ef90c3227e
We used to flag the _second_ packet in the first frame as the first. Subsequent frames worked as intended. BUG=1103 TEST=vie_auto_test --automated, rtp_rtcp_unittests Review URL: https://webrtc-codereview.appspot.com/964020 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3164 4adac7df-926f-26a2-2b94-8c16560cd09d
340 lines
11 KiB
C++
340 lines
11 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 "rtp_receiver_video.h"
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#include <cassert> //assert
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#include <cstring> // memcpy()
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#include <math.h>
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#include "critical_section_wrapper.h"
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#include "receiver_fec.h"
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#include "rtp_rtcp_impl.h"
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#include "rtp_utility.h"
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#include "trace.h"
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namespace webrtc {
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WebRtc_UWord32 BitRateBPS(WebRtc_UWord16 x )
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{
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return (x & 0x3fff) * WebRtc_UWord32(pow(10.0f,(2 + (x >> 14))));
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}
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RTPReceiverVideo::RTPReceiverVideo(const WebRtc_Word32 id,
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ModuleRtpRtcpImpl* owner)
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: _id(id),
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_criticalSectionReceiverVideo(
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CriticalSectionWrapper::CreateCriticalSection()),
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_currentFecFrameDecoded(false),
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_receiveFEC(NULL) {
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}
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RTPReceiverVideo::~RTPReceiverVideo() {
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delete _criticalSectionReceiverVideo;
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delete _receiveFEC;
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}
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ModuleRTPUtility::Payload* RTPReceiverVideo::RegisterReceiveVideoPayload(
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const char payloadName[RTP_PAYLOAD_NAME_SIZE],
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const WebRtc_Word8 payloadType,
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const WebRtc_UWord32 maxRate) {
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RtpVideoCodecTypes videoType = kRtpNoVideo;
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if (ModuleRTPUtility::StringCompare(payloadName, "VP8", 3)) {
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videoType = kRtpVp8Video;
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} else if (ModuleRTPUtility::StringCompare(payloadName, "I420", 4)) {
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videoType = kRtpNoVideo;
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} else if (ModuleRTPUtility::StringCompare(payloadName, "ULPFEC", 6)) {
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// store this
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if (_receiveFEC == NULL) {
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_receiveFEC = new ReceiverFEC(_id, this);
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}
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_receiveFEC->SetPayloadTypeFEC(payloadType);
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videoType = kRtpFecVideo;
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} else {
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return NULL;
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}
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ModuleRTPUtility::Payload* payload = new ModuleRTPUtility::Payload;
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payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
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strncpy(payload->name, payloadName, RTP_PAYLOAD_NAME_SIZE - 1);
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payload->typeSpecific.Video.videoCodecType = videoType;
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payload->typeSpecific.Video.maxRate = maxRate;
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payload->audio = false;
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return payload;
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}
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// we have no critext when calling this
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// we are not allowed to have any critsects when calling
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// CallbackOfReceivedPayloadData
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WebRtc_Word32 RTPReceiverVideo::ParseVideoCodecSpecific(
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WebRtcRTPHeader* rtpHeader,
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const WebRtc_UWord8* payloadData,
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const WebRtc_UWord16 payloadDataLength,
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const RtpVideoCodecTypes videoType,
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const bool isRED,
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const WebRtc_UWord8* incomingRtpPacket,
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const WebRtc_UWord16 incomingRtpPacketSize,
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const WebRtc_Word64 nowMS) {
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WebRtc_Word32 retVal = 0;
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_criticalSectionReceiverVideo->Enter();
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if (isRED) {
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if(_receiveFEC == NULL) {
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_criticalSectionReceiverVideo->Leave();
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return -1;
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}
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bool FECpacket = false;
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retVal = _receiveFEC->AddReceivedFECPacket(
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rtpHeader,
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incomingRtpPacket,
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payloadDataLength,
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FECpacket);
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if (retVal != -1) {
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retVal = _receiveFEC->ProcessReceivedFEC();
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}
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_criticalSectionReceiverVideo->Leave();
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if(retVal == 0 && FECpacket) {
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// Callback with the received FEC packet.
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// The normal packets are delivered after parsing.
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// This contains the original RTP packet header but with
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// empty payload and data length.
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rtpHeader->frameType = kFrameEmpty;
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// We need this for the routing.
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WebRtc_Word32 retVal = SetCodecType(videoType, rtpHeader);
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if(retVal != 0) {
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return retVal;
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}
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// Pass the length of FEC packets so that they can be accounted for in
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// the bandwidth estimator.
