static gboolean read_data(gst_app_t *app)
{
GstBuffer *buffer;
guint8 *ptr;
GstFlowReturn ret;
int iret;
char *vbuf;
char *abuf;
int res_len = 0;
iret = hu_aap_recv_process ();
if (iret != 0) {
printf("hu_aap_recv_process() iret: %d\n", iret);
g_main_loop_quit(app->loop);
return FALSE;
}
/* Is there a video buffer queued? */
vbuf = vid_read_head_buf_get (&res_len);
if (vbuf != NULL) {
//buffer = gst_buffer_new();
//gst_buffer_set_data(buffer, vbuf, res_len);
buffer = gst_buffer_new_and_alloc(res_len);
memcpy(GST_BUFFER_DATA(buffer),vbuf,res_len);
ret = gst_app_src_push_buffer(app->src, buffer);
if(ret != GST_FLOW_OK){
printf("push buffer returned %d for %d bytes \n", ret, res_len);
return FALSE;
}
}
/* Is there an audio buffer queued? */
abuf = aud_read_head_buf_get (&res_len);
if (abuf != NULL) {
//buffer = gst_buffer_new();
//gst_buffer_set_data(buffer, abuf, res_len);
buffer = gst_buffer_new_and_alloc(res_len);
memcpy(GST_BUFFER_DATA(buffer),abuf,res_len);
if (res_len <= 2048 + 96)
ret = gst_app_src_push_buffer((GstAppSrc *)au1_src, buffer);
else
ret = gst_app_src_push_buffer((GstAppSrc *)aud_src, buffer);
if(ret != GST_FLOW_OK){
printf("push buffer returned %d for %d bytes \n", ret, res_len);
return FALSE;
}
}
return TRUE;
}
bool CvVideoWriter_GStreamer::writeFrame( const IplImage * image )
{
CV_FUNCNAME("CvVideoWriter_GStreamer::writerFrame");
__BEGIN__;
if (input_pix_fmt == 1) {
if (image->nChannels != 3 || image->depth != IPL_DEPTH_8U) {
CV_ERROR(CV_StsUnsupportedFormat, "cvWriteFrame() needs images with depth = IPL_DEPTH_8U and nChannels = 3.");
}
}
else if (input_pix_fmt == 0) {
if (image->nChannels != 1 || image->depth != IPL_DEPTH_8U) {
CV_ERROR(CV_StsUnsupportedFormat, "cvWriteFrame() needs images with depth = IPL_DEPTH_8U and nChannels = 1.");
}
}
else {
assert(false);
}
int size;
size = image->imageSize;
buffer = gst_buffer_new_and_alloc (size);
//gst_buffer_set_data (buffer,(guint8*)image->imageData, size);
memcpy (GST_BUFFER_DATA(buffer),image->imageData, size);
gst_app_src_push_buffer(GST_APP_SRC(source),buffer);
//gst_buffer_unref(buffer);
//buffer = 0;
__END__;
return true;
}
void MediaPlayer::cb_source_need_data(GstAppSrc *src, guint length, gpointer user_data)
{
MediaPlayer * self = reinterpret_cast<MediaPlayer*>( user_data );
qint64 totalread = 0;
if ( !self->m_mediaIODevice->atEnd() )
{
GstBuffer * buffer = gst_buffer_new_and_alloc( length );
GstMapInfo map;
gst_buffer_map( buffer, &map, GST_MAP_WRITE );
totalread = self->m_mediaIODevice->read( (char*) map.data, length );
gst_buffer_unmap( buffer, &map );
if ( totalread > 0)
{
GstFlowReturn ret = gst_app_src_push_buffer( src, buffer );
if (ret == GST_FLOW_ERROR) {
qWarning()<<"appsrc: push buffer error";
} else if (ret == GST_FLOW_FLUSHING) {
qWarning()<<"appsrc: push buffer wrong state";
}
}
}
// We need to tell GStreamer this is end of stream
if ( totalread <= 0 )
gst_app_src_end_of_stream( src );
}
//--------------------------------------------------------------
void testApp::update(){
grabber.update();
if(grabber.isFrameNew()){
GstBuffer * buffer;
if( 1 ){
unsigned char * lpBits24 = grabber.getPixels();
memcpy(pixels.getPixels(), lpBits24, 640*480*3);
//for(int i=0;i<640*480;i++){
// int nPos24 = i*3;
//
// pixels[nPos24] = lpBits24[;
// pixels[nPos24+1] = (lpBits24[i*3]+lpBits24[i*3+1]+lpBits24[i*3+2])*.3333;
// pixels[nPos24+2] = 0;
//
//}
buffer = gst_app_buffer_new (pixels.getPixels(), 640*480*3, NULL, pixels.getPixels());
}
GstFlowReturn flow_return = gst_app_src_push_buffer(gstSrc, buffer);
if (flow_return != GST_FLOW_OK) {
ofLog(OF_LOG_WARNING,"error pushing buffer");
}
gst.update();
}
grabber.draw(0,0);
}
gboolean airplayRendererFeedData(GstElement *appsrc, guint size, gpointer *object) {
AirplayRenderer *self = AIRPLAY_RENDERER(object);
AirplayRendererPrivate *priv = AIRPLAY_RENDERER_GET_PRIVATE(self);
g_print("AirplayMediaPlayer: feed data %d\n", priv->seq);
AudioPkg *pkg = g_async_queue_try_pop(priv->bufferList);
