MPEG4ESVideoRTPSink
::MPEG4ESVideoRTPSink(UsageEnvironment& env, Groupsock* RTPgs, unsigned char rtpPayloadFormat, u_int32_t rtpTimestampFrequency,
u_int8_t profileAndLevelIndication, char const* configStr)
: VideoRTPSink(env, RTPgs, rtpPayloadFormat, rtpTimestampFrequency, "MP4V-ES"),
fVOPIsPresent(False), fProfileAndLevelIndication(profileAndLevelIndication), fFmtpSDPLine(NULL) {
fConfigBytes = parseGeneralConfigStr(configStr, fNumConfigBytes);
}
void MPEG4VideoStreamFramer
::setConfigInfo(u_int8_t profileAndLevelIndication, char const* configStr) {
fProfileAndLevelIndication = profileAndLevelIndication;
delete[] fConfigBytes;
fConfigBytes = parseGeneralConfigStr(configStr, fNumConfigBytes);
}
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);
}
void rtpCodecInitialize_video(demuxer_t* demuxer,
MediaSubsession* subsession,
unsigned& flags) {
flags = 0;
// Create a dummy video stream header
// to make the main MPlayer code happy:
sh_video_t* sh_video = new_sh_video(demuxer,0);
BITMAPINFOHEADER* bih
= (BITMAPINFOHEADER*)calloc(1,sizeof(BITMAPINFOHEADER));
bih->biSize = sizeof(BITMAPINFOHEADER);
sh_video->bih = bih;
demux_stream_t* d_video = demuxer->video;
d_video->sh = sh_video; sh_video->ds = d_video;
// Map known video MIME types to the BITMAPINFOHEADER parameters
// that this program uses. (Note that not all types need all
// of the parameters to be set.)
if (strcmp(subsession->codecName(), "MPV") == 0) {
flags |= RTPSTATE_IS_MPEG12_VIDEO;
} else if (strcmp(subsession->codecName(), "MP1S") == 0 ||
strcmp(subsession->codecName(), "MP2T") == 0) {
flags |= RTPSTATE_IS_MPEG12_VIDEO|RTPSTATE_IS_MULTIPLEXED;
} else if (strcmp(subsession->codecName(), "H263") == 0 ||
strcmp(subsession->codecName(), "H263-2000") == 0 ||
strcmp(subsession->codecName(), "H263-1998") == 0) {
bih->biCompression = sh_video->format
= mmioFOURCC('H','2','6','3');
needVideoFrameRate(demuxer, subsession);
} else if (strcmp(subsession->codecName(), "H264") == 0) {
bih->biCompression = sh_video->format
= mmioFOURCC('H','2','6','4');
unsigned int configLen = 0;
unsigned char* configData
= parseH264ConfigStr(subsession->fmtp_spropparametersets(), configLen);
sh_video->bih = bih = insertVideoExtradata(bih, configData, configLen);
delete[] configData;
#ifdef USE_LIBAVCODEC
av_register_codec_parser(&h264_parser);
h264parserctx = av_parser_init(CODEC_ID_H264);
#endif
needVideoFrameRate(demuxer, subsession);
} else if (strcmp(subsession->codecName(), "H261") == 0) {
bih->biCompression = sh_video->format
= mmioFOURCC('H','2','6','1');
needVideoFrameRate(demuxer, subsession);
} else if (strcmp(subsession->codecName(), "JPEG") == 0) {
bih->biCompression = sh_video->format
= mmioFOURCC('M','J','P','G');
needVideoFrameRate(demuxer, subsession);
} else if (strcmp(subsession->codecName(), "MP4V-ES") == 0) {
bih->biCompression = sh_video->format
= mmioFOURCC('m','p','4','v');
// For the codec to work correctly, it may need a 'VOL Header' to be
// inserted at the front of the data stream. Construct this from the
// "config" MIME parameter, which was present (hopefully) in the
// session's SDP description:
unsigned configLen;
unsigned char* configData
= parseGeneralConfigStr(subsession->fmtp_config(), configLen);
sh_video->bih = bih = insertVideoExtradata(bih, configData, configLen);
