void FileSink::afterGettingFrame(unsigned frameSize,
unsigned numTruncatedBytes,
struct timeval presentationTime) {
if (numTruncatedBytes > 0) {
envir() << "FileSink::afterGettingFrame(): The input frame data was too large for our buffer size ("
<< fBufferSize << "). "
<< numTruncatedBytes << " bytes of trailing data was dropped! Correct this by increasing the \"bufferSize\" parameter in the \"createNew()\" call to at least "
<< fBufferSize + numTruncatedBytes << "\n";
}
addData(fBuffer, frameSize, presentationTime);
if (fOutFid == NULL || fflush(fOutFid) == EOF) {
// The output file has closed. Handle this the same way as if the input source had closed:
if (fSource != NULL) fSource->stopGettingFrames();
onSourceClosure();
return;
}
if (fPerFrameFileNameBuffer != NULL) {
if (fOutFid != NULL) { fclose(fOutFid); fOutFid = NULL; }
}
// Then try getting the next frame:
continuePlaying();
}
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:
#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: " << (int)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
#ifdef DEBUG_PRINT_NPT
envir() << "\tNPT: " << fSubsession.getNormalPlayTime(presentationTime)<< "\n";
#endif
//envir()<<"*";
// Then continue, to request the next frame of data:
continuePlaying();
}
void afterGettingFrame(unsigned frameSize, unsigned numTruncatedBytes,
struct timeval presentationTime, unsigned durationInMicroseconds)
{
if (fStreamId != NULL) envir() << "Stream \"" << fStreamId << "\"; ";
envir() << fSubsession.mediumName() << "/" << fSubsession.codecName() << ":\tReceived " << frameSize << " bytes";
QByteArray frameBuffer((char*)fReceiveBuffer, frameSize);
//插入SPS PPS才能让H264解码器正确解码
QByteArray sps = sprop_parameter_sets;
QByteArray extraData;
QList<QByteArray> recodList = sps.split(',');
for (int i = 0; i < recodList.size(); ++i)
{
extraData.append(char(0x00));
extraData.append(char(0x00));
extraData.append(char(0x00));
extraData.append(char(0x01));
extraData += QByteArray::fromBase64(recodList.at(i));
}
QByteArray endMark = QByteArray(4, 0);
endMark[3] = 0x01;
frameBuffer.insert(0, extraData);
frameBuffer.insert(extraData.size(), endMark);
m_ffmpeg->decodeFrame((uint8_t*)frameBuffer.data(), frameBuffer.size(), presentationTime.tv_sec, presentationTime.tv_usec);
// Then continue, to request the next frame of data:
continuePlaying();
}
void CMemoryStreamSink::afterGettingFrame1(unsigned frameSize,struct timeval presentationTime)
{
CAutoLock BufferLock(&m_BufferLock);
OnRawData(fBuffer, frameSize);
//addData(fBuffer, frameSize, presentationTime);
// Then try getting the next frame:
continuePlaying();
}
void MFSD_DummySink::afterGettingFrame1() {
if (fReturnFirstSeenCode && fOurDemux.lastSeenSCR().isValid) {
// We were asked to return the first SCR that we saw, and we've seen one,
// so we're done. (Handle this as if the input source had closed.)
onSourceClosure(this);
return;
}
continuePlaying();
}
void DefaultSink::onAfterGettingFrame(unsigned frame_size,
unsigned truncated_bytes,
struct timeval const & presentation_time,
unsigned UNUSED_PARAM(duration_in_microseconds))
{
crLogIfD(_verbose, getFrameInfo(frame_size, truncated_bytes, presentation_time));
if (!_have_written_first_frame) {
// If we have NAL units encoded in "sprop parameter strings",
// prepend these to the file:
for (auto & param : _sprop_parameter_sets) {
unsigned int sprop_records_size = 0;
// Returns the binary value of each 'parameter set' specified in a "sprop-parameter-sets" string
// (in the SDP description for a H.264/RTP stream).
//
// The value is returned as an array (length "numSPropRecords") of "SPropRecord"s.
// This array is dynamically allocated by this routine, and must be delete[]d by the caller.
SPropRecord * sprop_records = parseSPropParameterSets(param.data(), sprop_records_size);
for (unsigned int i = 0; i < sprop_records_size; ++i) {
write(NAL_START_CODE, sizeof(NAL_START_CODE), presentation_time);
write(sprop_records[i].sPropBytes, sprop_records[i].sPropLength, presentation_time);
}
delete [] sprop_records;
}
_have_written_first_frame = true;
}
if (truncated_bytes > 0) {
auto const BUFFER_SIZE = _receive_buffer.size();
crLogW("DefaultSink::onAfterGettingFrame() The input frame data was too large for our buffer size ({})"
"{}bytes of trailing data was dropped!"