while(pkg == NULL) {
if(!priv->isRunning) return FALSE;
pkg = g_hash_table_lookup(priv->resendTable, &priv->seq);
g_print("Look up: %d %d\n", priv->seq, pkg == NULL);
if(pkg) {
priv->seq = (priv->seq + 1) % 65536;
break;
}
usleep(50000);
sched_yield();
pkg = g_async_queue_try_pop(priv->bufferList);
g_print("Sleep: %d\n", priv->seq);
}
GstBuffer *buffer = gst_buffer_new();
gst_buffer_set_data(buffer, pkg->data, pkg->length);
GST_BUFFER_SIZE(buffer) = pkg->length;
GST_BUFFER_MALLOCDATA(buffer) = pkg->data;
GST_BUFFER_DATA(buffer) = GST_BUFFER_MALLOCDATA(buffer);
gst_app_src_push_buffer((GstAppSrc *)appsrc, buffer);
g_free(pkg);
// gst_buffer_unref(buffer);
return TRUE;
}
CAMLprim value ocaml_gstreamer_appsrc_push_buffer_data(value _as, value _buf)
{
CAMLparam2(_as, _buf);
int buflen = Caml_ba_array_val(_buf)->dim[0];
appsrc *as = Appsrc_val(_as);
GstBuffer *gstbuf;
GstMapInfo map;
GstFlowReturn ret;
gboolean bret;
caml_release_runtime_system();
gstbuf = gst_buffer_new_and_alloc(buflen);
bret = gst_buffer_map(gstbuf, &map, GST_MAP_WRITE);
caml_acquire_runtime_system();
if(!bret) caml_raise_constant(*caml_named_value("gstreamer_exn_failure"));
memcpy(map.data, (unsigned char*)Caml_ba_data_val(_buf), buflen);
caml_release_runtime_system();
gst_buffer_unmap(gstbuf, &map);
ret = gst_app_src_push_buffer(as->appsrc, gstbuf);
caml_acquire_runtime_system();
if (ret != GST_FLOW_OK) caml_raise_constant(*caml_named_value("gstreamer_exn_failure"));
CAMLreturn(Val_unit);
}
void QGstAppSrc::pushDataToAppSrc()
{
if (!isStreamValid() || !m_setup)
return;
if (m_dataRequested && !m_enoughData) {
qint64 size;
if (m_dataRequestSize == (unsigned int)-1)
size = qMin(m_stream->bytesAvailable(), queueSize());
else
size = qMin(m_stream->bytesAvailable(), (qint64)m_dataRequestSize);
void *data = g_malloc(size);
GstBuffer* buffer = gst_app_buffer_new(data, size, g_free, data);
buffer->offset = m_stream->pos();
qint64 bytesRead = m_stream->read((char*)GST_BUFFER_DATA(buffer), size);
buffer->offset_end = buffer->offset + bytesRead - 1;
if (bytesRead > 0) {
m_dataRequested = false;
m_enoughData = false;
GstFlowReturn ret = gst_app_src_push_buffer (GST_APP_SRC (element()), buffer);
if (ret == GST_FLOW_ERROR) {
qWarning()<<"appsrc: push buffer error";
} else if (ret == GST_FLOW_WRONG_STATE) {
qWarning()<<"appsrc: push buffer wrong state";
} else if (ret == GST_FLOW_RESEND) {
qWarning()<<"appsrc: push buffer resend";
}
}
} else if (m_stream->atEnd()) {
sendEOS();
}
}
void AVMuxEncode::needVideoData()
{
GstFlowReturn ret;
GstBuffer *buffer;
m_videoSrcLock.lock();
if (m_videoSrcQ.empty()) {
m_videoSrcLock.unlock();
return;
}
AVMUX_QUEUEDATA *qd = m_videoSrcQ.dequeue();
QByteArray frame = qd->data;
m_lastQueuedVideoTimestamp = qd->timestamp;
m_lastQueuedVideoParam = qd->param;
delete qd;
buffer = gst_buffer_new_and_alloc(frame.length());
memcpy(GST_BUFFER_DATA(buffer), (unsigned char *)frame.data(), frame.length());
m_videoSrcLock.unlock();
ret = gst_app_src_push_buffer((GstAppSrc *)(m_appVideoSrc), buffer);
if (ret != GST_FLOW_OK) {
qDebug() << "video push error ";
}
}
void GStreamerReader::ReadAndPushData(guint aLength)
{
MediaResource* resource = mDecoder->GetResource();
NS_ASSERTION(resource, "Decoder has no media resource");
nsresult rv = NS_OK;
GstBuffer* buffer = gst_buffer_new_and_alloc(aLength);
guint8* data = GST_BUFFER_DATA(buffer);
uint32_t size = 0, bytesRead = 0;
while(bytesRead < aLength) {
rv = resource->Read(reinterpret_cast<char*>(data + bytesRead),
aLength - bytesRead, &size);
if (NS_FAILED(rv) || size == 0)
break;
bytesRead += size;
}
GST_BUFFER_SIZE(buffer) = bytesRead;
mByteOffset += bytesRead;
GstFlowReturn ret = gst_app_src_push_buffer(mSource, gst_buffer_ref(buffer));
if (ret != GST_FLOW_OK) {
LOG(PR_LOG_ERROR, ("ReadAndPushData push ret %s", gst_flow_get_name(ret)));
}
if (GST_BUFFER_SIZE (buffer) < aLength) {
/* If we read less than what we wanted, we reached the end */