needVideoFrameRate(demuxer, subsession);
} else if (strcmp(subsession->codecName(), "X-QT") == 0 ||
strcmp(subsession->codecName(), "X-QUICKTIME") == 0) {
// QuickTime generic RTP format, as described in
// http://developer.apple.com/quicktime/icefloe/dispatch026.html
// We can't initialize this stream until we've received the first packet
// that has QuickTime "sdAtom" information in the header. So, keep
// reading packets until we get one:
unsigned char* packetData; unsigned packetDataLen; float pts;
QuickTimeGenericRTPSource* qtRTPSource
= (QuickTimeGenericRTPSource*)(subsession->rtpSource());
unsigned fourcc;
do {
if (!awaitRTPPacket(demuxer, demuxer->video,
packetData, packetDataLen, pts)) {
return;
}
} while (!parseQTState_video(qtRTPSource->qtState, fourcc));
bih->biCompression = sh_video->format = fourcc;
bih->biWidth = qtRTPSource->qtState.width;
bih->biHeight = qtRTPSource->qtState.height;
uint8_t *pos = (uint8_t*)qtRTPSource->qtState.sdAtom + 86;
uint8_t *endpos = (uint8_t*)qtRTPSource->qtState.sdAtom
+ qtRTPSource->qtState.sdAtomSize;
while (pos+8 < endpos) {
unsigned atomLength = pos[0]<<24 | pos[1]<<16 | pos[2]<<8 | pos[3];
if (atomLength == 0 || atomLength > endpos-pos) break;
if ((!memcmp(pos+4, "avcC", 4) && fourcc==mmioFOURCC('a','v','c','1') ||
!memcmp(pos+4, "esds", 4) ||
!memcmp(pos+4, "SMI ", 4) && fourcc==mmioFOURCC('S','V','Q','3')) &&
atomLength > 8) {
sh_video->bih = bih =
insertVideoExtradata(bih, pos+8, atomLength-8);
break;
}
pos += atomLength;
}
needVideoFrameRate(demuxer, subsession);
} else {
fprintf(stderr,
"Unknown MPlayer format code for MIME type \"video/%s\"\n",
subsession->codecName());
}
}
void rtpCodecInitialize_audio(demuxer_t* demuxer,
MediaSubsession* subsession,
unsigned& flags) {
flags = 0;
// Create a dummy audio stream header
// to make the main MPlayer code happy:
sh_audio_t* sh_audio = new_sh_audio(demuxer,0);
WAVEFORMATEX* wf = (WAVEFORMATEX*)calloc(1,sizeof(WAVEFORMATEX));
sh_audio->wf = wf;
demux_stream_t* d_audio = demuxer->audio;
d_audio->sh = sh_audio; sh_audio->ds = d_audio;
wf->nChannels = subsession->numChannels();
// Map known audio MIME types to the WAVEFORMATEX parameters
// that this program uses. (Note that not all types need all
// of the parameters to be set.)
wf->nSamplesPerSec
= subsession->rtpSource()->timestampFrequency(); // by default
if (strcmp(subsession->codecName(), "MPA") == 0 ||
strcmp(subsession->codecName(), "MPA-ROBUST") == 0 ||
strcmp(subsession->codecName(), "X-MP3-DRAFT-00") == 0) {
wf->wFormatTag = sh_audio->format = 0x55;
// Note: 0x55 is for layer III, but should work for I,II also
wf->nSamplesPerSec = 0; // sample rate is deduced from the data
} else if (strcmp(subsession->codecName(), "AC3") == 0) {
wf->wFormatTag = sh_audio->format = 0x2000;
wf->nSamplesPerSec = 0; // sample rate is deduced from the data
} else if (strcmp(subsession->codecName(), "L16") == 0) {
wf->wFormatTag = sh_audio->format = 0x736f7774; // "twos"
wf->nBlockAlign = 1;
wf->wBitsPerSample = 16;
wf->cbSize = 0;
} else if (strcmp(subsession->codecName(), "L8") == 0) {
wf->wFormatTag = sh_audio->format = 0x20776172; // "raw "
wf->nBlockAlign = 1;
wf->wBitsPerSample = 8;
wf->cbSize = 0;
} else if (strcmp(subsession->codecName(), "PCMU") == 0) {
