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📄 mpeg1or2audiostreamframer.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.// A filter that breaks up an MPEG (1,2) audio elementary stream into frames// Implementation#include "MPEG1or2AudioStreamFramer.hh"#include "StreamParser.hh"#include "MP3Internals.hh"#include <GroupsockHelper.hh>////////// MPEG1or2AudioStreamParser definition //////////class MPEG1or2AudioStreamParser: public StreamParser {public:  MPEG1or2AudioStreamParser(MPEG1or2AudioStreamFramer* usingSource,			FramedSource* inputSource);  virtual ~MPEG1or2AudioStreamParser();public:  unsigned parse(unsigned& numTruncatedBytes);      // returns the size of the frame that was acquired, or 0 if none was  void registerReadInterest(unsigned char* to, unsigned maxSize);  MP3FrameParams const& currentFrame() const { return fCurrentFrame; }private:  unsigned char* fTo;  unsigned fMaxSize;  // Parameters of the most recently read frame:  MP3FrameParams fCurrentFrame; // also works for layer I or II};////////// MPEG1or2AudioStreamFramer implementation //////////MPEG1or2AudioStreamFramer::MPEG1or2AudioStreamFramer(UsageEnvironment& env, FramedSource* inputSource,			    Boolean syncWithInputSource)  : FramedFilter(env, inputSource),    fSyncWithInputSource(syncWithInputSource) {  reset();  fParser = new MPEG1or2AudioStreamParser(this, inputSource);}MPEG1or2AudioStreamFramer::~MPEG1or2AudioStreamFramer() {  delete fParser;}MPEG1or2AudioStreamFramer*MPEG1or2AudioStreamFramer::createNew(UsageEnvironment& env,				     FramedSource* inputSource,				     Boolean syncWithInputSource) {  // Need to add source type checking here???  #####  return new MPEG1or2AudioStreamFramer(env, inputSource, syncWithInputSource);}void MPEG1or2AudioStreamFramer::flushInput() {  reset();  fParser->flushInput();}void MPEG1or2AudioStreamFramer::reset() {  // Use the current wallclock time as the initial 'presentation time':  struct timeval timeNow;  gettimeofday(&timeNow, NULL);  resetPresentationTime(timeNow);}void MPEG1or2AudioStreamFramer::resetPresentationTime(struct timeval newPresentationTime) {  fNextFramePresentationTime = newPresentationTime;}void MPEG1or2AudioStreamFramer::doGetNextFrame() {  fParser->registerReadInterest(fTo, fMaxSize);  continueReadProcessing();}#define MILLION 1000000static unsigned const numSamplesByLayer[4] = {0, 384, 1152, 1152};struct timeval MPEG1or2AudioStreamFramer::currentFramePlayTime() const {  MP3FrameParams const& fr = fParser->currentFrame();  unsigned const numSamples = numSamplesByLayer[fr.layer];  unsigned const freq = fr.samplingFreq*(1 + fr.isMPEG2);  // result is numSamples/freq  unsigned const uSeconds    = ((numSamples*2*MILLION)/freq + 1)/2; // rounds to nearest integer  struct timeval result;  result.tv_sec = uSeconds/MILLION;  result.tv_usec = uSeconds%MILLION;  return result;}void MPEG1or2AudioStreamFramer::continueReadProcessing(void* clientData,			 unsigned char* /*ptr*/, unsigned /*size*/,			 struct timeval presentationTime) {  MPEG1or2AudioStreamFramer* framer = (MPEG1or2AudioStreamFramer*)clientData;  if (framer->fSyncWithInputSource) {    framer->resetPresentationTime(presentationTime);  }  framer->continueReadProcessing();}void MPEG1or2AudioStreamFramer::continueReadProcessing() {  unsigned acquiredFrameSize = fParser->parse(fNumTruncatedBytes);  if (acquiredFrameSize > 0) {    // We were able to acquire a frame from the input.    // It has already been copied to the reader's space.    fFrameSize = acquiredFrameSize;        // Also set the presentation time, and increment it for next time,    // based on the length of this frame:    fPresentationTime = fNextFramePresentationTime;    struct timeval framePlayTime = currentFramePlayTime();    fDurationInMicroseconds = framePlayTime.tv_sec*MILLION + framePlayTime.tv_usec;    fNextFramePresentationTime.tv_usec += framePlayTime.tv_usec;    fNextFramePresentationTime.tv_sec      += framePlayTime.tv_sec + fNextFramePresentationTime.tv_usec/MILLION;    fNextFramePresentationTime.tv_usec %= MILLION;    // Call our own 'after getting' function.  Because we're not a 'leaf'    // source, we can call this directly, without risking infinite recursion.    afterGetting(this);  } else {    // We were unable to parse a complete frame from the input, because:    // - we had to read more data from the source stream, or    // - the source stream has ended.  }}////////// MPEG1or2AudioStreamParser implementation //////////MPEG1or2AudioStreamParser::MPEG1or2AudioStreamParser(MPEG1or2AudioStreamFramer* usingSource,			FramedSource* inputSource)  : StreamParser(inputSource, FramedSource::handleClosure, usingSource,		 &MPEG1or2AudioStreamFramer::continueReadProcessing, usingSource) {}MPEG1or2AudioStreamParser::~MPEG1or2AudioStreamParser() {}void MPEG1or2AudioStreamParser::registerReadInterest(unsigned char* to,						 unsigned maxSize) {  fTo = to;  fMaxSize = maxSize;}unsigned MPEG1or2AudioStreamParser::parse(unsigned& numTruncatedBytes) {  try {    saveParserState();        // We expect a MPEG audio header (first 11 bits set to 1) at the start:    while (((fCurrentFrame.hdr = test4Bytes())&0xFFE00000) != 0xFFE00000) {      skipBytes(1);      saveParserState();    }        fCurrentFrame.setParamsFromHeader();        // Copy the frame to the requested destination:    unsigned frameSize = fCurrentFrame.frameSize + 4; // include header    if (frameSize > fMaxSize) {      numTruncatedBytes = frameSize - fMaxSize;      frameSize = fMaxSize;    } else {      numTruncatedBytes = 0;    }    getBytes(fTo, frameSize);    skipBytes(numTruncatedBytes);        return frameSize;  } catch (int /*e*/) {#ifdef DEBUG    fprintf(stderr, "MPEG1or2AudioStreamParser::parse() EXCEPTION (This is normal behavior - *not* an error)\n");#endif    return 0;  // the parsing got interrupted  }}

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