chromium/third_party/webrtc/pc/rtp_sender.h

/*
 *  Copyright 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.
 */

// This file contains classes that implement RtpSenderInterface.
// An RtpSender associates a MediaStreamTrackInterface with an underlying
// transport (provided by AudioProviderInterface/VideoProviderInterface)

#ifndef PC_RTP_SENDER_H_
#define PC_RTP_SENDER_H_

#include <stddef.h>
#include <stdint.h>

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

#include "absl/types/optional.h"
#include "api/crypto/frame_encryptor_interface.h"
#include "api/dtls_transport_interface.h"
#include "api/dtmf_sender_interface.h"
#include "api/frame_transformer_interface.h"
#include "api/media_stream_interface.h"
#include "api/media_types.h"
#include "api/rtc_error.h"
#include "api/rtp_parameters.h"
#include "api/rtp_sender_interface.h"
#include "api/scoped_refptr.h"
#include "api/sequence_checker.h"
#include "media/base/audio_source.h"
#include "media/base/media_channel.h"
#include "pc/dtmf_sender.h"
#include "pc/legacy_stats_collector_interface.h"
#include "rtc_base/checks.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/thread.h"
#include "rtc_base/thread_annotations.h"

namespace webrtc {

bool UnimplementedRtpParameterHasValue(const RtpParameters& parameters);

// Internal interface used by PeerConnection.
class RtpSenderInternal : public RtpSenderInterface { … };

// Shared implementation for RtpSenderInternal interface.
class RtpSenderBase : public RtpSenderInternal, public ObserverInterface { … };

// LocalAudioSinkAdapter receives data callback as a sink to the local
// AudioTrack, and passes the data to the sink of AudioSource.
class LocalAudioSinkAdapter : public AudioTrackSinkInterface,
                              public cricket::AudioSource { … };

class AudioRtpSender : public DtmfProviderInterface, public RtpSenderBase { … };

class VideoRtpSender : public RtpSenderBase { … };

}  // namespace webrtc

#endif  // PC_RTP_SENDER_H_