(
base::TimeDelta::FromMilliseconds(10), 255, false, &new_volume,
&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();
}
}
void VerifyDefaultComponents(MediaStreamAudioProcessor* audio_processor) {
webrtc::AudioProcessing* audio_processing =
audio_processor->audio_processing_.get();
#if defined(OS_ANDROID)
EXPECT_TRUE(audio_processing->echo_control_mobile()->is_enabled());
EXPECT_TRUE(audio_processing->echo_control_mobile()->routing_mode() ==
webrtc::EchoControlMobile::kSpeakerphone);
EXPECT_FALSE(audio_processing->echo_cancellation()->is_enabled());
#elif !defined(OS_IOS)
EXPECT_TRUE(audio_processing->echo_cancellation()->is_enabled());
EXPECT_TRUE(audio_processing->echo_cancellation()->suppression_level() ==
webrtc::EchoCancellation::kHighSuppression);
EXPECT_TRUE(audio_processing->echo_cancellation()->are_metrics_enabled());
EXPECT_TRUE(
audio_processing->echo_cancellation()->is_delay_logging_enabled());
#endif
EXPECT_TRUE(audio_processing->noise_suppression()->is_enabled());
EXPECT_TRUE(audio_processing->noise_suppression()->level() ==
webrtc::NoiseSuppression::kHigh);
EXPECT_TRUE(audio_processing->high_pass_filter()->is_enabled());
EXPECT_TRUE(audio_processing->gain_control()->is_enabled());
#if defined(OS_ANDROID) || defined(OS_IOS)
EXPECT_TRUE(audio_processing->gain_control()->mode() ==
webrtc::GainControl::kFixedDigital);
EXPECT_FALSE(audio_processing->voice_detection()->is_enabled());
#else
EXPECT_TRUE(audio_processing->gain_control()->mode() ==
webrtc::GainControl::kAdaptiveAnalog);
EXPECT_TRUE(audio_processing->voice_detection()->is_enabled());
EXPECT_TRUE(audio_processing->voice_detection()->likelihood() ==
webrtc::VoiceDetection::kVeryLowLikelihood);
#endif
}
media::AudioParameters params_;
};
TEST_F(MediaStreamAudioProcessorTest, WithoutAudioProcessing) {
// Setup the audio processor without enabling the flag.
blink::WebMediaConstraints constraints;
scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
new WebRtcAudioDeviceImpl());
scoped_refptr<MediaStreamAudioProcessor> audio_processor(
new talk_base::RefCountedObject<MediaStreamAudioProcessor>(
constraints, 0, MEDIA_DEVICE_AUDIO_CAPTURE,
webrtc_audio_device.get()));
EXPECT_FALSE(audio_processor->has_audio_processing());
audio_processor->OnCaptureFormatChanged(params_);
ProcessDataAndVerifyFormat(audio_processor,
params_.sample_rate(),
params_.channels(),
params_.sample_rate() / 100);
// Set |audio_processor| to NULL to make sure |webrtc_audio_device| outlives
// |audio_processor|.
audio_processor = NULL;
}
TEST_F(MediaStreamAudioProcessorTest, WithAudioProcessing) {
// Setup the audio processor with the flag enabled.
CommandLine::ForCurrentProcess()->AppendSwitch(
switches::kEnableAudioTrackProcessing);
blink::WebMediaConstraints constraints;
scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
new WebRtcAudioDeviceImpl());
scoped_refptr<MediaStreamAudioProcessor> audio_processor(
new talk_base::RefCountedObject<MediaStreamAudioProcessor>(
constraints, 0, MEDIA_DEVICE_AUDIO_CAPTURE,
webrtc_audio_device.get()));
EXPECT_TRUE(audio_processor->has_audio_processing());
audio_processor->OnCaptureFormatChanged(params_);
VerifyDefaultComponents(audio_processor);
ProcessDataAndVerifyFormat(audio_processor,
kAudioProcessingSampleRate,
kAudioProcessingNumberOfChannel,
kAudioProcessingSampleRate / 100);
// Set |audio_processor| to NULL to make sure |webrtc_audio_device| outlives
// |audio_processor|.
audio_processor = NULL;
}
TEST_F(MediaStreamAudioProcessorTest, VerifyTabCaptureWithoutAudioProcessing) {
// Setup the audio processor with enabling the flag.
CommandLine::ForCurrentProcess()->AppendSwitch(
switches::kEnableAudioTrackProcessing);
blink::WebMediaConstraints constraints;
scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
new WebRtcAudioDeviceImpl());
// Create MediaStreamAudioProcessor instance for MEDIA_TAB_AUDIO_CAPTURE type.
scoped_refptr<MediaStreamAudioProcessor> audio_processor(
new talk_base::RefCountedObject<MediaStreamAudioProcessor>(
constraints, 0, MEDIA_TAB_AUDIO_CAPTURE,
webrtc_audio_device.get()));
EXPECT_FALSE(audio_processor->has_audio_processing());
audio_processor->OnCaptureFormatChanged(params_);
ProcessDataAndVerifyFormat(audio_processor,
params_.sample_rate(),
params_.channels(),
params_.sample_rate() / 100);
// Create MediaStreamAudioProcessor instance for MEDIA_LOOPBACK_AUDIO_CAPTURE.
audio_processor =
new talk_base::RefCountedObject<MediaStreamAudioProcessor>(
constraints, 0, MEDIA_LOOPBACK_AUDIO_CAPTURE,
webrtc_audio_device.get());
EXPECT_FALSE(audio_processor->has_audio_processing());
// Set |audio_processor| to NULL to make sure |webrtc_audio_device| outlives
// |audio_processor|.
audio_processor = NULL;
}
TEST_F(MediaStreamAudioProcessorTest, TurnOffDefaultConstraints) {
// Setup the audio processor with enabling the flag.
CommandLine::ForCurrentProcess()->AppendSwitch(
switches::kEnableAudioTrackProcessing);
// Turn off the default constraints and pass it to MediaStreamAudioProcessor.
MockMediaConstraintFactory constraint_factory;
constraint_factory.DisableDefaultAudioConstraints();
scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
new WebRtcAudioDeviceImpl());
scoped_refptr<MediaStreamAudioProcessor> audio_processor(
new talk_base::RefCountedObject<MediaStreamAudioProcessor>(
constraint_factory.CreateWebMediaConstraints(), 0,
MEDIA_DEVICE_AUDIO_CAPTURE, webrtc_audio_device.get()));
EXPECT_FALSE(audio_processor->has_audio_processing());
audio_processor->OnCaptureFormatChanged(params_);
ProcessDataAndVerifyFormat(audio_processor,
params_.sample_rate(),
params_.channels(),
params_.sample_rate() / 100);
// Set |audio_processor| to NULL to make sure |webrtc_audio_device| outlives
// |audio_processor|.
audio_processor = NULL;
}
} // namespace content
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