amraudiortpsource.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

/**********
This library is free software; you can redistribute it and/or modify it under
the terms of the GNU Lesser General Public License as published by the
Free Software Foundation; either version 2.1 of the License, or (at your
option) any later version. (See <http://www.gnu.org/copyleft/lesser.html>.)

This library is distributed in the hope that it will be useful, but WITHOUT
ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS
FOR A PARTICULAR PURPOSE.  See the GNU Lesser General Public License for
more details.

You should have received a copy of the GNU Lesser General Public License
along 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.
// AMR Audio RTP Sources (RFC 3267)
// Implementation

#include "AMRAudioRTPSource.hh"
#include "MultiFramedRTPSource.hh"
#include "BitVector.hh"
#include <string.h>
#include <stdlib.h>

// This source is implemented internally by two separate sources:
// (i) a RTP source for the raw (and possibly interleaved) AMR frames, and
// (ii) a deinterleaving filter that reads from this.
// Define these two new classes here:

class RawAMRRTPSource: public MultiFramedRTPSource {
public:
  static RawAMRRTPSource*
  createNew(UsageEnvironment& env,
	    Groupsock* RTPgs, unsigned char rtpPayloadFormat,
	    Boolean isWideband, Boolean isOctetAligned,
	    Boolean isInterleaved, Boolean CRCsArePresent);

  Boolean isWideband() const { return fIsWideband; }
  unsigned char ILL() const { return fILL; }
  unsigned char ILP() const { return fILP; }
  unsigned TOCSize() const { return fTOCSize; } // total # of frames in the last pkt
  unsigned char* TOC() const { return fTOC; } // FT+Q value for each TOC entry
  unsigned& frameIndex() { return fFrameIndex; } // index of frame-block within pkt

private:
  RawAMRRTPSource(UsageEnvironment& env, Groupsock* RTPgs,
		  unsigned char rtpPayloadFormat,
		  Boolean isWideband, Boolean isOctetAligned,
		  Boolean isInterleaved, Boolean CRCsArePresent);
      // called only by createNew()

  virtual ~RawAMRRTPSource();

private:
  // redefined virtual functions:
  virtual Boolean processSpecialHeader(BufferedPacket* packet,
                                       unsigned& resultSpecialHeaderSize);
  virtual char const* MIMEtype() const; 

  virtual Boolean hasBeenSynchronizedUsingRTCP();

private:
  Boolean fIsWideband, fIsOctetAligned, fIsInterleaved, fCRCsArePresent;
  unsigned char fILL, fILP;
  unsigned fTOCSize;
  unsigned char* fTOC;
  unsigned fFrameIndex, fNumSuccessiveSyncedPackets;
};

class AMRDeinterleaver: public AMRAudioSource {
public:
  static AMRDeinterleaver*
  createNew(UsageEnvironment& env,
	    Boolean isWideband, unsigned numChannels, unsigned maxInterleaveGroupSize,
	    RawAMRRTPSource* inputSource);

private:
  AMRDeinterleaver(UsageEnvironment& env,
		   Boolean isWideband, unsigned numChannels,
		   unsigned maxInterleaveGroupSize, RawAMRRTPSource* inputSource);
      // called only by "createNew()" 

  virtual ~AMRDeinterleaver();

  static void afterGettingFrame(void* clientData, unsigned frameSize,
				unsigned numTruncatedBytes,
                                struct timeval presentationTime,
				unsigned durationInMicroseconds);
  void afterGettingFrame1(unsigned frameSize, struct timeval presentationTime); 

private:
  // Redefined virtual functions:
  void doGetNextFrame();

private:
  FramedSource* fInputSource;
  class AMRDeinterleavingBuffer* fDeinterleavingBuffer;
  Boolean fNeedAFrame;
};


////////// AMRAudioRTPSource implementation //////////

#define MAX_NUM_CHANNELS 20 // far larger than ever expected...
#define MAX_INTERLEAVING_GROUP_SIZE 1000 // far larger than ever expected...

AMRAudioSource*
AMRAudioRTPSource::createNew(UsageEnvironment& env,
			     Groupsock* RTPgs,
			     RTPSource*& resultRTPSource,
			     unsigned char rtpPayloadFormat,
			     Boolean isWideband,
			     unsigned numChannels,
			     Boolean isOctetAligned,
			     unsigned interleaving,
			     Boolean robustSortingOrder,
			     Boolean CRCsArePresent) {
  // Perform sanity checks on the input parameters:
  if (robustSortingOrder) {
    env << "AMRAudioRTPSource::createNew(): 'Robust sorting order' was specified, but we don't yet support this!\n";
    return NULL;
  } else if (numChannels > MAX_NUM_CHANNELS) {
    env << "AMRAudioRTPSource::createNew(): The \"number of channels\" parameter ("
	<< numChannels << ") is much too large!\n";
    return NULL;
  } else if (interleaving > MAX_INTERLEAVING_GROUP_SIZE) {
    env << "AMRAudioRTPSource::createNew(): The \"interleaving\" parameter ("
	<< interleaving << ") is much too large!\n";
    return NULL;
  }