wf->wFormatTag = sh_audio->format = 0x7;
wf->nAvgBytesPerSec = 8000;
wf->nBlockAlign = 1;
wf->wBitsPerSample = 8;
wf->cbSize = 0;
} else if (strcmp(subsession->codecName(), "PCMA") == 0) {
wf->wFormatTag = sh_audio->format = 0x6;
wf->nAvgBytesPerSec = 8000;
wf->nBlockAlign = 1;
wf->wBitsPerSample = 8;
wf->cbSize = 0;
} else if (strcmp(subsession->codecName(), "AMR") == 0) {
wf->wFormatTag = sh_audio->format = mmioFOURCC('s','a','m','r');
} else if (strcmp(subsession->codecName(), "AMR-WB") == 0) {
wf->wFormatTag = sh_audio->format = mmioFOURCC('s','a','w','b');
} else if (strcmp(subsession->codecName(), "GSM") == 0) {
wf->wFormatTag = sh_audio->format = mmioFOURCC('a','g','s','m');
wf->nAvgBytesPerSec = 1650;
wf->nBlockAlign = 33;
wf->wBitsPerSample = 16;
wf->cbSize = 0;
} else if (strcmp(subsession->codecName(), "QCELP") == 0) {
wf->wFormatTag = sh_audio->format = mmioFOURCC('Q','c','l','p');
wf->nAvgBytesPerSec = 1750;
wf->nBlockAlign = 35;
wf->wBitsPerSample = 16;
wf->cbSize = 0;
} else if (strcmp(subsession->codecName(), "MP4A-LATM") == 0) {
wf->wFormatTag = sh_audio->format = mmioFOURCC('m','p','4','a');
// For the codec to work correctly, it needs "AudioSpecificConfig"
// data, which is parsed from the "StreamMuxConfig" string that
// was present (hopefully) in the SDP description:
unsigned codecdata_len;
sh_audio->codecdata
= parseStreamMuxConfigStr(subsession->fmtp_config(),
codecdata_len);
sh_audio->codecdata_len = codecdata_len;
//faad doesn't understand LATM's data length field, so omit it
((MPEG4LATMAudioRTPSource*)subsession->rtpSource())->omitLATMDataLengthField();
} else if (strcmp(subsession->codecName(), "MPEG4-GENERIC") == 0) {
wf->wFormatTag = sh_audio->format = mmioFOURCC('m','p','4','a');
// For the codec to work correctly, it needs "AudioSpecificConfig"
// data, which was present (hopefully) in the SDP description:
unsigned codecdata_len;
sh_audio->codecdata
= parseGeneralConfigStr(subsession->fmtp_config(),
codecdata_len);
sh_audio->codecdata_len = codecdata_len;
} else if (strcmp(subsession->codecName(), "X-QT") == 0 ||
strcmp(subsession->codecName(), "X-QUICKTIME") == 0) {
// QuickTime generic RTP format, as described in
// http://developer.apple.com/quicktime/icefloe/dispatch026.html
// We can't initialize this stream until we've received the first packet
// that has QuickTime "sdAtom" information in the header. So, keep
// reading packets until we get one:
unsigned char* packetData; unsigned packetDataLen; float pts;
QuickTimeGenericRTPSource* qtRTPSource
= (QuickTimeGenericRTPSource*)(subsession->rtpSource());
unsigned fourcc, numChannels;
do {
if (!awaitRTPPacket(demuxer, demuxer->audio,
packetData, packetDataLen, pts)) {
return;
}
} while (!parseQTState_audio(qtRTPSource->qtState, fourcc, numChannels));
wf->wFormatTag = sh_audio->format = fourcc;
wf->nChannels = numChannels;
uint8_t *pos = (uint8_t*)qtRTPSource->qtState.sdAtom + 52;
uint8_t *endpos = (uint8_t*)qtRTPSource->qtState.sdAtom
+ qtRTPSource->qtState.sdAtomSize;
while (pos+8 < endpos) {
unsigned atomLength = pos[0]<<24 | pos[1]<<16 | pos[2]<<8 | pos[3];
if (atomLength == 0 || atomLength > endpos-pos) break;
if (!memcmp(pos+4, "wave", 4) && fourcc==mmioFOURCC('Q','D','M','2') &&
atomLength > 8 &&
atomLength <= INT_MAX) {
sh_audio->codecdata = (unsigned char*) malloc(atomLength-8);