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retVal = CallbackOfReceivedPayloadData(NULL, payloadDataLength,
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rtpHeader);
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}
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} else {
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// will leave the _criticalSectionReceiverVideo critsect
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retVal = ParseVideoCodecSpecificSwitch(rtpHeader,
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payloadData,
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payloadDataLength,
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videoType);
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}
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return retVal;
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}
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WebRtc_Word32 RTPReceiverVideo::BuildRTPheader(
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const WebRtcRTPHeader* rtpHeader,
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WebRtc_UWord8* dataBuffer) const {
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dataBuffer[0] = static_cast<WebRtc_UWord8>(0x80); // version 2
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dataBuffer[1] = static_cast<WebRtc_UWord8>(rtpHeader->header.payloadType);
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if (rtpHeader->header.markerBit) {
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dataBuffer[1] |= kRtpMarkerBitMask; // MarkerBit is 1
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}
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ModuleRTPUtility::AssignUWord16ToBuffer(dataBuffer + 2,
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rtpHeader->header.sequenceNumber);
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ModuleRTPUtility::AssignUWord32ToBuffer(dataBuffer + 4,
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rtpHeader->header.timestamp);
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ModuleRTPUtility::AssignUWord32ToBuffer(dataBuffer + 8,
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rtpHeader->header.ssrc);
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WebRtc_Word32 rtpHeaderLength = 12;
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// Add the CSRCs if any
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if (rtpHeader->header.numCSRCs > 0) {
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if (rtpHeader->header.numCSRCs > 16) {
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// error
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assert(false);
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}
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WebRtc_UWord8* ptr = &dataBuffer[rtpHeaderLength];
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for (WebRtc_UWord32 i = 0; i < rtpHeader->header.numCSRCs; ++i) {
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ModuleRTPUtility::AssignUWord32ToBuffer(ptr,
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rtpHeader->header.arrOfCSRCs[i]);
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ptr +=4;
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}
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dataBuffer[0] = (dataBuffer[0]&0xf0) | rtpHeader->header.numCSRCs;
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// Update length of header
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rtpHeaderLength += sizeof(WebRtc_UWord32)*rtpHeader->header.numCSRCs;
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}
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return rtpHeaderLength;
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}
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WebRtc_Word32 RTPReceiverVideo::ReceiveRecoveredPacketCallback(
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WebRtcRTPHeader* rtpHeader,
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const WebRtc_UWord8* payloadData,
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const WebRtc_UWord16 payloadDataLength) {
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// TODO(pwestin) Re-factor this to avoid the messy critsect handling.
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_criticalSectionReceiverVideo->Enter();
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_currentFecFrameDecoded = true;
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ModuleRTPUtility::Payload* payload = NULL;
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if (PayloadTypeToPayload(rtpHeader->header.payloadType, payload) != 0) {
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_criticalSectionReceiverVideo->Leave();
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return -1;
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}
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// here we can re-create the original lost packet so that we can use it for
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// the relay we need to re-create the RED header too
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WebRtc_UWord8 recoveredPacket[IP_PACKET_SIZE];
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WebRtc_UWord16 rtpHeaderLength = (WebRtc_UWord16)BuildRTPheader(
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rtpHeader, recoveredPacket);
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const WebRtc_UWord8 REDForFECHeaderLength = 1;
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// replace pltype
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recoveredPacket[1] &= 0x80; // reset
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recoveredPacket[1] += REDPayloadType(); // replace with RED payload type
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// add RED header
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recoveredPacket[rtpHeaderLength] = rtpHeader->header.payloadType;
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// f-bit always 0
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memcpy(recoveredPacket + rtpHeaderLength + REDForFECHeaderLength, payloadData,
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payloadDataLength);
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return ParseVideoCodecSpecificSwitch(
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rtpHeader,
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payloadData,
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payloadDataLength,
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payload->typeSpecific.Video.videoCodecType);
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}
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WebRtc_Word32 RTPReceiverVideo::SetCodecType(const RtpVideoCodecTypes videoType,
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WebRtcRTPHeader* rtpHeader) const {
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switch (videoType) {
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case kRtpNoVideo:
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rtpHeader->type.Video.codec = kRTPVideoGeneric;
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break;
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case kRtpVp8Video:
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rtpHeader->type.Video.codec = kRTPVideoVP8;
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break;
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case kRtpFecVideo:
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rtpHeader->type.Video.codec = kRTPVideoFEC;
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break;
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}
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return 0;
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}
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WebRtc_Word32 RTPReceiverVideo::ParseVideoCodecSpecificSwitch(
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WebRtcRTPHeader* rtpHeader,
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const WebRtc_UWord8* payloadData,
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const WebRtc_UWord16 payloadDataLength,
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const RtpVideoCodecTypes videoType) {
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WebRtc_Word32 retVal = SetCodecType(videoType, rtpHeader);
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if (retVal != 0) {
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_criticalSectionReceiverVideo->Leave();
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return retVal;
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}
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WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, _id, "%s(timestamp:%u)",
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__FUNCTION__, rtpHeader->header.timestamp);
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// All receive functions release _criticalSectionReceiverVideo before
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// returning.
