static int base_init(base *self, PyObject *args, PyObject *kwds)
{
// call tp_init of PyGObject
int args_len = PyTuple_Size(args);
PyObject *reduced_args = PySequence_GetSlice(args, 2, args_len);
if (PyGObject_Type->tp_init((PyObject *)self, reduced_args, kwds) < 0) return -1;
Py_DECREF(reduced_args);
// parse arguments
if (!PyArg_ParseTuple(args, "O!O!", PyGObject_Type, &(self->spectrum_element), PyGObject_Type, &(self->pipeline))) return -1;
// listen for spectrum messages
GstBus *gstbus = gst_pipeline_get_bus(GST_PIPELINE(self->pipeline->obj));
g_assert(gstbus != NULL);
gst_bus_add_signal_watch(gstbus);
Py_INCREF(self);
g_signal_connect(G_OBJECT(gstbus), "message::element", G_CALLBACK(on_message), self);
gst_object_unref(gstbus);
// get clock of pipeline
self->sync_clock = gst_pipeline_get_clock(GST_PIPELINE(self->pipeline->obj));
g_assert(self->sync_clock != NULL);
return 0;
}
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;
}
}
static gboolean create_pipeline(SpiceGstDecoder *decoder)
{
gchar *desc;
gboolean auto_enabled;
guint opt;
GstAppSinkCallbacks appsink_cbs = { NULL };
GError *err = NULL;
GstBus *bus;
auto_enabled = (g_getenv("SPICE_GSTVIDEO_AUTO") != NULL);
if (auto_enabled || !VALID_VIDEO_CODEC_TYPE(decoder->base.codec_type)) {
SPICE_DEBUG("Trying %s for codec type %d %s",
gst_opts[0].dec_name, decoder->base.codec_type,
(auto_enabled) ? "(SPICE_GSTVIDEO_AUTO is set)" : "");
opt = 0;
} else {
opt = decoder->base.codec_type;
}
/* - We schedule the frame display ourselves so set sync=false on appsink
* so the pipeline decodes them as fast as possible. This will also
* minimize the risk of frames getting lost when we rebuild the
* pipeline.
* - Set max-bytes=0 on appsrc so it does not drop frames that may be
* needed by those that follow.
*/
desc = g_strdup_printf("appsrc name=src is-live=true format=time max-bytes=0 block=true "
"%s ! %s ! videoconvert ! appsink name=sink "
"caps=video/x-raw,format=BGRx sync=false drop=false",
gst_opts[opt].dec_caps, gst_opts[opt].dec_name);
SPICE_DEBUG("GStreamer pipeline: %s", desc);
decoder->pipeline = gst_parse_launch_full(desc, NULL, GST_PARSE_FLAG_FATAL_ERRORS, &err);
g_free(desc);
if (!decoder->pipeline) {
spice_warning("GStreamer error: %s", err->message);
g_clear_error(&err);
return FALSE;
}
decoder->appsrc = GST_APP_SRC(gst_bin_get_by_name(GST_BIN(decoder->pipeline), "src"));
decoder->appsink = GST_APP_SINK(gst_bin_get_by_name(GST_BIN(decoder->pipeline), "sink"));
appsink_cbs.new_sample = new_sample;
gst_app_sink_set_callbacks(decoder->appsink, &appsink_cbs, decoder, NULL);
bus = gst_pipeline_get_bus(GST_PIPELINE(decoder->pipeline));
gst_bus_add_watch(bus, handle_pipeline_message, decoder);
gst_object_unref(bus);
decoder->clock = gst_pipeline_get_clock(GST_PIPELINE(decoder->pipeline));
if (gst_element_set_state(decoder->pipeline, GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE) {
SPICE_DEBUG("GStreamer error: Unable to set the pipeline to the playing state.");
free_pipeline(decoder);
return FALSE;
}
return TRUE;
}
GstClockTime ofxGstRTPServer::getTimeStamp(){