if (sh_audio->codecdata) {
memcpy(sh_audio->codecdata, pos+8, atomLength-8);
sh_audio->codecdata_len = atomLength-8;
}
break;
}
pos += atomLength;
}
} else {
fprintf(stderr,
"Unknown MPlayer format code for MIME type \"audio/%s\"\n",
subsession->codecName());
}
}
bool MtkRTSPClient::handSetup(char* resultString)
{
CHECK_NULL_COND(session, false);
CHECK_NULL_COND(rtsp::env, false);
bool bSuccess = false;
// Then, setup the "RTPSource"s for the session:
MediaSubsessionIterator iter(*(session));
MediaSubsession *subsession = NULL;
while ((subsession = iter.next()) != NULL)
{
if (subsession->readSource() == NULL)
{
LOG_ERR("warning");
continue; // was not initiated
}
if (subsession->sink != NULL)/*already be set*/
{
continue;
}
unsigned int type = getBufType(subsession);
if (type == 0)
{
LOG_ERR("error type=%d", type);
continue;
}
{
iSetupCount--;
/*set mediay info*/
setMediaInfo(subsession, type);
}
CmpbSink *sink = NULL;
if ((type != mediatype_audio) && (strcmp(subsession->codecName(), "H264") == 0))
{
sink = CmpbH264Sink::createNew(*env, *subsession, type, fileSinkBufferSize);
}
else if ((type == mediatype_audio) &&
((stMediaInfo.audioCodec == MEDIACODEC_AC3) ||
(stMediaInfo.audioCodec == MEDIACODEC_EAC3) ||
(stMediaInfo.audioCodec == MEDIACODEC_MPEG4_GENERIC)))
{
sink = CmpbAACSink::createNew(*env, *subsession, type, fileSinkBufferSize);
}
else if ((type == mediatype_audio) && (stMediaInfo.audioCodec == MEDIACODEC_MP4A_LATM))
{
sink = CmpbLATMSink::createNew(*env, *subsession, type, fileSinkBufferSize);
}
else
{
sink = CmpbSink::createNew(*env, *subsession, type, fileSinkBufferSize);
}
subsession->sink = sink;
if (subsession->sink == NULL)
{
LOG_ERR("error!");
}
else
{
#if 0 /*this should be remove to cmpb sink*/
if ((type != mediatype_audio) && (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);
sink->sendData(configData, configLen, timeNow);
delete[] configData;
}
#endif
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(subsessionAfterPlaying, subsession);
}
bSuccess = true;
}
break;
}
if (iSetupCount == 0)
{
mediaInfoReady();
}
return bSuccess ;
}
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 CLvRtspClient::InitAudio( MediaSubsession *pss )
{_STT();
if ( !m_pRtspClient )
{ setLastError( -200, oexT( "Invalid rtsp client object" ) );
return 0;
} // end if
if ( !pss )
{ setLastError( -201, oexT( "Invalid video object" ) );
return 0;
} // end if
// Create receiver for stream
if ( !pss->initiate() )
{ setLastError( -203, sqbind::oex2std( oexMks( oexT( "initiate() video stream failed : " ), oexT( " : " ), m_pEnv->getResultMsg() ) ) );
return 0;
} // end if
if ( !pss->rtpSource() )
{ setLastError( -204, sqbind::oex2std( oexMks( oexT( "RTP source is null : " ), oexT( " : " ), m_pEnv->getResultMsg() ) ) );
return 0;
} // end if
if ( oex::CStr8( "MP4A-LATM" ) == pss->codecName() )
{
((MPEG4LATMAudioRTPSource*)pss->rtpSource())->omitLATMDataLengthField();
const char *pCfg = pss->fmtp_config();
if ( pCfg )
{ unsigned elen = 0;
unsigned char *pExtra = parseStreamMuxConfigStr( pCfg, elen );
if ( pExtra && elen )
m_extraAudio.AppendBuffer( (const char*)pExtra, elen );
} // end if
} // end if
else if ( oex::CStr8( "MPEG4-GENERIC" ) == pss->codecName() )
{
const char *pCfg = pss->fmtp_config();