  // 'Bandwidth-efficient mode' precludes some other options:
  if (!isOctetAligned) {
    if (interleaving > 0 || robustSortingOrder || CRCsArePresent) {
      env << "AMRAudioRTPSource::createNew(): 'Bandwidth-efficient mode' was specified, along with interleaving, 'robust sorting order', and/or CRCs, so we assume 'octet-aligned mode' instead.\n";
      isOctetAligned = True;
    }
  } 

  Boolean isInterleaved;
  unsigned maxInterleaveGroupSize; // in frames (not frame-blocks)
  if (interleaving > 0) {
    isInterleaved = True;
    maxInterleaveGroupSize = interleaving*numChannels;
  } else {
    isInterleaved = False;
    maxInterleaveGroupSize = numChannels;
  }

  RawAMRRTPSource* rawRTPSource;
  resultRTPSource = rawRTPSource
    = RawAMRRTPSource::createNew(env, RTPgs, rtpPayloadFormat,
				 isWideband, isOctetAligned,
				 isInterleaved, CRCsArePresent);
  if (resultRTPSource == NULL) return NULL;

  AMRDeinterleaver* deinterleaver
    = AMRDeinterleaver::createNew(env, isWideband, numChannels,
				  maxInterleaveGroupSize, rawRTPSource);
  if (deinterleaver == NULL) {
    Medium::close(resultRTPSource);
    resultRTPSource = NULL;
  }

  return deinterleaver;
}


////////// AMRBufferedPacket and AMRBufferedPacketFactory //////////

// A subclass of BufferedPacket, used to separate out AMR frames.

class AMRBufferedPacket: public BufferedPacket {
public:
  AMRBufferedPacket(RawAMRRTPSource& ourSource);
  virtual ~AMRBufferedPacket();

private: // redefined virtual functions
  virtual unsigned nextEnclosedFrameSize(unsigned char*& framePtr,
					 unsigned dataSize);
private:
  RawAMRRTPSource& fOurSource;
};

class AMRBufferedPacketFactory: public BufferedPacketFactory {
private: // redefined virtual functions
  virtual BufferedPacket* createNewPacket(MultiFramedRTPSource* ourSource);
};


///////// RawAMRRTPSource implementation ////////

RawAMRRTPSource*
RawAMRRTPSource::createNew(UsageEnvironment& env, Groupsock* RTPgs,
			   unsigned char rtpPayloadFormat,
			   Boolean isWideband, Boolean isOctetAligned,
			   Boolean isInterleaved, Boolean CRCsArePresent) {
  return new RawAMRRTPSource(env, RTPgs, rtpPayloadFormat,
			     isWideband, isOctetAligned,
			     isInterleaved, CRCsArePresent);
}

RawAMRRTPSource
::RawAMRRTPSource(UsageEnvironment& env,
		  Groupsock* RTPgs, unsigned char rtpPayloadFormat,
		  Boolean isWideband, Boolean isOctetAligned,
		  Boolean isInterleaved, Boolean CRCsArePresent)
  : MultiFramedRTPSource(env, RTPgs, rtpPayloadFormat,
			 isWideband ? 16000 : 8000,
                         new AMRBufferedPacketFactory),
  fIsWideband(isWideband), fIsOctetAligned(isOctetAligned),
  fIsInterleaved(isInterleaved), fCRCsArePresent(CRCsArePresent),
  fILL(0), fILP(0), fTOCSize(0), fTOC(NULL), fFrameIndex(0),
  fNumSuccessiveSyncedPackets(0) {
}

RawAMRRTPSource::~RawAMRRTPSource() {
  delete[] fTOC;
}

#define FT_SPEECH_LOST 14
#define FT_NO_DATA 15

static void unpackBandwidthEfficientData(BufferedPacket* packet,
					 Boolean isWideband); // forward

Boolean RawAMRRTPSource
::processSpecialHeader(BufferedPacket* packet,
		       unsigned& resultSpecialHeaderSize) {
  // If the data is 'bandwidth-efficient', first unpack it so that it's
  // 'octet-aligned':
  if (!fIsOctetAligned) unpackBandwidthEfficientData(packet, fIsWideband);

  unsigned char* headerStart = packet->data();
  unsigned packetSize = packet->dataSize();

  // First, check whether this packet's RTP timestamp is synchronized:
  if (RTPSource::hasBeenSynchronizedUsingRTCP()) {
    ++fNumSuccessiveSyncedPackets;
  } else {
    fNumSuccessiveSyncedPackets = 0;
  }

  // There's at least a 1-byte header, containing the CMR:
  if (packetSize < 1) return False;
  resultSpecialHeaderSize = 1;

  if (fIsInterleaved) {
    // There's an extra byte, containing the interleave parameters:
    if (packetSize < 2) return False;

