void FileSource::visit(FileSink& v)
{
#if 0
// only works if target is a pipe.
loff_t offset = offset_;
result_ = ::splice(handle(), &offset, v.handle(), NULL, count_,
SPLICE_F_MOVE | SPLICE_F_NONBLOCK | SPLICE_F_MORE);
if (result_ > 0) {
offset_ = offset;
count_ -= result_;
}
#endif
char buf[8 * 1024];
result_ = pread(handle(), buf, sizeof(buf), offset_);
if (result_ <= 0)
return;
result_ = v.write(buf, result_);
if (result_ <= 0)
return;
offset_ += result_;
count_ -= result_;
}
void FileSink::afterGettingFrame(void* clientData, unsigned frameSize,
unsigned numTruncatedBytes,
struct timeval presentationTime,
unsigned /*durationInMicroseconds*/) {
FileSink* sink = (FileSink*)clientData;
sink->afterGettingFrame(frameSize, numTruncatedBytes, presentationTime);
}
void setupStreams() {
static MediaSubsessionIterator* setupIter = NULL;
if (setupIter == NULL) setupIter = new MediaSubsessionIterator(*session);
while ((subsession = setupIter->next()) != NULL) {
// We have another subsession left to set up:
if (subsession->clientPortNum() == 0) continue; // port # was not set
setupSubsession(subsession, streamUsingTCP, continueAfterSETUP);
return;
}
// We're done setting up subsessions.
delete setupIter;
if (!madeProgress) shutdown();
// Create output files:
if (createReceivers) {
#if 0 /*wayde*/
if (outputQuickTimeFile) {
// Create a "QuickTimeFileSink", to write to 'stdout':
qtOut = QuickTimeFileSink::createNew(*env, *session, "stdout",
fileSinkBufferSize,
movieWidth, movieHeight,
movieFPS,
packetLossCompensate,
syncStreams,
generateHintTracks,
generateMP4Format);
if (qtOut == NULL) {
*env << "Failed to create QuickTime file sink for stdout: " << env->getResultMsg();
shutdown();
}
qtOut->startPlaying(sessionAfterPlaying, NULL);
} else if (outputAVIFile) {
// Create an "AVIFileSink", to write to 'stdout':
aviOut = AVIFileSink::createNew(*env, *session, "stdout",
fileSinkBufferSize,
movieWidth, movieHeight,
movieFPS,
packetLossCompensate);
if (aviOut == NULL) {
*env << "Failed to create AVI file sink for stdout: " << env->getResultMsg();
shutdown();
}
aviOut->startPlaying(sessionAfterPlaying, NULL);
#endif /*wayde*/
} else {
// Create and start "FileSink"s for each subsession:
madeProgress = False;
MediaSubsessionIterator iter(*session);
while ((subsession = iter.next()) != NULL) {
if (subsession->readSource() == NULL) continue; // was not initiated
// Create an output file for each desired stream:
char outFileName[1000];
if (singleMedium == NULL) {
// Output file name is
// "<filename-prefix><medium_name>-<codec_name>-<counter>"
static unsigned streamCounter = 0;
snprintf(outFileName, sizeof outFileName, "%s%s-%s-%d",
fileNamePrefix, subsession->mediumName(),
subsession->codecName(), ++streamCounter);
} else {
sprintf(outFileName, "stdout");
}
FileSink* fileSink;
if (strcmp(subsession->mediumName(), "audio") == 0 &&
(strcmp(subsession->codecName(), "AMR") == 0 ||
strcmp(subsession->codecName(), "AMR-WB") == 0)) {
// For AMR audio streams, we use a special sink that inserts AMR frame hdrs:
fileSink = AMRAudioFileSink::createNew(*env, outFileName,
fileSinkBufferSize, oneFilePerFrame);
} else if (strcmp(subsession->mediumName(), "video") == 0 &&
(strcmp(subsession->codecName(), "H264") == 0)) {
// For H.264 video stream, we use a special sink that insert start_codes:
fileSink = H264VideoFileSink::createNew(*env, outFileName,
fileSinkBufferSize, oneFilePerFrame);
} else {
// Normal case:
fileSink = FileSink::createNew(*env, outFileName,
fileSinkBufferSize, oneFilePerFrame);
}
subsession->sink = fileSink;
if (subsession->sink == NULL) {
*env << "Failed to create FileSink for \"" << outFileName
<< "\": " << env->getResultMsg() << "\n";
} else {
if (singleMedium == NULL) {
*env << "Created output file: \"" << outFileName << "\"\n";
} else {
*env << "Outputting data from the \"" << subsession->mediumName()
<< "/" << subsession->codecName()
<< "\" subsession to 'stdout'\n";
}
if (strcmp(subsession->mediumName(), "video") == 0 &&
strcmp(subsession->codecName(), "MP4V-ES") == 0 &&
subsession->fmtp_config() != NULL) {
// For MPEG-4 video RTP streams, the 'config' information
// from the SDP description contains useful VOL etc. headers.
// Insert this data at the front of the output file:
unsigned configLen;
unsigned char* configData
= parseGeneralConfigStr(subsession->fmtp_config(), configLen);
struct timeval timeNow;
gettimeofday(&timeNow, NULL);
fileSink->addData(configData, configLen, timeNow);
delete[] configData;
}
subsession->sink->startPlaying(*(subsession->readSource()),
subsessionAfterPlaying,
subsession);
// Also set a handler to be called if a RTCP "BYE" arrives
// for this subsession:
if (subsession->rtcpInstance() != NULL) {
subsession->rtcpInstance()->setByeHandler(subsessionByeHandler,
subsession);
}
madeProgress = True;
}
}
if (!madeProgress) shutdown();
}
}
// Finally, start playing each subsession, to start the data flow:
if (duration == 0) {
if (scale > 0) duration = session->playEndTime() - initialSeekTime; // use SDP end time
else if (scale < 0) duration = initialSeekTime;
}
if (duration < 0) duration = 0.0;
endTime = initialSeekTime;
if (scale > 0) {
if (duration <= 0) endTime = -1.0f;
else endTime = initialSeekTime + duration;
} else {
endTime = initialSeekTime - duration;
if (endTime < 0) endTime = 0.0f;
}
startPlayingSession(session, initialSeekTime, endTime, scale, continueAfterPLAY);
}
/*------------------------------------------------------------------------------
* Look for the FileCast stream outputs in the config file
*----------------------------------------------------------------------------*/
void
DarkIce :: configFileCast ( const Config & config )
throw ( Exception )
{
// look for FileCast encoder output streams,
// sections [file-0], [file-1], ...
char stream[] = "file- ";
size_t streamLen = Util::strLen( stream);
unsigned int u;
for ( u = noAudioOuts; u < maxOutput; ++u ) {
const ConfigSection * cs;
// ugly hack to change the section name to "stream0", "stream1", etc.
