3c45dfd178
Review URL: http://webrtc-codereview.appspot.com/47005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@122 4adac7df-926f-26a2-2b94-8c16560cd09d
681 lines
23 KiB
C++
681 lines
23 KiB
C++
/*
|
|
* Copyright (c) 2011 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 <cassert> //assert
|
|
#include <cstring> // memcpy()
|
|
#include <math.h>
|
|
|
|
#include "rtp_receiver_video.h"
|
|
|
|
#include "trace.h"
|
|
#include "critical_section_wrapper.h"
|
|
#include "tick_util.h"
|
|
|
|
#include "receiver_fec.h"
|
|
|
|
namespace webrtc {
|
|
WebRtc_UWord32 BitRateBPS(WebRtc_UWord16 x )
|
|
{
|
|
return (x & 0x3fff) * WebRtc_UWord32(pow(10.0f,(2 + (x >> 14))));
|
|
}
|
|
|
|
RTPReceiverVideo::RTPReceiverVideo(const WebRtc_Word32 id,
|
|
ModuleRtpRtcpPrivate& callback):
|
|
_id(id),
|
|
_criticalSectionFeedback(*CriticalSectionWrapper::CreateCriticalSection()),
|
|
_cbVideoFeedback(NULL),
|
|
_cbPrivateFeedback(callback),
|
|
_criticalSectionReceiverVideo(*CriticalSectionWrapper::CreateCriticalSection()),
|
|
|
|
_completeFrame(false),
|
|
_receiveFEC(NULL),
|
|
_packetStartTimeMs(0),
|
|
_receivedBW(),
|
|
_estimatedBW(0),
|
|
_currentFecFrameDecoded(false),
|
|
_h263InverseLogic(false),
|
|
_overUseDetector(),
|
|
_videoBitRate(),
|
|
_lastBitRateChange(0),
|
|
_packetOverHead(28)
|
|
{
|
|
memset(_receivedBW, 0,sizeof(_receivedBW));
|
|
}
|
|
|
|
RTPReceiverVideo::~RTPReceiverVideo()
|
|
{
|
|
delete &_criticalSectionFeedback;
|
|
delete &_criticalSectionReceiverVideo;
|
|
delete _receiveFEC;
|
|
}
|
|
|
|
WebRtc_Word32
|
|
RTPReceiverVideo::Init()
|
|
{
|
|
_completeFrame = false;
|
|
_packetStartTimeMs = 0;
|
|
_estimatedBW = 0;
|
|
_currentFecFrameDecoded = false;
|
|
_packetOverHead = 28;
|
|
for (int i = 0; i < BW_HISTORY_SIZE; i++)
|
|
{
|
|
_receivedBW[i] = 0;
|
|
}
|
|
ResetOverUseDetector();
|
|
return 0;
|
|
}
|
|
|
|
void
|
|
RTPReceiverVideo::ChangeUniqueId(const WebRtc_Word32 id)
|
|
{
|
|
_id = id;
|
|
}
|
|
|
|
WebRtc_Word32
|
|
RTPReceiverVideo::RegisterIncomingVideoCallback(RtpVideoFeedback* incomingMessagesCallback)
|
|
{
|
|
CriticalSectionScoped lock(_criticalSectionFeedback);
|
|
_cbVideoFeedback = incomingMessagesCallback;
|
|
return 0;
|
|
}
|
|
|
|
void
|
|
RTPReceiverVideo::UpdateBandwidthManagement(const WebRtc_UWord32 minBitrateBps,
|
|
const WebRtc_UWord32 maxBitrateBps,
|
|
const WebRtc_UWord8 fractionLost,
|
|
const WebRtc_UWord16 roundTripTimeMs,
|
|
const WebRtc_UWord16 bwEstimateKbitMin,
|
|
const WebRtc_UWord16 bwEstimateKbitMax)
|
|
{
|
|
CriticalSectionScoped lock(_criticalSectionFeedback);
|
|
if(_cbVideoFeedback)
|
|
{
|
|
_cbVideoFeedback->OnNetworkChanged(_id, minBitrateBps, maxBitrateBps, fractionLost, roundTripTimeMs, bwEstimateKbitMin, bwEstimateKbitMax);
|
|
}
|
|
}
|
|
|
|
ModuleRTPUtility::Payload*
|
|
RTPReceiverVideo::RegisterReceiveVideoPayload(const WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE],
|
|
const WebRtc_Word8 payloadType,
|
|
const WebRtc_UWord32 maxRate)
|
|
{
|
|
RtpVideoCodecTypes videoType = kRtpNoVideo;
|
|
if (ModuleRTPUtility::StringCompare(payloadName, "VP8",3))
|
|
{
|
|
videoType = kRtpVp8Video;
|
|
|
|
} else if ((ModuleRTPUtility::StringCompare(payloadName, "H263-1998", 9)) ||
|
|
