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📄 ac3audiortpsink.cpp

📁 流媒体传输协议的实现代码,非常有用.可以支持rtsp mms等流媒体传输协议
💻 CPP
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/**********This library is free software; you can redistribute it and/or modify it underthe terms of the GNU Lesser General Public License as published by theFree Software Foundation; either version 2.1 of the License, or (at youroption) any later version. (See <http://www.gnu.org/copyleft/lesser.html>.)This library is distributed in the hope that it will be useful, but WITHOUTANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESSFOR A PARTICULAR PURPOSE.  See the GNU Lesser General Public License formore details.You should have received a copy of the GNU Lesser General Public Licensealong with this library; if not, write to the Free Software Foundation, Inc.,59 Temple Place, Suite 330, Boston, MA  02111-1307  USA**********/// "liveMedia"// Copyright (c) 1996-2004 Live Networks, Inc.  All rights reserved.// RTP sink for AC3 audio// Implementation#include "AC3AudioRTPSink.hh"AC3AudioRTPSink::AC3AudioRTPSink(UsageEnvironment& env, Groupsock* RTPgs,				 u_int8_t rtpPayloadFormat,				 u_int32_t rtpTimestampFrequency)  : AudioRTPSink(env, RTPgs, rtpPayloadFormat,		       rtpTimestampFrequency, "AC3") {}AC3AudioRTPSink::~AC3AudioRTPSink() {}AC3AudioRTPSink*AC3AudioRTPSink::createNew(UsageEnvironment& env, Groupsock* RTPgs,			   u_int8_t rtpPayloadFormat,			   u_int32_t rtpTimestampFrequency) {  return new AC3AudioRTPSink(env, RTPgs,			     rtpPayloadFormat, rtpTimestampFrequency);}Boolean AC3AudioRTPSink::frameCanAppearAfterPacketStart(unsigned char const* /*frameStart*/,                                 unsigned /*numBytesInFrame*/) const {  // (For now) allow at most 1 frame in a single packet:  return False;}void AC3AudioRTPSink::doSpecialFrameHandling(unsigned fragmentationOffset,			 unsigned char* frameStart,			 unsigned numBytesInFrame,			 struct timeval frameTimestamp,			 unsigned numRemainingBytes) {  // Update the "NDU" header.  // Also set the "Data Unit Header" for the frame, because we  // have already allotted space for this, by virtue of the fact that  // (for now) we pack only one frame in each RTP packet:  unsigned char headers[2];  headers[0] = numFramesUsedSoFar() + 1;  Boolean isFragment = numRemainingBytes > 0 || fragmentationOffset > 0;  unsigned const totalFrameSize    = fragmentationOffset + numBytesInFrame + numRemainingBytes;  unsigned const fiveEighthsPoint = totalFrameSize/2 + totalFrameSize/8;  Boolean haveFiveEighths    = fragmentationOffset == 0 && numBytesInFrame >= fiveEighthsPoint;  headers[1] = (isFragment<<5)|(haveFiveEighths<<4); // F|B      // Note: TYP==0, RDT==0 ???, T==0 ???  setSpecialHeaderBytes(headers, sizeof headers);  if (numRemainingBytes == 0) {    // This packet contains the last (or only) fragment of the frame.    // Set the RTP 'M' ('marker') bit:    setMarkerBit();  }  // Important: Also call our base class's doSpecialFrameHandling(),  // to set the packet's timestamp:  MultiFramedRTPSink::doSpecialFrameHandling(fragmentationOffset,					     frameStart, numBytesInFrame,					     frameTimestamp,					     numRemainingBytes);}unsigned AC3AudioRTPSink::specialHeaderSize() const {  // There's a 1 byte "NDU" header.  // There's also a 1-byte "Data Unit Header" preceding each frame in  // the RTP packet, but since we (for now) pack only one frame in  // each RTP packet, we also count this here:  return 1 + 1;}

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