/********** 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 .) 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA **********/ // "liveMedia" // Copyright (c) 1996-2015 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, 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); } 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 ::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); } 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(); // We handle only 4,8,16,20,24 bits-per-sample audio: if (fBitsPerSample%4 != 0 || fBitsPerSample < 4 || fBitsPerSample > 24 || fBitsPerSample == 12) { 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: resultSource = wavSource; // by default if (fAudioFormat == WA_PCM) { 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 if (fBitsPerSample == 20 || fBitsPerSample == 24) { // Add a filter that converts from little-endian to network (big-endian) order: resultSource = EndianSwap24::createNew(envir(), 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 const* mimeType; unsigned char payloadFormatCode = rtpPayloadTypeIfDynamic; // by default, unless a static RTP payload type can be used if (fAudioFormat == WA_PCM) { if (fBitsPerSample == 16) { if (fConvertToULaw) { mimeType = "PCMU"; if (fSamplingFrequency == 8000 && fNumChannels == 1) { payloadFormatCode = 0; // a static RTP payload type } } 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 if (fBitsPerSample == 20) { mimeType = "L20"; } else if (fBitsPerSample == 24) { mimeType = "L24"; } else { // fBitsPerSample == 8 (we assume that fBitsPerSample == 4 is only for WA_IMA_ADPCM) mimeType = "L8"; } } else if (fAudioFormat == WA_PCMU) { mimeType = "PCMU"; if (fSamplingFrequency == 8000 && fNumChannels == 1) { payloadFormatCode = 0; // a static RTP payload type } } else if (fAudioFormat == WA_PCMA) { mimeType = "PCMA"; if (fSamplingFrequency == 8000 && fNumChannels == 1) { payloadFormatCode = 8; // a static RTP payload type } } else if (fAudioFormat == WA_IMA_ADPCM) { mimeType = "DVI4"; // Use a static payload type, if one is defined: if (fNumChannels == 1) { if (fSamplingFrequency == 8000) { payloadFormatCode = 5; // a static RTP payload type } else if (fSamplingFrequency == 16000) { payloadFormatCode = 6; // a static RTP payload type } else if (fSamplingFrequency == 11025) { payloadFormatCode = 16; // a static RTP payload type } else if (fSamplingFrequency == 22050) { payloadFormatCode = 17; // a static RTP payload type } } } else { //unknown format break; } 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; }