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/*
* Copyright (c) 2012 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/rtp_rtcp/source/fec_receiver_impl.h"
#include <assert.h>
#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/system_wrappers/interface/logging.h"
// RFC 5109
namespace webrtc {
FecReceiver* FecReceiver::Create(RtpData* callback) {
return new FecReceiverImpl(callback);
}
FecReceiverImpl::FecReceiverImpl(RtpData* callback)
: crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
recovered_packet_callback_(callback),
fec_(new ForwardErrorCorrection()) {}
FecReceiverImpl::~FecReceiverImpl() {
while (!received_packet_list_.empty()) {
delete received_packet_list_.front();
received_packet_list_.pop_front();
}
if (fec_ != NULL) {
fec_->ResetState(&recovered_packet_list_);
delete fec_;
}
}
// 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
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// |F| block PT | timestamp offset | block length |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
//
//
// RFC 2198 RTP Payload for Redundant Audio Data September 1997
//
// The bits in the header are specified as follows:
//
// F: 1 bit First bit in header indicates whether another header block
// follows. If 1 further header blocks follow, if 0 this is the
// last header block.
// If 0 there is only 1 byte RED header
//
// block PT: 7 bits RTP payload type for this block.
//
// timestamp offset: 14 bits Unsigned offset of timestamp of this block
// relative to timestamp given in RTP header. The use of an unsigned
// offset implies that redundant data must be sent after the primary
// data, and is hence a time to be subtracted from the current
// timestamp to determine the timestamp of the data for which this
// block is the redundancy.
//
// block length: 10 bits Length in bytes of the corresponding data
// block excluding header.
int32_t FecReceiverImpl::AddReceivedRedPacket(
const RTPHeader& header, const uint8_t* incoming_rtp_packet,
size_t packet_length, uint8_t ulpfec_payload_type) {
CriticalSectionScoped cs(crit_sect_.get());
uint8_t REDHeaderLength = 1;
size_t payload_data_length = packet_length - header.headerLength;
// Add to list without RED header, aka a virtual RTP packet
// we remove the RED header
ForwardErrorCorrection::ReceivedPacket* received_packet =
new ForwardErrorCorrection::ReceivedPacket;
received_packet->pkt = new ForwardErrorCorrection::Packet;
// get payload type from RED header
uint8_t payload_type =
incoming_rtp_packet[header.headerLength] & 0x7f;
received_packet->is_fec = payload_type == ulpfec_payload_type;
received_packet->seq_num = header.sequenceNumber;
uint16_t blockLength = 0;
if (incoming_rtp_packet[header.headerLength] & 0x80) {
// f bit set in RED header
REDHeaderLength = 4;
uint16_t timestamp_offset =
(incoming_rtp_packet[header.headerLength + 1]) << 8;
timestamp_offset +=
incoming_rtp_packet[header.headerLength + 2];
timestamp_offset = timestamp_offset >> 2;
if (timestamp_offset != 0) {
// |timestampOffset| should be 0. However, it's possible this is the first
// location a corrupt payload can be caught, so don't assert.
LOG(LS_WARNING) << "Corrupt payload found.";
delete received_packet;
return -1;
}
blockLength =
(0x03 & incoming_rtp_packet[header.headerLength + 2]) << 8;
blockLength += (incoming_rtp_packet[header.headerLength + 3]);
// check next RED header
if (incoming_rtp_packet[header.headerLength + 4] & 0x80) {
// more than 2 blocks in packet not supported
delete received_packet;
assert(false);
return -1;
}
if (blockLength > payload_data_length - REDHeaderLength) {
// block length longer than packet
delete received_packet;
assert(false);
return -1;
}
}
ForwardErrorCorrection::ReceivedPacket* second_received_packet = NULL;
if (blockLength > 0) {
// handle block length, split into 2 packets
REDHeaderLength = 5;
// copy the RTP header
memcpy(received_packet->pkt->data, incoming_rtp_packet,
header.headerLength);
// replace the RED payload type
received_packet->pkt->data[1] &= 0x80; // reset the payload
received_packet->pkt->data[1] +=
payload_type; // set the media payload type
// copy the payload data
memcpy(
received_packet->pkt->data + header.headerLength,
incoming_rtp_packet + header.headerLength + REDHeaderLength,
blockLength);
received_packet->pkt->length = blockLength;
second_received_packet = new ForwardErrorCorrection::ReceivedPacket;
second_received_packet->pkt = new ForwardErrorCorrection::Packet;
second_received_packet->is_fec = true;
second_received_packet->seq_num = header.sequenceNumber;
// copy the FEC payload data
memcpy(second_received_packet->pkt->data,
incoming_rtp_packet + header.headerLength +
REDHeaderLength + blockLength,
payload_data_length - REDHeaderLength - blockLength);
second_received_packet->pkt->length =
payload_data_length - REDHeaderLength - blockLength;
} else if (received_packet->is_fec) {
// everything behind the RED header
memcpy(
received_packet->pkt->data,
incoming_rtp_packet + header.headerLength + REDHeaderLength,
payload_data_length - REDHeaderLength);
received_packet->pkt->length = payload_data_length - REDHeaderLength;
received_packet->ssrc =
RtpUtility::BufferToUWord32(&incoming_rtp_packet[8]);
} else {
// copy the RTP header
memcpy(received_packet->pkt->data, incoming_rtp_packet,
header.headerLength);
// replace the RED payload type
received_packet->pkt->data[1] &= 0x80; // reset the payload
received_packet->pkt->data[1] +=
payload_type; // set the media payload type
// copy the media payload data
memcpy(
received_packet->pkt->data + header.headerLength,
incoming_rtp_packet + header.headerLength + REDHeaderLength,
payload_data_length - REDHeaderLength);
received_packet->pkt->length =
header.headerLength + payload_data_length - REDHeaderLength;
}
if (received_packet->pkt->length == 0) {
delete second_received_packet;
delete received_packet;
return 0;
}
received_packet_list_.push_back(received_packet);
if (second_received_packet) {
received_packet_list_.push_back(second_received_packet);
}
return 0;
}
int32_t FecReceiverImpl::ProcessReceivedFec() {
crit_sect_->Enter();
if (!received_packet_list_.empty()) {
// Send received media packet to VCM.
if (!received_packet_list_.front()->is_fec) {
ForwardErrorCorrection::Packet* packet =
received_packet_list_.front()->pkt;
crit_sect_->Leave();
if (!recovered_packet_callback_->OnRecoveredPacket(packet->data,
packet->length)) {
return -1;
}
crit_sect_->Enter();
}
if (fec_->DecodeFEC(&received_packet_list_, &recovered_packet_list_) != 0) {
crit_sect_->Leave();
return -1;
}
assert(received_packet_list_.empty());
}
// Send any recovered media packets to VCM.
ForwardErrorCorrection::RecoveredPacketList::iterator it =
recovered_packet_list_.begin();
for (; it != recovered_packet_list_.end(); ++it) {
if ((*it)->returned) // Already sent to the VCM and the jitter buffer.
continue;
ForwardErrorCorrection::Packet* packet = (*it)->pkt;
crit_sect_->Leave();
if (!recovered_packet_callback_->OnRecoveredPacket(packet->data,
packet->length)) {
return -1;
}
crit_sect_->Enter();
(*it)->returned = true;
}
crit_sect_->Leave();
return 0;
}
} // namespace webrtc
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