static void
gst_app_sink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAppSink *appsink = GST_APP_SINK_CAST (object);
switch (prop_id) {
case PROP_CAPS:
gst_app_sink_set_caps (appsink, gst_value_get_caps (value));
break;
case PROP_EMIT_SIGNALS:
gst_app_sink_set_emit_signals (appsink, g_value_get_boolean (value));
break;
case PROP_MAX_BUFFERS:
gst_app_sink_set_max_buffers (appsink, g_value_get_uint (value));
break;
case PROP_DROP:
gst_app_sink_set_drop (appsink, g_value_get_boolean (value));
break;
case PROP_WAIT_ON_EOS:
gst_app_sink_set_wait_on_eos (appsink, g_value_get_boolean (value));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
CAMLprim value ocaml_gstreamer_appsink_emit_signals(value _as)
{
CAMLparam0();
appsink *as = Appsink_val(_as);
caml_release_runtime_system();
gst_app_sink_set_emit_signals(as->appsink, TRUE);
caml_acquire_runtime_system();
CAMLreturn(Val_unit);
}
Capture::Capture() {
// prepare the pipeline
pipeline = gst_pipeline_new("xvoverlay");
GstElement *src = gst_element_factory_make("videotestsrc", NULL);
GstElement *conv = gst_element_factory_make("videoconvert", NULL);
GstElement *sink = gst_element_factory_make("appsink", NULL);
gst_bin_add_many(GST_BIN(pipeline), src, conv, sink, NULL);
gst_element_link_many(src, conv, sink, NULL);
gst_app_sink_set_emit_signals(GST_APP_SINK(sink), TRUE);
g_signal_connect(sink, "new-sample", G_CALLBACK(newSample), (gpointer)this);
// run the pipeline
GstStateChangeReturn sret = gst_element_set_state(pipeline, GST_STATE_PLAYING);
g_print("Playing...\n");
if (sret == GST_STATE_CHANGE_FAILURE) {
gst_element_set_state(pipeline, GST_STATE_NULL);
gst_object_unref(pipeline);
// Exit application
QTimer::singleShot(0, QApplication::activeWindow(), SLOT(quit()));
}
}
/*!
* \brief OpenIMAJCapGStreamer::open Open the given file with gstreamer
* \param type CvCapture type. One of CAP_GSTREAMER_*
* \param filename Filename to open in case of CAP_GSTREAMER_FILE
* \return boolean. Specifies if opening was succesful.
*
* In case of CAP_GSTREAMER_V4L(2), a pipelin is constructed as follows:
* v4l2src ! autoconvert ! appsink
*
*
* The 'filename' parameter is not limited to filesystem paths, and may be one of the following:
*
* - a normal filesystem path:
* e.g. video.avi or /path/to/video.avi or C:\\video.avi
* - an uri:
* e.g. file:///path/to/video.avi or rtsp:///path/to/stream.asf
* - a gstreamer pipeline description:
* e.g. videotestsrc ! videoconvert ! appsink
* the appsink name should be either 'appsink0' (the default) or 'opencvsink'
*
* When dealing with a file, OpenIMAJCapGStreamer will not drop frames if the grabbing interval
* larger than the framerate period. (Unlike the uri or manual pipeline description, which assume
* a live source)
*
* The pipeline will only be started whenever the first frame is grabbed. Setting pipeline properties
* is really slow if we need to restart the pipeline over and over again.
*
*/
bool OpenIMAJCapGStreamer::open(const char* filename )
{
if(!isInited) {
//FIXME: threadsafety
gst_init (NULL, NULL);
isInited = true;
}
bool stream = false;
bool manualpipeline = false;
char *uri = NULL;
uridecodebin = NULL;
// test if we have a valid uri. If so, open it with an uridecodebin
// else, we might have a file or a manual pipeline.
// if gstreamer cannot parse the manual pipeline, we assume we were given and
// ordinary file path.
if(!gst_uri_is_valid(filename))
{
uri = realpath(filename, NULL);
stream = false;
if(uri)
{
uri = g_filename_to_uri(uri, NULL, NULL);
if(!uri) {
WARN("GStreamer: Error opening file\n");
close();
return false;
}
}
else
{
GError *err = NULL;
uridecodebin = gst_parse_launch(filename, &err);
if(!uridecodebin) {
//fprintf(stderr, "GStreamer: Error opening bin: %s\n", err->message);
//close();
return false;
}
stream = true;
manualpipeline = true;
}
} else {
stream = true;
uri = g_strdup(filename);
}
bool element_from_uri = false;
if(!uridecodebin)
{
// At this writing, the v4l2 element (and maybe others too) does not support caps renegotiation.
// This means that we cannot use an uridecodebin when dealing with v4l2, since setting
// capture properties will not work.