stream[streamLen-1] = '0' + (u - noAudioOuts);
if ( !(cs = config.get( stream)) ) {
break;
}
const char * str;
const char * format = 0;
AudioEncoder::BitrateMode bitrateMode;
unsigned int bitrate = 0;
double quality = 0.0;
const char * targetFileName = 0;
unsigned int sampleRate = 0;
int lowpass = 0;
int highpass = 0;
bool fileAddDate = false;
const char * fileDateFormat = 0;
format = cs->getForSure( "format", " missing in section ", stream);
if ( !Util::strEq( format, "vorbis")
&& !Util::strEq( format, "mp3")
&& !Util::strEq( format, "mp2")
&& !Util::strEq( format, "aac")
&& !Util::strEq( format, "aacp") ) {
throw Exception( __FILE__, __LINE__,
"unsupported stream format: ", format);
}
str = cs->getForSure("bitrate", " missing in section ", stream);
bitrate = Util::strToL( str);
targetFileName = cs->getForSure( "fileName",
" missing in section ",
stream);
str = cs->get( "fileAddDate");
fileAddDate = str ? (Util::strEq( str, "yes") ? true : false) : false;
fileDateFormat = cs->get( "fileDateFormat");
str = cs->get( "sampleRate");
sampleRate = str ? Util::strToL( str) : dsp->getSampleRate();
str = cs->get( "bitrate");
bitrate = str ? Util::strToL( str) : 0;
str = cs->get( "quality");
quality = str ? Util::strToD( str) : 0.0;
str = cs->getForSure( "bitrateMode",
" not specified in section ",
stream);
if ( Util::strEq( str, "cbr") ) {
bitrateMode = AudioEncoder::cbr;
if ( bitrate == 0 ) {
throw Exception( __FILE__, __LINE__,
"bitrate not specified for CBR encoding");
}
} else if ( Util::strEq( str, "abr") ) {
bitrateMode = AudioEncoder::abr;
if ( bitrate == 0 ) {
throw Exception( __FILE__, __LINE__,
"bitrate not specified for ABR encoding");
}
} else if ( Util::strEq( str, "vbr") ) {
bitrateMode = AudioEncoder::vbr;
if ( cs->get( "quality" ) == 0 ) {
throw Exception( __FILE__, __LINE__,
"quality not specified for VBR encoding");
}
} else {
throw Exception( __FILE__, __LINE__,
"invalid bitrate mode: ", str);
}
if (Util::strEq(format, "aac") && bitrateMode != AudioEncoder::abr) {
throw Exception(__FILE__, __LINE__,
"currently the AAC format only supports "
"average bitrate mode");
}
if (Util::strEq(format, "aacp") && bitrateMode != AudioEncoder::cbr) {
throw Exception(__FILE__, __LINE__,
"currently the AAC+ format only supports "
"constant bitrate mode");
}
str = cs->get( "lowpass");
lowpass = str ? Util::strToL( str) : 0;
str = cs->get( "highpass");
highpass = str ? Util::strToL( str) : 0;
// go on and create the things
// the underlying file
if ( fileAddDate ) {
if (fileDateFormat == 0) {
targetFileName = Util::fileAddDate( targetFileName);
}
else {
targetFileName = Util::fileAddDate( targetFileName,
fileDateFormat );
}
}
FileSink * targetFile = new FileSink( stream, targetFileName);
if ( !targetFile->exists() ) {
if ( !targetFile->create() ) {
throw Exception( __FILE__, __LINE__,
"can't create output file", targetFileName);
}
}
// streaming related stuff
audioOuts[u].socket = 0;
audioOuts[u].server = new FileCast( targetFile );
if ( Util::strEq( format, "mp3") ) {
#ifndef HAVE_LAME_LIB
throw Exception( __FILE__, __LINE__,
"DarkIce not compiled with lame support, "
"thus can't create mp3 stream: ",
stream);
#else
audioOuts[u].encoder = new LameLibEncoder(
audioOuts[u].server.get(),
dsp.get(),
bitrateMode,
bitrate,
quality,
sampleRate,
dsp->getChannel(),
lowpass,
highpass );
#endif // HAVE_TWOLAME_LIB
} else if ( Util::strEq( format, "mp2") ) {
#ifndef HAVE_TWOLAME_LIB
throw Exception( __FILE__, __LINE__,
"DarkIce not compiled with TwoLAME support, "
"thus can't create MPEG Audio Layer 2 stream: ",
stream);
#else
audioOuts[u].encoder = new TwoLameLibEncoder(
audioOuts[u].server.get(),
dsp.get(),
bitrateMode,
bitrate,
sampleRate,
dsp->getChannel() );
#endif // HAVE_TWOLAME_LIB
} else if ( Util::strEq( format, "vorbis") ) {
#ifndef HAVE_VORBIS_LIB
throw Exception( __FILE__, __LINE__,
"DarkIce not compiled with Ogg Vorbis support, "
"thus can't Ogg Vorbis stream: ",
stream);
#else
audioOuts[u].encoder = new VorbisLibEncoder(
audioOuts[u].server.get(),
dsp.get(),
bitrateMode,
bitrate,
quality,
dsp->getSampleRate(),
dsp->getChannel() );
#endif // HAVE_VORBIS_LIB
} else if ( Util::strEq( format, "aac") ) {
#ifndef HAVE_FAAC_LIB
throw Exception( __FILE__, __LINE__,
"DarkIce not compiled with AAC support, "
"thus can't aac stream: ",
stream);
#else
audioOuts[u].encoder = new FaacEncoder(
audioOuts[u].server.get(),
dsp.get(),
bitrateMode,
bitrate,
quality,
sampleRate,
dsp->getChannel());
#endif // HAVE_FAAC_LIB
} else if ( Util::strEq( format, "aacp") ) {
#ifndef HAVE_AACPLUS_LIB
throw Exception( __FILE__, __LINE__,
"DarkIce not compiled with AAC+ support, "
"thus can't aacplus stream: ",
stream);
#else
audioOuts[u].encoder = new aacPlusEncoder(
audioOuts[u].server.get(),
dsp.get(),
bitrateMode,
bitrate,
quality,
sampleRate,
dsp->getChannel());
#endif // HAVE_AACPLUS_LIB
} else {
throw Exception( __FILE__, __LINE__,
"Illegal stream format: ", format);
}
encConnector->attach( audioOuts[u].encoder.get());
}
noAudioOuts += u;
}
/*------------------------------------------------------------------------------
* Look for the ShoutCast stream outputs in the config file
*----------------------------------------------------------------------------*/
void
DarkIce :: configShoutCast ( const Config & config,
unsigned int bufferSecs )
throw ( Exception )
{
// look for Shoutcast encoder output streams,
// sections [shoutcast-0], [shoutcast-1], ...
char stream[] = "shoutcast- ";
size_t streamLen = Util::strLen( stream);
unsigned int u;
for ( u = noAudioOuts; u < maxOutput; ++u ) {
const ConfigSection * cs;
// ugly hack to change the section name to "stream0", "stream1", etc.
