This source file includes following definitions.
- ParsePacket
- ParseCommon
- ParseCast
#include "media/cast/rtp_receiver/rtp_parser/rtp_parser.h"
#include "base/big_endian.h"
#include "base/logging.h"
#include "media/cast/cast_defines.h"
#include "media/cast/rtp_receiver/rtp_receiver.h"
namespace media {
namespace cast {
static const size_t kRtpCommonHeaderLength = 12;
static const size_t kRtpCastHeaderLength = 7;
static const uint8 kCastKeyFrameBitMask = 0x80;
static const uint8 kCastReferenceFrameIdBitMask = 0x40;
RtpParser::RtpParser(const RtpParserConfig parser_config)
: parser_config_(parser_config) {}
RtpParser::~RtpParser() {}
bool RtpParser::ParsePacket(const uint8* packet,
size_t length,
RtpCastHeader* rtp_header) {
if (length == 0)
return false;
if (!ParseCommon(packet, length, rtp_header))
return false;
if (rtp_header->webrtc.header.payloadType == parser_config_.payload_type &&
rtp_header->webrtc.header.ssrc == parser_config_.ssrc) {
return ParseCast(packet + kRtpCommonHeaderLength,
length - kRtpCommonHeaderLength,
rtp_header);
}
return false;
}
bool RtpParser::ParseCommon(const uint8* packet,
size_t length,
RtpCastHeader* rtp_header) {
if (length < kRtpCommonHeaderLength)
return false;
uint8 version = packet[0] >> 6;
if (version != 2)
return false;
uint8 cc = packet[0] & 0x0f;
bool marker = ((packet[1] & 0x80) != 0);
int payload_type = packet[1] & 0x7f;
uint16 sequence_number;
uint32 rtp_timestamp, ssrc;
base::BigEndianReader big_endian_reader(
reinterpret_cast<const char*>(packet + 2), 10);
big_endian_reader.ReadU16(&sequence_number);
big_endian_reader.ReadU32(&rtp_timestamp);
big_endian_reader.ReadU32(&ssrc);
if (ssrc != parser_config_.ssrc)
return false;
rtp_header->webrtc.header.markerBit = marker;
rtp_header->webrtc.header.payloadType = payload_type;
rtp_header->webrtc.header.sequenceNumber = sequence_number;
rtp_header->webrtc.header.timestamp = rtp_timestamp;
rtp_header->webrtc.header.ssrc = ssrc;
rtp_header->webrtc.header.numCSRCs = cc;
uint8 csrc_octs = cc * 4;
rtp_header->webrtc.type.Audio.numEnergy = rtp_header->webrtc.header.numCSRCs;
rtp_header->webrtc.header.headerLength = kRtpCommonHeaderLength + csrc_octs;
rtp_header->webrtc.type.Audio.isCNG = false;
rtp_header->webrtc.type.Audio.channel = parser_config_.audio_channels;
return true;
}
bool RtpParser::ParseCast(const uint8* packet,
size_t length,
RtpCastHeader* rtp_header) {
if (length < kRtpCastHeaderLength)
return false;
const uint8* data_ptr = packet;
size_t data_length = length;
rtp_header->is_key_frame = !!(data_ptr[0] & kCastKeyFrameBitMask);
rtp_header->is_reference = !!(data_ptr[0] & kCastReferenceFrameIdBitMask);
rtp_header->frame_id = frame_id_wrap_helper_.MapTo32bitsFrameId(data_ptr[1]);
base::BigEndianReader big_endian_reader(
reinterpret_cast<const char*>(data_ptr + 2), 4);
big_endian_reader.ReadU16(&rtp_header->packet_id);
big_endian_reader.ReadU16(&rtp_header->max_packet_id);
if (rtp_header->is_reference) {
rtp_header->reference_frame_id =
reference_frame_id_wrap_helper_.MapTo32bitsFrameId(data_ptr[6]);
data_ptr += kRtpCastHeaderLength;
data_length -= kRtpCastHeaderLength;
} else {
data_ptr += kRtpCastHeaderLength - 1;
data_length -= kRtpCastHeaderLength - 1;
}
if (rtp_header->max_packet_id < rtp_header->packet_id)
return false;
OnReceivedPayloadData(data_ptr, data_length, *rtp_header);
return true;
}
}
}