| Index: content/renderer/media/webrtc_audio_processor_options.cc
|
| diff --git a/content/renderer/media/webrtc_audio_processor_options.cc b/content/renderer/media/webrtc_audio_processor_options.cc
|
| new file mode 100644
|
| index 0000000000000000000000000000000000000000..e7abe04febf022293840f5e74db7f7f4fa734bf1
|
| --- /dev/null
|
| +++ b/content/renderer/media/webrtc_audio_processor_options.cc
|
| @@ -0,0 +1,96 @@
|
| +// 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/webrtc_audio_processor_options.h"
|
| +
|
| +#include "base/files/file_path.h"
|
| +#include "base/logging.h"
|
| +#include "base/path_service.h"
|
| +#include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface.h"
|
| +#include "third_party/webrtc/modules/audio_processing/include/audio_processing.h"
|
| +
|
| +namespace content {
|
| +
|
| +bool GetPropertyFromConstraints(const MediaConstraintsInterface* constraints,
|
| + const std::string& key) {
|
| + bool value = false;
|
| + return webrtc::FindConstraint(constraints, key, &value, NULL) && value;
|
| +}
|
| +
|
| +// Extract all these methods to a helper class.
|
| +void EnableEchoCancellation(AudioProcessing* audio_processing) {
|
| +#if defined(IOS) || defined(ANDROID)
|
| + // Mobile devices are using AECM.
|
| + if (audio_processing->echo_control_mobile()->Enable(true))
|
| + NOTREACHED();
|
| +
|
| + if (audio_processing->echo_control_mobile()->set_routing_mode(
|
| + webrtc::EchoControlMobile::kSpeakerphone))
|
| + NOTREACHED();
|
| +#else
|
| + if (audio_processing->echo_cancellation()->Enable(true))
|
| + NOTREACHED();
|
| + if (audio_processing->echo_cancellation()->set_suppression_level(
|
| + webrtc::EchoCancellation::kHighSuppression))
|
| + NOTREACHED();
|
| +
|
| + // Enable the metrics for AEC.
|
| + if (audio_processing->echo_cancellation()->enable_metrics(true))
|
| + NOTREACHED();
|
| + if (audio_processing->echo_cancellation()->enable_delay_logging(true))
|
| + NOTREACHED();
|
| +#endif
|
| +}
|
| +
|
| +void EnableNoiseSuppression(AudioProcessing* audio_processing) {
|
| + if (audio_processing->noise_suppression()->set_level(
|
| + webrtc::NoiseSuppression::kHigh))
|
| + NOTREACHED();
|
| +
|
| + if (audio_processing->noise_suppression()->Enable(true))
|
| + NOTREACHED();
|
| +}
|
| +
|
| +void EnableHighPassFilter(AudioProcessing* audio_processing) {
|
| + if (audio_processing->high_pass_filter()->Enable(true))
|
| + NOTREACHED();
|
| +}
|
| +
|
| +// TODO(xians): stereo swapping
|
| +void EnableTypingDetection(AudioProcessing* audio_processing) {
|
| + if (audio_processing->voice_detection()->Enable(true))
|
| + NOTREACHED();
|
| +
|
| + if (audio_processing->voice_detection()->set_likelihood(
|
| + webrtc::VoiceDetection::kVeryLowLikelihood))
|
| + NOTREACHED();
|
| +}
|
| +
|
| +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(xians): 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 (audio_processing->StartDebugRecording(file.value().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
|
|
|