chromium/third_party/webrtc/call/audio_send_stream.h

/*
 *  Copyright (c) 2015 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 CALL_AUDIO_SEND_STREAM_H_
#define CALL_AUDIO_SEND_STREAM_H_

#include <memory>
#include <string>
#include <vector>

#include "absl/types/optional.h"
#include "api/audio/audio_processing_statistics.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/audio_encoder_factory.h"
#include "api/audio_codecs/audio_format.h"
#include "api/call/transport.h"
#include "api/crypto/crypto_options.h"
#include "api/crypto/frame_encryptor_interface.h"
#include "api/frame_transformer_interface.h"
#include "api/rtp_parameters.h"
#include "api/rtp_sender_interface.h"
#include "api/scoped_refptr.h"
#include "call/audio_sender.h"
#include "call/rtp_config.h"
#include "modules/rtp_rtcp/include/report_block_data.h"

namespace webrtc {

class AudioSendStream : public AudioSender { … };

}  // namespace webrtc

#endif  // CALL_AUDIO_SEND_STREAM_H_