(ModuleRTPUtility::StringCompare(payloadName, "H263-2000", 9)))
|
|
{
|
|
videoType = kRtpH2631998Video;
|
|
|
|
} else if (ModuleRTPUtility::StringCompare(payloadName, "H263", 4))
|
|
{
|
|
videoType = kRtpH263Video;
|
|
|
|
} else if (ModuleRTPUtility::StringCompare(payloadName, "MP4V-ES", 7))
|
|
{
|
|
videoType = kRtpMpeg4Video;
|
|
|
|
} else if (ModuleRTPUtility::StringCompare(payloadName, "I420", 4))
|
|
{
|
|
videoType = kRtpNoVideo;
|
|
|
|
} else if (ModuleRTPUtility::StringCompare(payloadName, "ULPFEC", 6))
|
|
{
|
|
// store this
|
|
if(_receiveFEC == NULL)
|
|
{
|
|
_receiveFEC = new ReceiverFEC(_id, this);
|
|
}
|
|
_receiveFEC->SetPayloadTypeFEC(payloadType);
|
|
videoType = kRtpFecVideo;
|
|
}else
|
|
{
|
|
return NULL;
|
|
}
|
|
|
|
ModuleRTPUtility::Payload* payload = new ModuleRTPUtility::Payload;
|
|
strncpy(payload->name, payloadName, RTP_PAYLOAD_NAME_SIZE);
|
|
payload->typeSpecific.Video.videoCodecType = videoType;
|
|
payload->typeSpecific.Video.maxRate = maxRate;
|
|
payload->audio = false;
|
|
return payload;
|
|
}
|
|
|
|
void RTPReceiverVideo::ResetOverUseDetector()
|
|
{
|
|
_overUseDetector.Reset();
|
|
_videoBitRate.Init();
|
|
_lastBitRateChange = 0;
|
|
}
|
|
|
|
// called under _criticalSectionReceiverVideo
|
|
WebRtc_UWord16
|
|
RTPReceiverVideo::EstimateBandwidth(const WebRtc_UWord16 bandwidth)
|
|
{
|
|
// received fragments
|
|
// estimate BW
|
|
|
|
WebRtc_UWord16 bwSort[BW_HISTORY_SIZE];
|
|
for(int i = 0; i < BW_HISTORY_SIZE-1; i++)
|
|
{
|
|
_receivedBW[i] = _receivedBW[i+1];
|
|
bwSort[i] = _receivedBW[i+1];
|
|
}
|
|
_receivedBW[BW_HISTORY_SIZE-1] = bandwidth;
|
|
bwSort[BW_HISTORY_SIZE-1] = bandwidth;
|
|
|
|
WebRtc_UWord16 temp;
|
|
for (int i = BW_HISTORY_SIZE-1; i >= 0; i--)
|
|
{
|
|
for (int j = 1; j <= i; j++)
|
|
{
|
|
if (bwSort[j-1] > bwSort[j])
|
|
{
|
|
temp = bwSort[j-1];
|
|
bwSort[j-1] = bwSort[j];
|
|
bwSort[j] = temp;
|
|
}
|
|
}
|
|
}
|
|
int zeroCount = 0;
|
|
for (; zeroCount < BW_HISTORY_SIZE; zeroCount++)
|
|
{
|
|
if (bwSort[zeroCount]!= 0)
|
|
{
|
|
break;
|
|
}
|
|
}
|
|
WebRtc_UWord32 indexMedian = (BW_HISTORY_SIZE -1) - (BW_HISTORY_SIZE-zeroCount)/2;
|
|
WebRtc_UWord16 bandwidthMedian = bwSort[indexMedian];
|
|
|
|
if (bandwidthMedian > 0)
|
|
{
|
|
if (_estimatedBW == bandwidth)
|
|
{
|
|
// don't trigger a callback
|
|
bandwidthMedian = 0;
|
|
} else
|
|
{
|
|
_estimatedBW = bandwidthMedian;
|
|
}
|
|
} else
|
|
{
|
|
// can't be negative
|
|
bandwidthMedian = 0;
|
|
}
|
|
|
|
return bandwidthMedian;
|
|
}
|
|
|
|
// we have no critext when calling this
|
|
// we are not allowed to have any critsects when calling CallbackOfReceivedPayloadData
|
|
WebRtc_Word32
|
|
RTPReceiverVideo::ParseVideoCodecSpecific(WebRtcRTPHeader* rtpHeader,
|
|
const WebRtc_UWord8* payloadData,
|
|
const WebRtc_UWord16 payloadDataLength,
|
|
const RtpVideoCodecTypes videoType,
|
|
const bool isRED,
|
|
const WebRtc_UWord8* incomingRtpPacket,
|
|
const WebRtc_UWord16 incomingRtpPacketSize)
|
|
{
|
|
WebRtc_Word32 retVal = 0;
|
|
|
|
_criticalSectionReceiverVideo.Enter();
|
|
|
|
_videoBitRate.Update(payloadDataLength, TickTime::MillisecondTimestamp());
|
|
|
|
// Add headers, ideally we would like to include for instance
|
|
// Ethernet header here as well.