gst_app_src_end_of_stream(mSource);
}
gst_buffer_unref(buffer);
}
void ofxGstRTPServer::sendAudioOut(PooledAudioFrame * pooledFrame){
GstClock * clock = gst_pipeline_get_clock(GST_PIPELINE(gst.getPipeline()));
gst_object_ref(clock);
GstClockTime now = gst_clock_get_time (clock) - gst_element_get_base_time(gst.getPipeline());
gst_object_unref (clock);
if(firstAudioFrame && !audioAutoTimestamp){
prevTimestampAudio = now;
firstAudioFrame = false;
return;
}
int size = pooledFrame->audioFrame._payloadDataLengthInSamples*2*pooledFrame->audioFrame._audioChannel;
GstBuffer * echoCancelledBuffer = gst_buffer_new_wrapped_full(GST_MEMORY_FLAG_READONLY,(void*)pooledFrame->audioFrame._payloadData,size,0,size,pooledFrame,(GDestroyNotify)&ofxWebRTCAudioPool::relaseFrame);
if(!audioAutoTimestamp){
GstClockTime duration = (pooledFrame->audioFrame._payloadDataLengthInSamples * GST_SECOND / pooledFrame->audioFrame._frequencyInHz);
GstClockTime now = prevTimestamp + duration;
GST_BUFFER_OFFSET(echoCancelledBuffer) = numFrameAudio++;
GST_BUFFER_OFFSET_END(echoCancelledBuffer) = numFrameAudio;
GST_BUFFER_DTS (echoCancelledBuffer) = now;
GST_BUFFER_PTS (echoCancelledBuffer) = now;
GST_BUFFER_DURATION(echoCancelledBuffer) = duration;
prevTimestampAudio = now;
}
GstFlowReturn flow_return = gst_app_src_push_buffer((GstAppSrc*)appSrcAudio, echoCancelledBuffer);
if (flow_return != GST_FLOW_OK) {
ofLogError(LOG_NAME) << "error pushing audio buffer: flow_return was " << flow_return;
}
}
int
main (int argc, char *argv[])
{
App *app = &s_app;
int i;
gst_init (&argc, &argv);
app->pipe = gst_pipeline_new (NULL);
g_assert (app->pipe);
app->src = gst_element_factory_make ("appsrc", NULL);
g_assert (app->src);
gst_bin_add (GST_BIN (app->pipe), app->src);
app->id = gst_element_factory_make ("identity", NULL);
g_assert (app->id);
gst_bin_add (GST_BIN (app->pipe), app->id);
app->sink = gst_element_factory_make ("appsink", NULL);
g_assert (app->sink);
gst_bin_add (GST_BIN (app->pipe), app->sink);
gst_element_link (app->src, app->id);
gst_element_link (app->id, app->sink);
gst_element_set_state (app->pipe, GST_STATE_PLAYING);
for (i = 0; i < 10; i++) {
GstBuffer *buf;
GstMapInfo map;
buf = gst_buffer_new_and_alloc (100);
gst_buffer_map (buf, &map, GST_MAP_WRITE);
memset (map.data, i, 100);
gst_buffer_unmap (buf, &map);
printf ("%d: pushing buffer for pointer %p, %p\n", i, map.data, buf);
gst_app_src_push_buffer (GST_APP_SRC (app->src), buf);
}
/* push EOS */
gst_app_src_end_of_stream (GST_APP_SRC (app->src));
/* _is_eos() does not block and returns TRUE if there is not currently an EOS
* to be retrieved */
while (!gst_app_sink_is_eos (GST_APP_SINK (app->sink))) {
GstSample *sample;
/* pull the next item, this can return NULL when there is no more data and
* EOS has been received */
sample = gst_app_sink_pull_sample (GST_APP_SINK (app->sink));
printf ("retrieved sample %p\n", sample);
if (sample)
gst_sample_unref (sample);
}
gst_element_set_state (app->pipe, GST_STATE_NULL);
return 0;
}
void VideoHttpBuffer::needData(int size)
{
Q_ASSERT(media);
/* Refactor to use gst_pad_alloc_buffer? Probably wouldn't provide any benefit. */
GstBuffer *buffer = gst_buffer_new_and_alloc(size);
int re = media->read(media->readPosition(), (char*)GST_BUFFER_DATA(buffer), size);
if (re < 0)
{
/* Error reporting is handled by MediaDownload for this case */
qDebug() << "VideoHttpBuffer: read error";
return;
}
else if (re == 0)
{
if (media->readPosition() >= media->fileSize() && media->isFinished())
{
qDebug() << "VideoHttpBuffer: end of stream";
gst_app_src_end_of_stream(m_element);
}
else
qDebug() << "VideoHttpBuffer: read aborted";
return;
}
GST_BUFFER_SIZE(buffer) = re;
GstFlowReturn flow = gst_app_src_push_buffer(m_element, buffer);
if (flow != GST_FLOW_OK)
qDebug() << "VideoHttpBuffer: Push result is" << flow;
}
void GStreamerReader::ReadAndPushData(guint aLength)
{
int64_t offset1 = mResource.Tell();
unused << offset1;
nsresult rv = NS_OK;
