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| 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 <string> |
| 12 |
| 13 #include "modules/include/module_common_types.h" |
| 14 #include "modules/rtp_rtcp/source/rtp_format_video_stereo.h" |
| 15 #include "modules/rtp_rtcp/source/rtp_packet_to_send.h" |
| 16 #include "rtc_base/logging.h" |
| 17 |
| 18 namespace webrtc { |
| 19 |
| 20 static const size_t kStereoHeaderMarkerLength = 1; |
| 21 static const size_t kStereoHeaderLength = sizeof(RTPVideoStereoInfo); |
| 22 |
| 23 RtpPacketizerStereo::RtpPacketizerStereo( |
| 24 size_t max_payload_len, |
| 25 size_t last_packet_reduction_len, |
| 26 const RTPVideoTypeHeader* rtp_type_header, |
| 27 const RTPVideoStereoInfo* stereoInfo) |
| 28 : max_payload_len_(max_payload_len - kStereoHeaderMarkerLength - |
| 29 kStereoHeaderLength), |
| 30 last_packet_reduction_len_(last_packet_reduction_len), |
| 31 packetizer_(RtpPacketizer::Create(stereoInfo->stereoCodecType, |
| 32 max_payload_len_, |
| 33 last_packet_reduction_len_, |
| 34 rtp_type_header, |
| 35 stereoInfo, |
| 36 kVideoFrameDelta)), |
| 37 stereoInfo_(stereoInfo) {} |
| 38 |
| 39 RtpPacketizerStereo::~RtpPacketizerStereo() {} |
| 40 |
| 41 size_t RtpPacketizerStereo::SetPayloadData( |
| 42 const uint8_t* payload_data, |
| 43 size_t payload_size, |
| 44 const RTPFragmentationHeader* fragmentation) { |
| 45 header_marker_ = RtpFormatVideoStereo::kFirstPacketBit; |
| 46 return packetizer_->SetPayloadData(payload_data, payload_size, fragmentation); |
| 47 } |
| 48 |
| 49 bool RtpPacketizerStereo::NextPacket(RtpPacketToSend* packet) { |
| 50 RTC_DCHECK(packet); |
| 51 const bool rv = packetizer_->NextPacket(packet); |
| 52 RTC_CHECK(rv); |
| 53 |
| 54 const bool first_packet = |
| 55 header_marker_ == RtpFormatVideoStereo::kFirstPacketBit; |
| 56 size_t header_length = first_packet |
| 57 ? kStereoHeaderMarkerLength + kStereoHeaderLength |
| 58 : kStereoHeaderMarkerLength; |
| 59 |
| 60 std::unique_ptr<RtpPacketToSend> copied_packet(new RtpPacketToSend(*packet)); |
| 61 uint8_t* wrapped_payload = |
| 62 packet->AllocatePayload(header_length + packet->payload_size()); |
| 63 RTC_DCHECK(wrapped_payload); |
| 64 wrapped_payload[0] = header_marker_; |
| 65 header_marker_ &= ~RtpFormatVideoStereo::kFirstPacketBit; |
| 66 if (first_packet) { |
| 67 memcpy(&wrapped_payload[kStereoHeaderMarkerLength], stereoInfo_, |
| 68 kStereoHeaderLength); |
| 69 } |
| 70 auto payload = copied_packet->payload(); |
| 71 memcpy(&wrapped_payload[header_length], payload.data(), payload.size()); |
| 72 return rv; |
| 73 } |
| 74 |
| 75 ProtectionType RtpPacketizerStereo::GetProtectionType() { |
| 76 return kProtectedPacket; |
| 77 } |
| 78 |
| 79 StorageType RtpPacketizerStereo::GetStorageType( |
| 80 uint32_t retransmission_settings) { |
| 81 return kDontRetransmit; |
| 82 } |
| 83 |
| 84 std::string RtpPacketizerStereo::ToString() { |
| 85 return "RtpPacketizerStereo"; |
| 86 } |
| 87 |
| 88 bool RtpDepacketizerStereo::Parse(ParsedPayload* parsed_payload, |
| 89 const uint8_t* payload_data, |
| 90 size_t payload_data_length) { |
| 91 assert(parsed_payload != NULL); |
| 92 if (payload_data_length == 0) { |
| 93 LOG(LS_ERROR) << "Empty payload."; |
| 94 return false; |
| 95 } |
| 96 |
| 97 uint8_t marker_header = *payload_data++; |
| 98 --payload_data_length; |
| 99 const bool first_packet = |
| 100 (marker_header & RtpFormatVideoStereo::kFirstPacketBit) != 0; |
| 101 |
| 102 if (first_packet) { |
| 103 memcpy(&parsed_payload->type.Video.stereoInfo, payload_data, |
| 104 kStereoHeaderLength); |
| 105 payload_data += kStereoHeaderLength; |
| 106 payload_data_length -= kStereoHeaderLength; |
| 107 } |
| 108 const bool rv = |
| 109 depacketizer_.Parse(parsed_payload, payload_data, payload_data_length); |
| 110 RTC_DCHECK(rv); |
| 111 RTC_DCHECK_EQ(parsed_payload->type.Video.is_first_packet_in_frame, |
| 112 first_packet); |
| 113 parsed_payload->type.Video.codec = kRtpVideoStereo; |
| 114 return rv; |
| 115 } |
| 116 } // namespace webrtc |
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