1 // Copyright 2013 The Chromium Authors. All rights reserved.
2 // Use of this source code is governed by a BSD-style license that can be
3 // found in the LICENSE file.
5 #include "media/cast/rtp_receiver/rtp_parser/rtp_parser.h"
7 #include "base/logging.h"
8 #include "media/cast/cast_defines.h"
9 #include "media/cast/rtp_receiver/rtp_receiver.h"
10 #include "net/base/big_endian.h"
15 static const size_t kRtpCommonHeaderLength = 12;
16 static const size_t kRtpCastHeaderLength = 7;
17 static const uint8 kCastKeyFrameBitMask = 0x80;
18 static const uint8 kCastReferenceFrameIdBitMask = 0x40;
20 RtpParser::RtpParser(RtpData* incoming_payload_callback,
21 const RtpParserConfig parser_config)
22 : data_callback_(incoming_payload_callback),
23 parser_config_(parser_config) {}
25 RtpParser::~RtpParser() {}
27 bool RtpParser::ParsePacket(const uint8* packet,
29 RtpCastHeader* rtp_header) {
32 // Get RTP general header.
33 if (!ParseCommon(packet, length, rtp_header))
35 if (rtp_header->webrtc.header.payloadType == parser_config_.payload_type &&
36 rtp_header->webrtc.header.ssrc == parser_config_.ssrc) {
37 return ParseCast(packet + kRtpCommonHeaderLength,
38 length - kRtpCommonHeaderLength,
41 // Not a valid payload type / ssrc combination.
45 bool RtpParser::ParseCommon(const uint8* packet,
47 RtpCastHeader* rtp_header) {
48 if (length < kRtpCommonHeaderLength)
50 uint8 version = packet[0] >> 6;
53 uint8 cc = packet[0] & 0x0f;
54 bool marker = ((packet[1] & 0x80) != 0);
55 int payload_type = packet[1] & 0x7f;
57 uint16 sequence_number;
58 uint32 rtp_timestamp, ssrc;
59 net::BigEndianReader big_endian_reader(packet + 2, 10);
60 big_endian_reader.ReadU16(&sequence_number);
61 big_endian_reader.ReadU32(&rtp_timestamp);
62 big_endian_reader.ReadU32(&ssrc);
64 if (ssrc != parser_config_.ssrc)
67 rtp_header->webrtc.header.markerBit = marker;
68 rtp_header->webrtc.header.payloadType = payload_type;
69 rtp_header->webrtc.header.sequenceNumber = sequence_number;
70 rtp_header->webrtc.header.timestamp = rtp_timestamp;
71 rtp_header->webrtc.header.ssrc = ssrc;
72 rtp_header->webrtc.header.numCSRCs = cc;
74 uint8 csrc_octs = cc * 4;
75 rtp_header->webrtc.type.Audio.numEnergy = rtp_header->webrtc.header.numCSRCs;
76 rtp_header->webrtc.header.headerLength = kRtpCommonHeaderLength + csrc_octs;
77 rtp_header->webrtc.type.Audio.isCNG = false;
78 rtp_header->webrtc.type.Audio.channel = parser_config_.audio_channels;
79 // TODO(pwestin): look at x bit and skip data.
83 bool RtpParser::ParseCast(const uint8* packet,
85 RtpCastHeader* rtp_header) {
86 if (length < kRtpCastHeaderLength)
90 const uint8* data_ptr = packet;
91 size_t data_length = length;
92 rtp_header->is_key_frame = (data_ptr[0] & kCastKeyFrameBitMask);
93 rtp_header->is_reference = (data_ptr[0] & kCastReferenceFrameIdBitMask);
94 rtp_header->frame_id = frame_id_wrap_helper_.MapTo32bitsFrameId(data_ptr[1]);
96 net::BigEndianReader big_endian_reader(data_ptr + 2, 4);
97 big_endian_reader.ReadU16(&rtp_header->packet_id);
98 big_endian_reader.ReadU16(&rtp_header->max_packet_id);
100 if (rtp_header->is_reference) {
101 rtp_header->reference_frame_id =
102 reference_frame_id_wrap_helper_.MapTo32bitsFrameId(data_ptr[6]);
103 data_ptr += kRtpCastHeaderLength;
104 data_length -= kRtpCastHeaderLength;
106 data_ptr += kRtpCastHeaderLength - 1;
107 data_length -= kRtpCastHeaderLength - 1;
110 if (rtp_header->max_packet_id < rtp_header->packet_id)
113 data_callback_->OnReceivedPayloadData(data_ptr, data_length, rtp_header);