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 void
gst_rtp_dtmf_prepare_timestamps (GstRTPDTMFSrc * dtmfsrc)
{
GstClock *clock;
GstClockTime base_time;
#ifdef MAEMO_BROKEN
base_time = 0;
#else
base_time = gst_element_get_base_time (GST_ELEMENT (dtmfsrc));
#endif
clock = gst_element_get_clock (GST_ELEMENT (dtmfsrc));
if (clock != NULL) {
dtmfsrc->timestamp = gst_clock_get_time (clock)
+ (MIN_INTER_DIGIT_INTERVAL * GST_MSECOND) - base_time;
dtmfsrc->start_timestamp = dtmfsrc->timestamp;
gst_object_unref (clock);
} else {
gchar *dtmf_name = gst_element_get_name (dtmfsrc);
GST_ERROR_OBJECT (dtmfsrc, "No clock set for element %s", dtmf_name);
dtmfsrc->timestamp = GST_CLOCK_TIME_NONE;
g_free (dtmf_name);
}
dtmfsrc->rtp_timestamp = dtmfsrc->ts_base +
gst_util_uint64_scale_int (gst_segment_to_running_time (&GST_BASE_SRC
(dtmfsrc)->segment, GST_FORMAT_TIME, dtmfsrc->timestamp),
dtmfsrc->clock_rate, GST_SECOND);
}
static gboolean
gst_rtp_dtmf_prepare_timestamps (GstRTPDTMFSrc * dtmfsrc)
{
GstClockTime last_stop;
GST_OBJECT_LOCK (dtmfsrc);
last_stop = dtmfsrc->last_stop;
GST_OBJECT_UNLOCK (dtmfsrc);
if (GST_CLOCK_TIME_IS_VALID (last_stop)) {
dtmfsrc->start_timestamp = last_stop;
} else {
GstClock *clock = gst_element_get_clock (GST_ELEMENT (dtmfsrc));
if (clock == NULL)
return FALSE;
dtmfsrc->start_timestamp = gst_clock_get_time (clock)
- gst_element_get_base_time (GST_ELEMENT (dtmfsrc));
gst_object_unref (clock);
}
/* If the last stop was in the past, then lets add the buffers together */
if (dtmfsrc->start_timestamp < dtmfsrc->timestamp)
dtmfsrc->start_timestamp = dtmfsrc->timestamp;
dtmfsrc->timestamp = dtmfsrc->start_timestamp;
dtmfsrc->rtp_timestamp = dtmfsrc->ts_base +
gst_util_uint64_scale_int (gst_segment_to_running_time (&GST_BASE_SRC
(dtmfsrc)->segment, GST_FORMAT_TIME, dtmfsrc->timestamp),
dtmfsrc->clock_rate, GST_SECOND);
return TRUE;
}
static void
gst_dtmf_prepare_timestamps (GstDTMFSrc * dtmfsrc)
{
GstClockTime last_stop;
GstClockTime timestamp;
GST_OBJECT_LOCK (dtmfsrc);
last_stop = dtmfsrc->last_stop;
GST_OBJECT_UNLOCK (dtmfsrc);
if (GST_CLOCK_TIME_IS_VALID (last_stop)) {
timestamp = last_stop;
} else {
GstClock *clock;
/* If there is no valid start time, lets use now as the start time */
clock = gst_element_get_clock (GST_ELEMENT (dtmfsrc));
if (clock != NULL) {
timestamp = gst_clock_get_time (clock)
- gst_element_get_base_time (GST_ELEMENT (dtmfsrc));
gst_object_unref (clock);
} else {
gchar *dtmf_name = gst_element_get_name (dtmfsrc);
GST_ERROR_OBJECT (dtmfsrc, "No clock set for element %s", dtmf_name);
dtmfsrc->timestamp = GST_CLOCK_TIME_NONE;
g_free (dtmf_name);
return;
}
}
/* Make sure the timestamp always goes forward */
if (timestamp > dtmfsrc->timestamp)
dtmfsrc->timestamp = timestamp;
}
static gboolean
gst_dshowvideosrc_push_buffer (guint8 * buffer, guint size, gpointer src_object,
GstClockTime duration)
{
GstDshowVideoSrc *src = GST_DSHOWVIDEOSRC (src_object);
GstBuffer *buf = NULL;
IPin *pPin = NULL;
HRESULT hres = S_FALSE;
AM_MEDIA_TYPE *pMediaType = NULL;
if (!buffer || size == 0 || !src) {
return FALSE;
}
/* create a new buffer assign to it the clock time as timestamp */
buf = gst_buffer_new_and_alloc (size);
GST_BUFFER_SIZE (buf) = size;
GstClock *clock = gst_element_get_clock (GST_ELEMENT (src));
GST_BUFFER_TIMESTAMP (buf) =
GST_CLOCK_DIFF (gst_element_get_base_time (GST_ELEMENT (src)), gst_clock_get_time (clock));
gst_object_unref (clock);
GST_BUFFER_DURATION (buf) = duration;
if (src->is_rgb) {
/* FOR RGB directshow decoder will return bottom-up BITMAP
* There is probably a way to get top-bottom video frames from
* the decoder...
*/
gint line = 0;
gint stride = size / src->height;
for (; line < src->height; line++) {
memcpy (GST_BUFFER_DATA (buf) + (line * stride),
buffer + (size - ((line + 1) * (stride))), stride);
}
} else {
memcpy (GST_BUFFER_DATA (buf), buffer, size);
}
GST_DEBUG ("push_buffer => pts %" GST_TIME_FORMAT "duration %"
GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
GST_TIME_ARGS (duration));
/* the negotiate() method already set caps on the source pad */
gst_buffer_set_caps (buf, GST_PAD_CAPS (GST_BASE_SRC_PAD (src)));
g_mutex_lock (src->buffer_mutex);
if (src->buffer != NULL)
gst_buffer_unref (src->buffer);
src->buffer = buf;
g_cond_signal (src->buffer_cond);
g_mutex_unlock (src->buffer_mutex);
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;
}
static gboolean
pause_play_stream (GstElement * bin, gint seconds)
{
gboolean punch_in;
GstStateChangeReturn ret;
GstClockTime now, base_time, running_time;
/* get current running time, we need this value to continue playback of
* non-live pipelines. */
now = gst_clock_get_time (theclock);
base_time = gst_element_get_base_time (bin);
running_time = now - base_time;
/* set the new bin to PAUSED, the parent bin will notice (because of the ASYNC
* message and will perform latency calculations again when going to PLAYING
* later. */
ret = gst_element_set_state (bin, GST_STATE_PAUSED);
switch (ret) {
case GST_STATE_CHANGE_NO_PREROLL:
/* live source, timestamps are running_time of the pipeline clock. */
punch_in = FALSE;
break;
case GST_STATE_CHANGE_SUCCESS:
/* success, no async state changes, same as async, timestamps start
* from 0 */
case GST_STATE_CHANGE_ASYNC:
/* no live source, bin will preroll. We have to punch it in because in
* this situation timestamps start from 0. */
punch_in = TRUE;
break;
default:
case GST_STATE_CHANGE_FAILURE:
return FALSE;
}
if (seconds)
g_usleep (seconds * G_USEC_PER_SEC);
if (punch_in) {
/* new bin has to be aligned with previous running_time. We do this by taking
* the current absolute clock time and calculating the base time that would
* give the previous running_time. We set this base_time on the bin before
