| Index: media/cast/rtp_sender/rtp_packetizer/test/rtp_header_parser.cc
|
| diff --git a/media/cast/rtp_sender/rtp_packetizer/test/rtp_header_parser.cc b/media/cast/rtp_sender/rtp_packetizer/test/rtp_header_parser.cc
|
| deleted file mode 100644
|
| index 5c1c9fe6d0989bffdd5823f6ee023fb76f84ce4c..0000000000000000000000000000000000000000
|
| --- a/media/cast/rtp_sender/rtp_packetizer/test/rtp_header_parser.cc
|
| +++ /dev/null
|
| @@ -1,87 +0,0 @@
|
| -// 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_sender/rtp_packetizer/test/rtp_header_parser.h"
|
| -
|
| -#include <cstddef>
|
| -
|
| -#include "net/base/big_endian.h"
|
| -
|
| -namespace media {
|
| -namespace cast {
|
| -
|
| -static const uint8 kCastKeyFrameBitMask = 0x80;
|
| -static const uint8 kCastReferenceFrameIdBitMask = 0x40;
|
| -static const size_t kRtpCommonHeaderLength = 12;
|
| -static const size_t kRtpCastHeaderLength = 12;
|
| -
|
| -
|
| -RtpHeaderParser::RtpHeaderParser(const uint8* rtp_data,
|
| - size_t rtp_data_length)
|
| - : rtp_data_begin_(rtp_data),
|
| - length_(rtp_data_length) {}
|
| -
|
| -RtpHeaderParser::~RtpHeaderParser() {}
|
| -
|
| -bool RtpHeaderParser::Parse(RtpCastHeader* parsed_packet) const {
|
| - if (length_ < kRtpCommonHeaderLength + kRtpCastHeaderLength)
|
| - return false;
|
| - if (!ParseCommon(parsed_packet)) return false;
|
| - return ParseCast(parsed_packet);
|
| -}
|
| -
|
| -bool RtpHeaderParser::ParseCommon(RtpCastHeader* parsed_packet) const {
|
| - const uint8 version = rtp_data_begin_[0] >> 6;
|
| - if (version != 2) {
|
| - return false;
|
| - }
|
| -
|
| - const uint8 num_csrcs = rtp_data_begin_[0] & 0x0f;
|
| - const bool marker = ((rtp_data_begin_[1] & 0x80) == 0) ? false : true;
|
| - const uint8 payload_type = rtp_data_begin_[1] & 0x7f;
|
| - const uint16 sequence_number = (rtp_data_begin_[2] << 8) +
|
| - rtp_data_begin_[3];
|
| -
|
| - const uint8* ptr = &rtp_data_begin_[4];
|
| -
|
| - net::BigEndianReader big_endian_reader(ptr, 8);
|
| - uint32 rtp_timestamp, ssrc;
|
| - big_endian_reader.ReadU32(&rtp_timestamp);
|
| - big_endian_reader.ReadU32(&ssrc);
|
| -
|
| - const uint8 csrc_octs = num_csrcs * 4;
|
| -
|
| - parsed_packet->webrtc.header.markerBit = marker;
|
| - parsed_packet->webrtc.header.payloadType = payload_type;
|
| - parsed_packet->webrtc.header.sequenceNumber = sequence_number;
|
| - parsed_packet->webrtc.header.timestamp = rtp_timestamp;
|
| - parsed_packet->webrtc.header.ssrc = ssrc;
|
| - parsed_packet->webrtc.header.numCSRCs = num_csrcs;
|
| -
|
| - parsed_packet->webrtc.type.Audio.numEnergy =
|
| - parsed_packet->webrtc.header.numCSRCs;
|
| -
|
| - parsed_packet->webrtc.header.headerLength = 12 + csrc_octs;
|
| - return true;
|
| -}
|
| -
|
| -bool RtpHeaderParser::ParseCast(RtpCastHeader* parsed_packet) const {
|
| - const uint8* data = rtp_data_begin_ + kRtpCommonHeaderLength;
|
| - parsed_packet->is_key_frame = (data[0] & kCastKeyFrameBitMask);
|
| - parsed_packet->is_reference = (data[0] & kCastReferenceFrameIdBitMask);
|
| - parsed_packet->frame_id = frame_id_wrap_helper_.MapTo32bitsFrameId(data[1]);
|
| -
|
| - net::BigEndianReader big_endian_reader(data + 2, 8);
|
| - big_endian_reader.ReadU16(&parsed_packet->packet_id);
|
| - big_endian_reader.ReadU16(&parsed_packet->max_packet_id);
|
| -
|
| - if (parsed_packet->is_reference) {
|
| - parsed_packet->reference_frame_id =
|
| - reference_frame_id_wrap_helper_.MapTo32bitsFrameId(data[6]);
|
| - }
|
| - return true;
|
| -}
|
| -
|
| -} // namespace cast
|
| -} // namespace media
|
|
|