stream[streamLen-1] = '0' + (u - noAudioOuts);
if ( !(cs = config.get( stream)) ) {
break;
}
#ifndef HAVE_LAME_LIB
throw Exception( __FILE__, __LINE__,
"DarkIce not compiled with lame support, "
"thus can't connect to ShoutCast, stream: ",
stream);
#else
const char * str;
unsigned int sampleRate = 0;
unsigned int channel = 0;
AudioEncoder::BitrateMode bitrateMode;
unsigned int bitrate = 0;
double quality = 0.0;
const char * server = 0;
unsigned int port = 0;
const char * password = 0;
const char * name = 0;
const char * url = 0;
const char * genre = 0;
bool isPublic = false;
const char * mountPoint = 0;
int lowpass = 0;
int highpass = 0;
const char * irc = 0;
const char * aim = 0;
const char * icq = 0;
const char * localDumpName = 0;
FileSink * localDumpFile = 0;
bool fileAddDate = false;
const char * fileDateFormat = 0;
str = cs->get( "sampleRate");
sampleRate = str ? Util::strToL( str) : dsp->getSampleRate();
str = cs->get( "channel");
channel = str ? Util::strToL( str) : dsp->getChannel();
str = cs->get( "bitrate");
bitrate = str ? Util::strToL( str) : 0;
str = cs->get( "quality");
quality = str ? Util::strToD( str) : 0.0;
str = cs->getForSure( "bitrateMode",
" not specified in section ",
stream);
if ( Util::strEq( str, "cbr") ) {
bitrateMode = AudioEncoder::cbr;
if ( bitrate == 0 ) {
throw Exception( __FILE__, __LINE__,
"bitrate not specified for CBR encoding");
}
if ( cs->get( "quality" ) == 0 ) {
throw Exception( __FILE__, __LINE__,
"quality not specified for CBR encoding");
}
} else if ( Util::strEq( str, "abr") ) {
bitrateMode = AudioEncoder::abr;
if ( bitrate == 0 ) {
throw Exception( __FILE__, __LINE__,
"bitrate not specified for ABR encoding");
}
} else if ( Util::strEq( str, "vbr") ) {
bitrateMode = AudioEncoder::vbr;
if ( cs->get( "quality" ) == 0 ) {
throw Exception( __FILE__, __LINE__,
"quality not specified for VBR encoding");
}
} else {
throw Exception( __FILE__, __LINE__,
"invalid bitrate mode: ", str);
}
server = cs->getForSure( "server", " missing in section ", stream);
str = cs->getForSure( "port", " missing in section ", stream);
port = Util::strToL( str);
password = cs->getForSure("password"," missing in section ",stream);
name = cs->get( "name");
mountPoint = cs->get( "mountPoint" );
url = cs->get( "url");
genre = cs->get( "genre");
str = cs->get( "public");
isPublic = str ? (Util::strEq( str, "yes") ? true : false) : false;
str = cs->get( "lowpass");
lowpass = str ? Util::strToL( str) : 0;
str = cs->get( "highpass");
highpass = str ? Util::strToL( str) : 0;
irc = cs->get( "irc");
aim = cs->get( "aim");
icq = cs->get( "icq");
str = cs->get("fileAddDate");
fileAddDate = str ? (Util::strEq( str, "yes") ? true : false) : false;
fileDateFormat = cs->get( "fileDateFormat");
localDumpName = cs->get( "localDumpFile");
// go on and create the things
// check for and create the local dump file if needed
if ( localDumpName != 0 ) {
if ( fileAddDate ) {
if (fileDateFormat == 0) {
localDumpName = Util::fileAddDate(localDumpName);
}
else {
localDumpName = Util::fileAddDate( localDumpName,
fileDateFormat );
}
}
localDumpFile = new FileSink( stream, localDumpName);
if ( !localDumpFile->exists() ) {
if ( !localDumpFile->create() ) {
reportEvent( 1, "can't create local dump file",
localDumpName);
localDumpFile = 0;
}
}
if ( fileAddDate ) {
delete[] localDumpFile;
}
}
// streaming related stuff
audioOuts[u].socket = new TcpSocket( server, port);
audioOuts[u].server = new ShoutCast( audioOuts[u].socket.get(),
password,
mountPoint,
bitrate,
name,
url,
genre,
isPublic,
irc,
aim,
icq,
localDumpFile,
bufferSecs );
audioOuts[u].encoder = new LameLibEncoder( audioOuts[u].server.get(),
dsp.get(),
bitrateMode,
bitrate,
quality,
sampleRate,
channel,
lowpass,
highpass );
encConnector->attach( audioOuts[u].encoder.get());
#endif // HAVE_LAME_LIB
}
noAudioOuts += u;
}
/*------------------------------------------------------------------------------
* Look for the IceCast2 stream outputs in the config file
*----------------------------------------------------------------------------*/
void
DarkIce :: configIceCast2 ( const Config & config,
unsigned int bufferSecs )
throw ( Exception )
{
// look for IceCast2 encoder output streams,
// sections [icecast2-0], [icecast2-1], ...
char stream[] = "icecast2- ";
size_t streamLen = Util::strLen( stream);
unsigned int u;
for ( u = noAudioOuts; u < maxOutput; ++u ) {
const ConfigSection * cs;
// ugly hack to change the section name to "stream0", "stream1", etc.
stream[streamLen-1] = '0' + (u - noAudioOuts);
if ( !(cs = config.get( stream)) ) {
break;
}
const char * str;
IceCast2::StreamFormat format;
unsigned int sampleRate = 0;
unsigned int channel = 0;
AudioEncoder::BitrateMode bitrateMode;
unsigned int bitrate = 0;
unsigned int maxBitrate = 0;
double quality = 0.0;
const char * server = 0;
unsigned int port = 0;
const char * password = 0;
const char * mountPoint = 0;
const char * name = 0;
const char * description = 0;
const char * url = 0;
const char * genre = 0;
bool isPublic = false;
int lowpass = 0;
int highpass = 0;
const char * localDumpName = 0;
FileSink * localDumpFile = 0;
bool fileAddDate = false;
const char * fileDateFormat = 0;
str = cs->getForSure( "format", " missing in section ", stream);
if ( Util::strEq( str, "vorbis") ) {
format = IceCast2::oggVorbis;
} else if ( Util::strEq( str, "mp3") ) {
format = IceCast2::mp3;
} else if ( Util::strEq( str, "mp2") ) {
format = IceCast2::mp2;
} else if ( Util::strEq( str, "aac") ) {
format = IceCast2::aac;
} else if ( Util::strEq( str, "aacp") ) {
format = IceCast2::aacp;
} else {
throw Exception( __FILE__, __LINE__,
"unsupported stream format: ", str);
}
str = cs->get( "sampleRate");
sampleRate = str ? Util::strToL( str) : dsp->getSampleRate();
str = cs->get( "channel");
channel = str ? Util::strToL( str) : dsp->getChannel();
// determine fixed bitrate or variable bitrate quality
str = cs->get( "bitrate");
bitrate = str ? Util::strToL( str) : 0;
str = cs->get( "maxBitrate");
maxBitrate = str ? Util::strToL( str) : 0;
str = cs->get( "quality");
quality = str ? Util::strToD( str) : 0.0;
str = cs->getForSure( "bitrateMode",
" not specified in section ",
stream);
if ( Util::strEq( str, "cbr") ) {
bitrateMode = AudioEncoder::cbr;
if ( bitrate == 0 ) {
throw Exception( __FILE__, __LINE__,
"bitrate not specified for CBR encoding");
}
} else if ( Util::strEq( str, "abr") ) {
bitrateMode = AudioEncoder::abr;
if ( bitrate == 0 ) {
throw Exception( __FILE__, __LINE__,
"bitrate not specified for ABR encoding");
}
} else if ( Util::strEq( str, "vbr") ) {
bitrateMode = AudioEncoder::vbr;
if ( cs->get( "quality" ) == 0 ) {
throw Exception( __FILE__, __LINE__,
"quality not specified for VBR encoding");
}
} else {
throw Exception( __FILE__, __LINE__,
"invalid bitrate mode: ", str);
}
server = cs->getForSure( "server", " missing in section ", stream);
str = cs->getForSure( "port", " missing in section ", stream);
port = Util::strToL( str);
password = cs->getForSure("password"," missing in section ",stream);