* setting it to PLAYING. */
now = gst_clock_get_time (theclock);
base_time = now - running_time;
gst_element_set_base_time (bin, base_time);
}
/* now set the pipeline to PLAYING */
gst_element_set_state (bin, GST_STATE_PLAYING);
return TRUE;
}
/* 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;
}
static void
gst_wrapper_camera_bin_reset_video_src_caps (GstWrapperCameraBinSrc * self,
GstCaps * caps)
{
GstClock *clock;
gint64 base_time;
GST_DEBUG_OBJECT (self, "Resetting src caps to %" GST_PTR_FORMAT, caps);
if (self->src_vid_src) {
clock = gst_element_get_clock (self->src_vid_src);
base_time = gst_element_get_base_time (self->src_vid_src);
gst_element_set_state (self->src_vid_src, GST_STATE_READY);
set_capsfilter_caps (self, caps);
self->drop_newseg = TRUE;
GST_DEBUG_OBJECT (self, "Bringing source up");
gst_element_sync_state_with_parent (self->src_vid_src);
if (clock) {
gst_element_set_clock (self->src_vid_src, clock);
gst_element_set_base_time (self->src_vid_src, base_time);
if (GST_IS_BIN (self->src_vid_src)) {
GstIterator *it =
gst_bin_iterate_elements (GST_BIN (self->src_vid_src));
gpointer item = NULL;
gboolean done = FALSE;
while (!done) {
switch (gst_iterator_next (it, &item)) {
case GST_ITERATOR_OK:
gst_element_set_base_time (GST_ELEMENT (item), base_time);
gst_object_unref (item);
break;
case GST_ITERATOR_RESYNC:
gst_iterator_resync (it);
break;
case GST_ITERATOR_ERROR:
done = TRUE;
break;
case GST_ITERATOR_DONE:
done = TRUE;
break;
}
}
gst_iterator_free (it);
}
gst_object_unref (clock);
}
}
}
static GstStateChangeReturn
gst_decklink_video_sink_stop_scheduled_playback (GstDecklinkVideoSink * self)
{
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
GstClockTime start_time;
HRESULT res;
GstClock *clock;
if (!self->output->started)
return ret;
clock = gst_element_get_clock (GST_ELEMENT_CAST (self));
if (clock) {
// FIXME: start time is the same for the complete pipeline,
// but what we need here is the start time of this element!
start_time = gst_element_get_base_time (GST_ELEMENT (self));
if (start_time != GST_CLOCK_TIME_NONE)
start_time = gst_clock_get_time (clock) - start_time;
// FIXME: This will probably not work
if (start_time == GST_CLOCK_TIME_NONE)
start_time = 0;
convert_to_internal_clock (self, &start_time, NULL);
// The start time is now the running time when we stopped
// playback
gst_object_unref (clock);
} else {
GST_WARNING_OBJECT (self,
"No clock, stopping scheduled playback immediately");
start_time = 0;
}
GST_DEBUG_OBJECT (self,
"Stopping scheduled playback at %" GST_TIME_FORMAT,
GST_TIME_ARGS (start_time));
g_mutex_lock (&self->output->lock);
self->output->started = FALSE;
g_mutex_unlock (&self->output->lock);
res = self->output->output->StopScheduledPlayback (start_time, 0, GST_SECOND);
if (res != S_OK) {
GST_ELEMENT_ERROR (self, STREAM, FAILED,
(NULL), ("Failed to stop scheduled playback: 0x%08x", res));
ret = GST_STATE_CHANGE_FAILURE;
}
self->internal_base_time = GST_CLOCK_TIME_NONE;
self->external_base_time = GST_CLOCK_TIME_NONE;
return ret;
}
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);
}
}
static gboolean
state_change(GstElement *newbin, guint seconds)
{
gboolean kick;
GstStateChangeReturn chret;
GstClockTime base;
GstClockTime now;
GstClockTime offset;
now = gst_clock_get_time(pipeclock);
base = gst_element_get_base_time(newbin);
offset = now -base;
/* change state and evaluate return */
chret = gst_element_set_state(newbin, GST_STATE_PAUSED);
switch (chret) {
case GST_STATE_CHANGE_NO_PREROLL:
/* live source, timestamps are running_time of the
* pipeline clock. */
g_printf("no preroll\n");
kick = FALSE;
break;
case GST_STATE_CHANGE_SUCCESS:
/* success, no async state changes,
* same as async, timestamps start
* * from 0 */
g_printf("changed immediately\n");
case GST_STATE_CHANGE_ASYNC:
/* no live source, bin will preroll.
* We have to punch it in because in
* this situation timestamps
* start from 0. */
g_printf("will change async\n");
kick = TRUE;
break;
default:
case GST_STATE_CHANGE_FAILURE:
return FALSE;
}
if(seconds)
g_usleep(seconds * G_USEC_PER_SEC);
if(kick){
now = gst_clock_get_time(pipeclock);
base = now - offset;
gst_element_set_base_time(newbin, base);
}
}
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);
}
static GstFlowReturn gst_imx_v4l2src_fill(GstPushSrc *src, GstBuffer *buf)
{
GstImxV4l2VideoSrc *v4l2src = GST_IMX_V4L2SRC(src);
GstClockTime ts;
GST_LOG_OBJECT(v4l2src, "fill");
ts = gst_clock_get_time(GST_ELEMENT(v4l2src)->clock);
if (ts != GST_CLOCK_TIME_NONE)
ts -= gst_element_get_base_time(GST_ELEMENT(v4l2src));
else
ts = v4l2src->count * v4l2src->time_per_frame;
v4l2src->count++;
GST_BUFFER_TIMESTAMP(buf) = ts;
GST_BUFFER_DURATION(buf) = v4l2src->time_per_frame;
return GST_FLOW_OK;
}
/* receive spectral data from element message */
static gboolean
message_handler (GstBus * bus, GstMessage * message, gpointer data)
{
if (message->type == GST_MESSAGE_ELEMENT) {
const GstStructure *s = gst_message_get_structure (message);
const gchar *name = gst_structure_get_name (s);
if (strcmp (name, "spectrum") == 0) {
GstElement *spectrum = GST_ELEMENT (GST_MESSAGE_SRC (message));
GstClockTime timestamp, duration;
GstClockTime waittime = GST_CLOCK_TIME_NONE;
if (gst_structure_get_clock_time (s, "running-time", ×tamp) &&
gst_structure_get_clock_time (s, "duration", &duration)) {
/* wait for middle of buffer */
waittime = timestamp + duration / 2;
} else if (gst_structure_get_clock_time (s, "endtime", ×tamp)) {
waittime = timestamp;
}
if (GST_CLOCK_TIME_IS_VALID (waittime)) {
GstClockID clock_id;
GstClockTime basetime = gst_element_get_base_time (spectrum);
gfloat *spect = g_new (gfloat, spect_bands);