|
|
const WebRtc_UWord16 packetSize = payloadDataLength + _packetOverHead +
|
|
rtpHeader->header.headerLength + rtpHeader->header.paddingLength;
|
|
_overUseDetector.Update(*rtpHeader, packetSize);
|
|
|
|
if (isRED)
|
|
{
|
|
if(_receiveFEC == NULL)
|
|
{
|
|
_criticalSectionReceiverVideo.Leave();
|
|
return -1;
|
|
}
|
|
if (rtpHeader->header.timestamp != TimeStamp())
|
|
{
|
|
// We have a new frame. Force a decode with the existing packets.
|
|
retVal = _receiveFEC->ProcessReceivedFEC(true);
|
|
_currentFecFrameDecoded = false;
|
|
}
|
|
|
|
bool FECpacket = false;
|
|
if(retVal != -1)
|
|
{
|
|
if (!_currentFecFrameDecoded)
|
|
{
|
|
retVal = _receiveFEC->AddReceivedFECPacket(rtpHeader, incomingRtpPacket, payloadDataLength, FECpacket);
|
|
|
|
if (retVal != -1 && (FECpacket || rtpHeader->header.markerBit))
|
|
{
|
|
// Only attempt a decode after receiving the last media packet.
|
|
retVal = _receiveFEC->ProcessReceivedFEC(false);
|
|
}
|
|
}else
|
|
{
|
|
_receiveFEC->AddReceivedFECInfo(rtpHeader,incomingRtpPacket, FECpacket);
|
|
}
|
|
}
|
|
_criticalSectionReceiverVideo.Leave();
|
|
|
|
if(retVal == 0 && FECpacket )
|
|
{
|
|
// callback with the received FEC packet, the normal packets are deliverd after parsing
|
|
// this contain the original RTP packet header but with empty payload and data length
|
|
rtpHeader->frameType = kFrameEmpty;
|
|
WebRtc_Word32 retVal = SetCodecType(videoType, rtpHeader); //we need this for the routing
|
|
if(retVal != 0)
|
|
{
|
|
return retVal;
|
|
}
|
|
retVal =CallbackOfReceivedPayloadData(NULL,
|
|
0,
|
|
rtpHeader);
|
|
}
|
|
}else
|
|
{
|
|
// will leave the _criticalSectionReceiverVideo critsect
|
|
retVal = ParseVideoCodecSpecificSwitch(rtpHeader,
|
|
payloadData,
|
|
payloadDataLength,
|
|
videoType);
|
|
}
|
|
|
|
// Update the remote rate control object and update the overuse
|
|
// detector with the current rate control region.