mountPoint = cs->getForSure( "mountPoint",
" missing in section ",
stream);
name = cs->get( "name");
description = cs->get( "description");
url = cs->get( "url");
genre = cs->get( "genre");
str = cs->get( "public");
isPublic = str ? (Util::strEq( str, "yes") ? true : false) : false;
str = cs->get( "lowpass");
lowpass = str ? Util::strToL( str) : 0;
str = cs->get( "highpass");
highpass = str ? Util::strToL( str) : 0;
str = cs->get( "fileAddDate");
fileAddDate = str ? (Util::strEq( str, "yes") ? true : false) : false;
fileDateFormat = cs->get( "fileDateFormat");
localDumpName = cs->get( "localDumpFile");
// go on and create the things
// check for and create the local dump file if needed
if ( localDumpName != 0 ) {
if ( fileAddDate ) {
if (fileDateFormat == 0) {
localDumpName = Util::fileAddDate(localDumpName);
}
else {
localDumpName = Util::fileAddDate( localDumpName,
fileDateFormat );
}
}
localDumpFile = new FileSink( stream, localDumpName);
if ( !localDumpFile->exists() ) {
if ( !localDumpFile->create() ) {
reportEvent( 1, "can't create local dump file",
localDumpName);
localDumpFile = 0;
}
}
if ( fileAddDate ) {
delete[] localDumpName;
}
}
// streaming related stuff
audioOuts[u].socket = new TcpSocket( server, port);
audioOuts[u].server = new IceCast2( audioOuts[u].socket.get(),
password,
mountPoint,
format,
bitrate,
name,
description,
url,
genre,
isPublic,
localDumpFile,
bufferSecs );
switch ( format ) {
case IceCast2::mp3:
#ifndef HAVE_LAME_LIB
throw Exception( __FILE__, __LINE__,
"DarkIce not compiled with lame support, "
"thus can't create mp3 stream: ",
stream);
#else
audioOuts[u].encoder = new LameLibEncoder(
audioOuts[u].server.get(),
dsp.get(),
bitrateMode,
bitrate,
quality,
sampleRate,
channel,
lowpass,
highpass );
#endif // HAVE_LAME_LIB
break;
case IceCast2::oggVorbis:
#ifndef HAVE_VORBIS_LIB
throw Exception( __FILE__, __LINE__,
"DarkIce not compiled with Ogg Vorbis support, "
"thus can't Ogg Vorbis stream: ",
stream);
#else
audioOuts[u].encoder = new VorbisLibEncoder(
audioOuts[u].server.get(),
dsp.get(),
bitrateMode,
bitrate,
quality,
sampleRate,
dsp->getChannel(),
maxBitrate);
#endif // HAVE_VORBIS_LIB
break;
case IceCast2::mp2:
#ifndef HAVE_TWOLAME_LIB
throw Exception( __FILE__, __LINE__,
"DarkIce not compiled with TwoLame support, "
"thus can't create mp2 stream: ",
stream);
#else
audioOuts[u].encoder = new TwoLameLibEncoder(
audioOuts[u].server.get(),
dsp.get(),
bitrateMode,
bitrate,
sampleRate,
channel );
#endif // HAVE_TWOLAME_LIB
break;
case IceCast2::aac:
#ifndef HAVE_FAAC_LIB
throw Exception( __FILE__, __LINE__,
"DarkIce not compiled with AAC support, "
"thus can't aac stream: ",
stream);
#else
audioOuts[u].encoder = new FaacEncoder(
audioOuts[u].server.get(),
dsp.get(),
bitrateMode,
bitrate,
quality,
sampleRate,
dsp->getChannel());
#endif // HAVE_FAAC_LIB
break;
case IceCast2::aacp:
#ifndef HAVE_AACPLUS_LIB
throw Exception( __FILE__, __LINE__,
"DarkIce not compiled with AAC+ support, "
"thus can't aacp stream: ",
stream);
#else
audioOuts[u].encoder = new aacPlusEncoder(
audioOuts[u].server.get(),
dsp.get(),
bitrateMode,
bitrate,
quality,
sampleRate,
channel );
#endif // HAVE_AACPLUS_LIB
break;
default:
throw Exception( __FILE__, __LINE__,
"Illegal stream format: ", format);
}
encConnector->attach( audioOuts[u].encoder.get());
}
noAudioOuts += u;
}
/*------------------------------------------------------------------------------
* Look for the ShoutCast stream outputs in the config file
*----------------------------------------------------------------------------*/
void
DarkIce :: configShoutCast ( const Config & config,
unsigned int bufferSecs )
throw ( Exception )
{
// look for IceCast encoder output streams,
// sections [shoutcast-0], [shoutcast-1], ...
char stream[] = "shoutcast- ";
size_t streamLen = Util::strLen( stream);
unsigned int u;
for ( u = 0; u < maxOutput; ++u ) {
const ConfigSection * cs;
// ugly hack to change the section name to "stream0", "stream1", etc.
stream[streamLen-1] = '0' + u;
if ( !(cs = config.get( stream)) ) {
break;
}
#ifndef HAVE_LAME_LIB
throw Exception( __FILE__, __LINE__,
"DarkIce not compiled with lame support, "
"thus can't connect to ShoutCast, stream: ",
stream);
#else
const char * str;
unsigned int sampleRate = 0;
unsigned int bitrate = 0;
const char * server = 0;
unsigned int port = 0;
const char * password = 0;
const char * name = 0;
const char * url = 0;
const char * genre = 0;
bool isPublic = false;
int lowpass = 0;
int highpass = 0;
const char * irc = 0;
const char * aim = 0;
const char * icq = 0;
const char * localDumpName = 0;
FileSink * localDumpFile = 0;
str = cs->get( "sampleRate");
sampleRate = str ? Util::strToL( str) : dsp->getSampleRate();
str = cs->getForSure("bitrate", " missing in section ", stream);
bitrate = Util::strToL( str);
server = cs->getForSure( "server", " missing in section ", stream);
str = cs->getForSure( "port", " missing in section ", stream);
port = Util::strToL( str);
password = cs->getForSure("password"," missing in section ",stream);
name = cs->get( "name");
url = cs->get( "url");
genre = cs->get( "genre");
str = cs->get( "public");
isPublic = str ? (Util::strEq( str, "yes") ? true : false) : false;
str = cs->get( "lowpass");
lowpass = str ? Util::strToL( str) : 0;
str = cs->get( "highpass");
highpass = str ? Util::strToL( str) : 0;
irc = cs->get( "irc");
aim = cs->get( "aim");
icq = cs->get( "icq");
localDumpName = cs->get( "localDumpFile");
// go on and create the things
// check for and create the local dump file if needed
if ( localDumpName != 0 ) {
localDumpFile = new FileSink( localDumpName);
if ( !localDumpFile->exists() ) {
if ( !localDumpFile->create() ) {
reportEvent( 1, "can't create local dump file",
localDumpName);
localDumpFile = 0;
}
}
}
// encoder related stuff
unsigned int bs = bufferSecs *
(dsp->getBitsPerSample() / 8) *
dsp->getChannel() *
dsp->getSampleRate();
reportEvent( 6, "using buffer size", bs);
// streaming related stuff
audioOuts[u].socket = new TcpSocket( server, port);
audioOuts[u].server = new ShoutCast( audioOuts[u].socket.get(),
password,
bitrate,
name,
url,
genre,
isPublic,
irc,
aim,
icq,
localDumpFile );
audioOuts[u].encoder = new LameLibEncoder( audioOuts[u].server.get(),
dsp.get(),
bitrate,
sampleRate,
dsp->getChannel(),
lowpass,
highpass );
encConnector->attach( audioOuts[u].encoder.get());
#endif // HAVE_LAME_LIB
}
noAudioOuts += u;
}
/*------------------------------------------------------------------------------
* Look for the IceCast2 stream outputs in the config file
*----------------------------------------------------------------------------*/
void
DarkIce :: configIceCast2 ( const Config & config,
unsigned int bufferSecs )
throw ( Exception )
{
// look for IceCast2 encoder output streams,
// sections [icecast2-0], [icecast2-1], ...
char stream[] = "icecast2- ";
size_t streamLen = Util::strLen( stream);
unsigned int u;
for ( u = 0; u < maxOutput; ++u ) {
const ConfigSection * cs;
// ugly hack to change the section name to "stream0", "stream1", etc.