GstBuffer* buffer = gst_buffer_new_and_alloc(aLength);
#if GST_VERSION_MAJOR >= 1
GstMapInfo info;
gst_buffer_map(buffer, &info, GST_MAP_WRITE);
guint8 *data = info.data;
#else
guint8* data = GST_BUFFER_DATA(buffer);
#endif
uint32_t size = 0, bytesRead = 0;
while(bytesRead < aLength) {
rv = mResource.Read(reinterpret_cast<char*>(data + bytesRead),
aLength - bytesRead, &size);
if (NS_FAILED(rv) || size == 0)
break;
bytesRead += size;
}
int64_t offset2 = mResource.Tell();
unused << offset2;
#if GST_VERSION_MAJOR >= 1
gst_buffer_unmap(buffer, &info);
gst_buffer_set_size(buffer, bytesRead);
#else
GST_BUFFER_SIZE(buffer) = bytesRead;
#endif
GstFlowReturn ret = gst_app_src_push_buffer(mSource, gst_buffer_ref(buffer));
if (ret != GST_FLOW_OK) {
LOG(LogLevel::Error, "ReadAndPushData push ret %s(%d)", gst_flow_get_name(ret), ret);
}
if (NS_FAILED(rv)) {
/* Terminate the stream if there is an error in reading */
LOG(LogLevel::Error, "ReadAndPushData read error, rv=%x", rv);
gst_app_src_end_of_stream(mSource);
} else if (bytesRead < aLength) {
/* If we read less than what we wanted, we reached the end */
LOG(LogLevel::Warning, "ReadAndPushData read underflow, "
"bytesRead=%u, aLength=%u, offset(%lld,%lld)",
bytesRead, aLength, offset1, offset2);
gst_app_src_end_of_stream(mSource);
}
gst_buffer_unref(buffer);
/* Ensure offset change is consistent in this function.
* If there are other stream operations on another thread at the same time,
* it will disturb the GStreamer state machine.
*/
MOZ_ASSERT(offset1 + bytesRead == offset2);
}
FlowReturn ApplicationSource::pushBuffer(const BufferPtr & buffer)
{
if (d->appSrc()) {
return static_cast<FlowReturn>(gst_app_src_push_buffer(d->appSrc(), gst_buffer_ref(buffer)));
} else {
return FlowWrongState;
}
}
void ofxGstRTPServer::newFrameDepth(ofPixels & pixels, GstClockTime timestamp){
// here we push new depth frames in the pipeline, it's important
// to timestamp them properly so gstreamer can sync them with the
// audio.
if(!bufferPoolDepth || !appSrcDepth) return;
GstClockTime now = timestamp;
if(!depthAutoTimestamp){
if(now==GST_CLOCK_TIME_NONE){
now = getTimeStamp();
}
if(firstDepthFrame){
prevTimestampDepth = now;
firstDepthFrame = false;
return;
}
}
// get a pixels buffer from the pool and copy the passed frame into it
PooledPixels<unsigned char> * pooledPixels = bufferPoolDepth->newBuffer();
//pooledPixels->swap(pixels);
*(ofPixels*)pooledPixels=pixels;
// wrap the pooled pixels into a gstreamer buffer and pass the release
// callback so when it's not needed anymore by gst we can return it to the pool
GstBuffer * buffer;
buffer = gst_buffer_new_wrapped_full (GST_MEMORY_FLAG_READONLY,pooledPixels->getPixels(), pooledPixels->size(), 0, pooledPixels->size(), pooledPixels, (GDestroyNotify)&ofxGstBufferPool<unsigned char>::relaseBuffer);
// timestamp the buffer, right now we are using:
// timestamp = current pipeline time - base time
// duration = timestamp - previousTimeStamp
// the duration is actually the duration of the previous frame
// but should be accurate enough
if(!depthAutoTimestamp){
GST_BUFFER_OFFSET(buffer) = numFrameDepth++;
GST_BUFFER_OFFSET_END(buffer) = numFrameDepth;
GST_BUFFER_DTS (buffer) = now;
GST_BUFFER_PTS (buffer) = now;
GST_BUFFER_DURATION(buffer) = now-prevTimestampDepth;
prevTimestampDepth = now;
}
if(sendDepthKeyFrame){
emitDepthKeyFrame();
}
// finally push the buffer into the pipeline through the appsrc element
GstFlowReturn flow_return = gst_app_src_push_buffer((GstAppSrc*)appSrcDepth, buffer);
if (flow_return != GST_FLOW_OK) {
ofLogError() << "error pushing depth buffer: flow_return was " << flow_return;
}
}
void ofxGstRTPServer::newFrameDepth(ofShortPixels & pixels, GstClockTime timestamp, float pixel_size, float distance){
//unsigned long long time = ofGetElapsedTimeMicros();
// here we push new depth frames in the pipeline, it's important
// to timestamp them properly so gstreamer can sync them with the
// audio.
