chromium/third_party/webrtc/call/rtp_video_sender.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_RTP_VIDEO_SENDER_H_
#define CALL_RTP_VIDEO_SENDER_H_

#include <map>
#include <memory>
#include <unordered_set>
#include <vector>

#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/call/transport.h"
#include "api/environment/environment.h"
#include "api/fec_controller.h"
#include "api/fec_controller_override.h"
#include "api/sequence_checker.h"
#include "api/video_codecs/video_encoder.h"
#include "call/rtp_config.h"
#include "call/rtp_payload_params.h"
#include "call/rtp_transport_controller_send_interface.h"
#include "call/rtp_video_sender_interface.h"
#include "modules/rtp_rtcp/include/flexfec_sender.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
#include "modules/rtp_rtcp/source/rtp_sender.h"
#include "modules/rtp_rtcp/source/rtp_sender_video.h"
#include "modules/rtp_rtcp/source/rtp_sequence_number_map.h"
#include "modules/rtp_rtcp/source/rtp_video_header.h"
#include "rtc_base/rate_limiter.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/thread_annotations.h"

namespace webrtc {

class FrameEncryptorInterface;
class RtpTransportControllerSendInterface;

namespace webrtc_internal_rtp_video_sender {
// RTP state for a single simulcast stream. Internal to the implementation of
// RtpVideoSender.
struct RtpStreamSender { … };

}  // namespace webrtc_internal_rtp_video_sender

// RtpVideoSender routes outgoing data to the correct sending RTP module, based
// on the simulcast layer in RTPVideoHeader.
class RtpVideoSender : public RtpVideoSenderInterface,
                       public VCMProtectionCallback,
                       public StreamFeedbackObserver { … };

}  // namespace webrtc

#endif  // CALL_RTP_VIDEO_SENDER_H_