stream[streamLen-1] = '0' + u;
if ( !(cs = config.get( stream)) ) {
break;
}
const char * str;
IceCast2::StreamFormat format;
unsigned int bitrate = 0;
const char * server = 0;
unsigned int port = 0;
const char * password = 0;
const char * mountPoint = 0;
const char * name = 0;
const char * description = 0;
const char * url = 0;
const char * genre = 0;
bool isPublic = false;
const char * localDumpName = 0;
FileSink * localDumpFile = 0;
str = cs->getForSure( "format", " missing in section ", stream);
if ( Util::strEq( str, "vorbis") ) {
format = IceCast2::oggVorbis;
} else if ( Util::strEq( str, "mp3") ) {
format = IceCast2::mp3;
// TODO: enable this format in the future, when icecast2
// supports it as well
throw Exception( __FILE__, __LINE__,
"unsupported stream format: ", str);
} else {
throw Exception( __FILE__, __LINE__,
"unsupported stream format: ", str);
}
str = cs->getForSure("bitrate", " missing in section ", stream);
bitrate = Util::strToL( str);
server = cs->getForSure( "server", " missing in section ", stream);
str = cs->getForSure( "port", " missing in section ", stream);
port = Util::strToL( str);
password = cs->getForSure("password"," missing in section ",stream);
mountPoint = cs->getForSure( "mountPoint",
" missing in section ",
stream);
name = cs->get( "name");
description = cs->get("description");
url = cs->get( "url");
genre = cs->get( "genre");
str = cs->get( "public");
isPublic = str ? (Util::strEq( str, "yes") ? true : false) : false;
localDumpName = cs->get( "localDumpFile");
// go on and create the things
// check for and create the local dump file if needed
if ( localDumpName != 0 ) {
localDumpFile = new FileSink( localDumpName);
if ( !localDumpFile->exists() ) {
if ( !localDumpFile->create() ) {
reportEvent( 1, "can't create local dump file",
localDumpName);
localDumpFile = 0;
}
}
}
// encoder related stuff
unsigned int bs = bufferSecs *
(dsp->getBitsPerSample() / 8) *
dsp->getChannel() *
dsp->getSampleRate();
reportEvent( 6, "using buffer size", bs);
// streaming related stuff
audioOuts[u].socket = new TcpSocket( server, port);
audioOuts[u].server = new IceCast2( audioOuts[u].socket.get(),
password,
mountPoint,
format,
bitrate,
name,
description,
url,
genre,
isPublic,
localDumpFile );
switch ( format ) {
case IceCast2::mp3:
#ifndef HAVE_LAME_LIB
throw Exception( __FILE__, __LINE__,
"DarkIce not compiled with lame support, "
"thus can't create mp3 stream: ",
stream);
#else
audioOuts[u].encoder = new LameLibEncoder(
audioOuts[u].server.get(),
dsp.get(),
bitrate,
dsp->getSampleRate(),
dsp->getChannel() );
#endif // HAVE_LAME_LIB
break;
case IceCast2::oggVorbis:
#ifndef HAVE_VORBIS_LIB
throw Exception( __FILE__, __LINE__,
"DarkIce not compiled with Ogg Vorbis support, "
"thus can't Ogg Vorbis stream: ",
stream);
#else
audioOuts[u].encoder = new VorbisLibEncoder(
audioOuts[u].server.get(),
dsp.get(),
bitrate,
dsp->getSampleRate(),
dsp->getChannel() );
#endif // HAVE_VORBIS_LIB
break;
default:
throw Exception( __FILE__, __LINE__,
"Illegal stream format: ", format);
}
encConnector->attach( audioOuts[u].encoder.get());
}
noAudioOuts += u;
}
void createOutputFiles(char const* periodicFilenameSuffix) {
char outFileName[1000];
if (outputQuickTimeFile || outputAVIFile) {
if (periodicFilenameSuffix[0] == '\0') {
// Normally (unless the '-P <interval-in-seconds>' option was given) we output to 'stdout':
sprintf(outFileName, "stdout");
} else {
// Otherwise output to a type-specific file name, containing "periodicFilenameSuffix":
char const* prefix = fileNamePrefix[0] == '\0' ? "output" : fileNamePrefix;
snprintf(outFileName, sizeof outFileName, "%s%s.%s", prefix, periodicFilenameSuffix,
outputAVIFile ? "avi" : generateMP4Format ? "mp4" : "mov");
}
if (outputQuickTimeFile) {
qtOut = QuickTimeFileSink::createNew(*env, *session, outFileName,
fileSinkBufferSize,
movieWidth, movieHeight,
movieFPS,
packetLossCompensate,
syncStreams,
generateHintTracks,
generateMP4Format);
if (qtOut == NULL) {
*env << "Failed to create a \"QuickTimeFileSink\" for outputting to \""
<< outFileName << "\": " << env->getResultMsg() << "\n";
shutdown();
} else {
*env << "Outputting to the file: \"" << outFileName << "\"\n";
}
qtOut->startPlaying(sessionAfterPlaying, NULL);
} else { // outputAVIFile
aviOut = AVIFileSink::createNew(*env, *session, outFileName,
fileSinkBufferSize,
movieWidth, movieHeight,
movieFPS,
packetLossCompensate);
if (aviOut == NULL) {
*env << "Failed to create an \"AVIFileSink\" for outputting to \""
<< outFileName << "\": " << env->getResultMsg() << "\n";
shutdown();
} else {
*env << "Outputting to the file: \"" << outFileName << "\"\n";
}
aviOut->startPlaying(sessionAfterPlaying, NULL);
}
} else {
// Create and start "FileSink"s for each subsession:
madeProgress = False;
MediaSubsessionIterator iter(*session);
while ((subsession = iter.next()) != NULL) {
if (subsession->readSource() == NULL) continue; // was not initiated
// Create an output file for each desired stream:
if (singleMedium == NULL || periodicFilenameSuffix[0] != '\0') {
// Output file name is
// "<filename-prefix><medium_name>-<codec_name>-<counter><periodicFilenameSuffix>"
static unsigned streamCounter = 0;
snprintf(outFileName, sizeof outFileName, "%s%s-%s-%d%s",
fileNamePrefix, subsession->mediumName(),
subsession->codecName(), ++streamCounter, periodicFilenameSuffix);
} else {
// When outputting a single medium only, we output to 'stdout
// (unless the '-P <interval-in-seconds>' option was given):
sprintf(outFileName, "stdout");
}
FileSink* fileSink;
if (strcmp(subsession->mediumName(), "audio") == 0 &&
(strcmp(subsession->codecName(), "AMR") == 0 ||
strcmp(subsession->codecName(), "AMR-WB") == 0)) {
// For AMR audio streams, we use a special sink that inserts AMR frame hdrs:
fileSink = AMRAudioFileSink::createNew(*env, outFileName,
fileSinkBufferSize, oneFilePerFrame);
} else if (strcmp(subsession->mediumName(), "video") == 0 &&
(strcmp(subsession->codecName(), "H264") == 0)) {
// For H.264 video stream, we use a special sink that adds 'start codes',
// and (at the start) the SPS and PPS NAL units:
fileSink = H264VideoFileSink::createNew(*env, outFileName,
subsession->fmtp_spropparametersets(),
fileSinkBufferSize, oneFilePerFrame);
} else if (strcmp(subsession->mediumName(), "video") == 0 &&
(strcmp(subsession->codecName(), "H265") == 0)) {
// For H.265 video stream, we use a special sink that adds 'start codes',
// and (at the start) the VPS, SPS, and PPS NAL units:
fileSink = H265VideoFileSink::createNew(*env, outFileName,
subsession->fmtp_spropvps(),
subsession->fmtp_spropsps(),
subsession->fmtp_sproppps(),
fileSinkBufferSize, oneFilePerFrame);
} else {
// Normal case:
fileSink = FileSink::createNew(*env, outFileName,
fileSinkBufferSize, oneFilePerFrame);
}
subsession->sink = fileSink;
if (subsession->sink == NULL) {
*env << "Failed to create FileSink for \"" << outFileName
<< "\": " << env->getResultMsg() << "\n";
} else {
if (singleMedium == NULL) {
*env << "Created output file: \"" << outFileName << "\"\n";
} else {
*env << "Outputting data from the \"" << subsession->mediumName()
<< "/" << subsession->codecName()
<< "\" subsession to \"" << outFileName << "\"\n";
}
if (strcmp(subsession->mediumName(), "video") == 0 &&
strcmp(subsession->codecName(), "MP4V-ES") == 0 &&
subsession->fmtp_config() != NULL) {
// For MPEG-4 video RTP streams, the 'config' information
// from the SDP description contains useful VOL etc. headers.