if(!gst.isLoaded()) return GST_CLOCK_TIME_NONE;
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);
return now;
}
int
main (int argc, char *argv[])
{
GstElement *bin;
GstElement *src, *spectrum, *sink;
GstBus *bus;
GtkWidget *appwindow;
gst_init (&argc, &argv);
gtk_init (&argc, &argv);
bin = gst_pipeline_new ("bin");
src = gst_element_factory_make (DEFAULT_AUDIOSRC, "src");
spectrum = gst_element_factory_make ("spectrum", "spectrum");
g_object_set (G_OBJECT (spectrum), "bands", spect_bands, "threshold", -80,
"post-messages", TRUE, NULL);
sink = gst_element_factory_make ("fakesink", "sink");
gst_bin_add_many (GST_BIN (bin), src, spectrum, sink, NULL);
if (!gst_element_link_many (src, spectrum, sink, NULL)) {
fprintf (stderr, "can't link elements\n");
exit (1);
}
bus = gst_element_get_bus (bin);
gst_bus_add_watch (bus, message_handler, NULL);
gst_object_unref (bus);
sync_clock = gst_pipeline_get_clock (GST_PIPELINE (bin));
appwindow = gtk_window_new (GTK_WINDOW_TOPLEVEL);
g_signal_connect (G_OBJECT (appwindow), "destroy",
G_CALLBACK (on_window_destroy), NULL);
drawingarea = gtk_drawing_area_new ();
gtk_widget_set_size_request (drawingarea, spect_bands, spect_height);
g_signal_connect (G_OBJECT (drawingarea), "configure-event",
G_CALLBACK (on_configure_event), (gpointer) spectrum);
gtk_container_add (GTK_CONTAINER (appwindow), drawingarea);
gtk_widget_show_all (appwindow);
gst_element_set_state (bin, GST_STATE_PLAYING);
gtk_main ();
gst_element_set_state (bin, GST_STATE_NULL);
gst_object_unref (sync_clock);
gst_object_unref (bin);
return 0;
}
static void on_new_decoded_pad(GstElement *introdec,
GstPad *srcpad,
gpointer data)
{
GstPadLinkReturn result;
GstPad *sinkpad;
GstCaps *new_pad_caps;
CustomData *cdata=(CustomData*)data;
GstElement *introbin=cdata->introbin;
new_pad_caps=gst_pad_query_caps(srcpad,NULL);
g_print("Caps:%s\n",gst_caps_to_string(new_pad_caps));
/* Setup src pad offset, sync with pipeline. */
gint64 pos2;
pos2=gst_element_get_base_time(cdata->pipeline);
GstClock *clock;
clock=gst_pipeline_get_clock(GST_PIPELINE(cdata->pipeline));
GstClockTime clock_time;
clock_time=gst_clock_get_time(clock);
gst_object_unref(clock);
// g_print("Pipeline times: base_time=%lld\n clock_time=%lld\n",
// pos2,clock_time);
gst_pad_set_offset(srcpad,clock_time-pos2);
cdata->introbin_offset=clock_time-pos2;
if(strncmp(gst_caps_to_string(new_pad_caps),"video",5)==0) {
GstElement *vqueue;
vqueue=gst_bin_get_by_name(GST_BIN(introbin),"introscale");
sinkpad=gst_element_get_static_pad(vqueue,"sink");
result=gst_pad_link(srcpad,sinkpad);
if(result!=GST_PAD_LINK_OK) {
g_printerr("Couldn't link introbin decodebin video pad...\n");
}
gst_object_unref(vqueue);
}
if(strncmp(gst_caps_to_string(new_pad_caps),"audio",5)==0) {
GstElement *arate;
arate=gst_bin_get_by_name(GST_BIN(introbin),"introaudiorate");
sinkpad=gst_element_get_static_pad(arate,"sink");
result=gst_pad_link(srcpad,sinkpad);
if(result!=GST_PAD_LINK_OK) {
GstCaps *peer_caps;
peer_caps=gst_pad_query_caps(sinkpad,NULL);
g_print("SinkCaps:%s\n",gst_caps_to_string(peer_caps));