|
|
_criticalSectionReceiverVideo.Enter();
|
|
const RateControlInput input(_overUseDetector.State(),
|
|
_videoBitRate.BitRate(
|
|
TickTime::MillisecondTimestamp()),
|
|
_overUseDetector.NoiseVar());
|
|
_criticalSectionReceiverVideo.Leave();
|
|
|
|
// Call the callback outside critical section
|
|
const RateControlRegion region = _cbPrivateFeedback.OnOverUseStateUpdate(input);
|
|
|
|
_criticalSectionReceiverVideo.Enter();
|
|
_overUseDetector.SetRateControlRegion(region);
|
|
_criticalSectionReceiverVideo.Leave();
|
|
|
|
return retVal;
|
|
}
|
|
|
|
WebRtc_Word32
|
|
RTPReceiverVideo::BuildRTPheader(const WebRtcRTPHeader* rtpHeader,
|
|
WebRtc_UWord8* dataBuffer) const
|
|
{
|
|
dataBuffer[0] = static_cast<WebRtc_UWord8>(0x80); // version 2
|
|
dataBuffer[1] = static_cast<WebRtc_UWord8>(rtpHeader->header.payloadType);
|
|
if (rtpHeader->header.markerBit)
|
|
{
|
|
dataBuffer[1] |= kRtpMarkerBitMask; // MarkerBit is 1
|
|
}
|
|
|
|
ModuleRTPUtility::AssignUWord16ToBuffer(dataBuffer+2, rtpHeader->header.sequenceNumber);
|
|
ModuleRTPUtility::AssignUWord32ToBuffer(dataBuffer+4, rtpHeader->header.timestamp);
|
|
ModuleRTPUtility::AssignUWord32ToBuffer(dataBuffer+8, rtpHeader->header.ssrc);
|
|
|
|
WebRtc_Word32 rtpHeaderLength = 12;
|
|
|
|
// Add the CSRCs if any
|
|
if (rtpHeader->header.numCSRCs > 0)
|
|
{
|
|
if(rtpHeader->header.numCSRCs > 16)
|
|
{
|
|
// error
|
|
assert(false);
|
|
}
|
|
WebRtc_UWord8* ptr = &dataBuffer[rtpHeaderLength];
|
|
for (WebRtc_UWord32 i = 0; i < rtpHeader->header.numCSRCs; ++i)
|
|
{
|
|
ModuleRTPUtility::AssignUWord32ToBuffer(ptr, rtpHeader->header.arrOfCSRCs[i]);
|
|
ptr +=4;
|
|
}
|
|
dataBuffer[0] = (dataBuffer[0]&0xf0) | rtpHeader->header.numCSRCs;
|
|
|
|
// Update length of header
|
|
rtpHeaderLength += sizeof(WebRtc_UWord32)*rtpHeader->header.numCSRCs;
|
|
}
|
|
return rtpHeaderLength;
|
|
}
|
|
|
|
WebRtc_Word32
|
|
RTPReceiverVideo::ReceiveRecoveredPacketCallback(WebRtcRTPHeader* rtpHeader,
|
|
const WebRtc_UWord8* payloadData,
|
|
const WebRtc_UWord16 payloadDataLength)
|
|
{
|
|
_criticalSectionReceiverVideo.Enter();
|
|
|
|
_currentFecFrameDecoded = true;
|
|
|
|
ModuleRTPUtility::Payload* payload = NULL;
|
|
if (PayloadTypeToPayload(rtpHeader->header.payloadType, payload) != 0)
|
|
{
|
|
return -1;
|
|
}
|
|
// here we can re-create the original lost packet so that we can use it for the relay
|
|
// we need to re-create the RED header too
|
|
WebRtc_UWord8 recoveredPacket[IP_PACKET_SIZE];
|
|
WebRtc_UWord16 rtpHeaderLength = (WebRtc_UWord16)BuildRTPheader(rtpHeader, recoveredPacket);
|
|
|
|
const WebRtc_UWord8 REDForFECHeaderLength = 1;
|
|
|
|
// replace pltype
|
|
recoveredPacket[1] &= 0x80; // reset
|
|
recoveredPacket[1] += REDPayloadType(); // replace with RED payload type
|
|
|
|
// add RED header
|
|
recoveredPacket[rtpHeaderLength] = rtpHeader->header.payloadType; // f-bit always 0
|
|
|
|
memcpy(recoveredPacket + rtpHeaderLength + REDForFECHeaderLength, payloadData, payloadDataLength);
|
|
|
|
return ParseVideoCodecSpecificSwitch(rtpHeader,
|
|