if(!appSrcDepth) return;
GstClockTime now = timestamp;
if(!depthAutoTimestamp){
if(now==GST_CLOCK_TIME_NONE){
now = getTimeStamp();
}
if(firstDepthFrame){
prevTimestampDepth = now;
firstDepthFrame = false;
return;
}
}
ofxDepthCompressedFrame frame = depthCompressor.newFrame(pixels,pixel_size,distance);
GstBuffer * buffer = gst_buffer_new_allocate(NULL,frame.compressedData().size()*sizeof(short),NULL);
GstMapInfo mapinfo;
gst_buffer_map(buffer,&mapinfo,GST_MAP_WRITE);
memcpy(mapinfo.data,&frame.compressedData()[0],frame.compressedData().size()*sizeof(short));
gst_buffer_unmap(buffer,&mapinfo);
// timestamp the buffer, right now we are using:
// timestamp = current pipeline time - base time
// duration = timestamp - previousTimeStamp
// the duration is actually the duration of the previous frame
// but should be accurate enough
if(!depthAutoTimestamp){
GST_BUFFER_OFFSET(buffer) = numFrameDepth++;
GST_BUFFER_OFFSET_END(buffer) = numFrameDepth;
GST_BUFFER_DTS (buffer) = now;
GST_BUFFER_PTS (buffer) = now;
GST_BUFFER_DURATION(buffer) = now-prevTimestampDepth;
prevTimestampDepth = now;
}
if(sendDepthKeyFrame){
//emitDepthKeyFrame();
}
// finally push the buffer into the pipeline through the appsrc element
GstFlowReturn flow_return = gst_app_src_push_buffer((GstAppSrc*)appSrcDepth, buffer);
if (flow_return != GST_FLOW_OK) {
ofLogError() << "error pushing depth buffer: flow_return was " << flow_return;
}
//cout << "sending depth buffer with " << pixels.getWidth() << "," << pixels.getHeight() << " csize: " << frame.compressedData().size() << endl;
//cout << ofGetElapsedTimeMicros() - time << endl;
}
/* spice_gst_decoder_queue_frame() queues the SpiceFrame for decoding and
* displaying. The steps it goes through are as follows:
*
* 1) A SpiceGstFrame is created to keep track of SpiceFrame and some additional
* metadata. The SpiceGstFrame is then pushed to the decoding_queue.
* 2) frame->data, which contains the compressed frame data, is reffed and
* wrapped in a GstBuffer which is pushed to the GStreamer pipeline for
* decoding.
* 3) As soon as the GStreamer pipeline no longer needs the compressed frame it
* will call frame->unref_data() to free it.
* 4) Once the decompressed frame is available the GStreamer pipeline calls
* new_sample() in the GStreamer thread.
* 5) new_sample() then matches the decompressed frame to a SpiceGstFrame from
* the decoding queue using the GStreamer timestamp information to deal with
* dropped frames. The SpiceGstFrame is popped from the decoding_queue.
* 6) new_sample() then attaches the decompressed frame to the SpiceGstFrame,
* pushes it to the display_queue and calls schedule_frame().
* 7) schedule_frame() then uses gstframe->frame->mm_time to arrange for
* display_frame() to be called, in the main thread, at the right time for
* the next frame.
* 8) display_frame() pops the first SpiceGstFrame from the display_queue and
* calls stream_display_frame().
* 9) display_frame() then frees the SpiceGstFrame, which frees the SpiceFrame
* and decompressed frame with it.
*/
static gboolean spice_gst_decoder_queue_frame(VideoDecoder *video_decoder,
SpiceFrame *frame, int latency)
{
SpiceGstDecoder *decoder = (SpiceGstDecoder*)video_decoder;
if (frame->size == 0) {
SPICE_DEBUG("got an empty frame buffer!");
frame->free(frame);
return TRUE;
}
if (frame->mm_time < decoder->last_mm_time) {
SPICE_DEBUG("new-frame-time < last-frame-time (%u < %u):"
" resetting stream",
frame->mm_time, decoder->last_mm_time);
/* Let GStreamer deal with the frame anyway */
}
decoder->last_mm_time = frame->mm_time;
if (latency < 0 &&
decoder->base.codec_type == SPICE_VIDEO_CODEC_TYPE_MJPEG) {
/* Dropping MJPEG frames has no impact on those that follow and
* saves CPU so do it.