g_printerr("Couldn't link introbin decodebin audio pad...\n");
gst_caps_unref(peer_caps);
}
gst_object_unref(arate);
}
}
void ofxGstRTPServer::emitDepthKeyFrame(){
GstClock * clock = gst_pipeline_get_clock(GST_PIPELINE(gst.getPipeline()));
gst_object_ref(clock);
GstClockTime time = gst_clock_get_time (clock);
GstClockTime now = time - gst_element_get_base_time(gst.getPipeline());
gst_object_unref (clock);
GstEvent * keyFrameEvent = gst_video_event_new_downstream_force_key_unit(now,
time,
now,
TRUE,
0);
gst_element_send_event(appSrcDepth,keyFrameEvent);
}
gboolean introbin_set_pad_offset(CustomData *data)
{
gint64 pos2;
pos2=gst_element_get_base_time(data->pipeline);
GstClock *clock;
clock=gst_pipeline_get_clock(GST_PIPELINE(data->pipeline));
GstClockTime clock_time;
clock_time=gst_clock_get_time(clock);
gst_object_unref(clock);
g_print("Pipeline times: base_time=%lld\n clock_time=%lld",
pos2,clock_time);
GstElement *dec=gst_bin_get_by_name(GST_BIN(data->introbin),"introdec");
GstPad *src_pad1,*src_pad2;
src_pad1=gst_element_get_static_pad(GST_ELEMENT(dec),"src_0");
gst_pad_set_offset(src_pad1,clock_time-pos2);
gst_object_unref(src_pad1);
src_pad2=gst_element_get_static_pad(GST_ELEMENT(dec),"src_1");
gst_pad_set_offset(src_pad2,clock_time-pos2);
gst_object_unref(src_pad2);
return TRUE;
}
int
main (int argc, char *argv[])
{
GstElement *bin;
GstElement *src, *spectrum, *audioconvert, *sink;
GstBus *bus;
GtkWidget *appwindow, *vbox, *widget;
gst_init (&argc, &argv);
gtk_init (&argc, &argv);
bin = gst_pipeline_new ("bin");
src = gst_element_factory_make ("audiotestsrc", "src");
g_object_set (G_OBJECT (src), "wave", 0, NULL);
spectrum = gst_element_factory_make ("spectrum", "spectrum");
g_object_set (G_OBJECT (spectrum), "bands", spect_bands, "threshold", -80,
"message", TRUE, NULL);
audioconvert = gst_element_factory_make ("audioconvert", "audioconvert");
sink = gst_element_factory_make (DEFAULT_AUDIOSINK, "sink");
gst_bin_add_many (GST_BIN (bin), src, spectrum, audioconvert, sink, NULL);
if (!gst_element_link_many (src, spectrum, audioconvert, sink, NULL)) {
fprintf (stderr, "can't link elements\n");
exit (1);
}
bus = gst_element_get_bus (bin);
gst_bus_add_watch (bus, message_handler, NULL);
gst_object_unref (bus);
sync_clock = gst_pipeline_get_clock (GST_PIPELINE (bin));
appwindow = gtk_window_new (GTK_WINDOW_TOPLEVEL);
g_signal_connect (G_OBJECT (appwindow), "destroy",
G_CALLBACK (on_window_destroy), NULL);
vbox = gtk_vbox_new (FALSE, 6);
widget = gtk_hscale_new_with_range (50.0, 20000.0, 10);
gtk_scale_set_draw_value (GTK_SCALE (widget), TRUE);
gtk_scale_set_value_pos (GTK_SCALE (widget), GTK_POS_TOP);
gtk_range_set_value (GTK_RANGE (widget), 440.0);
g_signal_connect (G_OBJECT (widget), "value-changed",
G_CALLBACK (on_frequency_changed), (gpointer) src);
gtk_box_pack_start (GTK_BOX (vbox), widget, FALSE, FALSE, 0);
drawingarea = gtk_drawing_area_new ();
gtk_widget_set_size_request (drawingarea, spect_bands, spect_height);
g_signal_connect (G_OBJECT (drawingarea), "configure-event",
G_CALLBACK (on_configure_event), (gpointer) spectrum);