// Insert this data at the front of the output file:
unsigned configLen;
unsigned char* configData
= parseGeneralConfigStr(subsession->fmtp_config(), configLen);
struct timeval timeNow;
gettimeofday(&timeNow, NULL);
fileSink->addData(configData, configLen, timeNow);
delete[] configData;
}
subsession->sink->startPlaying(*(subsession->readSource()),
subsessionAfterPlaying,
subsession);
// Also set a handler to be called if a RTCP "BYE" arrives
// for this subsession:
if (subsession->rtcpInstance() != NULL) {
subsession->rtcpInstance()->setByeHandler(subsessionByeHandler, subsession);
}
madeProgress = True;
}
}
if (!madeProgress) shutdown();
}
}
int main(int argc, char** argv) {
// Begin by setting up our usage environment:
TaskScheduler* scheduler = BasicTaskScheduler::createNew();
env = BasicUsageEnvironment::createNew(*scheduler);
progName = argv[0];
gettimeofday(&startTime, NULL);
#ifdef USE_SIGNALS
// Allow ourselves to be shut down gracefully by a SIGHUP or a SIGUSR1:
signal(SIGHUP, signalHandlerShutdown);
signal(SIGUSR1, signalHandlerShutdown);
#endif
unsigned short desiredPortNum = 0;
// unfortunately we can't use getopt() here, as Windoze doesn't have it
while (argc > 2) {
char* const opt = argv[1];
if (opt[0] != '-') usage();
switch (opt[1]) {
case 'p': { // specify start port number
int portArg;
if (sscanf(argv[2], "%d", &portArg) != 1) {
usage();
}
if (portArg <= 0 || portArg >= 65536 || portArg&1) {
*env << "bad port number: " << portArg
<< " (must be even, and in the range (0,65536))\n";
usage();
}
desiredPortNum = (unsigned short)portArg;
++argv; --argc;
break;
}
case 'r': { // do not receive data (instead, just 'play' the stream(s))
createReceivers = False;
break;
}
case 'q': { // output a QuickTime file (to stdout)
outputQuickTimeFile = True;
break;
}
case '4': { // output a 'mp4'-format file (to stdout)
outputQuickTimeFile = True;
generateMP4Format = True;
break;
}
case 'i': { // output an AVI file (to stdout)
outputAVIFile = True;
break;
}
case 'I': { // specify input interface...
NetAddressList addresses(argv[2]);
if (addresses.numAddresses() == 0) {
*env << "Failed to find network address for \"" << argv[2] << "\"";
break;
}
ReceivingInterfaceAddr = *(unsigned*)(addresses.firstAddress()->data());
++argv; --argc;
break;
}
case 'a': { // receive/record an audio stream only
audioOnly = True;
singleMedium = "audio";
break;
}
case 'v': { // receive/record a video stream only
videoOnly = True;
singleMedium = "video";
break;
}
case 'V': { // disable verbose output
verbosityLevel = 0;
break;
}
case 'd': { // specify duration, or how much to delay after end time
float arg;
if (sscanf(argv[2], "%g", &arg) != 1) {
usage();
}
if (argv[2][0] == '-') { // not "arg<0", in case argv[2] was "-0"
// a 'negative' argument was specified; use this for "durationSlop":
duration = 0; // use whatever's in the SDP
durationSlop = -arg;
} else {
duration = arg;
durationSlop = 0;
}
++argv; --argc;
break;
}
case 'D': { // specify maximum number of seconds to wait for packets:
if (sscanf(argv[2], "%u", &interPacketGapMaxTime) != 1) {
usage();
}
++argv; --argc;
break;
}
case 'c': { // play continuously
playContinuously = True;
break;
}
case 'S': { // specify an offset to use with "SimpleRTPSource"s
if (sscanf(argv[2], "%d", &simpleRTPoffsetArg) != 1) {
usage();
}
if (simpleRTPoffsetArg < 0) {
*env << "offset argument to \"-S\" must be >= 0\n";
usage();
}
++argv; --argc;
break;
}
case 'O': { // Don't send an "OPTIONS" request before "DESCRIBE"
sendOptionsRequest = False;
break;
}
case 'o': { // Send only the "OPTIONS" request to the server
sendOptionsRequestOnly = True;
break;
}
case 'm': { // output multiple files - one for each frame
oneFilePerFrame = True;
break;
}
case 'n': { // notify the user when the first data packet arrives
notifyOnPacketArrival = True;
break;
}
case 't': {
// stream RTP and RTCP over the TCP 'control' connection
if (controlConnectionUsesTCP) {
streamUsingTCP = True;
} else {
usage();
}
break;
}
case 'T': {
// stream RTP and RTCP over a HTTP connection
if (controlConnectionUsesTCP) {
if (argc > 3 && argv[2][0] != '-') {
// The next argument is the HTTP server port number:
if (sscanf(argv[2], "%hu", &tunnelOverHTTPPortNum) == 1
&& tunnelOverHTTPPortNum > 0) {
++argv; --argc;
break;
}
}
}
// If we get here, the option was specified incorrectly:
usage();
break;
}
case 'u': { // specify a username and password
username = argv[2];
password = argv[3];
argv+=2; argc-=2;
if (allowProxyServers && argc > 3 && argv[2][0] != '-') {
// The next argument is the name of a proxy server:
proxyServerName = argv[2];
++argv; --argc;
if (argc > 3 && argv[2][0] != '-') {
// The next argument is the proxy server port number:
if (sscanf(argv[2], "%hu", &proxyServerPortNum) != 1) {
usage();
}
++argv; --argc;
}
}
break;
}
case 'A': { // specify a desired audio RTP payload format
unsigned formatArg;
if (sscanf(argv[2], "%u", &formatArg) != 1
|| formatArg >= 96) {
usage();
}
desiredAudioRTPPayloadFormat = (unsigned char)formatArg;
++argv; --argc;
break;
}
case 'M': { // specify a MIME subtype for a dynamic RTP payload type
mimeSubtype = argv[2];
if (desiredAudioRTPPayloadFormat==0) desiredAudioRTPPayloadFormat =96;
++argv; --argc;
break;
}
case 'w': { // specify a width (pixels) for an output QuickTime or AVI movie
if (sscanf(argv[2], "%hu", &movieWidth) != 1) {
usage();
}
movieWidthOptionSet = True;
++argv; --argc;
break;
}
case 'h': { // specify a height (pixels) for an output QuickTime or AVI movie
if (sscanf(argv[2], "%hu", &movieHeight) != 1) {
usage();
}
movieHeightOptionSet = True;
++argv; --argc;
break;
}
case 'f': { // specify a frame rate (per second) for an output QT or AVI movie
if (sscanf(argv[2], "%u", &movieFPS) != 1) {
usage();
}
movieFPSOptionSet = True;
++argv; --argc;
break;
}
case 'F': { // specify a prefix for the audio and video output files
fileNamePrefix = argv[2];
++argv; --argc;
break;
}
case 'b': { // specify the size of buffers for "FileSink"s
if (sscanf(argv[2], "%u", &fileSinkBufferSize) != 1) {
usage();
}
++argv; --argc;
break;
}
case 'B': { // specify the size of input socket buffers
if (sscanf(argv[2], "%u", &socketInputBufferSize) != 1) {