// The solution (probably only until gstreamer 1.2) is to make an element from uri when dealing with v4l2.
gchar * protocol = gst_uri_get_protocol(uri);
if (!strcasecmp(protocol , "v4l2"))
{
uridecodebin = gst_element_make_from_uri(GST_URI_SRC, uri, "src", NULL);
element_from_uri = true;
}else{
uridecodebin = gst_element_factory_make ("uridecodebin", NULL);
g_object_set(G_OBJECT(uridecodebin),"uri",uri, NULL);
}
g_free(protocol);
if(!uridecodebin) {
//fprintf(stderr, "GStreamer: Error opening bin: %s\n", err->message);
close();
return false;
}
}
if(manualpipeline)
{
GstIterator *it = NULL;
it = gst_bin_iterate_sinks (GST_BIN(uridecodebin));
gboolean done = FALSE;
GstElement *element = NULL;
gchar* name = NULL;
GValue value = G_VALUE_INIT;
while (!done) {
switch (gst_iterator_next (it, &value)) {
case GST_ITERATOR_OK:
element = GST_ELEMENT (g_value_get_object (&value));
name = gst_element_get_name(element);
if (name){
if(strstr(name, "opencvsink") != NULL || strstr(name, "appsink") != NULL) {
sink = GST_ELEMENT ( gst_object_ref (element) );
done = TRUE;
}
g_free(name);
}
g_value_unset (&value);
break;
case GST_ITERATOR_RESYNC:
gst_iterator_resync (it);
break;
case GST_ITERATOR_ERROR:
case GST_ITERATOR_DONE:
done = TRUE;
break;
}
}
gst_iterator_free (it);
if (!sink){
//ERROR(1, "GStreamer: cannot find appsink in manual pipeline\n");
return false;
}
pipeline = uridecodebin;
}
else
{
pipeline = gst_pipeline_new (NULL);
// videoconvert (in 0.10: ffmpegcolorspace) automatically selects the correct colorspace
// conversion based on caps.
color = gst_element_factory_make(COLOR_ELEM, NULL);
sink = gst_element_factory_make("appsink", NULL);
gst_bin_add_many(GST_BIN(pipeline), uridecodebin, color, sink, NULL);
if(element_from_uri) {
if(!gst_element_link(uridecodebin, color)) {
//ERROR(1, "GStreamer: cannot link color -> sink\n");
gst_object_unref(pipeline);
return false;
}
}else{
g_signal_connect(uridecodebin, "pad-added", G_CALLBACK(newPad), color);
}
if(!gst_element_link(color, sink)) {
//ERROR(1, "GStreamer: cannot link color -> sink\n");
gst_object_unref(pipeline);
return false;
}
}
//TODO: is 1 single buffer really high enough?
gst_app_sink_set_max_buffers (GST_APP_SINK(sink), 1);
gst_app_sink_set_drop (GST_APP_SINK(sink), stream);
//do not emit signals: all calls will be synchronous and blocking
gst_app_sink_set_emit_signals (GST_APP_SINK(sink), 0);
// support 1 and 3 channel 8 bit data, as well as bayer (also 1 channel, 8bit)
caps = gst_caps_from_string("video/x-raw, format=(string){BGR, GRAY8}; video/x-bayer,format=(string){rggb,bggr,grbg,gbrg}");
gst_app_sink_set_caps(GST_APP_SINK(sink), caps);
gst_caps_unref(caps);
//we do not start recording here just yet.
// the user probably wants to set capture properties first, so start recording whenever the first frame is requested
return true;
}
void AVMuxEncode::deletePipelines()
{
m_videoSrcLock.lock();
m_audioSrcLock.lock();
m_videoSinkLock.lock();
m_audioSinkLock.lock();
if (m_videoPipeline != NULL) {
gst_element_set_state (m_videoPipeline, GST_STATE_NULL);
gst_app_sink_set_emit_signals((GstAppSink *)(m_appVideoSink), FALSE);
g_signal_handlers_disconnect_by_data(m_appVideoSink, this);
gst_object_unref(m_videoPipeline);
m_videoPipeline = NULL;
}
#ifdef GSTBUSMSG
if (m_videoBusWatch != -1) {
g_source_remove(m_videoBusWatch);
m_videoBusWatch = -1;
}
#endif
if (m_audioPipeline != NULL) {
gst_element_set_state (m_audioPipeline, GST_STATE_NULL);
gst_app_sink_set_emit_signals((GstAppSink *)(m_appAudioSink), FALSE);
g_signal_handlers_disconnect_by_data(m_appAudioSink, this);
gst_object_unref(m_audioPipeline);
m_audioPipeline = NULL;
