third_party/chromium/media/webrtc/helpers.h

// Copyright 2019 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.

#ifndef MEDIA_WEBRTC_HELPERS_H_
#define MEDIA_WEBRTC_HELPERS_H_

#include "base/component_export.h"
#include "base/files/file.h"
#include "build/build_config.h"
#include "media/base/audio_bus.h"
#include "media/base/audio_parameters.h"
#include "media/base/audio_processing.h"
#include "third_party/abseil-cpp/absl/types/optional.h"
#include "third_party/webrtc/modules/audio_processing/include/audio_processing.h"

namespace rtc {
class TaskQueue;
}  // namespace rtc

namespace media {

COMPONENT_EXPORT(MEDIA_WEBRTC)
webrtc::StreamConfig CreateStreamConfig(const AudioParameters& parameters);

// Tests whether the audio bus data can be treated as upmixed mono audio:
// Returns true if there is at most one channel or if each sample is identical
// in the first two channels.
COMPONENT_EXPORT(MEDIA_WEBRTC)
bool LeftAndRightChannelsAreSymmetric(const AudioBus& audio);

// Creates and configures a `webrtc::AudioProcessing` audio processing module
// (APM), based on the provided parameters and on features and field trials.
COMPONENT_EXPORT(MEDIA_WEBRTC)
rtc::scoped_refptr<webrtc::AudioProcessing> CreateWebRtcAudioProcessingModule(
    const AudioProcessingSettings& settings);

// Starts the echo cancellation dump in
// |audio_processing|. |worker_queue| must be kept alive until either
// |audio_processing| is destroyed, or
// StopEchoCancellationDump(audio_processing) is called.
COMPONENT_EXPORT(MEDIA_WEBRTC)
void StartEchoCancellationDump(webrtc::AudioProcessing* audio_processing,
                               base::File aec_dump_file,
                               rtc::TaskQueue* worker_queue);

// Stops the echo cancellation dump in |audio_processing|.
// This method has no impact if echo cancellation dump has not been started on
// |audio_processing|.
COMPONENT_EXPORT(MEDIA_WEBRTC)
void StopEchoCancellationDump(webrtc::AudioProcessing* audio_processing);

}  // namespace media

#endif  // MEDIA_WEBRTC_HELPERS_H_