const GValue *list;
const GValue *value;
guint i;
list = gst_structure_get_value (s, "magnitude");
for (i = 0; i < spect_bands; ++i) {
value = gst_value_list_get_value (list, i);
spect[i] = height_scale * g_value_get_float (value);
}
clock_id =
gst_clock_new_single_shot_id (sync_clock, waittime + basetime);
gst_clock_id_wait_async (clock_id, delayed_spectrum_update,
(gpointer) spect);
gst_clock_id_unref (clock_id);
}
}
}
return TRUE;
}
/* called when we need to give data to appsrc */
static void
need_data (GstElement * appsrc, guint unused, MyContext * ctx)
{
GstBuffer *buffer;
guint size;
GstFlowReturn ret;
size = 385 * 288 * 2;
buffer = gst_buffer_new_allocate (NULL, size, NULL);
/* this makes the image black/white */
gst_buffer_memset (buffer, 0, ctx->white ? 0xff : 0x0, size);
ctx->white = !ctx->white;
/* get timestamp from clock */
//ctx->timestamp = gst_clock_get_time (global_clock);
g_print("Time is: %"G_GUINT64_FORMAT" \n", ctx->timestamp);
//remove basetime
//ctx->timestamp -= gst_element_get_base_time (GST_ELEMENT (appsrc));
GST_BUFFER_PTS (buffer) = ctx->timestamp;
g_print("PTS: %"G_GUINT64_FORMAT" BASETIME %"G_GUINT64_FORMAT" SUM %"G_GUINT64_FORMAT " \n", ctx->timestamp, gst_element_get_base_time(appsrc),
( ctx->timestamp) + ( gst_element_get_base_time(appsrc)));
GST_BUFFER_DURATION (buffer) = gst_util_uint64_scale_int (1, GST_SECOND, 2);
ctx->timestamp += GST_BUFFER_DURATION (buffer);
g_signal_emit_by_name (appsrc, "push-buffer", buffer, &ret);
}
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;
}
static void
gst_dtmf_prepare_timestamps (GstDTMFSrc * dtmfsrc)
{
GstClock *clock;
GstClockTime base_time;
base_time = gst_element_get_base_time (GST_ELEMENT (dtmfsrc));
clock = gst_element_get_clock (GST_ELEMENT (dtmfsrc));
if (clock != NULL) {
#ifdef MAEMO_BROKEN
dtmfsrc->timestamp = gst_clock_get_time (clock);
#else
dtmfsrc->timestamp = gst_clock_get_time (clock) - base_time;
#endif
gst_object_unref (clock);
} else {
gchar *dtmf_name = gst_element_get_name (dtmfsrc);
GST_ERROR_OBJECT (dtmfsrc, "No clock set for element %s", dtmf_name);
dtmfsrc->timestamp = GST_CLOCK_TIME_NONE;
g_free (dtmf_name);
}
}
static GstFlowReturn
gst_rtp_dtmf_src_create (GstBaseSrc * basesrc, guint64 offset,
guint length, GstBuffer ** buffer)
{
GstRTPDTMFSrcEvent *event;
GstRTPDTMFSrc *dtmfsrc;
GstClock *clock;
GstClockID *clockid;
GstClockReturn clockret;
GstMessage *message;
GQueue messages = G_QUEUE_INIT;
dtmfsrc = GST_RTP_DTMF_SRC (basesrc);
do {
if (dtmfsrc->payload == NULL) {
GST_DEBUG_OBJECT (dtmfsrc, "popping");
event = g_async_queue_pop (dtmfsrc->event_queue);
GST_DEBUG_OBJECT (dtmfsrc, "popped %d", event->event_type);
switch (event->event_type) {
case RTP_DTMF_EVENT_TYPE_STOP:
GST_WARNING_OBJECT (dtmfsrc,
"Received a DTMF stop event when already stopped");
gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-dropped", event);
break;
case RTP_DTMF_EVENT_TYPE_START:
dtmfsrc->first_packet = TRUE;
dtmfsrc->last_packet = FALSE;
/* Set the redundancy on the first packet */
dtmfsrc->redundancy_count = dtmfsrc->packet_redundancy;
if (!gst_rtp_dtmf_prepare_timestamps (dtmfsrc))
goto no_clock;
g_queue_push_tail (&messages,
gst_dtmf_src_prepare_message (dtmfsrc, "dtmf-event-processed",
event));
dtmfsrc->payload = event->payload;
dtmfsrc->payload->duration =
dtmfsrc->ptime * dtmfsrc->clock_rate / 1000;
event->payload = NULL;
break;
case RTP_DTMF_EVENT_TYPE_PAUSE_TASK:
/*
* We're pushing it back because it has to stay in there until
* the task is really paused (and the queue will then be flushed
*/
GST_OBJECT_LOCK (dtmfsrc);
if (dtmfsrc->paused) {
g_async_queue_push (dtmfsrc->event_queue, event);
goto paused_locked;
}
GST_OBJECT_UNLOCK (dtmfsrc);
break;
}
gst_rtp_dtmf_src_event_free (event);
} else if (!dtmfsrc->first_packet && !dtmfsrc->last_packet &&
(dtmfsrc->timestamp - dtmfsrc->start_timestamp) / GST_MSECOND >=
MIN_PULSE_DURATION) {
GST_DEBUG_OBJECT (dtmfsrc, "try popping");
event = g_async_queue_try_pop (dtmfsrc->event_queue);
if (event != NULL) {
GST_DEBUG_OBJECT (dtmfsrc, "try popped %d", event->event_type);
switch (event->event_type) {
case RTP_DTMF_EVENT_TYPE_START:
GST_WARNING_OBJECT (dtmfsrc,
"Received two consecutive DTMF start events");
gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-dropped", event);
break;
case RTP_DTMF_EVENT_TYPE_STOP:
dtmfsrc->first_packet = FALSE;
dtmfsrc->last_packet = TRUE;
/* Set the redundancy on the last packet */
dtmfsrc->redundancy_count = dtmfsrc->packet_redundancy;
g_queue_push_tail (&messages,
gst_dtmf_src_prepare_message (dtmfsrc, "dtmf-event-processed",
event));
break;
case RTP_DTMF_EVENT_TYPE_PAUSE_TASK:
/*
* We're pushing it back because it has to stay in there until
* the task is really paused (and the queue will then be flushed)
*/
GST_DEBUG_OBJECT (dtmfsrc, "pushing pause_task...");
GST_OBJECT_LOCK (dtmfsrc);
if (dtmfsrc->paused) {
g_async_queue_push (dtmfsrc->event_queue, event);
goto paused_locked;
}
GST_OBJECT_UNLOCK (dtmfsrc);
break;
}
gst_rtp_dtmf_src_event_free (event);
}
}
} while (dtmfsrc->payload == NULL);
GST_DEBUG_OBJECT (dtmfsrc, "Processed events, now lets wait on the clock");
clock = gst_element_get_clock (GST_ELEMENT (basesrc));
if (!clock)
goto no_clock;
clockid = gst_clock_new_single_shot_id (clock, dtmfsrc->timestamp +
gst_element_get_base_time (GST_ELEMENT (dtmfsrc)));
gst_object_unref (clock);
GST_OBJECT_LOCK (dtmfsrc);
if (!dtmfsrc->paused) {
dtmfsrc->clockid = clockid;
GST_OBJECT_UNLOCK (dtmfsrc);
clockret = gst_clock_id_wait (clockid, NULL);
GST_OBJECT_LOCK (dtmfsrc);
if (dtmfsrc->paused)
clockret = GST_CLOCK_UNSCHEDULED;
} else {
clockret = GST_CLOCK_UNSCHEDULED;
}
gst_clock_id_unref (clockid);
dtmfsrc->clockid = NULL;
GST_OBJECT_UNLOCK (dtmfsrc);