usage();
}
++argv; --argc;
break;
}
// Note: The following option is deprecated, and may someday be removed:
case 'l': { // try to compensate for packet loss by repeating frames
packetLossCompensate = True;
break;
}
case 'y': { // synchronize audio and video streams
syncStreams = True;
break;
}
case 'H': { // generate hint tracks (as well as the regular data tracks)
generateHintTracks = True;
break;
}
case 'Q': { // output QOS measurements
qosMeasurementIntervalMS = 1000; // default: 1 second
if (argc > 3 && argv[2][0] != '-') {
// The next argument is the measurement interval,
// in multiples of 100 ms
if (sscanf(argv[2], "%u", &qosMeasurementIntervalMS) != 1) {
usage();
}
qosMeasurementIntervalMS *= 100;
++argv; --argc;
}
break;
}
case 's': { // specify initial seek time (trick play)
double arg;
if (sscanf(argv[2], "%lg", &arg) != 1 || arg < 0) {
usage();
}
initialSeekTime = arg;
++argv; --argc;
break;
}
case 'z': { // scale (trick play)
float arg;
if (sscanf(argv[2], "%g", &arg) != 1 || arg == 0.0f) {
usage();
}
scale = arg;
++argv; --argc;
break;
}
default: {
usage();
break;
}
}
++argv; --argc;
}
if (argc != 2) usage();
if (outputQuickTimeFile && outputAVIFile) {
*env << "The -i and -q (or -4) flags cannot both be used!\n";
usage();
}
Boolean outputCompositeFile = outputQuickTimeFile || outputAVIFile;
if (!createReceivers && outputCompositeFile) {
*env << "The -r and -q (or -4 or -i) flags cannot both be used!\n";
usage();
}
if (outputCompositeFile && !movieWidthOptionSet) {
*env << "Warning: The -q, -4 or -i option was used, but not -w. Assuming a video width of "
<< movieWidth << " pixels\n";
}
if (outputCompositeFile && !movieHeightOptionSet) {
*env << "Warning: The -q, -4 or -i option was used, but not -h. Assuming a video height of "
<< movieHeight << " pixels\n";
}
if (outputCompositeFile && !movieFPSOptionSet) {
*env << "Warning: The -q, -4 or -i option was used, but not -f. Assuming a video frame rate of "
<< movieFPS << " frames-per-second\n";
}
if (audioOnly && videoOnly) {
*env << "The -a and -v flags cannot both be used!\n";
usage();
}
if (sendOptionsRequestOnly && !sendOptionsRequest) {
*env << "The -o and -O flags cannot both be used!\n";
usage();
}
if (tunnelOverHTTPPortNum > 0) {
if (streamUsingTCP) {
*env << "The -t and -T flags cannot both be used!\n";
usage();
} else {
streamUsingTCP = True;
}
}
if (!createReceivers && notifyOnPacketArrival) {
*env << "Warning: Because we're not receiving stream data, the -n flag has no effect\n";
}
if (durationSlop < 0) {
// This parameter wasn't set, so use a default value.
// If we're measuring QOS stats, then don't add any slop, to avoid
// having 'empty' measurement intervals at the end.
durationSlop = qosMeasurementIntervalMS > 0 ? 0.0 : 5.0;
}
char* url = argv[1];
// Create our client object:
ourClient = createClient(*env, verbosityLevel, progName);
if (ourClient == NULL) {
*env << "Failed to create " << clientProtocolName
<< " client: " << env->getResultMsg() << "\n";
shutdown();
}
if (sendOptionsRequest) {
// Begin by sending an "OPTIONS" command:
char* optionsResponse
= getOptionsResponse(ourClient, url, username, password);
if (sendOptionsRequestOnly) {
if (optionsResponse == NULL) {
*env << clientProtocolName << " \"OPTIONS\" request failed: "
<< env->getResultMsg() << "\n";
} else {
*env << clientProtocolName << " \"OPTIONS\" request returned: "
<< optionsResponse << "\n";
}
shutdown();
}
delete[] optionsResponse;
}
// Open the URL, to get a SDP description:
char* sdpDescription
= getSDPDescriptionFromURL(ourClient, url, username, password,
proxyServerName, proxyServerPortNum,
desiredPortNum);
if (sdpDescription == NULL) {
*env << "Failed to get a SDP description from URL \"" << url
<< "\": " << env->getResultMsg() << "\n";
shutdown();
}
*env << "Opened URL \"" << url
<< "\", returning a SDP description:\n" << sdpDescription << "\n";
// Create a media session object from this SDP description:
session = MediaSession::createNew(*env, sdpDescription);
delete[] sdpDescription;
if (session == NULL) {
*env << "Failed to create a MediaSession object from the SDP description: " << env->getResultMsg() << "\n";
shutdown();
} else if (!session->hasSubsessions()) {
*env << "This session has no media subsessions (i.e., \"m=\" lines)\n";
shutdown();
}
// Then, setup the "RTPSource"s for the session:
MediaSubsessionIterator iter(*session);
MediaSubsession *subsession;
Boolean madeProgress = False;
char const* singleMediumToTest = singleMedium;
while ((subsession = iter.next()) != NULL) {
// If we've asked to receive only a single medium, then check this now:
if (singleMediumToTest != NULL) {
if (strcmp(subsession->mediumName(), singleMediumToTest) != 0) {
*env << "Ignoring \"" << subsession->mediumName()
<< "/" << subsession->codecName()
<< "\" subsession, because we've asked to receive a single " << singleMedium
<< " session only\n";
continue;
} else {
// Receive this subsession only
singleMediumToTest = "xxxxx";
// this hack ensures that we get only 1 subsession of this type
}
}
if (desiredPortNum != 0) {
subsession->setClientPortNum(desiredPortNum);
desiredPortNum += 2;
}
if (createReceivers) {
if (!subsession->initiate(simpleRTPoffsetArg)) {
*env << "Unable to create receiver for \"" << subsession->mediumName()
<< "/" << subsession->codecName()
<< "\" subsession: " << env->getResultMsg() << "\n";
} else {
*env << "Created receiver for \"" << subsession->mediumName()
<< "/" << subsession->codecName()
<< "\" subsession (client ports " << subsession->clientPortNum()
<< "-" << subsession->clientPortNum()+1 << ")\n";
madeProgress = True;
if (subsession->rtpSource() != NULL) {
// Because we're saving the incoming data, rather than playing
// it in real time, allow an especially large time threshold
// (1 second) for reordering misordered incoming packets:
unsigned const thresh = 1000000; // 1 second
subsession->rtpSource()->setPacketReorderingThresholdTime(thresh);
// Set the RTP source's OS socket buffer size as appropriate - either if we were explicitly asked (using -B),
// or if the desired FileSink buffer size happens to be larger than the current OS socket buffer size.
