Moxie Marlinspike d83a3d71bc Support for Signal calls.
Merge in RedPhone

// FREEBIE
2015-09-30 14:30:09 -07:00

168 lines
5.8 KiB
C++

/*
* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
#include <string.h>
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
namespace webrtc {
namespace test {
Packet::Packet(uint8_t* packet_memory,
size_t allocated_bytes,
double time_ms,
const RtpHeaderParser& parser)
: payload_memory_(packet_memory),
payload_(NULL),
packet_length_bytes_(allocated_bytes),
payload_length_bytes_(0),
virtual_packet_length_bytes_(allocated_bytes),
virtual_payload_length_bytes_(0),
time_ms_(time_ms) {
valid_header_ = ParseHeader(parser);
}
Packet::Packet(uint8_t* packet_memory,
size_t allocated_bytes,
size_t virtual_packet_length_bytes,
double time_ms,
const RtpHeaderParser& parser)
: payload_memory_(packet_memory),
payload_(NULL),
packet_length_bytes_(allocated_bytes),
payload_length_bytes_(0),
virtual_packet_length_bytes_(virtual_packet_length_bytes),
virtual_payload_length_bytes_(0),
time_ms_(time_ms) {
valid_header_ = ParseHeader(parser);
}
Packet::Packet(uint8_t* packet_memory, size_t allocated_bytes, double time_ms)
: payload_memory_(packet_memory),
payload_(NULL),
packet_length_bytes_(allocated_bytes),
payload_length_bytes_(0),
virtual_packet_length_bytes_(allocated_bytes),
virtual_payload_length_bytes_(0),
time_ms_(time_ms) {
scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
valid_header_ = ParseHeader(*parser);
}
Packet::Packet(uint8_t* packet_memory,
size_t allocated_bytes,
size_t virtual_packet_length_bytes,
double time_ms)
: payload_memory_(packet_memory),
payload_(NULL),
packet_length_bytes_(allocated_bytes),
payload_length_bytes_(0),
virtual_packet_length_bytes_(virtual_packet_length_bytes),
virtual_payload_length_bytes_(0),
time_ms_(time_ms) {
scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
valid_header_ = ParseHeader(*parser);
}
bool Packet::ExtractRedHeaders(std::list<RTPHeader*>* headers) const {
//
// 0 1 2 3
// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// |1| block PT | timestamp offset | block length |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// |1| ... |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// |0| block PT |
// +-+-+-+-+-+-+-+-+
//
assert(payload_);
const uint8_t* payload_ptr = payload_;
const uint8_t* payload_end_ptr = payload_ptr + payload_length_bytes_;
// Find all RED headers with the extension bit set to 1. That is, all headers
// but the last one.
while ((payload_ptr < payload_end_ptr) && (*payload_ptr & 0x80)) {
RTPHeader* header = new RTPHeader;
CopyToHeader(header);
header->payloadType = payload_ptr[0] & 0x7F;
uint32_t offset = (payload_ptr[1] << 6) + ((payload_ptr[2] & 0xFC) >> 2);
header->timestamp -= offset;
headers->push_front(header);
payload_ptr += 4;
}
// Last header.
assert(payload_ptr < payload_end_ptr);
if (payload_ptr >= payload_end_ptr) {
return false; // Payload too short.
}
RTPHeader* header = new RTPHeader;
CopyToHeader(header);
header->payloadType = payload_ptr[0] & 0x7F;
headers->push_front(header);
return true;
}
void Packet::DeleteRedHeaders(std::list<RTPHeader*>* headers) {
while (!headers->empty()) {
delete headers->front();
headers->pop_front();
}
}
void Packet::ConvertHeader(WebRtcRTPHeader* copy_to) const {
memcpy(&copy_to->header, &header_, sizeof(header_));
copy_to->frameType = kAudioFrameSpeech;
copy_to->type.Audio.numEnergy = 0;
copy_to->type.Audio.channel = 1;
copy_to->type.Audio.isCNG = false;
}
bool Packet::ParseHeader(const RtpHeaderParser& parser) {
bool valid_header = parser.Parse(
payload_memory_.get(), static_cast<int>(packet_length_bytes_), &header_);
assert(valid_header);
if (!valid_header) {
return false;
}
assert(header_.headerLength <= packet_length_bytes_);
payload_ = &payload_memory_[header_.headerLength];
assert(packet_length_bytes_ >= header_.headerLength);
payload_length_bytes_ = packet_length_bytes_ - header_.headerLength;
assert(virtual_packet_length_bytes_ >= header_.headerLength);
virtual_payload_length_bytes_ =
virtual_packet_length_bytes_ - header_.headerLength;
return true;
}
void Packet::CopyToHeader(RTPHeader* destination) const {
destination->markerBit = header_.markerBit;
destination->payloadType = header_.payloadType;
destination->sequenceNumber = header_.sequenceNumber;
destination->timestamp = header_.timestamp;
destination->ssrc = header_.ssrc;
destination->numCSRCs = header_.numCSRCs;
destination->paddingLength = header_.paddingLength;
destination->headerLength = header_.headerLength;
destination->payload_type_frequency = header_.payload_type_frequency;
memcpy(&destination->arrOfCSRCs,
&header_.arrOfCSRCs,
sizeof(header_.arrOfCSRCs));
memcpy(
&destination->extension, &header_.extension, sizeof(header_.extension));
}
} // namespace test
} // namespace webrtc