chromium/media/webrtc/helpers.cc

// Copyright 2019 The Chromium Authors
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.

#include "media/webrtc/helpers.h"

#include <string>

#include "base/feature_list.h"
#include "base/files/file_util.h"
#include "base/logging.h"
#include "base/metrics/field_trial_params.h"
#include "build/build_config.h"
#include "build/chromecast_buildflags.h"
#include "media/webrtc/webrtc_features.h"
#include "third_party/webrtc/modules/audio_processing/aec_dump/aec_dump_factory.h"
#include "third_party/webrtc/modules/audio_processing/include/audio_processing.h"

namespace media {
namespace {

Agc1Mode;

DownmixMethod;
const base::FeatureParam<DownmixMethod>::Option kDownmixMethodOptions[] = …;
constexpr DownmixMethod kDefaultDownmixMethod = …;
const base::FeatureParam<DownmixMethod> kWebRtcApmDownmixMethodParam = …;

void ConfigAutomaticGainControl(const AudioProcessingSettings& settings,
                                webrtc::AudioProcessing::Config& apm_config) { … }

}  // namespace

webrtc::StreamConfig CreateStreamConfig(const AudioParameters& parameters) { … }

void StartEchoCancellationDump(webrtc::AudioProcessing* audio_processing,
                               base::File aec_dump_file,
                               webrtc::TaskQueueBase* worker_queue) { … }

void StopEchoCancellationDump(webrtc::AudioProcessing* audio_processing) { … }

rtc::scoped_refptr<webrtc::AudioProcessing> CreateWebRtcAudioProcessingModule(
    const AudioProcessingSettings& settings) { … }
}  // namespace media