if ( pCfg )
{ unsigned elen = 0;
unsigned char *pExtra = parseGeneralConfigStr( pCfg, elen );
if ( pExtra && elen )
m_extraAudio.AppendBuffer( (const char*)pExtra, elen );
} // end if
} // end if
// Read extradata
const char *props = pss->fmtp_spropparametersets();
if ( props )
{ oex::TList< oex::CStr8 > lst = oex::CParser::Explode( props, oexT( "," ) );
for ( oex::TList< oex::CStr8 >::iterator it; lst.Next( it ); )
{ m_extraAudio.AppendBuffer( "\x00\x00\x01", 3 );
m_extraAudio.Mem().appendString( oex::CBase64::Decode( *it ) );
} // end for
} // end if
// Set minimum rx buffer size
if ( 2000000 > m_nRxBufferSize )
m_nRxBufferSize = 2000000;
// Set rx buffer size
int sn = pss->rtpSource()->RTPgs()->socketNum();
increaseReceiveBufferTo( *m_pEnv, sn, m_nRxBufferSize );
pss->rtpSource()->setPacketReorderingThresholdTime( 2000000 );
if ( pss->codecName() )
m_sAudioCodec = oexMbToStrPtr( pss->codecName() );
if ( !m_pRtspClient->setupMediaSubsession( *pss, False, False ) )
{ setLastError( -205, sqbind::oex2std( oexMks( oexT( "setupMediaSubsession() failed : " ), oexT( " : " ), m_pEnv->getResultMsg() ) ) );
return 0;
} // end if
// Save away important audio parameters
m_nAudioNumChannels = pss->numChannels();
m_nAudioRate = pss->rtpTimestampFrequency();
m_nAudioBps = 0;
m_pAs = new CAudioSink( *m_pEnv );
if ( !m_pAs )
{ setLastError( -206, sqbind::oex2std( oexMks( oexT( "CAudioSink::createNew() failed : " ), oexT( " : " ), m_pEnv->getResultMsg() ) ) );
return 0;
} // end if
m_pAs->setDataEvent( &m_evtData );
m_pAsPss = pss;
return 1;
}
void DummySink::afterGettingFrame(unsigned frameSize, unsigned numTruncatedBytes,
struct timeval presentationTime, unsigned /*durationInMicroseconds*/) {
// We've just received a frame of data. (Optionally) print out information about it:
if(numTruncatedBytes > 0)
{
printf("============== warnning, live555 truncate %d bytes =================\n", numTruncatedBytes);
}
#ifdef DEBUG_PRINT_EACH_RECEIVED_FRAME
if (fStreamId != NULL) envir() << "Stream \"" << fStreamId << "\"; ";
envir() << fSubsession.mediumName() << "/" << fSubsession.codecName() << ":\tReceived " << frameSize << " bytes";
if (numTruncatedBytes > 0) envir() << " (with " << numTruncatedBytes << " bytes truncated)";
char uSecsStr[6+1]; // used to output the 'microseconds' part of the presentation time
sprintf(uSecsStr, "%06u", (unsigned)presentationTime.tv_usec);
envir() << ".\tPresentation time: " << (unsigned)presentationTime.tv_sec << "." << uSecsStr;
if (fSubsession.rtpSource() != NULL && !fSubsession.rtpSource()->hasBeenSynchronizedUsingRTCP()) {
envir() << "!"; // mark the debugging output to indicate that this presentation time is not RTCP-synchronized
}
envir() << "\n";
#endif
// 分辨率已经发生变化，不再上报数据
if (_drop)
{
printf("############################### drop frame\n");
return;
}
if(_cb)
{
RtspFrameInfo info;
info.videoFPS = fSubsession.videoFPS();
info.videoWidth = fSubsession.videoWidth();
info.videoHeight = fSubsession.videoHeight();
info.frequency = fSubsession.rtpTimestampFrequency();
info.channels = fSubsession.numChannels();
info.profile_level_id = fSubsession.fmtp_profile_level_id();
strncpy((char*)&(info.mediaName), fSubsession.mediumName(), sizeof(info.mediaName));
strncpy((char*)&(info.codecName), fSubsession.codecName(), sizeof(info.codecName));