while ((message = g_queue_pop_head (&messages)) != NULL)
gst_element_post_message (GST_ELEMENT (dtmfsrc), message);
if (clockret == GST_CLOCK_UNSCHEDULED) {
goto paused;
}
send_last:
if (dtmfsrc->dirty)
if (!gst_rtp_dtmf_src_negotiate (basesrc))
return GST_FLOW_NOT_NEGOTIATED;
/* create buffer to hold the payload */
*buffer = gst_rtp_dtmf_src_create_next_rtp_packet (dtmfsrc);
if (dtmfsrc->redundancy_count)
dtmfsrc->redundancy_count--;
/* Only the very first one has a marker */
dtmfsrc->first_packet = FALSE;
/* This is the end of the event */
if (dtmfsrc->last_packet == TRUE && dtmfsrc->redundancy_count == 0) {
g_slice_free (GstRTPDTMFPayload, dtmfsrc->payload);
dtmfsrc->payload = NULL;
dtmfsrc->last_packet = FALSE;
}
return GST_FLOW_OK;
paused_locked:
GST_OBJECT_UNLOCK (dtmfsrc);
paused:
if (dtmfsrc->payload) {
dtmfsrc->first_packet = FALSE;
dtmfsrc->last_packet = TRUE;
/* Set the redundanc on the last packet */
dtmfsrc->redundancy_count = dtmfsrc->packet_redundancy;
goto send_last;
} else {
return GST_FLOW_FLUSHING;
}
no_clock:
GST_ELEMENT_ERROR (dtmfsrc, STREAM, MUX, ("No available clock"),
("No available clock"));
gst_pad_pause_task (GST_BASE_SRC_PAD (dtmfsrc));
return GST_FLOW_ERROR;
}
static void
gst_decklink_video_sink_start_scheduled_playback (GstElement * element)
{
GstDecklinkVideoSink *self = GST_DECKLINK_VIDEO_SINK_CAST (element);
GstClockTime start_time;
HRESULT res;
bool active;
if (self->output->video_enabled && (!self->output->audiosink
|| self->output->audio_enabled)
&& (GST_STATE (self) == GST_STATE_PLAYING
|| GST_STATE_PENDING (self) == GST_STATE_PLAYING)) {
GstClock *clock = NULL;
clock = gst_element_get_clock (element);
if (!clock) {
GST_ELEMENT_ERROR (self, STREAM, FAILED, (NULL),
("Scheduled playback supposed to start but we have no clock"));
return;
}
// Need to unlock to get the clock time
g_mutex_unlock (&self->output->lock);
// FIXME: start time is the same for the complete pipeline,
// but what we need here is the start time of this element!
start_time = gst_element_get_base_time (element);
if (start_time != GST_CLOCK_TIME_NONE)
start_time = gst_clock_get_time (clock) - start_time;
// FIXME: This will probably not work
if (start_time == GST_CLOCK_TIME_NONE)
start_time = 0;
// Current times of internal and external clock when we go to
// playing. We need this to convert the pipeline running time
// to the running time of the hardware
//
// We can't use the normal base time for the external clock
// because we might go to PLAYING later than the pipeline
self->internal_base_time =
gst_clock_get_internal_time (self->output->clock);
self->external_base_time = gst_clock_get_internal_time (clock);
convert_to_internal_clock (self, &start_time, NULL);
g_mutex_lock (&self->output->lock);
// Check if someone else started in the meantime
if (self->output->started) {
gst_object_unref (clock);
return;
}
active = false;
self->output->output->IsScheduledPlaybackRunning (&active);
if (active) {
GST_DEBUG_OBJECT (self, "Stopping scheduled playback");
self->output->started = FALSE;
res = self->output->output->StopScheduledPlayback (0, 0, 0);
if (res != S_OK) {
GST_ELEMENT_ERROR (self, STREAM, FAILED,
(NULL), ("Failed to stop scheduled playback: 0x%08x", res));
gst_object_unref (clock);
return;
}
}
GST_DEBUG_OBJECT (self,
"Starting scheduled playback at %" GST_TIME_FORMAT,
GST_TIME_ARGS (start_time));
res =
self->output->output->StartScheduledPlayback (start_time,
GST_SECOND, 1.0);
if (res != S_OK) {
GST_ELEMENT_ERROR (self, STREAM, FAILED,
(NULL), ("Failed to start scheduled playback: 0x%08x", res));
gst_object_unref (clock);
return;
}
self->output->started = TRUE;
self->output->clock_restart = TRUE;
// Need to unlock to get the clock time
g_mutex_unlock (&self->output->lock);
// Sample the clocks again to get the most accurate values
// after we started scheduled playback
self->internal_base_time =
gst_clock_get_internal_time (self->output->clock);
self->external_base_time = gst_clock_get_internal_time (clock);
g_mutex_lock (&self->output->lock);
gst_object_unref (clock);
} else {
GST_DEBUG_OBJECT (self, "Not starting scheduled playback yet");
}
}
static void
convert_to_internal_clock (GstDecklinkVideoSink * self,
GstClockTime * timestamp, GstClockTime * duration)
{
GstClock *clock, *audio_clock;
g_assert (timestamp != NULL);
clock = gst_element_get_clock (GST_ELEMENT_CAST (self));
audio_clock = gst_decklink_output_get_audio_clock (self->output);
if (clock && clock != self->output->clock && clock != audio_clock) {
GstClockTime internal, external, rate_n, rate_d;
gst_clock_get_calibration (self->output->clock, &internal, &external,
&rate_n, &rate_d);
if (self->internal_base_time != GST_CLOCK_TIME_NONE) {
GstClockTime external_timestamp = *timestamp;
GstClockTime base_time;
// Convert to the running time corresponding to both clock times
if (internal < self->internal_base_time)
internal = 0;
else
internal -= self->internal_base_time;
if (external < self->external_base_time)
external = 0;
else
external -= self->external_base_time;
// Convert timestamp to the "running time" since we started scheduled
// playback, that is the difference between the pipeline's base time
// and our own base time.
base_time = gst_element_get_base_time (GST_ELEMENT_CAST (self));
if (base_time > self->external_base_time)
base_time = 0;
else
base_time = self->external_base_time - base_time;
if (external_timestamp < base_time)
external_timestamp = 0;
else
external_timestamp = external_timestamp - base_time;
// Get the difference in the external time, note
// that the running time is external time.
// Then scale this difference and offset it to
// our internal time. Now we have the running time
// according to our internal clock.