payloadData,
|
|
payloadDataLength,
|
|
payload->typeSpecific.Video.videoCodecType);
|
|
}
|
|
|
|
WebRtc_Word32
|
|
RTPReceiverVideo::SetCodecType(const RtpVideoCodecTypes videoType,
|
|
WebRtcRTPHeader* rtpHeader) const
|
|
{
|
|
switch (videoType)
|
|
{
|
|
case kRtpNoVideo:
|
|
rtpHeader->type.Video.codec = kRTPVideoGeneric;
|
|
break;
|
|
case kRtpVp8Video:
|
|
rtpHeader->type.Video.codec = kRTPVideoVP8;
|
|
break;
|
|
case kRtpH263Video:
|
|
rtpHeader->type.Video.codec = kRTPVideoH263;
|
|
break;
|
|
case kRtpH2631998Video:
|
|
rtpHeader->type.Video.codec = kRTPVideoH263;
|
|
break;
|
|
case kRtpMpeg4Video:
|
|
rtpHeader->type.Video.codec = kRTPVideoMPEG4;
|
|
break;
|
|
case kRtpFecVideo:
|
|
rtpHeader->type.Video.codec = kRTPVideoFEC;
|
|
break;
|
|
default:
|
|
assert(((void)"ParseCodecSpecific videoType can not be unknown here!", false));
|
|
return -1;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
|
|
WebRtc_Word32
|
|
RTPReceiverVideo::ParseVideoCodecSpecificSwitch(WebRtcRTPHeader* rtpHeader,
|
|
const WebRtc_UWord8* payloadData,
|
|
const WebRtc_UWord16 payloadDataLength,
|
|
const RtpVideoCodecTypes videoType)
|
|
{
|
|
WebRtc_Word32 retVal = SetCodecType(videoType, rtpHeader);
|
|
if(retVal != 0)
|
|
{
|
|
return retVal;
|
|
}
|
|
|
|
// all receive functions release _criticalSectionReceiverVideo before returning
|
|
switch (videoType)
|
|
{
|
|
case kRtpNoVideo:
|
|
retVal = ReceiveGenericCodec(rtpHeader, payloadData, payloadDataLength);
|
|
break;
|
|
case kRtpVp8Video:
|
|
retVal = ReceiveVp8Codec(rtpHeader, payloadData, payloadDataLength);
|
|
break;
|
|
case kRtpH263Video:
|
|
retVal = ReceiveH263Codec(rtpHeader, payloadData, payloadDataLength);
|
|
break;
|
|
case kRtpH2631998Video:
|
|
retVal = ReceiveH2631998Codec(rtpHeader,payloadData, payloadDataLength);
|
|
break;
|
|
case kRtpMpeg4Video:
|
|
retVal = ReceiveMPEG4Codec(rtpHeader,payloadData, payloadDataLength);
|
|
break;
|
|
default:
|
|
_criticalSectionReceiverVideo.Leave();
|
|
assert(((void)"ParseCodecSpecific videoType can not be unknown here!", false));
|
|
return -1;
|
|
}
|
|
return retVal;
|
|
}
|
|
|
|
WebRtc_Word32
|
|
RTPReceiverVideo::ReceiveH263Codec(WebRtcRTPHeader* rtpHeader,
|
|
const WebRtc_UWord8* payloadData,
|
|
const WebRtc_UWord16 payloadDataLength)
|
|
{
|
|
ModuleRTPUtility::RTPPayloadParser rtpPayloadParser(kRtpH263Video,
|
|
payloadData,
|
|
payloadDataLength);
|
|
ModuleRTPUtility::RTPPayload parsedPacket;
|
|
const bool success = rtpPayloadParser.Parse(parsedPacket);
|
|
|
|
// from here down we only work on local data
|
|
_criticalSectionReceiverVideo.Leave();
|
|
|
|
if (!success)
|
|
{
|
|
return -1;
|
|
}
|
|
if (IP_PACKET_SIZE < parsedPacket.info.H263.dataLength + parsedPacket.info.H263.insert2byteStartCode? 2:0)
|
|
{
|
|
return -1;
|
|
}
|
|
return ReceiveH263CodecCommon(parsedPacket, rtpHeader);
|
|
}
|
|
|
|
WebRtc_Word32
|
|
RTPReceiverVideo::ReceiveH2631998Codec(WebRtcRTPHeader* rtpHeader,
|
|
const WebRtc_UWord8* payloadData,
|
|
const WebRtc_UWord16 payloadDataLength)
|
|
{
|
|