*/
SPICE_DEBUG("dropping a late MJPEG frame");
frame->free(frame);
return TRUE;
}
if (decoder->pipeline == NULL) {
/* An error occurred, causing the GStreamer pipeline to be freed */
spice_warning("An error occurred, stopping the video stream");
return FALSE;
}
/* ref() the frame data for the buffer */
frame->ref_data(frame->data_opaque);
GstBuffer *buffer = gst_buffer_new_wrapped_full(GST_MEMORY_FLAG_PHYSICALLY_CONTIGUOUS,
frame->data, frame->size, 0, frame->size,
frame->data_opaque, frame->unref_data);
GST_BUFFER_DURATION(buffer) = GST_CLOCK_TIME_NONE;
GST_BUFFER_DTS(buffer) = GST_CLOCK_TIME_NONE;
GST_BUFFER_PTS(buffer) = gst_clock_get_time(decoder->clock) - gst_element_get_base_time(decoder->pipeline) + ((uint64_t)MAX(0, latency)) * 1000 * 1000;
g_mutex_lock(&decoder->queues_mutex);
g_queue_push_tail(decoder->decoding_queue, create_gst_frame(buffer, frame));
g_mutex_unlock(&decoder->queues_mutex);
if (gst_app_src_push_buffer(decoder->appsrc, buffer) != GST_FLOW_OK) {
SPICE_DEBUG("GStreamer error: unable to push frame of size %u", frame->size);
stream_dropped_frame_on_playback(decoder->base.stream);
}
return TRUE;
}
int
main (int argc, char *argv[])
{
App *app = &s_app;
int i;
gst_init (&argc, &argv);
app->pipe = gst_pipeline_new (NULL);
g_assert (app->pipe);
app->src = gst_element_factory_make ("appsrc", NULL);
g_assert (app->src);
gst_bin_add (GST_BIN (app->pipe), app->src);
app->id = gst_element_factory_make ("identity", NULL);
g_assert (app->id);
gst_bin_add (GST_BIN (app->pipe), app->id);
app->sink = gst_element_factory_make ("appsink", NULL);
g_assert (app->sink);
gst_bin_add (GST_BIN (app->pipe), app->sink);
gst_element_link (app->src, app->id);
gst_element_link (app->id, app->sink);
gst_element_set_state (app->pipe, GST_STATE_PLAYING);
for (i = 0; i < 10; i++) {
GstBuffer *buf;
void *data;
data = malloc (100);
memset (data, i, 100);
buf = gst_app_buffer_new (data, 100, dont_eat_my_chicken_wings, data);
printf ("%d: creating buffer for pointer %p, %p\n", i, data, buf);
gst_app_src_push_buffer (GST_APP_SRC (app->src), buf);
}
gst_app_src_end_of_stream (GST_APP_SRC (app->src));
while (!gst_app_sink_is_eos (GST_APP_SINK (app->sink))) {
GstBuffer *buf;
buf = gst_app_sink_pull_buffer (GST_APP_SINK (app->sink));
printf ("retrieved buffer %p\n", buf);
gst_buffer_unref (buf);
}
gst_element_set_state (app->pipe, GST_STATE_NULL);
return 0;
}
static int gst_video_push(struct videnc_state *st, const uint8_t *src,
size_t size)
{
GstBuffer *buffer;
int ret = 0;
if (!st) {
return EINVAL;
}
if (!size) {
warning("gst_video: push: eos returned %d at %d\n",
ret, __LINE__);
gst_app_src_end_of_stream((GstAppSrc *)st->source);
return ret;
}
/* Wait "start feed". */
pthread_mutex_lock(&st->mutex);
if (st->bwait) {
#define WAIT_TIME_SECONDS 5
struct timespec ts;
struct timeval tp;
gettimeofday(&tp, NULL);
ts.tv_sec = tp.tv_sec;
ts.tv_nsec = tp.tv_usec * 1000;
ts.tv_sec += WAIT_TIME_SECONDS;
/* Wait. */
ret = pthread_cond_timedwait(&st->wait, &st->mutex, &ts);
if (ETIMEDOUT == ret) {
warning("gst_video: Raw frame is lost"
" because of timeout\n");
return ret;
}
}
pthread_mutex_unlock(&st->mutex);
/* Create a new empty buffer */
buffer = gst_buffer_new();
GST_BUFFER_MALLOCDATA(buffer) = (guint8 *)src;
GST_BUFFER_SIZE(buffer) = (guint)size;
GST_BUFFER_DATA(buffer) = GST_BUFFER_MALLOCDATA(buffer);
ret = gst_app_src_push_buffer((GstAppSrc *)st->source, buffer);
if (ret != GST_FLOW_OK) {
warning("gst_video: push buffer returned"
" %d for %d bytes \n", ret, size);
return ret;
}
return ret;
}
void VideoReceiver::consumeVideo(QByteArray *media)
{
qDebug() << "In" << __FUNCTION__;
GstBuffer *buffer = gst_buffer_new_and_alloc(media->length());
// FIXME: zero copy?
