diff options
Diffstat (limited to 'media/cast/rtp_receiver/rtp_parser/rtp_parser.cc')
-rw-r--r-- | media/cast/rtp_receiver/rtp_parser/rtp_parser.cc | 107 |
1 files changed, 107 insertions, 0 deletions
diff --git a/media/cast/rtp_receiver/rtp_parser/rtp_parser.cc b/media/cast/rtp_receiver/rtp_parser/rtp_parser.cc new file mode 100644 index 0000000..0eb691b --- /dev/null +++ b/media/cast/rtp_receiver/rtp_parser/rtp_parser.cc @@ -0,0 +1,107 @@ +// Copyright 2013 The Chromium Authors. All rights reserved. +// Use of this source code is governed by a BSD-style license that can be +// found in the LICENSE file. + +#include "media/cast/rtp_receiver/rtp_parser/rtp_parser.h" + +#include "base/logging.h" +#include "media/cast/cast_defines.h" +#include "media/cast/rtp_receiver/rtp_receiver.h" +#include "net/base/big_endian.h" + +namespace media { +namespace cast { + +static const int kRtpCommonHeaderLength = 12; +static const int kRtpCastHeaderLength = 7; +static const uint8 kCastKeyFrameBitMask = 0x80; +static const uint8 kCastReferenceFrameIdBitMask = 0x40; + +RtpParser::RtpParser(RtpData* incoming_payload_callback, + const RtpParserConfig parser_config) + : data_callback_(incoming_payload_callback), + parser_config_(parser_config) { +} + +RtpParser::~RtpParser() { +} + +bool RtpParser::ParsePacket(const uint8* packet, int length, + RtpCastHeader* rtp_header) { + if (length == 0) return false; + // Get RTP general header. + 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); + } + // Not a valid payload type / ssrc combination. + return false; +} + +bool RtpParser::ParseCommon(const uint8* packet, + int 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; + net::BigEndianReader big_endian_reader(packet + 2, 80); + big_endian_reader.ReadU16(&sequence_number); + big_endian_reader.ReadU32(&rtp_timestamp); + big_endian_reader.ReadU32(&ssrc); + + 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, + int length, + RtpCastHeader* rtp_header) { + if (length < kRtpCastHeaderLength) return false; + // Extract header. + const uint8* data_ptr = packet; + int data_length = length; + rtp_header->is_key_frame = (data_ptr[0] & kCastKeyFrameBitMask); + rtp_header->is_reference = (data_ptr[0] & kCastReferenceFrameIdBitMask); + rtp_header->frame_id = data_ptr[1]; + + net::BigEndianReader big_endian_reader(data_ptr + 2, 32); + 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 = 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; + } + data_callback_->OnReceivedPayloadData(data_ptr, data_length, rtp_header); + return true; +} + +} // namespace cast +} // namespace media |