//
// For the duration we just scale
*timestamp =
gst_clock_unadjust_with_calibration (NULL, external_timestamp,
internal, external, rate_n, rate_d);
GST_LOG_OBJECT (self,
"Converted %" GST_TIME_FORMAT " to %" GST_TIME_FORMAT " (internal: %"
GST_TIME_FORMAT " external %" GST_TIME_FORMAT " rate: %lf)",
GST_TIME_ARGS (external_timestamp), GST_TIME_ARGS (*timestamp),
GST_TIME_ARGS (internal), GST_TIME_ARGS (external),
((gdouble) rate_n) / ((gdouble) rate_d));
if (duration) {
GstClockTime external_duration = *duration;
*duration = gst_util_uint64_scale (external_duration, rate_d, rate_n);
GST_LOG_OBJECT (self,
"Converted duration %" GST_TIME_FORMAT " to %" GST_TIME_FORMAT
" (internal: %" GST_TIME_FORMAT " external %" GST_TIME_FORMAT
" rate: %lf)", GST_TIME_ARGS (external_duration),
GST_TIME_ARGS (*duration), GST_TIME_ARGS (internal),
GST_TIME_ARGS (external), ((gdouble) rate_n) / ((gdouble) rate_d));
}
} else {
GST_LOG_OBJECT (self, "No clock conversion needed, not started yet");
}
} else {
GST_LOG_OBJECT (self, "No clock conversion needed, same clocks");
}
}
static gboolean
gst_dshowvideosrc_push_buffer (guint8 * buffer, guint size, gpointer src_object,
GstClockTime duration)
{
GstDshowVideoSrc *src = GST_DSHOWVIDEOSRC (src_object);
GstBuffer *buf = NULL;
GstMapInfo map;
IPin *pPin = NULL;
HRESULT hres = S_FALSE;
AM_MEDIA_TYPE *pMediaType = NULL;
if (!buffer || size == 0 || !src) {
return FALSE;
}
/* create a new buffer assign to it the clock time as timestamp */
buf = gst_buffer_new_and_alloc (size);
gst_buffer_set_size(buf, size);
GstClock *clock = gst_element_get_clock (GST_ELEMENT (src));
GST_BUFFER_PTS (buf) =
GST_CLOCK_DIFF (gst_element_get_base_time (GST_ELEMENT (src)), gst_clock_get_time (clock));
//GST_BUFFER_DTS(buf) = GST_BUFFER_PTS (buf);
GST_BUFFER_DTS(buf) = GST_CLOCK_TIME_NONE;
GST_BUFFER_OFFSET(buf) = src->offset++;
GST_BUFFER_OFFSET_END(buf) = src->offset;
GST_BUFFER_FLAG_SET(buf, GST_BUFFER_FLAG_LIVE);
gst_object_unref (clock);
GST_BUFFER_DURATION (buf) = duration;
gst_buffer_map(buf, &map, GST_MAP_WRITE);
if (src->is_rgb) {
/* FOR RGB directshow decoder will return bottom-up BITMAP
* There is probably a way to get top-bottom video frames from
* the decoder...
*/
gint line = 0;
gint stride = size / src->height;
for (; line < src->height; line++) {
memcpy (map.data + (line * stride),
buffer + (size - ((line + 1) * (stride))), stride);
}
} else {
memcpy (map.data, buffer, size);
}
gst_buffer_unmap(buf, &map);
src->time += duration;
gst_object_sync_values (GST_OBJECT (src), src->time);
GST_DEBUG ("push_buffer => pts %" GST_TIME_FORMAT "duration %"
GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
GST_TIME_ARGS (duration));
g_mutex_lock (&src->buffer_mutex);
if (src->buffer != NULL)
gst_buffer_unref (src->buffer);
src->buffer = buf;
g_cond_signal (&src->buffer_cond);
g_mutex_unlock (&src->buffer_mutex);
return TRUE;
}
static GstFlowReturn
gst_gdiscreencapsrc_create (GstPushSrc * push_src, GstBuffer ** buf)
{
GstGDIScreenCapSrc *src = GST_GDISCREENCAPSRC (push_src);
GstBuffer *new_buf;
GstFlowReturn res;
gint new_buf_size;
GstClock *clock;
GstClockTime time;
GstClockTime base_time;
if (G_UNLIKELY (!src->info.bmiHeader.biWidth ||
!src->info.bmiHeader.biHeight)) {
GST_ELEMENT_ERROR (src, CORE, NEGOTIATION, (NULL),
("format wasn't negotiated before create function"));
return GST_FLOW_NOT_NEGOTIATED;
} else if (G_UNLIKELY (src->rate_numerator == 0 && src->frames == 1)) {
GST_DEBUG_OBJECT (src, "eos: 0 framerate, frame %d", (gint) src->frames);
return GST_FLOW_UNEXPECTED;
}
new_buf_size = GST_ROUND_UP_4 (src->info.bmiHeader.biWidth * 3) *
(-src->info.bmiHeader.biHeight);
GST_LOG_OBJECT (src,
"creating buffer of %lu bytes with %dx%d image for frame %d",
new_buf_size, src->info.bmiHeader.biWidth,
-src->info.bmiHeader.biHeight, (gint) src->frames);
res =
gst_pad_alloc_buffer_and_set_caps (GST_BASE_SRC_PAD (src),
GST_BUFFER_OFFSET_NONE, new_buf_size,
GST_PAD_CAPS (GST_BASE_SRC_PAD (push_src)), &new_buf);
if (res != GST_FLOW_OK) {
GST_DEBUG_OBJECT (src, "could not allocate buffer, reason %s",
gst_flow_get_name (res));
return res;
}
clock = gst_element_get_clock (GST_ELEMENT (src));
if (clock) {
/* Calculate sync time. */
GstClockTime frame_time =
gst_util_uint64_scale_int (src->frames * GST_SECOND,
src->rate_denominator, src->rate_numerator);
time = gst_clock_get_time (clock);
base_time = gst_element_get_base_time (GST_ELEMENT (src));
GST_BUFFER_TIMESTAMP (new_buf) = MAX (time - base_time, frame_time);
} else {
GST_BUFFER_TIMESTAMP (new_buf) = GST_CLOCK_TIME_NONE;
}
/* Do screen capture and put it into buffer... */
gst_gdiscreencapsrc_screen_capture (src, new_buf);
if (src->rate_numerator) {
GST_BUFFER_DURATION (new_buf) =
gst_util_uint64_scale_int (GST_SECOND,
src->rate_denominator, src->rate_numerator);
if (clock) {
GST_BUFFER_DURATION (new_buf) =
MAX (GST_BUFFER_DURATION (new_buf),
gst_clock_get_time (clock) - time);
}
} else {
/* NONE means forever */
GST_BUFFER_DURATION (new_buf) = GST_CLOCK_TIME_NONE;
}
GST_BUFFER_OFFSET (new_buf) = src->frames;
src->frames++;
GST_BUFFER_OFFSET_END (new_buf) = src->frames;
gst_object_unref (clock);
*buf = new_buf;
return GST_FLOW_OK;
}
static GstFlowReturn
gst_dtmf_src_create (GstBaseSrc * basesrc, guint64 offset,
guint length, GstBuffer ** buffer)
{
GstBuffer *buf = NULL;
GstDTMFSrcEvent *event;
