static void
print_clocking_info (GstElement * element)
{
if (!gst_element_requires_clock (element) &&
!(gst_element_provides_clock (element) &&
gst_element_get_clock (element))) {
n_print ("\n");
n_print ("Element has no clocking capabilities.");
return;
}
n_print ("\n");
n_print ("Clocking Interaction:\n");
if (gst_element_requires_clock (element)) {
n_print (" element requires a clock\n");
}
if (gst_element_provides_clock (element)) {
GstClock *clock;
clock = gst_element_get_clock (element);
if (clock)
n_print (" element provides a clock: %s\n", GST_OBJECT_NAME (clock));
else
n_print (" element is supposed to provide a clock but returned NULL\n");
}
}
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;
}
MbEvent *
handle_state_change_message (GstMessage *message)
{
MbEvent *mb_event = NULL;
GstState old_state, new_state;
GstElement *source = (GstElement *) message->src;
gst_message_parse_state_changed (message, &old_state, &new_state, NULL);
if (new_state == GST_STATE_PLAYING)
{
if (source == _mb_global_data.pipeline)
{
g_mutex_lock (&(_mb_global_data.mutex));
if (_mb_global_data.initialized == FALSE)
{
if (_mb_global_data.clock_provider == NULL)
_mb_global_data.clock_provider =
gst_element_get_clock(_mb_global_data.pipeline);
_mb_global_data.initialized = TRUE;
mb_event = create_app_event (MB_APP_INIT_DONE);
}
g_mutex_unlock (&(_mb_global_data.mutex));
}
else
{
if (strcmp (G_OBJECT_TYPE_NAME(G_OBJECT (source)), "GstBin") == 0)
mb_event = create_state_change_event (MB_BEGIN,
GST_ELEMENT_NAME (source));
}
}
return mb_event;
}
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);
}
/* this is the first function where pipeline is built */
void
setup_streamer(int narg, char *sarg[])
{
/* the message bus */
GstBus *bus;
/* this init the global sources structures */
init_global_sources();
init_global_outputs();
pipeline = gst_pipeline_new("pipeline");
bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
gst_bus_add_watch (bus, cb_papaya_pipeline_bus, loop);
gst_object_unref (bus);
switch (STREAM_TYPE){
case AUDIO_PLAYLIST:
case AUDIO_LIVE:
papaya_audio_tree_build();
break;
case VIDEO_PLAYLIST:
case VIDEO_LIVE:
case XORG_LIVE:
case DVB_LIVE:
case DV1394_LIVE:
papaya_video_tree_build();
break;
}
/* after having built the chain, let's
* play some media */
gst_element_set_state (pipeline, GST_STATE_PLAYING);
pipeclock = gst_element_get_clock(pipeline);
}
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
print_clocking_info (GstElement * element)
{
gboolean requires_clock, provides_clock;
requires_clock =
GST_OBJECT_FLAG_IS_SET (element, GST_ELEMENT_FLAG_REQUIRE_CLOCK);
provides_clock =
GST_OBJECT_FLAG_IS_SET (element, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
if (!requires_clock && !provides_clock) {
n_print ("\n");
n_print ("Element has no clocking capabilities.\n");
return;
}
n_print ("\n");
n_print ("Clocking Interaction:\n");
if (requires_clock) {
n_print (" element requires a clock\n");
}
if (provides_clock) {
GstClock *clock;
clock = gst_element_get_clock (element);
if (clock) {
n_print (" element provides a clock: %s\n", GST_OBJECT_NAME (clock));
gst_object_unref (clock);
} else
n_print (" element is supposed to provide a clock but returned NULL\n");
}
}
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;
}
/* Method: clock
* Returns: the clock of the element (as a Gst::Clock object), or nil
* if this element does not provide a clock.