// (The latter case is a heuristic, on the assumption that if the user asked for a large FileSink buffer size,
// then the input data rate may be large enough to justify increasing the OS socket buffer size also.)
int socketNum = subsession->rtpSource()->RTPgs()->socketNum();
unsigned curBufferSize = getReceiveBufferSize(*env, socketNum);
if (socketInputBufferSize > 0 || fileSinkBufferSize > curBufferSize) {
unsigned newBufferSize = socketInputBufferSize > 0 ? socketInputBufferSize : fileSinkBufferSize;
newBufferSize = setReceiveBufferTo(*env, socketNum, newBufferSize);
if (socketInputBufferSize > 0) { // The user explicitly asked for the new socket buffer size; announce it:
*env << "Changed socket receive buffer size for the \""
<< subsession->mediumName()
<< "/" << subsession->codecName()
<< "\" subsession from "
<< curBufferSize << " to "
<< newBufferSize << " bytes\n";
}
}
}
}
} else {
if (subsession->clientPortNum() == 0) {
*env << "No client port was specified for the \""
<< subsession->mediumName()
<< "/" << subsession->codecName()
<< "\" subsession. (Try adding the \"-p <portNum>\" option.)\n";
} else {
madeProgress = True;
}
}
}
if (!madeProgress) shutdown();
// Perform additional 'setup' on each subsession, before playing them:
setupStreams();
// Create output files:
if (createReceivers) {
if (outputQuickTimeFile) {
// Create a "QuickTimeFileSink", to write to 'stdout':
qtOut = QuickTimeFileSink::createNew(*env, *session, "stdout",
fileSinkBufferSize,
movieWidth, movieHeight,
movieFPS,
packetLossCompensate,
syncStreams,
generateHintTracks,
generateMP4Format);
if (qtOut == NULL) {
*env << "Failed to create QuickTime file sink for stdout: " << env->getResultMsg();
shutdown();
}
qtOut->startPlaying(sessionAfterPlaying, NULL);
} else if (outputAVIFile) {
// Create an "AVIFileSink", to write to 'stdout':
aviOut = AVIFileSink::createNew(*env, *session, "stdout",
fileSinkBufferSize,
movieWidth, movieHeight,
movieFPS,
packetLossCompensate);
if (aviOut == NULL) {
*env << "Failed to create AVI file sink for stdout: " << env->getResultMsg();
shutdown();
}
aviOut->startPlaying(sessionAfterPlaying, NULL);
} else {
// Create and start "FileSink"s for each subsession:
madeProgress = False;
iter.reset();
while ((subsession = iter.next()) != NULL) {
if (subsession->readSource() == NULL) continue; // was not initiated
// Create an output file for each desired stream:
char outFileName[1000];
if (singleMedium == NULL) {
// Output file name is
// "<filename-prefix><medium_name>-<codec_name>-<counter>"
static unsigned streamCounter = 0;
snprintf(outFileName, sizeof outFileName, "%s%s-%s-%d",
fileNamePrefix, subsession->mediumName(),
subsession->codecName(), ++streamCounter);
} else {
sprintf(outFileName, "stdout");
}
FileSink* fileSink;
if (strcmp(subsession->mediumName(), "audio") == 0 &&
(strcmp(subsession->codecName(), "AMR") == 0 ||
strcmp(subsession->codecName(), "AMR-WB") == 0)) {
// For AMR audio streams, we use a special sink that inserts AMR frame hdrs:
fileSink = AMRAudioFileSink::createNew(*env, outFileName,
fileSinkBufferSize, oneFilePerFrame);
} else if (strcmp(subsession->mediumName(), "video") == 0 &&
(strcmp(subsession->codecName(), "H264") == 0)) {
// For H.264 video stream, we use a special sink that insert start_codes:
fileSink = H264VideoFileSink::createNew(*env, outFileName,
fileSinkBufferSize, oneFilePerFrame);
} else {
// Normal case:
fileSink = FileSink::createNew(*env, outFileName,
fileSinkBufferSize, oneFilePerFrame);
}
subsession->sink = fileSink;
if (subsession->sink == NULL) {
*env << "Failed to create FileSink for \"" << outFileName
<< "\": " << env->getResultMsg() << "\n";
} else {
if (singleMedium == NULL) {
*env << "Created output file: \"" << outFileName << "\"\n";
} else {
*env << "Outputting data from the \"" << subsession->mediumName()
<< "/" << subsession->codecName()
<< "\" subsession to 'stdout'\n";
}
if (strcmp(subsession->mediumName(), "video") == 0 &&
strcmp(subsession->codecName(), "MP4V-ES") == 0 &&
subsession->fmtp_config() != NULL) {
// For MPEG-4 video RTP streams, the 'config' information
// from the SDP description contains useful VOL etc. headers.
// Insert this data at the front of the output file:
unsigned configLen;
unsigned char* configData
= parseGeneralConfigStr(subsession->fmtp_config(), configLen);
struct timeval timeNow;
gettimeofday(&timeNow, NULL);
fileSink->addData(configData, configLen, timeNow);
delete[] configData;
}
subsession->sink->startPlaying(*(subsession->readSource()),
subsessionAfterPlaying,
subsession);
// Also set a handler to be called if a RTCP "BYE" arrives
// for this subsession:
if (subsession->rtcpInstance() != NULL) {
subsession->rtcpInstance()->setByeHandler(subsessionByeHandler,
subsession);
}
madeProgress = True;
}
}
if (!madeProgress) shutdown();
}
}
// Finally, start playing each subsession, to start the data flow:
startPlayingStreams();
env->taskScheduler().doEventLoop(); // does not return
return 0; // only to prevent compiler warning
}
int main(int argc, char *argv[]) {
Opt opt;
opt.registerOpt(P_DeviceName, "-d", false, true, "specify DV4Mini device name", "/dev/ttyACM0");
opt.registerOpt(P_Frequency, "-f", false, true, "set frequency in Hz", "430312500");
opt.registerOpt(P_WriteRxToFile, "-w", false, true, "write received data to file");
opt.registerOpt(P_TxFromFile, "-tx", false, true, "transmit data from file");
opt.registerOpt(P_SimMode, "-s", false, false, "simulate without real hardware");
opt.registerOpt(P_LogLevel, "-ll", false, true, "log level");
opt.registerOpt(P_DebugMode, "-debug", false, false, "set debug mode");
if(!opt.parse(argc, argv)) {
opt.printHelp();
return 1;
}
bool bDebug = opt.get(P_DebugMode);
bool bSimMode = opt.get(P_SimMode);
if(bDebug) opt.dump();
std::string sDeviceName = opt.getString(P_DeviceName);
int iFrequency_Hz = opt.getInt(P_Frequency);
DV4Mini mini;
if(bSimMode) {
sDeviceName = "/dev/null";
mini.setSimulationMode(true);
}
mini.setLogFile(stdout, opt.getInt(P_LogLevel));
if(!mini.open(sDeviceName.c_str())) {
perror(sDeviceName.c_str());
return 1;
}
mini.setFrequency(iFrequency_Hz);
printf("Using device \"%s\" at %d Hz (%.3f kHz).\n", sDeviceName.c_str(), iFrequency_Hz, iFrequency_Hz / 1000.);
mini.setLED(true);
FileSink fsink;
if(opt.get(P_WriteRxToFile)) {
const char *pszWriteFileName = opt.getString(P_WriteRxToFile).c_str();
if(!fsink.open(pszWriteFileName)) {
perror(pszWriteFileName);
return 1;
}
mini.setRxSink(&fsink);
printf("Writing received data to file \"%s\".\n", pszWriteFileName);
}
if(opt.get(P_TxFromFile)) {
const char *pszReadFileName = opt.getString(P_TxFromFile).c_str();
FILE *pFile = fopen(pszReadFileName, "r");
if(!pFile) {
perror(pszReadFileName);
return 1;
}
printf("Sending file \"%s\"", pszReadFileName);
BYTE buf[100];
while(true) {
putchar('.');
int iBytes = fread(buf, 1, sizeof buf, pFile);
if(iBytes <= 0) break;
mini.transmit(buf, iBytes);
usleep(10);
}
puts("done!\n");
mini.flush();
fclose(pFile);
mini.close();
return 0;
}
#if 1
for(int i = 0; i < 5; ++i) {
sleep(1);
printf("RSSI=%d\n", mini.getRSSI());
}
#else
for(int i = 0; i < 50; ++i) {
usleep(100);
printf("RSSI=%d\n", mini.getRSSI());
}
#endif
mini.close();
return 0;
}