info.timestamp = presentationTime;
if(fSubsession.rtpSource() != NULL && !fSubsession.rtpSource()->hasBeenSynchronizedUsingRTCP())
{
info.syncUseRTCP = false;
}
else
{
info.syncUseRTCP = true;
}
if(strcmp(fSubsession.mediumName(), "audio") == 0)
{
if (strcmp(fSubsession.codecName(), "MPEG4-GENERIC") == 0)
{
info.isHeader = 0;
_cb(_channel, frameSize, (char*)fReceiveBuffer, info);
}
else if (strcmp(fSubsession.codecName(), "L16") == 0)
{
int i = fSubsession.numChannels();
info.isHeader = 0;
_cb(_channel, frameSize, (char*)fReceiveBuffer, info);
}
}
else if(strcmp(fSubsession.mediumName(), "video") == 0)
{
if(strcmp(fSubsession.codecName(), "H264") == 0)
{
unsigned char start_code[4] = {0x00, 0x00, 0x00, 0x01};
if(!_sentHeader)
{
_sentHeader = true;
unsigned numSpropRecords;
if(fSubsession.fmtp_spropparametersets() && 0 < strlen(fSubsession.fmtp_spropparametersets()))
{
SPropRecord* sPropRecords = parseSPropParameterSets(fSubsession.fmtp_spropparametersets(), numSpropRecords);
printf("====================== proparamset: [%d]%s =================\n", numSpropRecords, fSubsession.fmtp_spropparametersets());
if(numSpropRecords > 0)
{
int headerLen = 0;
int validRecordNum = 0;
for(unsigned int i = 0; i < numSpropRecords; i++)
{
printf("spropparameter first byte = %x\n", sPropRecords[i].sPropBytes[0]);
if(((sPropRecords[i].sPropBytes[0] & 0x1f) == 7) || ((sPropRecords[i].sPropBytes[0] & 0x1f) == 8))
{
headerLen += sPropRecords[i].sPropLength;
validRecordNum += 1;
}
}
headerLen += sizeof(start_code) * validRecordNum;
char* headerData = new char[headerLen];
int offset = 0;
for(unsigned int i = 0; i < numSpropRecords; i++)
{
if(((sPropRecords[i].sPropBytes[0] & 0x1f) == 7) || ((sPropRecords[i].sPropBytes[0] & 0x1f) == 8))
{
memcpy(headerData + offset, start_code, 4);
offset += 4;
memcpy(headerData + offset, sPropRecords[i].sPropBytes, sPropRecords[i].sPropLength);
offset += sPropRecords[i].sPropLength;
}
}
uint16_t w = 0;
uint16_t h = 0;
if (H264Parse::GetResolution((uint8_t*)headerData, headerLen, &w, &h))
{
_w = w;
_h = h;
}
info.isHeader = 1;
_cb(_channel, headerLen, headerData, info);
delete [] headerData;
}
}
}
else
{
if ((fReceiveBuffer[0] & 0x1f) == 7)
{
uint16_t w = 0;
uint16_t h = 0;
if (H264Parse::GetResolution((uint8_t*)fReceiveBuffer, frameSize, &w, &h))
{
if (_w == 0 || _h == 0)
{
_w = w;
_h = h;
}
else if ((_w != w) || (_h != h))
{
printf("=====33333333========= %dx%d, %dx%d\n", _w, _h, w, h);
_drop = true;
}
}
}
}
if (!_drop)
{
info.isHeader = 0;
char* newData = new char[sizeof(start_code) + frameSize];
memcpy(newData, start_code, sizeof(start_code));
memcpy(newData + sizeof(start_code), (char*)fReceiveBuffer, frameSize);
_cb(_channel, frameSize + sizeof(start_code), newData, info);
delete [] newData;
}
}
else if(strcmp(fSubsession.codecName(), "MP4V-ES") == 0)
{
#ifdef SEND_CONFIG_HEADER
unsigned configLen;
unsigned char* configData = parseGeneralConfigStr(fSubsession.fmtp_config(), configLen);
info.isHeader = 1;
_cb(_channel, configLen, (char*)configData, info);
#endif
info.isHeader = 0;
_cb(_channel, frameSize, (char*)fReceiveBuffer, info);
}
else
{
info.isHeader = 0;
_cb(_channel, frameSize, (char*)fReceiveBuffer, info);
}
}
}
// Then continue, to request the next frame of data:
continuePlaying();
}