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
|
/*
* Copyright (c) 2023 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_PACKETIZER_H265_H_
#define MODULES_RTP_RTCP_SOURCE_RTP_PACKETIZER_H265_H_
#include <cstddef>
#include <cstdint>
#include <deque>
#include <queue>
#include "api/array_view.h"
#include "modules/rtp_rtcp/source/rtp_format.h"
#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
namespace webrtc {
class RtpPacketizerH265 : public RtpPacketizer {
public:
// Initialize with payload from encoder.
// The payload_data must be exactly one encoded H.265 frame.
// For H265 we only support tx-mode SRST.
RtpPacketizerH265(ArrayView<const uint8_t> payload, PayloadSizeLimits limits);
RtpPacketizerH265(const RtpPacketizerH265&) = delete;
RtpPacketizerH265& operator=(const RtpPacketizerH265&) = delete;
~RtpPacketizerH265() override;
size_t NumPackets() const override;
// Get the next payload with H.265 payload header.
// Write payload and set marker bit of the `packet`.
// Returns true on success or false if there was no payload to packetize.
bool NextPacket(RtpPacketToSend* rtp_packet) override;
private:
struct PacketUnit {
ArrayView<const uint8_t> source_fragment;
bool first_fragment = false;
bool last_fragment = false;
bool aggregated = false;
uint16_t header = 0;
};
std::deque<ArrayView<const uint8_t>> input_fragments_;
std::queue<PacketUnit> packets_;
bool GeneratePackets();
bool PacketizeFu(size_t fragment_index);
int PacketizeAp(size_t fragment_index);
void NextAggregatePacket(RtpPacketToSend* rtp_packet);
void NextFragmentPacket(RtpPacketToSend* rtp_packet);
const PayloadSizeLimits limits_;
size_t num_packets_left_ = 0;
};
} // namespace webrtc
#endif // MODULES_RTP_RTCP_SOURCE_RTP_PACKETIZER_H265_H_
|
