chromium/third_party/webrtc/audio/channel_send_frame_transformer_delegate.h

/*
 *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
 *
 *  Use of this source code is governed by a BSD-style license
 *  that can be found in the LICENSE file in the root of the source
 *  tree. An additional intellectual property rights grant can be found
 *  in the file PATENTS.  All contributing project authors may
 *  be found in the AUTHORS file in the root of the source tree.
 */

#ifndef AUDIO_CHANNEL_SEND_FRAME_TRANSFORMER_DELEGATE_H_
#define AUDIO_CHANNEL_SEND_FRAME_TRANSFORMER_DELEGATE_H_

#include <memory>
#include <string>

#include "api/frame_transformer_interface.h"
#include "api/sequence_checker.h"
#include "api/task_queue/task_queue_base.h"
#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
#include "rtc_base/buffer.h"
#include "rtc_base/synchronization/mutex.h"

namespace webrtc {

// Delegates calls to FrameTransformerInterface to transform frames, and to
// ChannelSend to send the transformed frames using `send_frame_callback_` on
// the `encoder_queue_`.
// OnTransformedFrame() can be called from any thread, the delegate ensures
// thread-safe access to the ChannelSend callback.
class ChannelSendFrameTransformerDelegate : public TransformedFrameCallback { … };

std::unique_ptr<TransformableAudioFrameInterface> CloneSenderAudioFrame(
    TransformableAudioFrameInterface* original);

}  // namespace webrtc
#endif  // AUDIO_CHANNEL_SEND_FRAME_TRANSFORMER_DELEGATE_H_