1 /*
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
3 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
11 #include "modules/rtp_rtcp/source/rtp_format_video_generic.h"
12
13 #include <string.h>
14
15 #include "absl/types/optional.h"
16 #include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
17 #include "rtc_base/checks.h"
18 #include "rtc_base/logging.h"
19
20 namespace webrtc {
21
22 static const size_t kGenericHeaderLength = 1;
23 static const size_t kExtendedHeaderLength = 2;
24
RtpPacketizerGeneric(rtc::ArrayView<const uint8_t> payload,PayloadSizeLimits limits,const RTPVideoHeader & rtp_video_header)25 RtpPacketizerGeneric::RtpPacketizerGeneric(
26 rtc::ArrayView<const uint8_t> payload,
27 PayloadSizeLimits limits,
28 const RTPVideoHeader& rtp_video_header)
29 : remaining_payload_(payload) {
30 BuildHeader(rtp_video_header);
31
32 limits.max_payload_len -= header_size_;
33 payload_sizes_ = SplitAboutEqually(payload.size(), limits);
34 current_packet_ = payload_sizes_.begin();
35 }
36
RtpPacketizerGeneric(rtc::ArrayView<const uint8_t> payload,PayloadSizeLimits limits)37 RtpPacketizerGeneric::RtpPacketizerGeneric(
38 rtc::ArrayView<const uint8_t> payload,
39 PayloadSizeLimits limits)
40 : header_size_(0), remaining_payload_(payload) {
41 payload_sizes_ = SplitAboutEqually(payload.size(), limits);
42 current_packet_ = payload_sizes_.begin();
43 }
44
45 RtpPacketizerGeneric::~RtpPacketizerGeneric() = default;
46
NumPackets() const47 size_t RtpPacketizerGeneric::NumPackets() const {
48 return payload_sizes_.end() - current_packet_;
49 }
50
NextPacket(RtpPacketToSend * packet)51 bool RtpPacketizerGeneric::NextPacket(RtpPacketToSend* packet) {
52 RTC_DCHECK(packet);
53 if (current_packet_ == payload_sizes_.end())
54 return false;
55
56 size_t next_packet_payload_len = *current_packet_;
57
58 uint8_t* out_ptr =
59 packet->AllocatePayload(header_size_ + next_packet_payload_len);
60 RTC_CHECK(out_ptr);
61
62 if (header_size_ > 0) {
63 memcpy(out_ptr, header_, header_size_);
64 // Remove first-packet bit, following packets are intermediate.
65 header_[0] &= ~RtpFormatVideoGeneric::kFirstPacketBit;
66 }
67
68 memcpy(out_ptr + header_size_, remaining_payload_.data(),
69 next_packet_payload_len);
70
71 remaining_payload_ = remaining_payload_.subview(next_packet_payload_len);
72
73 ++current_packet_;
74
75 // Packets left to produce and data left to split should end at the same time.
76 RTC_DCHECK_EQ(current_packet_ == payload_sizes_.end(),
77 remaining_payload_.empty());
78
79 packet->SetMarker(remaining_payload_.empty());
80 return true;
81 }
82
BuildHeader(const RTPVideoHeader & rtp_video_header)83 void RtpPacketizerGeneric::BuildHeader(const RTPVideoHeader& rtp_video_header) {
84 header_size_ = kGenericHeaderLength;
85 header_[0] = RtpFormatVideoGeneric::kFirstPacketBit;
86 if (rtp_video_header.frame_type == VideoFrameType::kVideoFrameKey) {
87 header_[0] |= RtpFormatVideoGeneric::kKeyFrameBit;
88 }
89 if (const auto* generic_header = absl::get_if<RTPVideoHeaderLegacyGeneric>(
90 &rtp_video_header.video_type_header)) {
91 // Store bottom 15 bits of the picture id. Only 15 bits are used for
92 // compatibility with other packetizer implemenetations.
93 uint16_t picture_id = generic_header->picture_id;
94 header_[0] |= RtpFormatVideoGeneric::kExtendedHeaderBit;
95 header_[1] = (picture_id >> 8) & 0x7F;
96 header_[2] = picture_id & 0xFF;
97 header_size_ += kExtendedHeaderLength;
98 }
99 }
100 } // namespace webrtc
101