}
#ifdef GSTBUSMSG
if (m_audioBusWatch != -1) {
g_source_remove(m_audioBusWatch);
m_audioBusWatch = -1;
}
#endif
if (m_videoCaps != NULL)
g_free(m_videoCaps);
if (m_audioCaps != NULL)
g_free(m_audioCaps);
m_videoCaps = NULL;
m_audioCaps = NULL;
m_appAudioSink = NULL;
m_appAudioSrc = NULL;
m_appVideoSink = NULL;
m_appAudioSrc = NULL;
while (!m_videoSrcQ.empty())
delete m_videoSrcQ.dequeue();
while (!m_audioSrcQ.empty())
delete m_audioSrcQ.dequeue();
while (!m_videoSinkQ.empty())
delete m_videoSinkQ.dequeue();
while (!m_audioSinkQ.empty())
delete m_audioSinkQ.dequeue();
m_audioTimestamp = -1;
m_videoSrcLock.unlock();
m_audioSrcLock.unlock();
m_videoSinkLock.unlock();
m_audioSinkLock.unlock();
}
bool AVMuxEncode::newPipelines(SYNTRO_AVPARAMS *avParams)
{
gchar *videoLaunch;
gchar *audioLaunch;
GError *error = NULL;
GstStateChangeReturn ret;
m_avParams = *avParams;
printf("width=%d, height=%d, rate=%d\n", avParams->videoWidth, avParams->videoHeight, avParams->videoFramerate);
printf("channels=%d, rate=%d, size=%d\n", avParams->audioChannels, avParams->audioSampleRate, avParams->audioSampleSize);
// Construct the pipelines
g_pipelineIndex++;
if (m_AVMode == AVMUXENCODE_AV_TYPE_RTPMP4) {
videoLaunch = g_strdup_printf (
" appsrc name=videoSrc%d ! jpegdec ! queue ! ffenc_mpeg4 bitrate=%d "
" ! queue ! rtpmp4vpay pt=96 ! queue ! appsink name=videoSink%d"
, g_pipelineIndex, m_videoCompressionRate, g_pipelineIndex);
} else {
videoLaunch = g_strdup_printf (
" appsrc name=videoSrc%d ! jpegdec ! queue ! x264enc bitrate=%d tune=zerolatency rc-lookahead=0"
" ! queue ! rtph264pay pt=96 ! queue ! appsink name=videoSink%d"
, g_pipelineIndex, m_videoCompressionRate / 1000, g_pipelineIndex);
}
m_videoPipeline = gst_parse_launch(videoLaunch, &error);
g_free(videoLaunch);
if (error != NULL) {
g_print ("could not construct video pipeline: %s\n", error->message);
g_error_free (error);
m_videoPipeline = NULL;
return false;
}
audioLaunch = g_strdup_printf (
#ifdef GST_IMX6
" appsrc name=audioSrc%d ! faac bitrate=%d ! rtpmp4apay pt=97 ! appsink name=audioSink%d "
#else
" appsrc name=audioSrc%d ! faac bitrate=%d ! rtpmp4apay pt=97 min-ptime=1000000000 ! appsink name=audioSink%d "
#endif
, g_pipelineIndex, m_audioCompressionRate, g_pipelineIndex);
m_audioPipeline = gst_parse_launch(audioLaunch, &error);
g_free(audioLaunch);
if (error != NULL) {
g_print ("could not construct audio pipeline: %s\n", error->message);
g_error_free (error);
gst_object_unref(m_videoPipeline);
m_videoPipeline = NULL;
m_audioPipeline = NULL;
return false;
}
// find the appsrcs and appsinks
gchar *videoSink = g_strdup_printf("videoSink%d", g_pipelineIndex);
if ((m_appVideoSink = gst_bin_get_by_name (GST_BIN (m_videoPipeline), videoSink)) == NULL) {
g_printerr("Unable to find video appsink\n");
g_free(videoSink);
deletePipelines();
return false;
}
g_free(videoSink);
gchar *videoSrc = g_strdup_printf("videoSrc%d", g_pipelineIndex);
if ((m_appVideoSrc = gst_bin_get_by_name (GST_BIN (m_videoPipeline), videoSrc)) == NULL) {
g_printerr("Unable to find video appsrc\n");
g_free(videoSrc);
deletePipelines();
return false;
}
g_free(videoSrc);
gchar *audioSink = g_strdup_printf("audioSink%d", g_pipelineIndex);
if ((m_appAudioSink = gst_bin_get_by_name (GST_BIN (m_audioPipeline), audioSink)) == NULL) {
g_printerr("Unable to find audio appsink\n");
g_free(audioSink);
deletePipelines();
return false;
}
g_free(audioSink);
gchar *audioSrc = g_strdup_printf("audioSrc%d", g_pipelineIndex);
if ((m_appAudioSrc = gst_bin_get_by_name (GST_BIN (m_audioPipeline), audioSrc)) == NULL) {
g_printerr("Unable to find audio appsrc\n");