"Correct this by increasing the 'bufferSize' parameter in the 'createNew()' call to at least {}",
BUFFER_SIZE, truncated_bytes, BUFFER_SIZE + truncated_bytes);
}
// Write the input data to the file, with the start code in front:
write(NAL_START_CODE, sizeof(NAL_START_CODE), presentation_time);
write(_receive_buffer.data(), frame_size, presentation_time);
if (isClosed()) {
// The output file has closed.
// Handle this the same way as if the input source had closed:
if (fSource != nullptr) {
fSource->stopGettingFrames();
}
onSourceClosure();
return;
}
// Then continue, to request the next frame of data:
continuePlaying();
}
void QueueSink::afterGettingFrame(unsigned frameSize, struct timeval presentationTime)
{
if (frame != NULL){
frame->setLength(frameSize);
frame->newOriginTime();
frame->setPresentationTime(std::chrono::system_clock::now());
frame->setSequenceNumber(++seqNum);
fWriter->addFrame();
}
continuePlaying();
}
Boolean AVIFileSink::startPlaying(afterPlayingFunc* afterFunc,
void* afterClientData) {
// Make sure we're not already being played:
if (fAreCurrentlyBeingPlayed) {
envir().setResultMsg("This sink has already been played");
return False;
}
fAreCurrentlyBeingPlayed = True;
fAfterFunc = afterFunc;
fAfterClientData = afterClientData;
return continuePlaying();
}
void CH264Frame::afterGettingFrame(unsigned frameSize,
unsigned numTruncatedBytes,
timeval presentationTime)
{
static int nFrameTotailNum = 100;
// 判断是否缓冲区太小
if (numTruncatedBytes > 0)
{
envir() << "H264Frame::afterGettingFrame(): The input frame data was too large for our buffer size ("
<< nBufferSize << "). "
<< numTruncatedBytes << " bytes of trailing data was dropped! Correct this by increasing the \"bufferSize\" parameter in the \"createNew()\" call to at least "
<< nBufferSize + numTruncatedBytes << "\n";
}
// 处理帧数据，然后调用回调函数,最后清空缓冲区
bool bIFrame = false;
nUsedBuffSize += frameSize;
if (((pWriteBuffer[0] & I_FRAME_HEAD_NO) == I_FRAME_HEAD_FIRST)
|| ((pWriteBuffer[0] & I_FRAME_HEAD_NO) == I_FRAME_HEAD_SECOND))
{
pWriteBuffer += frameSize; // 直到65(即I帧)之后才写入文件
}
else
{
if ((pWriteBuffer[0] & I_FRAME_HEAD_NO) == I_FRAME_HEAD_THIRD)
{
bIFrame = true; // I帧
}
++nFrameNum;
if (NULL != m_pFunCallBack)
{
m_pFunCallBack(fBuffer, nUsedBuffSize, bIFrame, nFrameNum, presentationTime, false, pFunCallBackData);
}
Clean();
}
// 测试主动暂停
// if (--nFrameTotailNum < 1)
// {
// this->stopPlaying();
// }
continuePlaying();
}
void ProxyMediaSink::afterGettingFrame(unsigned frameSize, unsigned numTruncatedBytes,
struct timeval presentationTime,
unsigned durationInMicroseconds)
{
if (numTruncatedBytes == 0)
{
bool isRtcpSynced =
_subsession.rtpSource() && _subsession.rtpSource()->hasBeenSynchronizedUsingRTCP();
_mediaPacketQueue.push(
MediaPacketSample(_receiveBuffer, frameSize, presentationTime, isRtcpSynced));
}
else
{
}
continuePlaying();
}
void ssc::ConsoleDataSink::afterGettingFrame(unsigned int frameSize, unsigned int numTruncatedBytes,
timeval presentationTime, unsigned int durationInMicroseconds)
{
envir() << mSubSession.mediumName() << "/" << mSubSession.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
std::snprintf(uSecsStr, sizeof(uSecsStr), "%06u", (unsigned)presentationTime.tv_usec);
envir() << ".\tPresentation time: " << (int)presentationTime.tv_sec << "." << uSecsStr << "\n";
if (mSubSession.rtpSource() && !mSubSession.rtpSource()->hasBeenSynchronizedUsingRTCP()) {
envir() << "!"; // mark the debugging output to indicate that this presentation time is not RTCP-synchronized
}
// Then continue, to request the next frame of data
continuePlaying();
}
void CH264StreamDecodeSink::afterGettingFrame1(unsigned frameSize, timeval presentationTime)
{
// if ( !log_file )
// {
// log_file = fopen(".\\lzxlog.txt", "w");
// }
// if ( log_file )
