Changed handling of padding data.
Review URL: http://webrtc-codereview.appspot.com/331008 git-svn-id: http://webrtc.googlecode.com/svn/trunk@1252 4adac7df-926f-26a2-2b94-8c16560cd09d
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@ -804,12 +804,6 @@ RTPReceiver::IncomingRTPPacket(WebRtcRTPHeader* rtpHeader,
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}
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}
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}
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if(length - rtpHeader->header.headerLength == 0)
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{
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// ok keepalive packet
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return 0;
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}
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WebRtc_Word8 firstPayloadByte = 0;
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if(length > 0)
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{
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@ -829,9 +823,23 @@ RTPReceiver::IncomingRTPPacket(WebRtcRTPHeader* rtpHeader,
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audioSpecific.channels = 0;
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audioSpecific.frequency = 0;
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if(CheckPayloadChanged(rtpHeader, firstPayloadByte, isRED, audioSpecific, videoSpecific) == -1)
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if (CheckPayloadChanged(rtpHeader,
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firstPayloadByte,
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isRED,
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audioSpecific,
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videoSpecific) == -1)
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{
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WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id, "%s received invalid payloadtype", __FUNCTION__);
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if (length - rtpHeader->header.headerLength == 0)
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{
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// ok keepalive packet
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WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, _id,
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"%s received keepalive",
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__FUNCTION__);
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return 0;
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}
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WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id,
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"%s received invalid payloadtype",
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__FUNCTION__);
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return -1;
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}
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CheckCSRC(rtpHeader);
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@ -847,7 +855,7 @@ RTPReceiver::IncomingRTPPacket(WebRtcRTPHeader* rtpHeader,
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payloadDataLength,
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audioSpecific,
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isRED);
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}else
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} else
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{
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retVal = ParseVideoCodecSpecific(rtpHeader,
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payloadData,
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@ -231,7 +231,8 @@ RTPReceiverVideo::ParseVideoCodecSpecific(WebRtcRTPHeader* rtpHeader,
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_criticalSectionReceiverVideo->Enter();
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_videoBitRate.Update(payloadDataLength, nowMS);
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_videoBitRate.Update(payloadDataLength + rtpHeader->header.paddingLength,
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nowMS);
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// Add headers, ideally we would like to include for instance
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// Ethernet header here as well.
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@ -647,14 +648,21 @@ RTPReceiverVideo::ReceiveVp8Codec(WebRtcRTPHeader* rtpHeader,
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const WebRtc_UWord8* payloadData,
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const WebRtc_UWord16 payloadDataLength)
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{
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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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bool success;
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ModuleRTPUtility::RTPPayload parsedPacket;
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const bool success = rtpPayloadParser.Parse(parsedPacket);
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if (payloadDataLength == 0)
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{
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success = true;
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parsedPacket.info.VP8.dataLength = 0;
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} else
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{
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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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@ -665,6 +673,12 @@ RTPReceiverVideo::ReceiveVp8Codec(WebRtcRTPHeader* rtpHeader,
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if (parsedPacket.info.VP8.dataLength == 0)
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{
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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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{
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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) ? kVideoFrameKey : kVideoFrameDelta;
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@ -545,50 +545,52 @@ RTPSenderVideo::SendGeneric(const WebRtc_Word8 payloadType,
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return 0;
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}
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WebRtc_Word32
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RTPSenderVideo::SendPadData(const WebRtcRTPHeader* rtpHeader,
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const WebRtc_UWord32 bytes)
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{
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const WebRtc_UWord16 rtpHeaderLength = _rtpSender.RTPHeaderLength();
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WebRtc_UWord32 maxLength = _rtpSender.MaxPayloadLength() -
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FECPacketOverhead() - rtpHeaderLength;
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WebRtc_UWord8 dataBuffer[IP_PACKET_SIZE];
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void RTPSenderVideo::SendPadData(WebRtc_Word8 payload_type,
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WebRtc_UWord32 capture_timestamp,
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WebRtc_Word32 bytes) {
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// Max in the RFC 3550 is 255 bytes, we limit it to be modulus 32 for SRTP.
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int max_length = 224;
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WebRtc_UWord8 data_buffer[IP_PACKET_SIZE];
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if(bytes<maxLength)
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for (; bytes > 0; bytes -= max_length) {
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WebRtc_Word32 header_length;
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{
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// For a small packet don't spend too much time
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maxLength = bytes;
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CriticalSectionScoped cs(_sendVideoCritsect);
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// Correct seq num, timestamp and payload type.
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header_length = _rtpSender.BuildRTPheader(
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data_buffer,
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payload_type,
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false, // No markerbit.
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capture_timestamp,
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true, // Timestamp provided.
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true); // Increment sequence number.
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}
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data_buffer[0] |= 0x20; // Set padding bit.
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WebRtc_Word32* data =
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reinterpret_cast<WebRtc_Word32*>(&(data_buffer[header_length]));
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{
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CriticalSectionScoped cs(_sendVideoCritsect);
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// Send paded data
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// Correct seq num, time stamp and payloadtype
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// We reuse the last seq number
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_rtpSender.BuildRTPheader(dataBuffer, rtpHeader->header.payloadType,
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false, 0, false, false);
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// Version 0 to be compatible with old ViE
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dataBuffer[0] &= !0x80;
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// Set relay SSRC
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ModuleRTPUtility::AssignUWord32ToBuffer(dataBuffer+8,
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rtpHeader->header.ssrc);
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// Start at 12
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WebRtc_Word32* data = (WebRtc_Word32*)&(dataBuffer[12]);
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// Build data buffer
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for(WebRtc_UWord32 j = 0; j < ((maxLength>>2)-4) && j < (bytes>>4); j++)
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{
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data[j] = rand();
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}
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int padding_bytes_in_packet = max_length;
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if (bytes < max_length) {
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padding_bytes_in_packet = (bytes + 16) & 0xffe0; // Keep our modulus 32.
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}
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// Min
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WebRtc_UWord16 length = (WebRtc_UWord16)(bytes<maxLength?bytes:maxLength);
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if (padding_bytes_in_packet < 32) {
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// Sanity don't send empty packets.
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return;
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}
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// Fill data buffer with random data.
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for(int j = 0; j < (padding_bytes_in_packet >> 2); j++) {
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data[j] = rand();
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}
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// Set number of padding bytes in the last byte of the packet.
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data_buffer[header_length + padding_bytes_in_packet - 1] =
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padding_bytes_in_packet;
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// Send the packet
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return _rtpSender.SendToNetwork(dataBuffer, length, rtpHeaderLength, true);
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_rtpSender.SendToNetwork(data_buffer,
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padding_bytes_in_packet,
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header_length);
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}
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}
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WebRtc_Word32
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@ -68,8 +68,9 @@ public:
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WebRtc_UWord32 MaxConfiguredBitrateVideo() const;
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WebRtc_Word32 SendPadData(const WebRtcRTPHeader* rtpHeader,
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const WebRtc_UWord32 bytes);
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void SendPadData(WebRtc_Word8 payload_type,
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WebRtc_UWord32 capture_timestamp,
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WebRtc_Word32 bytes);
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// FEC
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WebRtc_Word32 SetGenericFECStatus(const bool enable,
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