g_free(audioSrc);
deletePipelines();
return false;
}
g_free(audioSrc);
g_signal_connect (m_appVideoSink, "new-buffer", G_CALLBACK (newVideoSinkData), this);
gst_app_sink_set_emit_signals((GstAppSink *)(m_appVideoSink), TRUE);
g_signal_connect (m_appAudioSink, "new-buffer", G_CALLBACK (newAudioSinkData), this);
gst_app_sink_set_emit_signals((GstAppSink *)(m_appAudioSink), TRUE);
gst_app_src_set_caps((GstAppSrc *) (m_appVideoSrc),
gst_caps_new_simple ("image/jpeg",
"width", G_TYPE_INT, m_avParams.videoWidth,
"height", G_TYPE_INT, m_avParams.videoHeight,
"framerate", GST_TYPE_FRACTION, m_avParams.videoFramerate, 1,
NULL));
gst_app_src_set_stream_type((GstAppSrc *)(m_appVideoSrc), GST_APP_STREAM_TYPE_STREAM);
gst_app_src_set_caps((GstAppSrc *) (m_appAudioSrc),
gst_caps_new_simple ("audio/x-raw-int",
"width", G_TYPE_INT, (gint)m_avParams.audioSampleSize,
"depth", G_TYPE_INT, (gint)m_avParams.audioSampleSize,
"channels" ,G_TYPE_INT, (gint)m_avParams.audioChannels,
"rate",G_TYPE_INT, m_avParams.audioSampleRate,
"endianness",G_TYPE_INT,(gint)1234,
"signed", G_TYPE_BOOLEAN, (gboolean)TRUE,
NULL));
gst_app_src_set_stream_type((GstAppSrc *)(m_appAudioSrc), GST_APP_STREAM_TYPE_STREAM);
g_signal_connect(m_appAudioSrc, "need-data", G_CALLBACK (needAudioSrcData), this);
ret = gst_element_set_state (m_videoPipeline, GST_STATE_PLAYING);
if (ret == GST_STATE_CHANGE_FAILURE) {
g_printerr ("Unable to set the video pipeline to the play state.\n");
deletePipelines();
return false;
}
#ifdef GSTBUSMSG
GstBus *bus;
bus = gst_pipeline_get_bus(GST_PIPELINE (m_videoPipeline));
m_videoBusWatch = gst_bus_add_watch (bus, videoBusMessage, this);
gst_object_unref (bus);
bus = gst_pipeline_get_bus(GST_PIPELINE (m_audioPipeline));
m_audioBusWatch = gst_bus_add_watch (bus, audioBusMessage, this);
gst_object_unref (bus);
#endif
ret = gst_element_set_state (m_audioPipeline, GST_STATE_PLAYING);
if (ret == GST_STATE_CHANGE_FAILURE) {
g_printerr ("Unable to set the audio pipeline to the play state.\n");
deletePipelines();
return false;
}
m_videoInterval = (1000 * PIPELINE_MS_TO_NS) / m_avParams.videoFramerate;
m_nextVideoTime = QDateTime::currentMSecsSinceEpoch() * PIPELINE_MS_TO_NS;
qDebug() << "Pipelines established";
m_pipelinesActive = true;
return true;
}
/*!
* \brief CvCapture_GStreamer::open Open the given file with gstreamer
* \param type CvCapture type. One of CV_CAP_GSTREAMER_*
* \param filename Filename to open in case of CV_CAP_GSTREAMER_FILE
* \return boolean. Specifies if opening was succesful.
*
* In case of CV_CAP_GSTREAMER_V4L(2), a pipelin is constructed as follows:
* v4l2src ! autoconvert ! appsink
*
*
* The 'filename' parameter is not limited to filesystem paths, and may be one of the following:
*
* - a normal filesystem path:
* e.g. video.avi or /path/to/video.avi or C:\\video.avi
* - an uri:
* e.g. file:///path/to/video.avi or rtsp:///path/to/stream.asf
* - a gstreamer pipeline description:
* e.g. videotestsrc ! videoconvert ! appsink
* the appsink name should be either 'appsink0' (the default) or 'opencvsink'
*
* When dealing with a file, CvCapture_GStreamer will not drop frames if the grabbing interval
* larger than the framerate period. (Unlike the uri or manual pipeline description, which assume
* a live source)
*
* The pipeline will only be started whenever the first frame is grabbed. Setting pipeline properties
* is really slow if we need to restart the pipeline over and over again.
*
* TODO: the 'type' parameter is imo unneeded. for v4l2, filename 'v4l2:///dev/video0' can be used.
* I expect this to be the same for CV_CAP_GSTREAMER_1394. Is anyone actually still using v4l (v1)?