ModuleRTPUtility::RTPPayloadParser rtpPayloadParser(kRtpH2631998Video,
|
|
payloadData,
|
|
payloadDataLength);
|
|
|
|
ModuleRTPUtility::RTPPayload parsedPacket;
|
|
const bool success = rtpPayloadParser.Parse(parsedPacket);
|
|
if (!success)
|
|
{
|
|
_criticalSectionReceiverVideo.Leave();
|
|
return -1;
|
|
}
|
|
if (IP_PACKET_SIZE < parsedPacket.info.H263.dataLength + parsedPacket.info.H263.insert2byteStartCode? 2:0)
|
|
{
|
|
_criticalSectionReceiverVideo.Leave();
|
|
return -1;
|
|
}
|
|
// from here down we only work on local data
|
|
_criticalSectionReceiverVideo.Leave();
|
|
|
|
return ReceiveH263CodecCommon(parsedPacket, rtpHeader);
|
|
}
|
|
|
|
WebRtc_Word32
|
|
RTPReceiverVideo::ReceiveH263CodecCommon(ModuleRTPUtility::RTPPayload& parsedPacket,
|
|
WebRtcRTPHeader* rtpHeader)
|
|
{
|
|
rtpHeader->frameType = (parsedPacket.frameType == ModuleRTPUtility::kIFrame) ? kVideoFrameKey : kVideoFrameDelta;
|
|
if (_h263InverseLogic) // Microsoft H263 bug
|
|
{
|
|
if (rtpHeader->frameType == kVideoFrameKey)
|
|
rtpHeader->frameType = kVideoFrameDelta;
|
|
else
|
|
rtpHeader->frameType = kVideoFrameKey;
|
|
}
|
|
rtpHeader->type.Video.isFirstPacket = parsedPacket.info.H263.hasPictureStartCode;
|
|
|
|
// if p == 0
|
|
// it's a follow-on packet, hence it's not independently decodable
|
|
rtpHeader->type.Video.codecHeader.H263.independentlyDecodable = parsedPacket.info.H263.hasPbit;
|
|
|
|
if (parsedPacket.info.H263.hasPictureStartCode)
|
|
{
|
|
rtpHeader->type.Video.width = parsedPacket.info.H263.frameWidth;
|
|
rtpHeader->type.Video.height = parsedPacket.info.H263.frameHeight;
|
|
} else
|
|
{
|
|
rtpHeader->type.Video.width = 0;
|
|
rtpHeader->type.Video.height = 0;
|
|
}
|
|
rtpHeader->type.Video.codecHeader.H263.bits = (parsedPacket.info.H263.startBits > 0)?true:false;
|
|
|
|
// copy to a local buffer
|
|
WebRtc_UWord8 dataBuffer[IP_PACKET_SIZE];
|
|
WebRtc_UWord16 dataLength = 0;
|
|
|
|
// we need to copy since we modify the first byte
|
|
if(parsedPacket.info.H263.insert2byteStartCode)
|
|
{
|
|
dataBuffer[0] = 0;
|
|
dataBuffer[1] = 0;
|
|
memcpy(dataBuffer+2, parsedPacket.info.H263.data, parsedPacket.info.H263.dataLength);
|
|
dataLength = 2 + parsedPacket.info.H263.dataLength;
|
|
} else
|
|
{
|
|
memcpy(dataBuffer, parsedPacket.info.H263.data, parsedPacket.info.H263.dataLength);
|
|
dataLength = parsedPacket.info.H263.dataLength;
|
|
}
|
|
|
|
if(parsedPacket.info.H263.dataLength > 0)
|
|
{
|
|
if(parsedPacket.info.H263.startBits > 0)
|
|
{
|
|
// make sure that the ignored start bits are zero
|
|
dataBuffer[0] &= (0xff >> parsedPacket.info.H263.startBits);
|
|
}
|
|
if(parsedPacket.info.H263.endBits > 0)
|
|
{
|
|
// make sure that the ignored end bits are zero
|
|
dataBuffer[parsedPacket.info.H263.dataLength -1] &= ((0xff << parsedPacket.info.H263.endBits) & 0xff);
|
|
}
|
|
}
|
|
|
|
return CallbackOfReceivedPayloadData(dataBuffer, dataLength, rtpHeader);
|
|
}
|
|
|
|
WebRtc_Word32
|
|
RTPReceiverVideo::ReceiveMPEG4Codec(WebRtcRTPHeader* rtpHeader,
|
|
const WebRtc_UWord8* payloadData,
|
|
