rtp_sender_video.h | Explore in Territory

/*
 *  Copyright (c) 2012 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 MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_H_
#define MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_H_

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

#include "absl/strings/string_view.h"
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/frame_transformer_interface.h"
#include "api/scoped_refptr.h"
#include "api/sequence_checker.h"
#include "api/task_queue/task_queue_base.h"
#include "api/task_queue/task_queue_factory.h"
#include "api/transport/rtp/dependency_descriptor.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
#include "api/video/video_codec_type.h"
#include "api/video/video_frame_type.h"
#include "api/video/video_layers_allocation.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/absolute_capture_time_sender.h"
#include "modules/rtp_rtcp/source/active_decode_targets_helper.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
#include "modules/rtp_rtcp/source/rtp_sender.h"
#include "modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.h"
#include "modules/rtp_rtcp/source/rtp_video_header.h"
#include "modules/rtp_rtcp/source/video_fec_generator.h"
#include "rtc_base/bitrate_tracker.h"
#include "rtc_base/frequency_tracker.h"
#include "rtc_base/one_time_event.h"
#include "rtc_base/race_checker.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/thread_annotations.h"

namespace webrtc {

class FrameEncryptorInterface;
class RtpPacketizer;
class RtpPacketToSend;

// kConditionallyRetransmitHigherLayers allows retransmission of video frames
// in higher layers if either the last frame in that layer was too far back in
// time, or if we estimate that a new frame will be available in a lower layer
// in a shorter time than it would take to request and receive a retransmission.
enum RetransmissionMode : uint8_t { … };

class RTPSenderVideo : public RTPVideoFrameSenderInterface { … };

}  // namespace webrtc

#endif  // MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_H_
chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h
