void WAVAudioFileServerMediaSubsession
::setStreamSourceScale(FramedSource* inputSource, float scale) {
int iScale = (int)scale;
WAVAudioFileSource* wavSource;
if (fBitsPerSample > 8) {
// "inputSource" is a filter; its input source is the original WAV file source:
wavSource = (WAVAudioFileSource*)(((FramedFilter*)inputSource)->inputSource());
} else {
// "inputSource" is the original WAV file source:
wavSource = (WAVAudioFileSource*)inputSource;
}
wavSource->setScaleFactor(iScale);
}
void WAVAudioFileServerMediaSubsession
::seekStreamSource(FramedSource* inputSource, float seekNPT) {
WAVAudioFileSource* wavSource;
if (fBitsPerSample == 16) {
// "inputSource" is a filter; its input source is the original WAV file source:
wavSource = (WAVAudioFileSource*)(((FramedFilter*)inputSource)->inputSource());
} else {
// "inputSource" is the original WAV file source:
wavSource = (WAVAudioFileSource*)inputSource;
}
unsigned seekSampleNumber = (unsigned)(seekNPT*fSamplingFrequency);
unsigned seekByteNumber = (seekSampleNumber*fNumChannels*fBitsPerSample)/8;
wavSource->seekToPCMByte(seekByteNumber);
}
void WAVAudioFileServerMediaSubsession
::setStreamSourceDuration(FramedSource* inputSource, double streamDuration, u_int64_t& numBytes) {
WAVAudioFileSource* wavSource;
if (fBitsPerSample > 8) {
// "inputSource" is a filter; its input source is the original WAV file source:
wavSource = (WAVAudioFileSource*)(((FramedFilter*)inputSource)->inputSource());
} else {
// "inputSource" is the original WAV file source:
wavSource = (WAVAudioFileSource*)inputSource;
}
unsigned numDurationSamples = (unsigned)(streamDuration*fSamplingFrequency);
unsigned numDurationBytes = numDurationSamples*((fNumChannels*fBitsPerSample)/8);
numBytes = (u_int64_t)numDurationBytes;
wavSource->limitNumBytesToStream(numDurationBytes);
}
void WAVAudioFileServerMediaSubsession
::seekStreamSource(FramedSource* inputSource, double& seekNPT, double streamDuration, u_int64_t& numBytes) {
WAVAudioFileSource* wavSource;
if (fBitsPerSample > 8) {
// "inputSource" is a filter; its input source is the original WAV file source:
wavSource = (WAVAudioFileSource*)(((FramedFilter*)inputSource)->inputSource());
} else {
// "inputSource" is the original WAV file source:
wavSource = (WAVAudioFileSource*)inputSource;
}
unsigned seekSampleNumber = (unsigned)(seekNPT*fSamplingFrequency);
unsigned seekByteNumber = seekSampleNumber*((fNumChannels*fBitsPerSample)/8);
wavSource->seekToPCMByte(seekByteNumber);
setStreamSourceDuration(inputSource, streamDuration, numBytes);
}
WAVAudioFileSource*
WAVAudioFileSource::createNew(UsageEnvironment& env, char const* fileName) {
do {
FILE* fid = OpenInputFile(env, fileName);
if (fid == NULL) break;
WAVAudioFileSource* newSource = new WAVAudioFileSource(env, fid);
if (newSource != NULL && newSource->bitsPerSample() == 0) {
// The WAV file header was apparently invalid.
Medium::close(newSource);
break;
}
newSource->fFileSize = (unsigned)GetFileSize(fileName, fid);
return newSource;
} while (0);
return NULL;
}
FramedSource* WAVAudioFileServerMediaSubsession
::createNewStreamSource(unsigned /*clientSessionId*/, unsigned& estBitrate) {
FramedSource* resultSource = NULL;
do {
WAVAudioFileSource* wavSource
= WAVAudioFileSource::createNew(envir(), fFileName);
if (wavSource == NULL) break;
// Get attributes of the audio source:
fAudioFormat = wavSource->getAudioFormat();
fBitsPerSample = wavSource->bitsPerSample();
if (fBitsPerSample != 8 && fBitsPerSample != 16) {
envir() << "The input file contains " << fBitsPerSample
<< " bit-per-sample audio, which we don't handle\n";
break;
}
fSamplingFrequency = wavSource->samplingFrequency();
fNumChannels = wavSource->numChannels();
unsigned bitsPerSecond
= fSamplingFrequency*fBitsPerSample*fNumChannels;
fFileDuration = (float)((8.0*wavSource->numPCMBytes())
/(fSamplingFrequency*fNumChannels*fBitsPerSample));
// Add in any filter necessary to transform the data prior to streaming:
if (fBitsPerSample == 16) {
// Note that samples in the WAV audio file are in little-endian order.
if (fConvertToULaw) {
// Add a filter that converts from raw 16-bit PCM audio
// to 8-bit u-law audio:
resultSource
= uLawFromPCMAudioSource::createNew(envir(), wavSource, 1/*little-endian*/);
bitsPerSecond /= 2;
} else {
// Add a filter that converts from little-endian to network (big-endian) order:
resultSource = EndianSwap16::createNew(envir(), wavSource);
}
} else { // fBitsPerSample == 8
// Don't do any transformation; send the 8-bit PCM data 'as is':
resultSource = wavSource;
}
estBitrate = (bitsPerSecond+500)/1000; // kbps
return resultSource;
} while (0);
// An error occurred:
Medium::close(resultSource);
return NULL;
}
void play() {
// Open the file as a 'WAV' file:
WAVAudioFileSource* pcmSource
= WAVAudioFileSource::createNew(*env, inputFileName);
if (pcmSource == NULL) {
*env << "Unable to open file \"" << inputFileName
<< "\" as a WAV audio file source: "
<< env->getResultMsg() << "\n";
exit(1);
}
// Get attributes of the audio source:
unsigned char const bitsPerSample = pcmSource->bitsPerSample();
if (bitsPerSample != 8 && bitsPerSample != 16) {
*env << "The input file contains " << bitsPerSample
<< " bit-per-sample audio, which we don't handle\n";
exit(1);
}
sessionState.source = pcmSource;
unsigned const samplingFrequency = pcmSource->samplingFrequency();
unsigned char const numChannels = pcmSource->numChannels();
unsigned bitsPerSecond
= samplingFrequency*bitsPerSample*numChannels;
*env << "Audio source parameters:\n\t" << samplingFrequency << " Hz, ";
*env << bitsPerSample << " bits-per-sample, ";
*env << numChannels << " channels => ";
*env << bitsPerSecond << " bits-per-second\n";
// Add in any filter necessary to transform the data prior to streaming.