*
*/
bool CvCapture_GStreamer::open( int type, const char* filename )
{
CV_FUNCNAME("cvCaptureFromCAM_GStreamer");
__BEGIN__;
gst_initializer::init();
bool file = false;
bool stream = false;
bool manualpipeline = false;
char *uri = NULL;
uridecodebin = NULL;
GstElementFactory * testfac;
GstStateChangeReturn status;
if (type == CV_CAP_GSTREAMER_V4L){
testfac = gst_element_factory_find("v4lsrc");
if (!testfac){
return false;
}
g_object_unref(G_OBJECT(testfac));
filename = "v4lsrc ! "COLOR_ELEM" ! appsink";
}
if (type == CV_CAP_GSTREAMER_V4L2){
testfac = gst_element_factory_find("v4l2src");
if (!testfac){
return false;
}
g_object_unref(G_OBJECT(testfac));
filename = "v4l2src ! "COLOR_ELEM" ! appsink";
}
// test if we have a valid uri. If so, open it with an uridecodebin
// else, we might have a file or a manual pipeline.
// if gstreamer cannot parse the manual pipeline, we assume we were given and
// ordinary file path.
if(!gst_uri_is_valid(filename))
{
uri = realpath(filename, NULL);
stream = false;
if(uri)
{
uri = g_filename_to_uri(uri, NULL, NULL);
if(uri)
{
file = true;
}
else
{
CV_WARN("GStreamer: Error opening file\n");
close();
return false;
}
}
else
{
GError *err = NULL;
uridecodebin = gst_parse_launch(filename, &err);
if(!uridecodebin)
{
fprintf(stderr, "GStreamer: Error opening bin: %s\n", err->message);
return false;
}
stream = true;
manualpipeline = true;
}
} else {
stream = true;
uri = g_strdup(filename);
}
bool element_from_uri = false;
if(!uridecodebin)
{
// At this writing, the v4l2 element (and maybe others too) does not support caps renegotiation.
// This means that we cannot use an uridecodebin when dealing with v4l2, since setting
// capture properties will not work.
// The solution (probably only until gstreamer 1.2) is to make an element from uri when dealing with v4l2.
gchar * protocol = gst_uri_get_protocol(uri);
if (!strcasecmp(protocol , "v4l2"))
{
#if GST_VERSION_MAJOR == 0
uridecodebin = gst_element_make_from_uri(GST_URI_SRC, uri, "src");
#else
uridecodebin = gst_element_make_from_uri(GST_URI_SRC, uri, "src", NULL);
#endif
element_from_uri = true;
}else{
uridecodebin = gst_element_factory_make("uridecodebin", NULL);
g_object_set(G_OBJECT(uridecodebin), "uri", uri, NULL);
}
g_free(protocol);
if(!uridecodebin) {
//fprintf(stderr, "GStreamer: Error opening bin: %s\n", err->message);
close();
return false;
}
}
if(manualpipeline)
{
GstIterator *it = NULL;
#if GST_VERSION_MAJOR == 0
it = gst_bin_iterate_sinks(GST_BIN(uridecodebin));
if(gst_iterator_next(it, (gpointer *)&sink) != GST_ITERATOR_OK) {
CV_ERROR(CV_StsError, "GStreamer: cannot find appsink in manual pipeline\n");
return false;
}
#else
it = gst_bin_iterate_sinks (GST_BIN(uridecodebin));
gboolean done = FALSE;
GstElement *element = NULL;
gchar* name = NULL;
GValue value = G_VALUE_INIT;
while (!done) {
switch (gst_iterator_next (it, &value)) {
case GST_ITERATOR_OK:
element = GST_ELEMENT (g_value_get_object (&value));
name = gst_element_get_name(element);
if (name){
if(strstr(name, "opencvsink") != NULL || strstr(name, "appsink") != NULL) {
sink = GST_ELEMENT ( gst_object_ref (element) );
done = TRUE;
}
g_free(name);
}
g_value_unset (&value);
break;
case GST_ITERATOR_RESYNC:
gst_iterator_resync (it);
break;
case GST_ITERATOR_ERROR:
case GST_ITERATOR_DONE:
done = TRUE;
break;
}
}
gst_iterator_free (it);
if (!sink){
CV_ERROR(CV_StsError, "GStreamer: cannot find appsink in manual pipeline\n");
return false;
}
#endif
pipeline = uridecodebin;
}
else
{
pipeline = gst_pipeline_new(NULL);
// videoconvert (in 0.10: ffmpegcolorspace, in 1.x autovideoconvert)
//automatically selects the correct colorspace conversion based on caps.
color = gst_element_factory_make(COLOR_ELEM, NULL);
sink = gst_element_factory_make("appsink", NULL);
gst_bin_add_many(GST_BIN(pipeline), uridecodebin, color, sink, NULL);
if(element_from_uri) {
if(!gst_element_link(uridecodebin, color)) {
CV_ERROR(CV_StsError, "GStreamer: cannot link color -> sink\n");
gst_object_unref(pipeline);
pipeline = NULL;
return false;
}
}else{
g_signal_connect(uridecodebin, "pad-added", G_CALLBACK(newPad), color);
}
if(!gst_element_link(color, sink)) {
CV_ERROR(CV_StsError, "GStreamer: cannot link color -> sink\n");
gst_object_unref(pipeline);
pipeline = NULL;
return false;
}
}
//TODO: is 1 single buffer really high enough?