memcpy(GST_BUFFER_DATA(buffer), media->data(), media->length());
if (gst_app_src_push_buffer(GST_APP_SRC(source), buffer) != GST_FLOW_OK) {
qWarning("Error with gst_app_src_push_buffer");
}
}
void
shmdata_any_writer_push_data (shmdata_any_writer_t * context,
void *data,
int size,
unsigned long long timestamp,
void (*done_with_data) (void *),
void *user_data)
{
GstBuffer *buf;
buf = gst_app_buffer_new (data, size, done_with_data, user_data);
GST_BUFFER_TIMESTAMP (buf) = (GstClockTime) (timestamp);
gst_app_src_push_buffer (GST_APP_SRC (context->src_), buf);
}
extern int
feed_buffer_to_gst (const char *audio, size_t b_len, GNUNET_gstData * d)
{
GstBuffer *b;
gchar *bufspace;
GstFlowReturn flow;
GNUNET_log (GNUNET_ERROR_TYPE_DEBUG,
"Feeding %u bytes to GStreamer\n",
(unsigned int) b_len);
bufspace = g_memdup (audio, b_len);
b = gst_buffer_new_wrapped (bufspace, b_len);
if (NULL == b)
{
GNUNET_log (GNUNET_ERROR_TYPE_DEBUG,
"Failed to wrap a buffer\n");
g_free (bufspace);
return GNUNET_SYSERR;
}
if (GST_APP_SRC(d->appsrc) == NULL)
exit(10);
flow = gst_app_src_push_buffer (GST_APP_SRC(d->appsrc), b);
/* They all return GNUNET_OK, because currently player stops when
* data stops coming. This might need to be changed for the player
* to also stop when pipeline breaks.
*/
switch (flow)
{
case GST_FLOW_OK:
GNUNET_log (GNUNET_ERROR_TYPE_DEBUG,
"Fed %u bytes to the pipeline\n",
(unsigned int) b_len);
break;
case GST_FLOW_FLUSHING:
/* buffer was dropped, because pipeline state is not PAUSED or PLAYING */
GNUNET_log (GNUNET_ERROR_TYPE_INFO,
"Dropped a buffer\n");
break;
case GST_FLOW_EOS:
/* end of stream */
GNUNET_log (GNUNET_ERROR_TYPE_INFO,
"EOS\n");
break;
default:
GNUNET_log (GNUNET_ERROR_TYPE_WARNING,
"Unexpected push result\n");
break;
}
return GNUNET_OK;
}
CAMLprim value ocaml_gstreamer_appsrc_push_buffer(value _as, value _buf)
{
CAMLparam2(_as, _buf);
appsrc *as = Appsrc_val(_as);
GstBuffer *gstbuf = Buffer_val(_buf);
GstFlowReturn ret;
caml_release_runtime_system();
/* The reference will be eaten by push_buffer */
gst_buffer_ref(gstbuf);
ret = gst_app_src_push_buffer(as->appsrc, gstbuf);
caml_acquire_runtime_system();
if (ret != GST_FLOW_OK) caml_raise_constant(*caml_named_value("gstreamer_exn_failure"));
CAMLreturn(Val_unit);
}
static GstFlowReturn
on_new_sample_from_source (GstAppSink * elt, gpointer user_data)
{
ProgramData *data = (ProgramData *) user_data;
GstSample *sample;
GstBuffer *buffer;
GstElement *source;
sample = gst_app_sink_pull_sample (GST_APP_SINK (elt));
buffer = gst_sample_get_buffer (sample);
source = gst_bin_get_by_name (GST_BIN (data->sink), "testsource");
gst_app_src_push_buffer (GST_APP_SRC (source), gst_buffer_ref (buffer));
gst_sample_unref (sample);
g_object_unref (source);
return GST_FLOW_OK;
}
/* C920 Live Src - Handle a H.264 Buffer
* -------------------------------------
* Callback that's called from the device. Use the data and length to create a GstBuffer
*/
static void c920_live_src_handle_buffer(gconstpointer data, guint length, gpointer userdata)
{
g_return_if_fail(C920_IS_LIVE_SRC(userdata));
C920LiveSrc *self = C920_LIVE_SRC(userdata);
if (!self->priv->queue_full)
{
GstBuffer *buffer = gst_buffer_new();
if (buffer)
{
buffer->size = length;
buffer->malloc_data = buffer->data = g_malloc(length);
memcpy(buffer->malloc_data, data, length);
gst_app_src_push_buffer(GST_APP_SRC(self), buffer);
}
}
}
static void udp_streaming (Encoder *encoder, GstBuffer *buffer)
{
gsize buffer_size;
gssize offset;
GstFlowReturn ret;
offset = 0;
buffer_size = gst_buffer_get_size (buffer);
while (buffer_size != 0) {
if ((encoder->cache_size == 0) && (buffer_size < 1316)) {
encoder->cache_7x188 = gst_buffer_copy_region (buffer, GST_BUFFER_COPY_MEMORY, offset, buffer_size);