gtk_box_pack_start (GTK_BOX (vbox), drawingarea, TRUE, TRUE, 0);
gtk_container_add (GTK_CONTAINER (appwindow), vbox);
gtk_widget_show_all (appwindow);
gst_element_set_state (bin, GST_STATE_PLAYING);
gtk_main ();
gst_element_set_state (bin, GST_STATE_NULL);
gst_object_unref (sync_clock);
gst_object_unref (bin);
return 0;
}
gint
main (gint argc, gchar ** argv)
{
gint res = 1;
GstElement *src, *sink;
GstElement *bin;
GstController *ctrl;
GstInterpolationControlSource *csource1, *csource2;
GstClock *clock;
GstClockID clock_id;
GstClockReturn wait_ret;
GValue vol = { 0, };
gst_init (&argc, &argv);
gst_controller_init (&argc, &argv);
/* build pipeline */
bin = gst_pipeline_new ("pipeline");
clock = gst_pipeline_get_clock (GST_PIPELINE (bin));
src = gst_element_factory_make ("audiotestsrc", "gen_audio");
if (!src) {
GST_WARNING ("need audiotestsrc from gst-plugins-base");
goto Error;
}
sink = gst_element_factory_make ("autoaudiosink", "play_audio");
if (!sink) {
GST_WARNING ("need autoaudiosink from gst-plugins-base");
goto Error;
}
gst_bin_add_many (GST_BIN (bin), src, sink, NULL);
if (!gst_element_link (src, sink)) {
GST_WARNING ("can't link elements");
goto Error;
}
/* square wave
g_object_set (G_OBJECT(src), "wave", 1, NULL);
*/
/* add a controller to the source */
if (!(ctrl = gst_controller_new (G_OBJECT (src), "freq", "volume", NULL))) {
GST_WARNING ("can't control source element");
goto Error;
}
csource1 = gst_interpolation_control_source_new ();
csource2 = gst_interpolation_control_source_new ();
gst_controller_set_control_source (ctrl, "volume",
GST_CONTROL_SOURCE (csource1));
gst_controller_set_control_source (ctrl, "freq",
GST_CONTROL_SOURCE (csource2));
/* Set interpolation mode */
gst_interpolation_control_source_set_interpolation_mode (csource1,
GST_INTERPOLATE_LINEAR);
gst_interpolation_control_source_set_interpolation_mode (csource2,
GST_INTERPOLATE_LINEAR);
/* set control values */
g_value_init (&vol, G_TYPE_DOUBLE);
g_value_set_double (&vol, 0.0);
gst_interpolation_control_source_set (csource1, 0 * GST_SECOND, &vol);
g_value_set_double (&vol, 1.0);
gst_interpolation_control_source_set (csource1, 5 * GST_SECOND, &vol);
g_object_unref (csource1);
g_value_set_double (&vol, 220.0);
gst_interpolation_control_source_set (csource2, 0 * GST_SECOND, &vol);
g_value_set_double (&vol, 3520.0);
gst_interpolation_control_source_set (csource2, 3 * GST_SECOND, &vol);
g_value_set_double (&vol, 440.0);
gst_interpolation_control_source_set (csource2, 6 * GST_SECOND, &vol);
g_object_unref (csource2);
clock_id =
gst_clock_new_single_shot_id (clock,
gst_clock_get_time (clock) + (7 * GST_SECOND));
/* run for 7 seconds */
if (gst_element_set_state (bin, GST_STATE_PLAYING)) {
if ((wait_ret = gst_clock_id_wait (clock_id, NULL)) != GST_CLOCK_OK) {
GST_WARNING ("clock_id_wait returned: %d", wait_ret);
}
gst_element_set_state (bin, GST_STATE_NULL);
}
/* cleanup */
g_object_unref (G_OBJECT (ctrl));
gst_clock_id_unref (clock_id);
gst_object_unref (G_OBJECT (clock));
gst_object_unref (G_OBJECT (bin));
res = 0;
Error:
return (res);
}