gst_app_sink_set_max_buffers (GST_APP_SINK(sink), 1);
gst_app_sink_set_drop (GST_APP_SINK(sink), stream);
//do not emit signals: all calls will be synchronous and blocking
gst_app_sink_set_emit_signals (GST_APP_SINK(sink), 0);
#if GST_VERSION_MAJOR == 0
caps = gst_caps_new_simple("video/x-raw-rgb",
"bpp", G_TYPE_INT, 24,
"red_mask", G_TYPE_INT, 0x0000FF,
"green_mask", G_TYPE_INT, 0x00FF00,
"blue_mask", G_TYPE_INT, 0xFF0000,
NULL);
#else
// support 1 and 3 channel 8 bit data, as well as bayer (also 1 channel, 8bit)
caps = gst_caps_from_string("video/x-raw, format=(string){BGR, GRAY8}; video/x-bayer,format=(string){rggb,bggr,grbg,gbrg}");
#endif
gst_app_sink_set_caps(GST_APP_SINK(sink), caps);
gst_caps_unref(caps);
// For video files only: set pipeline to PAUSED state to get its duration
if (file)
{
status = gst_element_set_state(GST_ELEMENT(pipeline), GST_STATE_PAUSED);
if (status == GST_STATE_CHANGE_ASYNC)
{
// wait for status update
GstState st1;
GstState st2;
status = gst_element_get_state(pipeline, &st1, &st2, GST_CLOCK_TIME_NONE);
}
if (status == GST_STATE_CHANGE_FAILURE)
{
handleMessage(pipeline);
gst_object_unref(pipeline);
pipeline = NULL;
CV_ERROR(CV_StsError, "GStreamer: unable to start pipeline\n");
return false;
}
GstFormat format;
format = GST_FORMAT_DEFAULT;
#if GST_VERSION_MAJOR == 0
if(!gst_element_query_duration(sink, &format, &duration))
#else
if(!gst_element_query_duration(sink, format, &duration))
#endif
{
handleMessage(pipeline);
CV_WARN("GStreamer: unable to query duration of stream");
duration = -1;
return true;
}
}
else
{
duration = -1;
}
__END__;
return true;
}
bool GStreamerCameraFrameSourceImpl::InitializeGstPipeLine()
{
GstStateChangeReturn status;
end = true;
pipeline = GST_PIPELINE(gst_pipeline_new(NULL));
if (pipeline == NULL)
{
printf("Cannot create Gstreamer pipeline\n");
return false;
}
bus = gst_pipeline_get_bus(GST_PIPELINE (pipeline));
// create v4l2src
GstElement * v4l2src = gst_element_factory_make("v4l2src", NULL);
if (v4l2src == NULL)
{
printf("Cannot create v4l2src\n");
FinalizeGstPipeLine();
return false;
}
std::ostringstream cameraDev;
cameraDev << "/dev/video" << cameraIdx;
g_object_set(G_OBJECT(v4l2src), "device", cameraDev.str().c_str(), NULL);
gst_bin_add(GST_BIN(pipeline), v4l2src);
// create color convert element
GstElement * color = gst_element_factory_make(COLOR_ELEM, NULL);
if (color == NULL)
{
printf("Cannot create %s element\n", COLOR_ELEM);
FinalizeGstPipeLine();
return false;
}
gst_bin_add(GST_BIN(pipeline), color);
// create appsink element
sink = gst_element_factory_make("appsink", NULL);
if (sink == NULL)
{
printf("Cannot create appsink element\n");
FinalizeGstPipeLine();
return false;
}
gst_bin_add(GST_BIN(pipeline), sink);
// if initial values for FrameSource::Parameters are not
// specified, let's set them manually to prevent very huge images
if (configuration.frameWidth == (vx_uint32)-1)
configuration.frameWidth = 1920;
if (configuration.frameHeight == (vx_uint32)-1)
configuration.frameHeight = 1080;
if (configuration.fps == (vx_uint32)-1)
configuration.fps = 30;
#if GST_VERSION_MAJOR == 0
GstCaps* caps_v42lsrc = gst_caps_new_simple ("video/x-raw-rgb",
"width", GST_TYPE_INT_RANGE, 1, (int)configuration.frameWidth,
"height", GST_TYPE_INT_RANGE, 1, (int)configuration.frameHeight,
"framerate", GST_TYPE_FRACTION, (int)configuration.fps,
NULL);
#else
std::ostringstream stream;
stream << "video/x-raw, format=(string){RGB, GRAY8}, width=[1," << configuration.frameWidth <<
"], height=[1," << configuration.frameHeight << "], framerate=" << configuration.fps << "/1;";
GstCaps* caps_v42lsrc = gst_caps_from_string(stream.str().c_str());
#endif
if (caps_v42lsrc == NULL)
{
printf("Failed to create caps\n");
FinalizeGstPipeLine();