    // Get the interleaving parameters, and check them for validity:
    unsigned char const secondByte = headerStart[1];
    fILL = (secondByte&0xF0)>>4;
    fILP = secondByte&0x0F;
    if (fILP > fILL) return False; // invalid
    ++resultSpecialHeaderSize;
  }
#ifdef DEBUG
  fprintf(stderr, "packetSize: %d, ILL: %d, ILP: %d\n", packetSize, fILL, fILP);
#endif
  fFrameIndex = 0; // initially

  // Next, there's a "Payload Table of Contents" (one byte per entry):
  unsigned numFramesPresent = 0, numNonEmptyFramesPresent = 0;
  unsigned tocStartIndex = resultSpecialHeaderSize;
  Boolean F;
  do {
    if (resultSpecialHeaderSize >= packetSize) return False;
    unsigned char const tocByte = headerStart[resultSpecialHeaderSize++];
    F = (tocByte&0x80) != 0;
    unsigned char const FT = (tocByte&0x78) >> 3;
#ifdef DEBUG
    unsigned char Q = (tocByte&0x04)>>2;
    fprintf(stderr, "\tTOC entry: F %d, FT %d, Q %d\n", F, FT, Q);
#endif
    ++numFramesPresent;
    if (FT != FT_SPEECH_LOST && FT != FT_NO_DATA) ++numNonEmptyFramesPresent;
  } while (F);
#ifdef DEBUG
  fprintf(stderr, "TOC contains %d entries (%d non-empty)\n", numFramesPresent, numNonEmptyFramesPresent);
#endif

  // Now that we know the size of the TOC, fill in our copy:
  if (numFramesPresent > fTOCSize) {
    delete[] fTOC;
    fTOC = new unsigned char[numFramesPresent];
  }
  fTOCSize = numFramesPresent;
  for (unsigned i = 0; i < fTOCSize; ++i) {
    unsigned char const tocByte = headerStart[tocStartIndex + i];
    fTOC[i] = tocByte&0x7C; // clear everything except the F and Q fields
  }

  if (fCRCsArePresent) {
    // 'numNonEmptyFramesPresent' CRC bytes will follow.
    // Note: we currently don't check the CRCs for validity #####
    resultSpecialHeaderSize += numNonEmptyFramesPresent;
#ifdef DEBUG
    fprintf(stderr, "Ignoring %d following CRC bytes\n", numNonEmptyFramesPresent);
#endif
    if (resultSpecialHeaderSize > packetSize) return False;
  }
#ifdef DEBUG
  fprintf(stderr, "Total special header size: %d\n", resultSpecialHeaderSize);
#endif

  return True;
} 

char const* RawAMRRTPSource::MIMEtype() const {
  return fIsWideband ? "audio/AMR-WB" : "audio/AMR-WB";
}

Boolean RawAMRRTPSource::hasBeenSynchronizedUsingRTCP() {
  // Don't report ourselves as being synchronized until we've received
  // at least a complete interleave cycle of synchronized packets.
  // This ensures that the receiver is currently getting a frame from
  // a packet that was synchronized.
  if (fNumSuccessiveSyncedPackets > (unsigned)(fILL+1)) {
    fNumSuccessiveSyncedPackets = fILL + 2; // prevents overflow
    return True;
  }
  return False;
}


///// AMRBufferedPacket and AMRBufferedPacketFactory implementation

AMRBufferedPacket::AMRBufferedPacket(RawAMRRTPSource& ourSource)
  : fOurSource(ourSource) {
}

AMRBufferedPacket::~AMRBufferedPacket() {
}

// The mapping from the "FT" field to frame size.
// Values of 65535 are invalid.
#define FT_INVALID 65535
static unsigned short frameBytesFromFT[16] = {
  12, 13, 15, 17,
  19, 20, 26, 31,
  5, FT_INVALID, FT_INVALID, FT_INVALID,
  FT_INVALID, FT_INVALID, FT_INVALID, 0
};
static unsigned short frameBytesFromFTWideband[16] = {
  17, 23, 32, 36,
  40, 46, 50, 58,
  60, 5, FT_INVALID, FT_INVALID,
  FT_INVALID, FT_INVALID, 0, 0
};

unsigned AMRBufferedPacket::
  nextEnclosedFrameSize(unsigned char*& framePtr, unsigned dataSize) {
  if (dataSize == 0) return 0; // sanity check

  // The size of the AMR frame is determined by the corresponding 'FT' value
  // in the packet's Table of Contents.
  unsigned const tocIndex = fOurSource.frameIndex();
  if (tocIndex >= fOurSource.TOCSize()) return 0; // sanity check

  unsigned char const tocByte = fOurSource.TOC()[tocIndex];
  unsigned char const FT = (tocByte&0x78) >> 3;
  // ASSERT: FT < 16
  unsigned short frameSize
    = fOurSource.isWideband() ? frameBytesFromFTWideband[FT] : frameBytesFromFT[FT];
  if (frameSize == FT_INVALID) {