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Changed handling of padding data.

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Review URL: http://webrtc-codereview.appspot.com/331008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1252 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pwestin@webrtc.org 2011-12-20 15:56:17 +00:00
parent dbba1f969f
commit 061fa5b828
4 changed files with 82 additions and 57 deletions
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24
src/modules/rtp_rtcp/source/rtp_receiver.cc
View File

@ -804,12 +804,6 @@ RTPReceiver::IncomingRTPPacket(WebRtcRTPHeader* rtpHeader,
}
}
}
if(length - rtpHeader->header.headerLength == 0)
{
// ok keepalive packet
return 0;
}
WebRtc_Word8 firstPayloadByte = 0;
if(length > 0)
{
@ -829,9 +823,23 @@ RTPReceiver::IncomingRTPPacket(WebRtcRTPHeader* rtpHeader,
audioSpecific.channels = 0;
audioSpecific.frequency = 0;
if(CheckPayloadChanged(rtpHeader, firstPayloadByte, isRED, audioSpecific, videoSpecific) == -1)
if (CheckPayloadChanged(rtpHeader,
firstPayloadByte,
isRED,
audioSpecific,
videoSpecific) == -1)
{
WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id, "%s received invalid payloadtype", __FUNCTION__);
if (length - rtpHeader->header.headerLength == 0)
{
// ok keepalive packet
WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, _id,
"%s received keepalive",
__FUNCTION__);
return 0;
}
WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id,
"%s received invalid payloadtype",
__FUNCTION__);
return -1;
}
CheckCSRC(rtpHeader);

22
src/modules/rtp_rtcp/source/rtp_receiver_video.cc
View File

@ -231,7 +231,8 @@ RTPReceiverVideo::ParseVideoCodecSpecific(WebRtcRTPHeader* rtpHeader,
_criticalSectionReceiverVideo->Enter();
_videoBitRate.Update(payloadDataLength, nowMS);
_videoBitRate.Update(payloadDataLength + rtpHeader->header.paddingLength,
nowMS);
// Add headers, ideally we would like to include for instance
// Ethernet header here as well.
@ -646,15 +647,22 @@ WebRtc_Word32
RTPReceiverVideo::ReceiveVp8Codec(WebRtcRTPHeader* rtpHeader,
const WebRtc_UWord8* payloadData,
const WebRtc_UWord16 payloadDataLength)
{
bool success;
ModuleRTPUtility::RTPPayload parsedPacket;
if (payloadDataLength == 0)
{
success = true;
parsedPacket.info.VP8.dataLength = 0;
} else
{
ModuleRTPUtility::RTPPayloadParser rtpPayloadParser(kRtpVp8Video,
payloadData,
payloadDataLength,
_id);
ModuleRTPUtility::RTPPayload parsedPacket;
const bool success = rtpPayloadParser.Parse(parsedPacket);
success = rtpPayloadParser.Parse(parsedPacket);
}
// from here down we only work on local data
_criticalSectionReceiverVideo->Leave();
@ -665,6 +673,12 @@ RTPReceiverVideo::ReceiveVp8Codec(WebRtcRTPHeader* rtpHeader,
if (parsedPacket.info.VP8.dataLength == 0)
{
// we have an "empty" VP8 packet, it's ok, could be one way video
// Inform the jitter buffer about this packet.
rtpHeader->frameType = kFrameEmpty;
if (CallbackOfReceivedPayloadData(NULL, 0, rtpHeader) != 0)
{
return -1;
}
return 0;
}
rtpHeader->frameType = (parsedPacket.frameType == ModuleRTPUtility::kIFrame) ? kVideoFrameKey : kVideoFrameDelta;

72
src/modules/rtp_rtcp/source/rtp_sender_video.cc
View File

@ -545,50 +545,52 @@ RTPSenderVideo::SendGeneric(const WebRtc_Word8 payloadType,
return 0;
}
WebRtc_Word32
RTPSenderVideo::SendPadData(const WebRtcRTPHeader* rtpHeader,
const WebRtc_UWord32 bytes)
{
const WebRtc_UWord16 rtpHeaderLength = _rtpSender.RTPHeaderLength();
WebRtc_UWord32 maxLength = _rtpSender.MaxPayloadLength() -
FECPacketOverhead() - rtpHeaderLength;
WebRtc_UWord8 dataBuffer[IP_PACKET_SIZE];
if(bytes<maxLength)
{
// For a small packet don't spend too much time
maxLength = bytes;
}
void RTPSenderVideo::SendPadData(WebRtc_Word8 payload_type,
WebRtc_UWord32 capture_timestamp,
WebRtc_Word32 bytes) {
// Max in the RFC 3550 is 255 bytes, we limit it to be modulus 32 for SRTP.
int max_length = 224;
WebRtc_UWord8 data_buffer[IP_PACKET_SIZE];
for (; bytes > 0; bytes -= max_length) {
WebRtc_Word32 header_length;
{
CriticalSectionScoped cs(_sendVideoCritsect);
// Send paded data
// Correct seq num, time stamp and payloadtype
// We reuse the last seq number
_rtpSender.BuildRTPheader(dataBuffer, rtpHeader->header.payloadType,
false, 0, false, false);
// Correct seq num, timestamp and payload type.
header_length = _rtpSender.BuildRTPheader(
data_buffer,
payload_type,
false, // No markerbit.
capture_timestamp,
true, // Timestamp provided.
true); // Increment sequence number.
// Version 0 to be compatible with old ViE
dataBuffer[0] &= !0x80;
}
data_buffer[0] |= 0x20; // Set padding bit.
WebRtc_Word32* data =
reinterpret_cast<WebRtc_Word32*>(&(data_buffer[header_length]));
// Set relay SSRC
ModuleRTPUtility::AssignUWord32ToBuffer(dataBuffer+8,
rtpHeader->header.ssrc);
// Start at 12
WebRtc_Word32* data = (WebRtc_Word32*)&(dataBuffer[12]);
// Build data buffer
for(WebRtc_UWord32 j = 0; j < ((maxLength>>2)-4) && j < (bytes>>4); j++)
{
int padding_bytes_in_packet = max_length;
if (bytes < max_length) {
padding_bytes_in_packet = (bytes + 16) & 0xffe0; // Keep our modulus 32.
}
if (padding_bytes_in_packet < 32) {
// Sanity don't send empty packets.
return;
}
// Fill data buffer with random data.
for(int j = 0; j < (padding_bytes_in_packet >> 2); j++) {
data[j] = rand();
}
}
// Min
WebRtc_UWord16 length = (WebRtc_UWord16)(bytes<maxLength?bytes:maxLength);
// Set number of padding bytes in the last byte of the packet.
data_buffer[header_length + padding_bytes_in_packet - 1] =
padding_bytes_in_packet;
// Send the packet
return _rtpSender.SendToNetwork(dataBuffer, length, rtpHeaderLength, true);
_rtpSender.SendToNetwork(data_buffer,
padding_bytes_in_packet,
header_length);
}
}
WebRtc_Word32

5
src/modules/rtp_rtcp/source/rtp_sender_video.h
View File

@ -68,8 +68,9 @@ public:
WebRtc_UWord32 MaxConfiguredBitrateVideo() const;
WebRtc_Word32 SendPadData(const WebRtcRTPHeader* rtpHeader,
const WebRtc_UWord32 bytes);
void SendPadData(WebRtc_Word8 payload_type,
WebRtc_UWord32 capture_timestamp,
WebRtc_Word32 bytes);
// FEC
WebRtc_Word32 SetGenericFECStatus(const bool enable,