const WebRtc_UWord16 payloadDataLength)
|
|
{
|
|
ModuleRTPUtility::RTPPayloadParser rtpPayloadParser(kRtpMpeg4Video,
|
|
payloadData,
|
|
payloadDataLength);
|
|
|
|
ModuleRTPUtility::RTPPayload parsedPacket;
|
|
const bool success = rtpPayloadParser.Parse(parsedPacket);
|
|
if (!success)
|
|
{
|
|
_criticalSectionReceiverVideo.Leave();
|
|
return -1;
|
|
}
|
|
// from here down we only work on local data
|
|
_criticalSectionReceiverVideo.Leave();
|
|
|
|
rtpHeader->frameType = (parsedPacket.frameType == ModuleRTPUtility::kIFrame) ? kVideoFrameKey : kVideoFrameDelta;
|
|
rtpHeader->type.Video.isFirstPacket = parsedPacket.info.MPEG4.isFirstPacket;
|
|
|
|
if(CallbackOfReceivedPayloadData(parsedPacket.info.MPEG4.data,
|
|
parsedPacket.info.MPEG4.dataLength,
|
|
rtpHeader) != 0)
|
|
{
|
|
return -1;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
WebRtc_Word32
|
|
RTPReceiverVideo::ReceiveVp8Codec(WebRtcRTPHeader* rtpHeader,
|
|
const WebRtc_UWord8* payloadData,
|
|
const WebRtc_UWord16 payloadDataLength)
|
|
{
|
|
ModuleRTPUtility::RTPPayloadParser rtpPayloadParser(kRtpVp8Video,
|
|
payloadData,
|
|
payloadDataLength);
|
|
|
|
ModuleRTPUtility::RTPPayload parsedPacket;
|
|
const bool success = rtpPayloadParser.Parse(parsedPacket);
|
|
|
|
// from here down we only work on local data
|
|
_criticalSectionReceiverVideo.Leave();
|
|
|
|
if (!success)
|
|
{
|
|
return -1;
|
|
}
|
|
if (parsedPacket.info.VP8.dataLength == 0)
|
|
{
|
|
// we have an "empty" VP8 packet, it's ok, could be one way video
|
|
return 0;
|
|
}
|
|
rtpHeader->frameType = (parsedPacket.frameType == ModuleRTPUtility::kIFrame) ? kVideoFrameKey : kVideoFrameDelta;
|
|
|
|
rtpHeader->type.Video.codecHeader.VP8.startBit = parsedPacket.info.VP8.startFragment; // Start of partition
|
|
rtpHeader->type.Video.codecHeader.VP8.stopBit= parsedPacket.info.VP8.stopFragment; // Stop of partition
|
|
|
|
rtpHeader->type.Video.isFirstPacket = parsedPacket.info.VP8.beginningOfFrame;
|
|
|
|
if(CallbackOfReceivedPayloadData(parsedPacket.info.VP8.data,
|
|
parsedPacket.info.VP8.dataLength,
|
|
rtpHeader) != 0)
|
|
{
|
|
return -1;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
|
|
WebRtc_Word32
|
|
RTPReceiverVideo::ReceiveGenericCodec(WebRtcRTPHeader* rtpHeader,
|
|
const WebRtc_UWord8* payloadData,
|
|
const WebRtc_UWord16 payloadDataLength)
|
|
{
|
|
rtpHeader->frameType = kVideoFrameKey;
|
|
|
|
if(((SequenceNumber() + 1) == rtpHeader->header.sequenceNumber) &&
|
|
(TimeStamp() != rtpHeader->header.timestamp))
|
|
{
|
|
rtpHeader->type.Video.isFirstPacket = true;
|
|
}
|
|
_criticalSectionReceiverVideo.Leave();
|
|
|
|
if(CallbackOfReceivedPayloadData(payloadData, payloadDataLength, rtpHeader) != 0)
|
|
{
|
|
return -1;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
WebRtc_Word32 RTPReceiverVideo::SetH263InverseLogic(const bool enable)
|
|
{
|
|
_h263InverseLogic = enable;
|
|
return 0;
|
|
}
|
|
|
|
void RTPReceiverVideo::SetPacketOverHead(WebRtc_UWord16 packetOverHead)
|
|
{
|
|
_packetOverHead = packetOverHead;
|
|
}
|
|
} // namespace webrtc
|