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switch (videoType) {
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case kRtpNoVideo:
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return ReceiveGenericCodec(rtpHeader, payloadData, payloadDataLength);
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case kRtpVp8Video:
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return ReceiveVp8Codec(rtpHeader, payloadData, payloadDataLength);
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case kRtpFecVideo:
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break;
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}
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_criticalSectionReceiverVideo->Leave();
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return -1;
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}
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WebRtc_Word32 RTPReceiverVideo::ReceiveVp8Codec(
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WebRtcRTPHeader* rtpHeader,
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const WebRtc_UWord8* payloadData,
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const WebRtc_UWord16 payloadDataLength) {
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bool success;
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ModuleRTPUtility::RTPPayload parsedPacket;
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if (payloadDataLength == 0) {
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success = true;
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parsedPacket.info.VP8.dataLength = 0;
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} else {
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ModuleRTPUtility::RTPPayloadParser rtpPayloadParser(kRtpVp8Video,
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payloadData,
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payloadDataLength,
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_id);
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success = rtpPayloadParser.Parse(parsedPacket);
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}
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// from here down we only work on local data
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_criticalSectionReceiverVideo->Leave();
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if (!success) {
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return -1;
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}
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if (parsedPacket.info.VP8.dataLength == 0) {
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// we have an "empty" VP8 packet, it's ok, could be one way video
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// Inform the jitter buffer about this packet.
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rtpHeader->frameType = kFrameEmpty;
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if (CallbackOfReceivedPayloadData(NULL, 0, rtpHeader) != 0) {
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return -1;
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}
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return 0;
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}
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rtpHeader->frameType = (parsedPacket.frameType == ModuleRTPUtility::kIFrame) ?
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kVideoFrameKey : kVideoFrameDelta;
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RTPVideoHeaderVP8 *toHeader = &rtpHeader->type.Video.codecHeader.VP8;
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ModuleRTPUtility::RTPPayloadVP8 *fromHeader = &parsedPacket.info.VP8;
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rtpHeader->type.Video.isFirstPacket = fromHeader->beginningOfPartition
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&& (fromHeader->partitionID == 0);
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toHeader->nonReference = fromHeader->nonReferenceFrame;
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toHeader->pictureId = fromHeader->hasPictureID ? fromHeader->pictureID :
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kNoPictureId;
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toHeader->tl0PicIdx = fromHeader->hasTl0PicIdx ? fromHeader->tl0PicIdx :
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kNoTl0PicIdx;
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if (fromHeader->hasTID) {
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toHeader->temporalIdx = fromHeader->tID;
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toHeader->layerSync = fromHeader->layerSync;
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} else {
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toHeader->temporalIdx = kNoTemporalIdx;
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toHeader->layerSync = false;
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}
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toHeader->keyIdx = fromHeader->hasKeyIdx ? fromHeader->keyIdx : kNoKeyIdx;
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toHeader->frameWidth = fromHeader->frameWidth;
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toHeader->frameHeight = fromHeader->frameHeight;
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toHeader->partitionId = fromHeader->partitionID;
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toHeader->beginningOfPartition = fromHeader->beginningOfPartition;
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if(CallbackOfReceivedPayloadData(parsedPacket.info.VP8.data,
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parsedPacket.info.VP8.dataLength,
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rtpHeader) != 0) {
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return -1;
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}
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return 0;
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}
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WebRtc_Word32 RTPReceiverVideo::ReceiveGenericCodec(
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WebRtcRTPHeader* rtpHeader,
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const WebRtc_UWord8* payloadData,
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const WebRtc_UWord16 payloadDataLength) {
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rtpHeader->frameType = kVideoFrameKey;
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bool isFirstPacketInFrame =
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(SequenceNumber() + 1) == rtpHeader->header.sequenceNumber &&
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TimeStamp() != rtpHeader->header.timestamp;
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if (isFirstPacketInFrame || HaveNotReceivedPackets()) {
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rtpHeader->type.Video.isFirstPacket = true;
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}
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_criticalSectionReceiverVideo->Leave();
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if (CallbackOfReceivedPayloadData(payloadData, payloadDataLength,
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rtpHeader) != 0) {
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return -1;
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}
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return 0;
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}
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} // namespace webrtc
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