// {
//// fprintf( log_file, "afterGettingFrame1\n" );
// }
// 处理数据。
m_nDataLen = frameSize + ( m_pData - m_pBuffer );
m_timeStamp = presentationTime;
DecodeData();
continuePlaying();
}
Boolean MediaSink::startPlaying(MediaSource& source,
afterPlayingFunc* afterFunc,
void* afterClientData) {
// Make sure we're not already being played:
if (fSource != NULL) {
envir().setResultMsg("This sink is already being played");
return False;
}
// Make sure our source is compatible:
if (!sourceIsCompatibleWithUs(source)) {
envir().setResultMsg("MediaSink::startPlaying(): source is not compatible!");
return False;
}
fSource = (FramedSource*)&source;
fAfterFunc = afterFunc;
fAfterClientData = afterClientData;
return continuePlaying();
}
void HTTPSink::afterGettingFrame1(unsigned frameSize,
struct timeval /*presentationTime*/) {
// Write the data back to our client socket (if we have one):
if (fClientSocket >= 0 && isUseableFrame(fBuffer, frameSize)) {
int sendResult
= send(fClientSocket, (char*)(&fBuffer[0]), frameSize, 0);
if (sendResult < 0) {
int err = envir().getErrno();
if (err != EWOULDBLOCK) {
// The client appears to have gone; close him down,
// and consider ourselves done:
ourOnSourceClosure(this);
return;
}
}
}
// Then try getting the next frame:
continuePlaying();
}
void FileSink::afterGettingFrame1(unsigned frameSize,
struct timeval presentationTime) {
addData(fBuffer, frameSize, presentationTime);
if (fOutFid == NULL || fflush(fOutFid) == EOF) {
// The output file has closed. Handle this the same way as if the
// input source had closed:
onSourceClosure(this);
stopPlaying();
return;
}
if (fPerFrameFileNameBuffer != NULL) {
if (fOutFid != NULL) { fclose(fOutFid); fOutFid = NULL; }
}
// Then try getting the next frame:
continuePlaying();
}
// If you don't want to see debugging output for each received frame, then comment out the following line:
//#define DEBUG_PRINT_EACH_RECEIVED_FRAME 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:
#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: " << (int)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
}
#ifdef DEBUG_PRINT_NPT
envir() << "\tNPT: " << fSubsession.getNormalPlayTime(presentationTime);
#endif
envir() << "\n";
#endif
calcNPTandInform();
fAimerClient->m_nRecvBytes += frameSize;
fAimerClient->m_nRecvBytesSec += frameSize;
if ( strcmp( fSubsession.mediumName(), "VIDEO" ) == 0 ||
strcmp( fSubsession.mediumName(), "video" ) == 0 ) {
#if 0
LARGE_INTEGER stfreq;
QueryPerformanceFrequency(&stfreq);
static double m_dfFreq = (double)stfreq.QuadPart;
LARGE_INTEGER startTime;
static LONGLONG m_nStartTime = 0;
static LONGLONG nNowTime = 0;
QueryPerformanceCounter(&startTime);
m_nStartTime = (LONGLONG)startTime.QuadPart;
double dfMinus = (double)(m_nStartTime - nNowTime);
LONGLONG nMs = (uint64_t)((dfMinus / m_dfFreq) * 1000);
fprintf(stderr, "diif = %lld\n", nMs);
nNowTime = m_nStartTime;
#endif
fAimerClient->m_nRecvVideoBytes += frameSize;
vector<IDataSink *>::iterator iter = fAimerClient->m_aVecDataSink[STREAM_VIDEO].begin();
for ( ; iter != fAimerClient->m_aVecDataSink[STREAM_VIDEO].end(); ++iter ) {
(*iter)->SendData( fReceiveBuffer, frameSize, presentationTime,
fSubsession.mediumName(), fSubsession.codecName() );
}
//fprintf(stderr, "Video time %ds, %dms\n", presentationTime.tv_sec, (presentationTime.tv_usec / 1000));
}
if ( strcmp( fSubsession.mediumName(), "AUDIO" ) == 0 ||
strcmp( fSubsession.mediumName(), "audio" ) == 0 ) {
vector<IDataSink *>::iterator iter = fAimerClient->m_aVecDataSink[STREAM_AUDIO].begin();
for ( ; iter != fAimerClient->m_aVecDataSink[STREAM_AUDIO].end(); ++iter ) {
(*iter)->SendData( fReceiveBuffer, frameSize, presentationTime,
fSubsession.mediumName(), fSubsession.codecName(), fSubsession.fmtp_config() );