encoder->cache_size = buffer_size;
break;
} else if (encoder->cache_size == 0) {
/* buffer_size >= 1316 */
encoder->cache_7x188 = gst_buffer_copy_region (buffer, GST_BUFFER_COPY_MEMORY, offset, 1316);
offset += 1316;
buffer_size -= 1316;
} else if (encoder->cache_size + buffer_size >= 1316) {
gsize size;
gst_buffer_ref (buffer);
size = 1316 - encoder->cache_size;
encoder->cache_7x188 = gst_buffer_append_region (encoder->cache_7x188, buffer, offset, size);
offset += 1316 - encoder->cache_size;
buffer_size -= 1316 - encoder->cache_size;
encoder->cache_size = 0;
} else {
/* encoder->cache_size + buffer_size < 1316 */
gst_buffer_ref (buffer);
encoder->cache_7x188 = gst_buffer_append_region (encoder->cache_7x188, buffer, offset, buffer_size);
encoder->cache_size += buffer_size;
break;
}
ret = gst_app_src_push_buffer ((GstAppSrc *)encoder->appsrc, encoder->cache_7x188);
if (ret != GST_FLOW_OK) {
GST_ERROR ("appsrc push buffer failure, return %s.", gst_flow_get_name (ret));
gst_buffer_unref (encoder->cache_7x188);
}
encoder->cache_size = 0;
}
}
// FIXME: Use gst_app_src_push_sample() instead when we switch to the appropriate GStreamer version.
static GstFlowReturn pushSample(GstAppSrc* appsrc, GstSample* sample)
{
g_return_val_if_fail(GST_IS_SAMPLE(sample), GST_FLOW_ERROR);
GstCaps* caps = gst_sample_get_caps(sample);
if (caps)
gst_app_src_set_caps(appsrc, caps);
else
GST_WARNING_OBJECT(appsrc, "received sample without caps");
GstBuffer* buffer = gst_sample_get_buffer(sample);
if (UNLIKELY(!buffer)) {
GST_WARNING_OBJECT(appsrc, "received sample without buffer");
return GST_FLOW_OK;
}
// gst_app_src_push_buffer() steals the reference, we need an additional one.
return gst_app_src_push_buffer(appsrc, gst_buffer_ref(buffer));
}
void
shmdata_any_writer_push_data_with_duration (shmdata_any_writer_t * context,
void *data,
int size,
unsigned long long timestamp,
unsigned long long duration,
unsigned long long offset,
unsigned long long offset_end,
void (*done_with_data) (void *),
void *user_data)
{
GstBuffer *buf;
buf = gst_app_buffer_new (data, size, done_with_data, user_data);
GST_BUFFER_TIMESTAMP (buf) = (GstClockTime) (timestamp);
GST_BUFFER_DURATION (buf) = (GstClockTime) (duration);
GST_BUFFER_OFFSET (buf) = (GstClockTime) (offset);
GST_BUFFER_OFFSET_END (buf) = (GstClockTime) (offset_end);
gst_app_src_push_buffer (GST_APP_SRC (context->src_), buf);
}
bool Player::Backend::push_data(Packet& data, Offset current_offset) {
GstBuffer * buffer = gst_buffer_new_allocate(
nullptr, data.size()*Config::word_size, nullptr);
Offset packet_size = Offset(data.size());
GST_BUFFER_PTS(buffer) = current_offset.to_time(Config::sample_rate).value();
GST_BUFFER_DURATION(buffer) = packet_size.to_time(Config::sample_rate).value();
GST_BUFFER_OFFSET(buffer) = current_offset.value();
GST_BUFFER_OFFSET_END(buffer) = (current_offset+packet_size).value();
auto size = data.size() * Config::word_size;
auto rsize = gst_buffer_fill(buffer, 0, static_cast<void *>(data.data()), size);
assert(size==rsize);
auto ret = gst_app_src_push_buffer(GST_APP_SRC(appsrc_), buffer);
if ( ret != GST_FLOW_OK) {
throw BadFlowException("bad flow while pushing buffer");
}
return true;
}
static void
symmetry_test_assert_passthrough (SymmetryTest * st, GstBuffer * in)
{
gpointer copy;
gsize data_size;
GstSample *out;
gst_buffer_extract_dup (in, 0, -1, ©, &data_size);
fail_unless (gst_app_src_push_buffer (st->sink_src, in) == GST_FLOW_OK);
in = NULL;
out = gst_app_sink_pull_sample (st->src_sink);
fail_unless (out != NULL);
fail_unless (gst_buffer_get_size (gst_sample_get_buffer (out)) == data_size);
fail_unless (gst_buffer_memcmp (gst_sample_get_buffer (out), 0, copy,
data_size) == 0);
g_free (copy);
gst_sample_unref (out);
}