// (This is where any audio compression would get added.)
char const* mimeType;
unsigned char payloadFormatCode;
if (bitsPerSample == 16) {
#ifdef CONVERT_TO_ULAW
// Add a filter that converts from raw 16-bit PCM audio (in little-endian order)
// to 8-bit u-law audio:
sessionState.source
= uLawFromPCMAudioSource::createNew(*env, pcmSource, 1/*little-endian*/);
if (sessionState.source == NULL) {
*env << "Unable to create a u-law filter from the PCM audio source: "
<< env->getResultMsg() << "\n";
exit(1);
}
bitsPerSecond /= 2;
mimeType = "PCMU";
if (samplingFrequency == 8000 && numChannels == 1) {
payloadFormatCode = 0; // a static RTP payload type
} else {
payloadFormatCode = 96; // a dynamic RTP payload type
}
*env << "Converting to 8-bit u-law audio for streaming => "
<< bitsPerSecond << " bits-per-second\n";
#else
// The 16-bit samples in WAV files are in little-endian order.
// Add a filter that converts them to network (i.e., big-endian) order:
sessionState.source = EndianSwap16::createNew(*env, pcmSource);
if (sessionState.source == NULL) {
*env << "Unable to create a little->bit-endian order filter from the PCM audio source: "
<< env->getResultMsg() << "\n";
exit(1);
}
mimeType = "L16";
if (samplingFrequency == 44100 && numChannels == 2) {
payloadFormatCode = 10; // a static RTP payload type
} else if (samplingFrequency == 44100 && numChannels == 1) {
payloadFormatCode = 11; // a static RTP payload type
} else {
payloadFormatCode = 96; // a dynamic RTP payload type
}
*env << "Converting to network byte order for streaming\n";
#endif
} else { // bitsPerSample == 8
// Don't do any transformation; send the 8-bit PCM data 'as is':
mimeType = "L8";
payloadFormatCode = 96; // a dynamic RTP payload type
}
// Create 'groupsocks' for RTP and RTCP:
struct in_addr destinationAddress;
destinationAddress.s_addr = chooseRandomIPv4SSMAddress(*env);
// Note: This is a multicast address. If you wish instead to stream
// using unicast, then you should use the "testOnDemandRTSPServer"
// test program - not this test program - as a model.
const unsigned short rtpPortNum = 2222;
const unsigned short rtcpPortNum = rtpPortNum+1;
const unsigned char ttl = 255;
const Port rtpPort(rtpPortNum);
const Port rtcpPort(rtcpPortNum);
sessionState.rtpGroupsock
= new Groupsock(*env, destinationAddress, rtpPort, ttl);
sessionState.rtpGroupsock->multicastSendOnly(); // we're a SSM source
sessionState.rtcpGroupsock
= new Groupsock(*env, destinationAddress, rtcpPort, ttl);
sessionState.rtcpGroupsock->multicastSendOnly(); // we're a SSM source
// Create an appropriate audio RTP sink (using "SimpleRTPSink")
// from the RTP 'groupsock':
sessionState.sink
= SimpleRTPSink::createNew(*env, sessionState.rtpGroupsock,
payloadFormatCode, samplingFrequency,
"audio", mimeType, numChannels);
// Create (and start) a 'RTCP instance' for this RTP sink:
const unsigned estimatedSessionBandwidth = bitsPerSecond/1000;
// in kbps; for RTCP b/w share
const unsigned maxCNAMElen = 100;
unsigned char CNAME[maxCNAMElen+1];
gethostname((char*)CNAME, maxCNAMElen);
CNAME[maxCNAMElen] = '\0'; // just in case
sessionState.rtcpInstance
= RTCPInstance::createNew(*env, sessionState.rtcpGroupsock,
estimatedSessionBandwidth, CNAME,
sessionState.sink, NULL /* we're a server */,
True /* we're a SSM source*/);
// Note: This starts RTCP running automatically
// Create and start a RTSP server to serve this stream:
sessionState.rtspServer = RTSPServer::createNew(*env, 8554);
if (sessionState.rtspServer == NULL) {
*env << "Failed to create RTSP server: " << env->getResultMsg() << "\n";
exit(1);
}
ServerMediaSession* sms
= ServerMediaSession::createNew(*env, "testStream", inputFileName,
"Session streamed by \"testWAVAudiotreamer\"", True/*SSM*/);
sms->addSubsession(PassiveServerMediaSubsession::createNew(*sessionState.sink, sessionState.rtcpInstance));
sessionState.rtspServer->addServerMediaSession(sms);
char* url = sessionState.rtspServer->rtspURL(sms);
*env << "Play this stream using the URL \"" << url << "\"\n";
delete[] url;
// Finally, start the streaming:
*env << "Beginning streaming...\n";
sessionState.sink->startPlaying(*sessionState.source, afterPlaying, NULL);
}