GstDTMFSrc *dtmfsrc;
GstClock *clock;
GstClockID *clockid;
GstClockReturn clockret;
dtmfsrc = GST_DTMF_SRC (basesrc);
do {
if (dtmfsrc->last_event == NULL) {
GST_DEBUG_OBJECT (dtmfsrc, "popping");
event = g_async_queue_pop (dtmfsrc->event_queue);
GST_DEBUG_OBJECT (dtmfsrc, "popped %d", event->event_type);
switch (event->event_type) {
case DTMF_EVENT_TYPE_STOP:
GST_WARNING_OBJECT (dtmfsrc,
"Received a DTMF stop event when already stopped");
gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-dropped", event);
break;
case DTMF_EVENT_TYPE_START:
gst_dtmf_prepare_timestamps (dtmfsrc);
event->packet_count = 0;
dtmfsrc->last_event = event;
event = NULL;
gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-processed",
dtmfsrc->last_event);
break;
case DTMF_EVENT_TYPE_PAUSE_TASK:
/*
* We're pushing it back because it has to stay in there until
* the task is really paused (and the queue will then be flushed)
*/
GST_DEBUG_OBJECT (dtmfsrc, "pushing pause_task...");
GST_OBJECT_LOCK (dtmfsrc);
if (dtmfsrc->paused) {
g_async_queue_push (dtmfsrc->event_queue, event);
goto paused_locked;
}
GST_OBJECT_UNLOCK (dtmfsrc);
break;
}
if (event)
g_slice_free (GstDTMFSrcEvent, event);
} else if (dtmfsrc->last_event->packet_count * dtmfsrc->interval >=
MIN_DUTY_CYCLE) {
event = g_async_queue_try_pop (dtmfsrc->event_queue);
if (event != NULL) {
switch (event->event_type) {
case DTMF_EVENT_TYPE_START:
GST_WARNING_OBJECT (dtmfsrc,
"Received two consecutive DTMF start events");
gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-dropped", event);
break;
case DTMF_EVENT_TYPE_STOP:
g_slice_free (GstDTMFSrcEvent, dtmfsrc->last_event);
dtmfsrc->last_event = NULL;
gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-processed", event);
break;
case DTMF_EVENT_TYPE_PAUSE_TASK:
/*
* We're pushing it back because it has to stay in there until
* the task is really paused (and the queue will then be flushed)
*/
GST_DEBUG_OBJECT (dtmfsrc, "pushing pause_task...");
GST_OBJECT_LOCK (dtmfsrc);
if (dtmfsrc->paused) {
g_async_queue_push (dtmfsrc->event_queue, event);
goto paused_locked;
}
GST_OBJECT_UNLOCK (dtmfsrc);
break;
}
g_slice_free (GstDTMFSrcEvent, event);
}
}
} while (dtmfsrc->last_event == NULL);
GST_LOG_OBJECT (dtmfsrc, "end event check, now wait for the proper time");
clock = gst_element_get_clock (GST_ELEMENT (basesrc));
clockid = gst_clock_new_single_shot_id (clock, dtmfsrc->timestamp +
gst_element_get_base_time (GST_ELEMENT (dtmfsrc)));
gst_object_unref (clock);
GST_OBJECT_LOCK (dtmfsrc);
if (!dtmfsrc->paused) {
dtmfsrc->clockid = clockid;
GST_OBJECT_UNLOCK (dtmfsrc);
clockret = gst_clock_id_wait (clockid, NULL);
GST_OBJECT_LOCK (dtmfsrc);
if (dtmfsrc->paused)
clockret = GST_CLOCK_UNSCHEDULED;
} else {
clockret = GST_CLOCK_UNSCHEDULED;
}
gst_clock_id_unref (clockid);
dtmfsrc->clockid = NULL;
GST_OBJECT_UNLOCK (dtmfsrc);
if (clockret == GST_CLOCK_UNSCHEDULED) {
goto paused;
}
buf = gst_dtmf_src_create_next_tone_packet (dtmfsrc, dtmfsrc->last_event);
GST_LOG_OBJECT (dtmfsrc, "Created buffer of size %" G_GSIZE_FORMAT,
gst_buffer_get_size (buf));
*buffer = buf;
return GST_FLOW_OK;
paused_locked:
GST_OBJECT_UNLOCK (dtmfsrc);
paused:
if (dtmfsrc->last_event) {
GST_DEBUG_OBJECT (dtmfsrc, "Stopping current event");
/* Don't forget to release the stream lock */
g_slice_free (GstDTMFSrcEvent, dtmfsrc->last_event);
dtmfsrc->last_event = NULL;
}
return GST_FLOW_FLUSHING;
}
static GstFlowReturn
new_sample_cb (GstElement * appsink, gpointer user_data)
{
GstElement *appsrc = GST_ELEMENT (user_data);
GstFlowReturn ret;
GstSample *sample;
GstBuffer *buffer;
GstClockTime *base_time;
GstPad *src, *sink;
g_signal_emit_by_name (appsink, "pull-sample", &sample);
if (sample == NULL)
return GST_FLOW_OK;
buffer = gst_sample_get_buffer (sample);
if (buffer == NULL) {
ret = GST_FLOW_OK;
goto end;
}
gst_buffer_ref (buffer);
buffer = gst_buffer_make_writable (buffer);
KMS_ELEMENT_LOCK (GST_OBJECT_PARENT (appsrc));
base_time =
g_object_get_data (G_OBJECT (GST_OBJECT_PARENT (appsrc)), BASE_TIME_DATA);
if (base_time == NULL) {
GstClock *clock;
clock = gst_element_get_clock (appsrc);
base_time = g_slice_new0 (GstClockTime);
g_object_set_data_full (G_OBJECT (GST_OBJECT_PARENT (appsrc)),
BASE_TIME_DATA, base_time, release_gst_clock);
*base_time =
gst_clock_get_time (clock) - gst_element_get_base_time (appsrc);
g_object_unref (clock);
GST_DEBUG ("Setting base time to: %" G_GUINT64_FORMAT, *base_time);
}
src = gst_element_get_static_pad (appsrc, "src");
sink = gst_pad_get_peer (src);
if (sink != NULL) {
if (GST_OBJECT_FLAG_IS_SET (sink, GST_PAD_FLAG_EOS)) {
GST_INFO_OBJECT (sink, "Sending flush events");
gst_pad_send_event (sink, gst_event_new_flush_start ());
gst_pad_send_event (sink, gst_event_new_flush_stop (FALSE));
}
g_object_unref (sink);
}
g_object_unref (src);
if (GST_BUFFER_PTS_IS_VALID (buffer))
buffer->pts += *base_time;
if (GST_BUFFER_DTS_IS_VALID (buffer))
buffer->dts += *base_time;
KMS_ELEMENT_UNLOCK (GST_OBJECT_PARENT (appsrc));
// TODO: Do something to fix a possible previous EOS event
g_signal_emit_by_name (appsrc, "push-buffer", buffer, &ret);
gst_buffer_unref (buffer);
if (ret != GST_FLOW_OK) {
/* something wrong */
GST_ERROR ("Could not send buffer to appsrc %s. Cause: %s",
GST_ELEMENT_NAME (appsrc), gst_flow_get_name (ret));
}
end:
if (sample != NULL)
gst_sample_unref (sample);
return ret;
