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/**********
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) 1996-2005 Live Networks, Inc. All rights reserved.
// A 'ServerMediaSubsession' object that creates new, unicast, "RTPSink"s
// on demand, from an WAV audio file.
// Implementation
#include "WAVAudioFileServerMediaSubsession.hh"
#include "WAVAudioFileSource.hh"
#include "uLawAudioFilter.hh"
#include "SimpleRTPSink.hh"
WAVAudioFileServerMediaSubsession* WAVAudioFileServerMediaSubsession
::createNew(UsageEnvironment& env, char const* fileName, Boolean reuseFirstSource,
Boolean convertToULaw) {
return new WAVAudioFileServerMediaSubsession(env, fileName,
reuseFirstSource, convertToULaw);
}
WAVAudioFileServerMediaSubsession
::WAVAudioFileServerMediaSubsession(UsageEnvironment& env, char const* fileName,
Boolean reuseFirstSource, Boolean convertToULaw)
: FileServerMediaSubsession(env, fileName, reuseFirstSource),
fConvertToULaw(convertToULaw) {
}
WAVAudioFileServerMediaSubsession
::~WAVAudioFileServerMediaSubsession() {
}
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
::setStreamSourceScale(FramedSource* inputSource, float scale) {
int iScale = (int)scale;
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;
}
wavSource->setScaleFactor(iScale);
}
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:
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;
}
RTPSink* WAVAudioFileServerMediaSubsession
::createNewRTPSink(Groupsock* rtpGroupsock,
unsigned char rtpPayloadTypeIfDynamic,
FramedSource* /*inputSource*/) {
do {
char* mimeType;
unsigned char payloadFormatCode;
if (fBitsPerSample == 16) {
if (fConvertToULaw) {
mimeType = "PCMU";
if (fSamplingFrequency == 8000 && fNumChannels == 1) {
payloadFormatCode = 0; // a static RTP payload type
} else {
payloadFormatCode = rtpPayloadTypeIfDynamic;
}
} else {
mimeType = "L16";
if (fSamplingFrequency == 44100 && fNumChannels == 2) {
payloadFormatCode = 10; // a static RTP payload type
} else if (fSamplingFrequency == 44100 && fNumChannels == 1) {
payloadFormatCode = 11; // a static RTP payload type
} else {
payloadFormatCode = rtpPayloadTypeIfDynamic;
}
}
} else { // fBitsPerSample == 8
mimeType = "L8";
payloadFormatCode = rtpPayloadTypeIfDynamic;
}
return SimpleRTPSink::createNew(envir(), rtpGroupsock,
payloadFormatCode, fSamplingFrequency,
"audio", mimeType, fNumChannels);
} while (0);
// An error occurred:
return NULL;
}
void WAVAudioFileServerMediaSubsession::testScaleFactor(float& scale) {
if (fFileDuration <= 0.0) {
// The file is non-seekable, so is probably a live input source.
// We don't support scale factors other than 1
scale = 1;
} else {
// We support any integral scale, other than 0
int iScale = scale < 0.0 ? (int)(scale - 0.5) : (int)(scale + 0.5); // round
if (iScale == 0) iScale = 1;
scale = (float)iScale;
}
}
float WAVAudioFileServerMediaSubsession::duration() const {
return fFileDuration;
}
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