
This is accomplished by breaking out RTX and FEC/RED functionality from the RTP module and keeping track of the base payload type, that is the payload type received when not receiving RTX. Enables retransmissions over RTX by default in the loopback test. BUG=1811 TESTS=voe/vie_auto_test --automated and trybots. R=mflodman@webrtc.org, pbos@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2154004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4692 4adac7df-926f-26a2-2b94-8c16560cd09d
250 lines
8.7 KiB
C++
250 lines
8.7 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/modules/rtp_rtcp/source/fec_receiver_impl.h"
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#include <assert.h>
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#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
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#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
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#include "webrtc/system_wrappers/interface/scoped_ptr.h"
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#include "webrtc/system_wrappers/interface/trace.h"
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// RFC 5109
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namespace webrtc {
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FecReceiver* FecReceiver::Create(int32_t id, RtpData* callback) {
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return new FecReceiverImpl(id, callback);
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}
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FecReceiverImpl::FecReceiverImpl(const int32_t id, RtpData* callback)
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: id_(id),
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crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
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recovered_packet_callback_(callback),
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fec_(new ForwardErrorCorrection(id)) {}
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FecReceiverImpl::~FecReceiverImpl() {
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while (!received_packet_list_.empty()) {
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delete received_packet_list_.front();
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received_packet_list_.pop_front();
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}
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if (fec_ != NULL) {
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fec_->ResetState(&recovered_packet_list_);
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delete fec_;
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}
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}
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// 0 1 2 3
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// 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
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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// |F| block PT | timestamp offset | block length |
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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//
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//
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// RFC 2198 RTP Payload for Redundant Audio Data September 1997
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//
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// The bits in the header are specified as follows:
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//
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// F: 1 bit First bit in header indicates whether another header block
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// follows. If 1 further header blocks follow, if 0 this is the
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// last header block.
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// If 0 there is only 1 byte RED header
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//
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// block PT: 7 bits RTP payload type for this block.
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//
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// timestamp offset: 14 bits Unsigned offset of timestamp of this block
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// relative to timestamp given in RTP header. The use of an unsigned
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// offset implies that redundant data must be sent after the primary
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// data, and is hence a time to be subtracted from the current
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// timestamp to determine the timestamp of the data for which this
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// block is the redundancy.
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//
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// block length: 10 bits Length in bytes of the corresponding data
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// block excluding header.
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int32_t FecReceiverImpl::AddReceivedRedPacket(
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const RTPHeader& header, const uint8_t* incoming_rtp_packet,
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int packet_length, uint8_t ulpfec_payload_type) {
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CriticalSectionScoped cs(crit_sect_.get());
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uint8_t REDHeaderLength = 1;
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uint16_t payload_data_length = packet_length - header.headerLength;
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// Add to list without RED header, aka a virtual RTP packet
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// we remove the RED header
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ForwardErrorCorrection::ReceivedPacket* received_packet =
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new ForwardErrorCorrection::ReceivedPacket;
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received_packet->pkt = new ForwardErrorCorrection::Packet;
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// get payload type from RED header
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uint8_t payload_type =
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incoming_rtp_packet[header.headerLength] & 0x7f;
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received_packet->is_fec = payload_type == ulpfec_payload_type;
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received_packet->seq_num = header.sequenceNumber;
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uint16_t blockLength = 0;
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if (incoming_rtp_packet[header.headerLength] & 0x80) {
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// f bit set in RED header
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REDHeaderLength = 4;
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uint16_t timestamp_offset =
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(incoming_rtp_packet[header.headerLength + 1]) << 8;
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timestamp_offset +=
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incoming_rtp_packet[header.headerLength + 2];
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timestamp_offset = timestamp_offset >> 2;
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if (timestamp_offset != 0) {
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// |timestampOffset| should be 0. However, it's possible this is the first
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// location a corrupt payload can be caught, so don't assert.