}
static GstFlowReturn
gst_fake_src_create (GstBaseSrc * basesrc, guint64 offset, guint length,
GstBuffer ** ret)
{
GstFakeSrc *src;
GstBuffer *buf;
GstClockTime time;
gsize size;
src = GST_FAKE_SRC (basesrc);
buf = gst_fake_src_create_buffer (src, &size);
GST_BUFFER_OFFSET (buf) = offset;
if (src->datarate > 0) {
time = (src->bytes_sent * GST_SECOND) / src->datarate;
GST_BUFFER_DURATION (buf) = size * GST_SECOND / src->datarate;
} else if (gst_base_src_is_live (basesrc)) {
GstClock *clock;
clock = gst_element_get_clock (GST_ELEMENT (src));
if (clock) {
time = gst_clock_get_time (clock);
time -= gst_element_get_base_time (GST_ELEMENT (src));
gst_object_unref (clock);
} else {
/* not an error not to have a clock */
time = GST_CLOCK_TIME_NONE;
}
} else {
time = GST_CLOCK_TIME_NONE;
}
GST_BUFFER_DTS (buf) = time;
GST_BUFFER_PTS (buf) = time;
if (!src->silent) {
gchar dts_str[64], pts_str[64], dur_str[64];
gchar *flag_str;
GST_OBJECT_LOCK (src);
g_free (src->last_message);
if (GST_BUFFER_DTS (buf) != GST_CLOCK_TIME_NONE) {
g_snprintf (dts_str, sizeof (dts_str), "%" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_DTS (buf)));
} else {
g_strlcpy (dts_str, "none", sizeof (dts_str));
}
if (GST_BUFFER_PTS (buf) != GST_CLOCK_TIME_NONE) {
g_snprintf (pts_str, sizeof (pts_str), "%" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_PTS (buf)));
} else {
g_strlcpy (pts_str, "none", sizeof (pts_str));
}
if (GST_BUFFER_DURATION (buf) != GST_CLOCK_TIME_NONE) {
g_snprintf (dur_str, sizeof (dur_str), "%" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
} else {
g_strlcpy (dur_str, "none", sizeof (dur_str));
}
flag_str = gst_buffer_get_flags_string (buf);
src->last_message =
g_strdup_printf ("create ******* (%s:%s) (%u bytes, dts: %s, pts:%s"
", duration: %s, offset: %" G_GINT64_FORMAT ", offset_end: %"
G_GINT64_FORMAT ", flags: %08x %s) %p",
GST_DEBUG_PAD_NAME (GST_BASE_SRC_CAST (src)->srcpad), (guint) size,
dts_str, pts_str, dur_str, GST_BUFFER_OFFSET (buf),
GST_BUFFER_OFFSET_END (buf), GST_MINI_OBJECT_CAST (buf)->flags,
flag_str, buf);
g_free (flag_str);
GST_OBJECT_UNLOCK (src);
g_object_notify_by_pspec ((GObject *) src, pspec_last_message);
}
if (src->signal_handoffs) {
GST_LOG_OBJECT (src, "pre handoff emit");
g_signal_emit (src, gst_fake_src_signals[SIGNAL_HANDOFF], 0, buf,
basesrc->srcpad);
GST_LOG_OBJECT (src, "post handoff emit");
}
src->bytes_sent += size;
*ret = buf;
return GST_FLOW_OK;
}
static GstFlowReturn
gst_dasf_src_create (GstAudioSrc *audiosrc,
guint64 offset,
guint length,
GstBuffer **buffer)
{
GstDasfSrc* self = GST_DASF_SRC (audiosrc);
GstBaseAudioSrc *baseaudiosrc = GST_BASE_AUDIO_SRC (self);
GstBuffer* gst_buffer = NULL;
OMX_BUFFERHEADERTYPE* omx_buffer = NULL;
GstDasfSrcPrivate* priv = GST_DASF_SRC_GET_PRIVATE (self);
GstGooAudioFilter* me = self->peer_element;
GST_DEBUG ("");
if (me->component->cur_state != OMX_StateExecuting)
{
return GST_FLOW_UNEXPECTED;
}
GST_DEBUG ("goo stuff");
{
omx_buffer = goo_port_grab_buffer (me->outport);
if (gst_pad_alloc_buffer (GST_BASE_SRC_PAD (self),
priv->outcount,
omx_buffer->nFilledLen,
GST_PAD_CAPS (GST_BASE_SRC_PAD (self)),
&gst_buffer) == GST_FLOW_OK)
{
if (GST_IS_GOO_BUFFER (gst_buffer))
{
memcpy (GST_BUFFER_DATA (gst_buffer),
omx_buffer->pBuffer,
omx_buffer->nFilledLen);
goo_component_release_buffer (me->component,
omx_buffer);
}
else
{
gst_buffer_unref (gst_buffer);
gst_buffer = (GstBuffer*) gst_goo_buffer_new ();
gst_goo_buffer_set_data (gst_buffer,
me->component,
omx_buffer);
}
}
else
{
goto fail;
}
}
GST_DEBUG ("gst stuff");
{
GstClock* clock = NULL;
GstClockTime timestamp, duration;
clock = gst_element_get_clock (GST_ELEMENT (self));
timestamp = gst_clock_get_time (clock);
timestamp -= gst_element_get_base_time (GST_ELEMENT (self));
gst_object_unref (clock);
GST_BUFFER_TIMESTAMP (gst_buffer) = gst_util_uint64_scale_int (GST_SECOND, priv->outcount, 50);
/* Set 20 millisecond duration */
duration = gst_util_uint64_scale_int
(GST_SECOND,
1,
50);
GST_BUFFER_DURATION (gst_buffer) = duration;
GST_BUFFER_OFFSET (gst_buffer) = priv->outcount++;
GST_BUFFER_OFFSET_END (gst_buffer) = priv->outcount;
gst_buffer_set_caps (gst_buffer,
GST_PAD_CAPS (GST_BASE_SRC_PAD (self)));
}
beach:
*buffer = gst_buffer;
return GST_FLOW_OK;
fail:
if (G_LIKELY (*buffer))
{
gst_buffer_unref (*buffer);
}
return GST_FLOW_ERROR;
}
void
gst_decklink_video_src_convert_to_external_clock (GstDecklinkVideoSrc * self,
GstClockTime * timestamp, GstClockTime * duration)
{
GstClock *clock;
g_assert (timestamp != NULL);
if (*timestamp == GST_CLOCK_TIME_NONE)
return;
clock = gst_element_get_clock (GST_ELEMENT_CAST (self));
if (clock && clock != self->input->clock) {
GstClockTime internal, external, rate_n, rate_d;
GstClockTimeDiff external_start_time_diff;
gst_clock_get_calibration (self->input->clock, &internal, &external,
&rate_n, &rate_d);
if (rate_n != rate_d && self->internal_base_time != GST_CLOCK_TIME_NONE) {
GstClockTime internal_timestamp = *timestamp;
// Convert to the running time corresponding to both clock times
internal -= self->internal_base_time;
external -= self->external_base_time;
// Get the difference in the internal time, note
// that the capture time is internal time.
// Then scale this difference and offset it to
// our external time. Now we have the running time
// according to our external clock.