}
//fprintf(stderr, "Audio time %ds, %dms\n", presentationTime.tv_sec, (presentationTime.tv_usec / 1000));
}
// Then continue, to request the next frame of data:
continuePlaying();
}
void StreamMediaSink::afterGettingFrame(unsigned frameSize, unsigned numTruncatedBytes, struct timeval presentationTime, unsigned /*durationInMicroseconds*/)
{
// We've just received a frame of data. (Optionally) print out information about it:
#ifdef DEBUG_PRINT_EACH_RECEIVED_FRAME
if (m_fStreamId != NULL) {
envir() << "Stream \"" << m_fStreamId << "\"; ";
}
envir() << m_fSubsession.mediumName() << "/" << m_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: " << (int) presentationTime.tv_sec << "." << uSecsStr;
if (m_fSubsession.rtpSource() != NULL && !m_fSubsession.rtpSource()->hasBeenSynchronizedUsingRTCP()) {
envir() << "!"; // mark the debugging output to indicate that this presentation time is not RTCP-synchronized
}
#ifdef DEBUG_PRINT_NPT
envir() << "\tNPT: " << m_fSubsession.getNormalPlayTime(presentationTime);
#endif
envir() << "\n";
#endif
m_avPacket.size = frameSize + 4;
m_avPacket.data = m_buffer;
int gotFrame = 0;
int len = 0;
while (m_avPacket.size > 0) {
len = avcodec_decode_video2(m_avCodecContext, m_avFrame, &gotFrame, &m_avPacket);
if (len < 0) {
break;
}
if (gotFrame) {
envir() << "Decoded Frame: " << ++m_idx << " Picture Type: " << av_get_picture_type_char(m_avFrame->pict_type) << " Key Frame: " << m_avFrame->key_frame << "\n";
envir() << "showFrame: " << showFrame() << "\n";
SDL_PollEvent(&m_event);
switch (m_event.type) {
case SDL_QUIT:
SDL_Quit();
exit(0);
break;
default:
break;
}
#if defined(WRITE_RAW)
if (m_avFrame->key_frame) {
writeRaw(m_idx);
}
#endif
#if defined(WRITE_JPEG)
//if (m_avFrame->pict_type == AV_PICTURE_TYPE_I) {
writeJPEG(m_idx);
//}
#endif
}
if (m_avPacket.data) {
m_avPacket.size -= len;
m_avPacket.data += len;
}
}
// Then continue, to request the next frame of data:
continuePlaying();
}
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();
}
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:
#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: " << (int)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
}
#ifdef DEBUG_PRINT_NPT
envir() << "\tNPT: " << fSubsession.getNormalPlayTime(presentationTime);
#endif
envir() << "\n";
#endif
if (strcmp(fSubsession.codecName(),"H264") == 0) {
avpkt.data = fReceiveBufferAV;
// r2sprop();
// r2sprop2();
// avpkt.size = (int)fReceiveBuffer[0];
avpkt.size = frameSize + 4;
// avpkt.size = frameSize;
if (avpkt.size != 0) {
memcpy (fReceiveBufferAV + 4, fReceiveBuffer, frameSize);
avpkt.data = fReceiveBufferAV; //+2;
// avpkt.data = fReceiveBuffer; //+2;
len = avcodec_decode_video2 (c, picture, &got_picture, &avpkt);
if (len < 0) {
envir() << "Error while decoding frame" << frame;
// exit(6);
}
if (got_picture) {
// do something with it
SDL_LockYUVOverlay(bmp);
AVPicture pict;
pict.data[0] = bmp->pixels[0];
pict.data[1] = bmp->pixels[2];
pict.data[2] = bmp->pixels[1];
pict.linesize[0] = bmp->pitches[0];
pict.linesize[1] = bmp->pitches[2];
pict.linesize[2] = bmp->pitches[1];
struct SwsContext *sws;
sws = sws_getContext(
c->width,
c->height,
PIX_FMT_YUV420P,
c->width,
c->height,
PIX_FMT_YUV420P,
SWS_BICUBIC,
NULL,
NULL,
NULL
);
sws_scale(
sws,
picture->data,
picture->linesize,
0,
c->height,
pict.data,
pict.linesize
);
SDL_UnlockYUVOverlay(bmp);
rect.x = 0;
rect.y = 0;
rect.w = c->width;
rect.h = c->height;
SDL_DisplayYUVOverlay(bmp, &rect);
/*
char fname[256]={0};
sprintf(fname, "OriginalYUV%d.pgm",frame);
pgm_save (
picture->data[0],
picture->linesize[0],
c->width,
c->height,
fname
);
*/
frame ++;
} else {
envir() << "no picture :( !\n";
}
}
}
// Then continue, to request the next frame of data:
continuePlaying();
}