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WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, id_,
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"Corrupt payload found in %s", __FUNCTION__);
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delete received_packet;
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return -1;
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}
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blockLength =
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(0x03 & incoming_rtp_packet[header.headerLength + 2]) << 8;
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blockLength += (incoming_rtp_packet[header.headerLength + 3]);
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// check next RED header
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if (incoming_rtp_packet[header.headerLength + 4] & 0x80) {
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// more than 2 blocks in packet not supported
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delete received_packet;
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assert(false);
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return -1;
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}
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if (blockLength > payload_data_length - REDHeaderLength) {
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// block length longer than packet
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delete received_packet;
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assert(false);
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return -1;
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}
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}
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ForwardErrorCorrection::ReceivedPacket* second_received_packet = NULL;
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if (blockLength > 0) {
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// handle block length, split into 2 packets
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REDHeaderLength = 5;
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// copy the RTP header
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memcpy(received_packet->pkt->data, incoming_rtp_packet,
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header.headerLength);
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// replace the RED payload type
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received_packet->pkt->data[1] &= 0x80; // reset the payload
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received_packet->pkt->data[1] +=
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payload_type; // set the media payload type
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// copy the payload data
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memcpy(
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received_packet->pkt->data + header.headerLength,
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incoming_rtp_packet + header.headerLength + REDHeaderLength,
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blockLength);
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received_packet->pkt->length = blockLength;
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second_received_packet = new ForwardErrorCorrection::ReceivedPacket;
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second_received_packet->pkt = new ForwardErrorCorrection::Packet;
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second_received_packet->is_fec = true;
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second_received_packet->seq_num = header.sequenceNumber;
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// copy the FEC payload data
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memcpy(second_received_packet->pkt->data,
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incoming_rtp_packet + header.headerLength +
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REDHeaderLength + blockLength,
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payload_data_length - REDHeaderLength - blockLength);
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second_received_packet->pkt->length =
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payload_data_length - REDHeaderLength - blockLength;
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} else if (received_packet->is_fec) {
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// everything behind the RED header
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memcpy(
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received_packet->pkt->data,
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incoming_rtp_packet + header.headerLength + REDHeaderLength,
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payload_data_length - REDHeaderLength);
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received_packet->pkt->length = payload_data_length - REDHeaderLength;
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received_packet->ssrc =
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ModuleRTPUtility::BufferToUWord32(&incoming_rtp_packet[8]);
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} else {
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// copy the RTP header
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memcpy(received_packet->pkt->data, incoming_rtp_packet,
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header.headerLength);
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// replace the RED payload type
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received_packet->pkt->data[1] &= 0x80; // reset the payload
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received_packet->pkt->data[1] +=
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payload_type; // set the media payload type
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// copy the media payload data
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memcpy(
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received_packet->pkt->data + header.headerLength,
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incoming_rtp_packet + header.headerLength + REDHeaderLength,
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payload_data_length - REDHeaderLength);
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received_packet->pkt->length =
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header.headerLength + payload_data_length - REDHeaderLength;
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}
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if (received_packet->pkt->length == 0) {
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delete second_received_packet;
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delete received_packet;
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return 0;
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}
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received_packet_list_.push_back(received_packet);
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if (second_received_packet) {
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received_packet_list_.push_back(second_received_packet);
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}
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return 0;
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}
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int32_t FecReceiverImpl::ProcessReceivedFec() {
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crit_sect_->Enter();
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if (!received_packet_list_.empty()) {
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// Send received media packet to VCM.
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if (!received_packet_list_.front()->is_fec) {
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ForwardErrorCorrection::Packet* packet =
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received_packet_list_.front()->pkt;
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crit_sect_->Leave();
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if (!recovered_packet_callback_->OnRecoveredPacket(packet->data,
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packet->length)) {
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return -1;
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}
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crit_sect_->Enter();
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}
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if (fec_->DecodeFEC(&received_packet_list_, &recovered_packet_list_) != 0) {
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return -1;
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}
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assert(received_packet_list_.empty());
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}
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// Send any recovered media packets to VCM.
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ForwardErrorCorrection::RecoveredPacketList::iterator it =
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recovered_packet_list_.begin();
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for (; it != recovered_packet_list_.end(); ++it) {
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if ((*it)->returned) // Already sent to the VCM and the jitter buffer.
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continue;
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ForwardErrorCorrection::Packet* packet = (*it)->pkt;
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crit_sect_->Leave();
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if (!recovered_packet_callback_->OnRecoveredPacket(packet->data,
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packet->length)) {
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return -1;
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}
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crit_sect_->Enter();
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(*it)->returned = true;
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}
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crit_sect_->Leave();
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return 0;
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}
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} // namespace webrtc
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