//
// For the duration we just scale
*timestamp =
gst_clock_adjust_with_calibration (NULL, internal_timestamp, internal,
external, rate_n, rate_d);
GST_LOG_OBJECT (self,
"Converted %" GST_TIME_FORMAT " to %" GST_TIME_FORMAT " (external: %"
GST_TIME_FORMAT " internal %" GST_TIME_FORMAT " rate: %lf)",
GST_TIME_ARGS (internal_timestamp), GST_TIME_ARGS (*timestamp),
GST_TIME_ARGS (external), GST_TIME_ARGS (internal),
((gdouble) rate_n) / ((gdouble) rate_d));
if (duration) {
GstClockTime internal_duration = *duration;
*duration = gst_util_uint64_scale (internal_duration, rate_d, rate_n);
GST_LOG_OBJECT (self,
"Converted duration %" GST_TIME_FORMAT " to %" GST_TIME_FORMAT
" (external: %" GST_TIME_FORMAT " internal %" GST_TIME_FORMAT
" rate: %lf)", GST_TIME_ARGS (internal_duration),
GST_TIME_ARGS (*duration), GST_TIME_ARGS (external),
GST_TIME_ARGS (internal), ((gdouble) rate_n) / ((gdouble) rate_d));
}
} else {
GST_LOG_OBJECT (self, "No clock conversion needed, relative rate is 1.0");
}
// Add the diff between the external time when we
// went to playing and the external time when the
// pipeline went to playing. Otherwise we will
// always start outputting from 0 instead of the
// current running time.
external_start_time_diff =
gst_element_get_base_time (GST_ELEMENT_CAST (self));
external_start_time_diff =
self->external_base_time - external_start_time_diff;
*timestamp += external_start_time_diff;
} else {
GST_LOG_OBJECT (self, "No clock conversion needed, same clocks");
}
}
static GstFlowReturn
gst_gdiscreencapsrc_create (GstPushSrc * push_src, GstBuffer ** buf)
{
GstGDIScreenCapSrc *src = GST_GDISCREENCAPSRC (push_src);
GstBuffer *new_buf;
gint new_buf_size;
GstClock *clock;
GstClockTime buf_time, buf_dur;
guint64 frame_number;
if (G_UNLIKELY (!src->info.bmiHeader.biWidth ||
!src->info.bmiHeader.biHeight)) {
GST_ELEMENT_ERROR (src, CORE, NEGOTIATION, (NULL),
("format wasn't negotiated before create function"));
return GST_FLOW_NOT_NEGOTIATED;
}
new_buf_size = GST_ROUND_UP_4 (src->info.bmiHeader.biWidth * 3) *
(-src->info.bmiHeader.biHeight);
GST_LOG_OBJECT (src,
"creating buffer of %d bytes with %dx%d image",
new_buf_size, (gint) src->info.bmiHeader.biWidth,
(gint) (-src->info.bmiHeader.biHeight));
new_buf = gst_buffer_new_and_alloc (new_buf_size);
clock = gst_element_get_clock (GST_ELEMENT (src));
if (clock != NULL) {
GstClockTime time, base_time;
/* Calculate sync time. */
time = gst_clock_get_time (clock);
base_time = gst_element_get_base_time (GST_ELEMENT (src));
buf_time = time - base_time;
if (src->rate_numerator) {
frame_number = gst_util_uint64_scale (buf_time,
src->rate_numerator, GST_SECOND * src->rate_denominator);
} else {
frame_number = -1;
}
} else {
buf_time = GST_CLOCK_TIME_NONE;
frame_number = -1;
}
if (frame_number != -1 && frame_number == src->frame_number) {
GstClockID id;
GstClockReturn ret;
/* Need to wait for the next frame */
frame_number += 1;
/* Figure out what the next frame time is */
buf_time = gst_util_uint64_scale (frame_number,
src->rate_denominator * GST_SECOND, src->rate_numerator);
id = gst_clock_new_single_shot_id (clock,
buf_time + gst_element_get_base_time (GST_ELEMENT (src)));
GST_OBJECT_LOCK (src);
src->clock_id = id;
GST_OBJECT_UNLOCK (src);
GST_DEBUG_OBJECT (src, "Waiting for next frame time %" G_GUINT64_FORMAT,
buf_time);
ret = gst_clock_id_wait (id, NULL);
GST_OBJECT_LOCK (src);
gst_clock_id_unref (id);
src->clock_id = NULL;
if (ret == GST_CLOCK_UNSCHEDULED) {
/* Got woken up by the unlock function */
GST_OBJECT_UNLOCK (src);
return GST_FLOW_FLUSHING;
}
GST_OBJECT_UNLOCK (src);
/* Duration is a complete 1/fps frame duration */
buf_dur =
gst_util_uint64_scale_int (GST_SECOND, src->rate_denominator,
src->rate_numerator);
} else if (frame_number != -1) {
GstClockTime next_buf_time;
GST_DEBUG_OBJECT (src, "No need to wait for next frame time %"
G_GUINT64_FORMAT " next frame = %" G_GINT64_FORMAT " prev = %"
G_GINT64_FORMAT, buf_time, frame_number, src->frame_number);
next_buf_time = gst_util_uint64_scale (frame_number + 1,
src->rate_denominator * GST_SECOND, src->rate_numerator);
/* Frame duration is from now until the next expected capture time */
buf_dur = next_buf_time - buf_time;
} else {
buf_dur = GST_CLOCK_TIME_NONE;
}
src->frame_number = frame_number;
GST_BUFFER_TIMESTAMP (new_buf) = buf_time;
GST_BUFFER_DURATION (new_buf) = buf_dur;
/* Do screen capture and put it into buffer... */
gst_gdiscreencapsrc_screen_capture (src, new_buf);
gst_object_unref (clock);
*buf = new_buf;
return GST_FLOW_OK;
}
static GstFlowReturn
gst_genicamsrc_create (GstPushSrc * psrc, GstBuffer ** buf)
{
GstGenicamSrc *src = GST_GENICAM_SRC (psrc);
guint32 dropped_frames = 0;
GstClock *clock;
GstClockTime clock_time;
GST_LOG_OBJECT (src, "create");
*buf = gst_genicamsrc_get_buffer (src);
if (!*buf) {
return GST_FLOW_ERROR;
}
clock = gst_element_get_clock (GST_ELEMENT (src));
clock_time = gst_clock_get_time (clock);
gst_object_unref (clock);
/* check for dropped frames and disrupted signal */
//dropped_frames = (circ_handle.FrameCount - src->last_frame_count) - 1;
if (dropped_frames > 0) {
src->total_dropped_frames += dropped_frames;
GST_WARNING_OBJECT (src, "Dropped %d frames (%d total)", dropped_frames,
src->total_dropped_frames);
} else if (dropped_frames < 0) {
GST_WARNING_OBJECT (src, "Frame count non-monotonic, signal disrupted?");
}
//src->last_frame_count = circ_handle.FrameCount;
/* create GstBuffer then release circ buffer back to acquisition */
//*buf = gst_genicamsrc_create_buffer_from_circ_handle (src, &circ_handle);
//ret =
// BiCirStatusSet (src->board, &src->buffer_array, circ_handle, BIAVAILABLE);
//if (ret != BI_OK) {
// GST_ELEMENT_ERROR (src, RESOURCE, FAILED,
// ("Failed to release buffer: %s", gst_genicamsrc_get_error_string (src,
// ret)), (NULL));
// return GST_FLOW_ERROR;
//}
/* TODO: understand why timestamps for circ_handle are sometimes 0 */
//GST_BUFFER_TIMESTAMP (*buf) =
// GST_CLOCK_DIFF (gst_element_get_base_time (GST_ELEMENT (src)),
// src->acq_start_time + circ_handle.HiResTimeStamp.totalSec * GST_SECOND);
GST_BUFFER_TIMESTAMP (*buf) =
GST_CLOCK_DIFF (gst_element_get_base_time (GST_ELEMENT (src)),
clock_time);
//GST_BUFFER_OFFSET (*buf) = circ_handle.FrameCount - 1;
if (src->stop_requested) {
if (*buf != NULL) {
gst_buffer_unref (*buf);
*buf = NULL;
}
return GST_FLOW_FLUSHING;
}
return GST_FLOW_OK;
error:
return GST_FLOW_ERROR;
}