return false;
}
// link elements
if (!gst_element_link_filtered(v4l2src, color, caps_v42lsrc))
{
printf("GStreamer: cannot link v4l2src -> color using caps\n");
FinalizeGstPipeLine();
gst_caps_unref(caps_v42lsrc);
return false;
}
gst_caps_unref(caps_v42lsrc);
// link elements
if (!gst_element_link(color, sink))
{
printf("GStreamer: cannot link color -> appsink\n");
FinalizeGstPipeLine();
return false;
}
gst_app_sink_set_max_buffers (GST_APP_SINK(sink), 1);
gst_app_sink_set_drop (GST_APP_SINK(sink), true);
// do not emit signals: all calls will be synchronous and blocking
gst_app_sink_set_emit_signals (GST_APP_SINK(sink), 0);
#if GST_VERSION_MAJOR == 0
GstCaps* caps_appsink = gst_caps_new_simple("video/x-raw-rgb",
"bpp", G_TYPE_INT, 24,
"red_mask", G_TYPE_INT, 0xFF0000,
"green_mask", G_TYPE_INT, 0x00FF00,
"blue_mask", G_TYPE_INT, 0x0000FF,
NULL);
#else
// support 1 and 3 channel 8 bit data
GstCaps* caps_appsink = gst_caps_from_string("video/x-raw, format=(string){RGB, GRAY8};");
#endif
gst_app_sink_set_caps(GST_APP_SINK(sink), caps_appsink);
gst_caps_unref(caps_appsink);
// Force pipeline to play video as fast as possible, ignoring system clock
gst_pipeline_use_clock(pipeline, NULL);
status = gst_element_set_state(GST_ELEMENT(pipeline), GST_STATE_PLAYING);
handleGStreamerMessages();
if (status == GST_STATE_CHANGE_ASYNC)
{
// wait for status update
status = gst_element_get_state(GST_ELEMENT(pipeline), NULL, NULL, GST_CLOCK_TIME_NONE);
}
if (status == GST_STATE_CHANGE_FAILURE)
{
printf("GStreamer: unable to start playback\n");
FinalizeGstPipeLine();
return false;
}
std::unique_ptr<GstPad, GStreamerObjectDeleter> pad(gst_element_get_static_pad(color, "src"));
#if GST_VERSION_MAJOR == 0
std::unique_ptr<GstCaps, GStreamerObjectDeleter> bufferCaps(gst_pad_get_caps(pad.get()));
#else
std::unique_ptr<GstCaps, GStreamerObjectDeleter> bufferCaps(gst_pad_get_current_caps(pad.get()));
#endif
const GstStructure *structure = gst_caps_get_structure(bufferCaps.get(), 0);
int width, height;
if (!gst_structure_get_int(structure, "width", &width))
{
handleGStreamerMessages();
printf("Cannot query video width\n");
}
if (!gst_structure_get_int(structure, "height", &height))
{
handleGStreamerMessages();
printf("Cannot query video height\n");
}
configuration.frameWidth = static_cast<vx_uint32>(width);
configuration.frameHeight = static_cast<vx_uint32>(height);
gint num = 0, denom = 1;
if (!gst_structure_get_fraction(structure, "framerate", &num, &denom))
{
handleGStreamerMessages();
printf("Cannot query video fps\n");
}
configuration.fps = static_cast<float>(num) / denom;
end = false;
return true;
}
void ofxGstRTPServer::play(){
// pass the pipeline to the gstUtils so it starts everything
gst.setPipelineWithSink(pipelineStr,"",true);
// get the rtp and rtpc elements from the pipeline so we can read their properties
// during execution
rtpbin = gst.getGstElementByName("rtpbin");
vRTPsink = gst.getGstElementByName("vrtpsink");
vRTPCsink = gst.getGstElementByName("vrtcpsink");
vRTPCsrc = gst.getGstElementByName("vrtcpsrc");
aRTPsink = gst.getGstElementByName("artpsink");
aRTPCsink = gst.getGstElementByName("artcpsink");
aRTPCsrc = gst.getGstElementByName("artcpsrc");
dRTPsink = gst.getGstElementByName("drtpsink");
dRTPCsink = gst.getGstElementByName("drtcpsink");
dRTPCsrc = gst.getGstElementByName("drtcpsrc");
oRTPsink = gst.getGstElementByName("ortpsink");
oRTPCsink = gst.getGstElementByName("ortcpsink");
oRTPCsrc = gst.getGstElementByName("ortcpsrc");
vEncoder = gst.getGstElementByName("vencoder");
dEncoder = gst.getGstElementByName("dencoder");
aEncoder = gst.getGstElementByName("aencoder");
appSrcVideoRGB = gst.getGstElementByName("appsrcvideo");
appSrcDepth = gst.getGstElementByName("appsrcdepth");
