1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
|
/*
* 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 WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_H_
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_H_
#include <list>
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/bitrate.h"
#include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h"
#include "webrtc/modules/rtp_rtcp/source/producer_fec.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "webrtc/modules/rtp_rtcp/source/video_codec_information.h"
#include "webrtc/typedefs.h"
namespace webrtc {
class CriticalSectionWrapper;
struct RtpPacket;
class RTPSenderVideo {
public:
RTPSenderVideo(Clock* clock, RTPSenderInterface* rtpSender);
virtual ~RTPSenderVideo();
virtual RtpVideoCodecTypes VideoCodecType() const;
size_t FECPacketOverhead() const;
static RtpUtility::Payload* CreateVideoPayload(
const char payloadName[RTP_PAYLOAD_NAME_SIZE],
const int8_t payloadType,
const uint32_t maxBitRate);
int32_t SendVideo(const RtpVideoCodecTypes videoType,
const FrameType frameType,
const int8_t payloadType,
const uint32_t captureTimeStamp,
int64_t capture_time_ms,
const uint8_t* payloadData,
const size_t payloadSize,
const RTPFragmentationHeader* fragmentation,
const RTPVideoHeader* rtpHdr);
int32_t SendRTPIntraRequest();
void SetVideoCodecType(RtpVideoCodecTypes type);
void SetMaxConfiguredBitrateVideo(const uint32_t maxBitrate);
uint32_t MaxConfiguredBitrateVideo() const;
// FEC
void SetGenericFECStatus(const bool enable,
const uint8_t payloadTypeRED,
const uint8_t payloadTypeFEC);
void GenericFECStatus(bool* enable,
uint8_t* payloadTypeRED,
uint8_t* payloadTypeFEC) const;
void SetFecParameters(const FecProtectionParams* delta_params,
const FecProtectionParams* key_params);
void ProcessBitrate();
uint32_t VideoBitrateSent() const;
uint32_t FecOverheadRate() const;
int SelectiveRetransmissions() const;
void SetSelectiveRetransmissions(uint8_t settings);
private:
void SendVideoPacket(uint8_t* dataBuffer,
const size_t payloadLength,
const size_t rtpHeaderLength,
uint16_t seq_num,
const uint32_t capture_timestamp,
int64_t capture_time_ms,
StorageType storage);
void SendVideoPacketAsRed(uint8_t* dataBuffer,
const size_t payloadLength,
const size_t rtpHeaderLength,
uint16_t video_seq_num,
const uint32_t capture_timestamp,
int64_t capture_time_ms,
StorageType media_packet_storage,
bool protect);
RTPSenderInterface& _rtpSender;
// Should never be held when calling out of this class.
const rtc::scoped_ptr<CriticalSectionWrapper> crit_;
RtpVideoCodecTypes _videoType;
uint32_t _maxBitrate;
int32_t _retransmissionSettings GUARDED_BY(crit_);
// FEC
ForwardErrorCorrection fec_;
bool fec_enabled_ GUARDED_BY(crit_);
int8_t red_payload_type_ GUARDED_BY(crit_);
int8_t fec_payload_type_ GUARDED_BY(crit_);
FecProtectionParams delta_fec_params_ GUARDED_BY(crit_);
FecProtectionParams key_fec_params_ GUARDED_BY(crit_);
ProducerFec producer_fec_ GUARDED_BY(crit_);
// Bitrate used for FEC payload, RED headers, RTP headers for FEC packets
// and any padding overhead.
Bitrate _fecOverheadRate;
// Bitrate used for video payload and RTP headers
Bitrate _videoBitrate;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_H_
|