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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 "content/renderer/media/media_stream_audio_processor_options.h"
#include "base/files/file_path.h"
#include "base/logging.h"
#include "base/path_service.h"
#include "base/strings/utf_string_conversions.h"
#include "content/renderer/media/rtc_media_constraints.h"
#include "media/audio/audio_parameters.h"
#include "third_party/WebKit/public/platform/WebMediaConstraints.h"
#include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface.h"
#include "third_party/webrtc/modules/audio_processing/include/audio_processing.h"
namespace content {
namespace {
// Constant constraint keys which enables default audio constraints on
// mediastreams with audio.
struct {
const char* key;
const char* value;
} 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 },
// TODO(xians): Verify if it is OK to set typing detection to kValueFalse as
// default.
{ webrtc::MediaConstraintsInterface::kTypingNoiseDetection,
webrtc::MediaConstraintsInterface::kValueFalse },
};
} // namespace
void ApplyFixedAudioConstraints(RTCMediaConstraints* constraints) {
for (size_t i = 0; i < ARRAYSIZE_UNSAFE(kDefaultAudioConstraints); ++i) {
bool already_set_value;
if (!webrtc::FindConstraint(constraints, kDefaultAudioConstraints[i].key,
&already_set_value, NULL)) {
constraints->AddMandatory(kDefaultAudioConstraints[i].key,
kDefaultAudioConstraints[i].value, false);
} else {
DVLOG(1) << "Constraint " << kDefaultAudioConstraints[i].key
<< " already set to " << already_set_value;
}
}
}
bool NeedsAudioProcessing(const blink::WebMediaConstraints& constraints,
int effects) {
RTCMediaConstraints native_constraints(constraints);
ApplyFixedAudioConstraints(&native_constraints);
if (effects & media::AudioParameters::ECHO_CANCELLER) {
// If platform echo cancellator is enabled, disable the software AEC.
native_constraints.AddMandatory(
MediaConstraintsInterface::kEchoCancellation,
MediaConstraintsInterface::kValueFalse, true);
}
for (size_t i = 0; i < ARRAYSIZE_UNSAFE(kDefaultAudioConstraints); ++i) {
bool value = false;
if (webrtc::FindConstraint(&native_constraints,
kDefaultAudioConstraints[i].key, &value, NULL) &&
value) {
return true;
}
}
return false;
}
bool GetPropertyFromConstraints(const MediaConstraintsInterface* constraints,
const std::string& key) {
bool value = false;
return webrtc::FindConstraint(constraints, key, &value, NULL) && value;
}
void EnableEchoCancellation(AudioProcessing* audio_processing) {
#if defined(OS_IOS)
// On iOS, VPIO provides built-in EC and AGC.
return;
#elif defined(OS_ANDROID)
// Mobile devices are using AECM.
int err = audio_processing->echo_control_mobile()->Enable(true);
err |= audio_processing->echo_control_mobile()->set_routing_mode(
webrtc::EchoControlMobile::kSpeakerphone);
CHECK_EQ(err, 0);
#else
int err = audio_processing->echo_cancellation()->Enable(true);
err |= audio_processing->echo_cancellation()->set_suppression_level(
webrtc::EchoCancellation::kHighSuppression);
// Enable the metrics for AEC.
err |= audio_processing->echo_cancellation()->enable_metrics(true);
err |= audio_processing->echo_cancellation()->enable_delay_logging(true);
CHECK_EQ(err, 0);
#endif
}
void EnableNoiseSuppression(AudioProcessing* audio_processing) {
int err = audio_processing->noise_suppression()->set_level(
webrtc::NoiseSuppression::kHigh);
err |= audio_processing->noise_suppression()->Enable(true);
CHECK_EQ(err, 0);
}
void EnableHighPassFilter(AudioProcessing* audio_processing) {
CHECK_EQ(audio_processing->high_pass_filter()->Enable(true), 0);
}
// TODO(xians): stereo swapping
void EnableTypingDetection(AudioProcessing* audio_processing) {
int err = audio_processing->voice_detection()->Enable(true);
err |= audio_processing->voice_detection()->set_likelihood(
webrtc::VoiceDetection::kVeryLowLikelihood);
CHECK_EQ(err, 0);
}
void EnableExperimentalEchoCancellation(AudioProcessing* audio_processing) {
webrtc::Config config;
config.Set<webrtc::DelayCorrection>(new webrtc::DelayCorrection(true));
audio_processing->SetExtraOptions(config);
}
void StartAecDump(AudioProcessing* audio_processing) {
// TODO(grunell): Figure out a more suitable directory for the audio dump
// data.
base::FilePath path;
#if defined(CHROMEOS)
PathService::Get(base::DIR_TEMP, &path);
#elif defined(ANDROID)
path = base::FilePath(FILE_PATH_LITERAL("sdcard"));
#else
PathService::Get(base::DIR_EXE, &path);
#endif
base::FilePath file = path.Append(FILE_PATH_LITERAL("audio.aecdump"));
#if defined(OS_WIN)
const std::string file_name = base::WideToUTF8(file.value());
#else
const std::string file_name = file.value();
#endif
if (audio_processing->StartDebugRecording(file_name.c_str()))
DLOG(ERROR) << "Fail to start AEC debug recording";
}
void StopAecDump(AudioProcessing* audio_processing) {
if (audio_processing->StopDebugRecording())
DLOG(ERROR) << "Fail to stop AEC debug recording";
}
} // namespace content
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