appSrcOsc = gst.getGstElementByName("appsrcosc");
#if ENABLE_ECHO_CANCEL
if(echoCancel && audioChannelReady){
appSrcAudio = gst.getGstElementByName("audioechosrc");
if(appSrcAudio){
gst_app_src_set_stream_type((GstAppSrc*)appSrcAudio,GST_APP_STREAM_TYPE_STREAM);
}
#ifdef TARGET_LINUX
gstAudioIn.setPipelineWithSink("pulsesrc stream-properties=\"props,media.role=phone\" name=audiocapture ! audio/x-raw,format=S16LE,rate=44100,channels=1 ! audioresample ! audioconvert ! audio/x-raw,format=S16LE,rate=32000,channels=1 ! appsink name=audioechosink");
volume = gstAudioIn.getGstElementByName("audiocapture");
#elif defined(TARGET_OSX)
// for osx we specify the output format since osxaudiosrc doesn't report the formats supported by the hw
// FIXME: we should detect the format somehow and set it automatically
gstAudioIn.setPipelineWithSink("osxaudiosrc name=audiocapture ! audio/x-raw,rate=44100,channels=1 ! volume name=volume ! audioresample ! audioconvert ! audio/x-raw,format=S16LE,rate=32000,channels=1 ! appsink name=audioechosink");
volume = gstAudioIn.getGstElementByName("volume");
#endif
appSinkAudio = gstAudioIn.getGstElementByName("audioechosink");
audiocapture = gstAudioIn.getGstElementByName("audiocapture");
// set callbacks to receive audio data
GstAppSinkCallbacks gstCallbacks;
gstCallbacks.eos = &on_eos_from_audio;
gstCallbacks.new_preroll = &on_new_preroll_from_audio;
gstCallbacks.new_sample = &on_new_buffer_from_audio;
gst_app_sink_set_callbacks(GST_APP_SINK(appSinkAudio), &gstCallbacks, this, NULL);
gst_app_sink_set_emit_signals(GST_APP_SINK(appSinkAudio),0);
}
#endif
#if ENABLE_NAT_TRANSVERSAL
if(videoStream){
g_object_set(G_OBJECT(vRTPsink),"agent",videoStream->getAgent(),"stream",videoStream->getStreamID(),"component",1,NULL);
g_object_set(G_OBJECT(vRTPCsink),"agent",videoStream->getAgent(),"stream",videoStream->getStreamID(),"component",2,NULL);
g_object_set(G_OBJECT(vRTPCsrc),"agent",videoStream->getAgent(),"stream",videoStream->getStreamID(),"component",3,NULL);
}
if(depthStream){
g_object_set(G_OBJECT(dRTPsink),"agent",depthStream->getAgent(),"stream",depthStream->getStreamID(),"component",1,NULL);
g_object_set(G_OBJECT(dRTPCsink),"agent",depthStream->getAgent(),"stream",depthStream->getStreamID(),"component",2,NULL);
g_object_set(G_OBJECT(dRTPCsrc),"agent",depthStream->getAgent(),"stream",depthStream->getStreamID(),"component",3,NULL);
}
if(audioStream){
g_object_set(G_OBJECT(aRTPsink),"agent",audioStream->getAgent(),"stream",audioStream->getStreamID(),"component",1,NULL);
g_object_set(G_OBJECT(aRTPCsink),"agent",audioStream->getAgent(),"stream",audioStream->getStreamID(),"component",2,NULL);
g_object_set(G_OBJECT(aRTPCsrc),"agent",audioStream->getAgent(),"stream",audioStream->getStreamID(),"component",3,NULL);
}
if(oscStream){
g_object_set(G_OBJECT(oRTPsink),"agent",oscStream->getAgent(),"stream",oscStream->getStreamID(),"component",1,NULL);
g_object_set(G_OBJECT(oRTPCsink),"agent",oscStream->getAgent(),"stream",oscStream->getStreamID(),"component",2,NULL);
g_object_set(G_OBJECT(oRTPCsrc),"agent",oscStream->getAgent(),"stream",oscStream->getStreamID(),"component",3,NULL);
}
#endif
if(appSrcVideoRGB) gst_app_src_set_stream_type((GstAppSrc*)appSrcVideoRGB,GST_APP_STREAM_TYPE_STREAM);
if(appSrcDepth) gst_app_src_set_stream_type((GstAppSrc*)appSrcDepth,GST_APP_STREAM_TYPE_STREAM);
if(appSrcOsc) gst_app_src_set_stream_type((GstAppSrc*)appSrcOsc,GST_APP_STREAM_TYPE_STREAM);
g_signal_connect(rtpbin,"on-new-ssrc",G_CALLBACK(&ofxGstRTPServer::on_new_ssrc_handler),this);
#if ENABLE_ECHO_CANCEL
if(echoCancel && audioChannelReady){
gstAudioIn.startPipeline();
gstAudioIn.play();
}
#endif
gst.startPipeline();
gst.play();
ofAddListener(ofEvents().update,this,&ofxGstRTPServer::update);
}