*/
static VALUE
rg_clock(VALUE self)
{
GstClock *clock;
clock = gst_element_get_clock(SELF(self));
return clock != NULL ? RGST_CLOCK_NEW(clock)
: Qnil;
}
static guint64 get_gst_time_us(GstScreamQueue *self)
{
GstClock *clock = NULL;
GstClockTime time = 0;
clock = gst_element_get_clock(GST_ELEMENT(self));
if (G_LIKELY(clock)) {
time = gst_clock_get_time(clock);
}
return time / 1000;
}
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
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_src_change_state (GstElement * element,
GstStateChange transition)
{
GstDecklinkVideoSrc *self = GST_DECKLINK_VIDEO_SRC_CAST (element);
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
if (!gst_decklink_video_src_open (self)) {
ret = GST_STATE_CHANGE_FAILURE;
goto out;
}
if (self->mode == GST_DECKLINK_MODE_AUTO &&
self->video_format != GST_DECKLINK_VIDEO_FORMAT_AUTO) {
GST_WARNING_OBJECT (self, "Warning: mode=auto and format!=auto may \
not work");
}
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
g_mutex_lock (&self->input->lock);
self->input->clock_start_time = GST_CLOCK_TIME_NONE;
self->input->clock_epoch += self->input->clock_last_time;
self->input->clock_last_time = 0;
self->input->clock_offset = 0;
g_mutex_unlock (&self->input->lock);
gst_element_post_message (element,
gst_message_new_clock_provide (GST_OBJECT_CAST (element),
self->input->clock, TRUE));
self->flushing = FALSE;
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:{
GstClock *clock;
clock = gst_element_get_clock (GST_ELEMENT_CAST (self));
if (clock) {
if (clock != self->input->clock) {
gst_clock_set_master (self->input->clock, clock);
}
gst_object_unref (clock);
} else {
GST_ELEMENT_ERROR (self, STREAM, FAILED,
(NULL), ("Need a clock to go to PLAYING"));
ret = GST_STATE_CHANGE_FAILURE;
}
break;
}
default:
break;
}
static void
gst_decklink_video_src_start_streams (GstElement * element)
{
GstDecklinkVideoSrc *self = GST_DECKLINK_VIDEO_SRC_CAST (element);
HRESULT res;
if (self->input->video_enabled && (!self->input->audiosrc
|| self->input->audio_enabled)
&& (GST_STATE (self) == GST_STATE_PLAYING
|| GST_STATE_PENDING (self) == GST_STATE_PLAYING)) {
GstClock *clock;
clock = gst_element_get_clock (element);
if (!clock) {
GST_ELEMENT_ERROR (self, STREAM, FAILED, (NULL),
("Streams supposed to start but we have no clock"));
return;
}
GST_DEBUG_OBJECT (self, "Starting streams");
res = self->input->input->StartStreams ();
if (res != S_OK) {
GST_ELEMENT_ERROR (self, STREAM, FAILED,
(NULL), ("Failed to start streams: 0x%08x", res));
gst_object_unref (clock);
return;
}
self->input->started = TRUE;
self->input->clock_restart = TRUE;
// Need to unlock to get the clock time
g_mutex_unlock (&self->input->lock);
// 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->input->clock);
self->external_base_time = gst_clock_get_internal_time (clock);
gst_object_unref (clock);
g_mutex_lock (&self->input->lock);
} else {
GST_DEBUG_OBJECT (self, "Not starting streams yet");
}
}
/**
* @brief Handle buffer data.
* @return None
* @param self pointer to GstTensorSink2
* @param buffer pointer to GstBuffer to be handled
*/
static void
gst_tensor_sink2_render_buffer (GstTensorSink2 * self, GstBuffer * buffer)
{
GstClockTime now = GST_CLOCK_TIME_NONE;
guint signal_rate;
gboolean notify = FALSE;
g_return_if_fail (GST_IS_TENSOR_SINK2 (self));
signal_rate = gst_tensor_sink2_get_signal_rate (self);
if (signal_rate) {
GstClock *clock;
GstClockTime render_time;
GstClockTime last_render_time;
clock = gst_element_get_clock (GST_ELEMENT (self));
if (clock) {
now = gst_clock_get_time (clock);
last_render_time = gst_tensor_sink2_get_last_render_time (self);
/** time for next signal */
render_time = (1000 / signal_rate) * GST_MSECOND + last_render_time;
if (!GST_CLOCK_TIME_IS_VALID (last_render_time) ||
GST_CLOCK_DIFF (now, render_time) <= 0) {
/** send data after render time, or firstly received buffer */
notify = TRUE;
}
gst_object_unref (clock);
}
} else {
/** send data if signal rate is 0 */
notify = TRUE;
}
if (notify) {
gst_tensor_sink2_set_last_render_time (self, now);
if (gst_tensor_sink2_get_emit_signal (self)) {
silent_debug ("Emit signal for new data [%" GST_TIME_FORMAT "] rate [%d]",
GST_TIME_ARGS (now), signal_rate);
g_signal_emit (self, _tensor_sink2_signals[SIGNAL_NEW_DATA], 0, buffer);
}
}
}
int
main (int argc, char *argv[])
{
GstBus *bus;
gst_init (&argc, &argv);
loop = g_main_loop_new (NULL, TRUE);
pipeline = gst_pipeline_new ("pipeline");
/* setup message handling */
bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
g_signal_connect (bus, "message::error", (GCallback) message_received,
pipeline);
g_signal_connect (bus, "message::warning", (GCallback) message_received,
pipeline);
g_signal_connect (bus, "message::eos", (GCallback) eos_message_received,
pipeline);
/* we set the pipeline to PLAYING, this will distribute a default clock and
* start running. no preroll is needed */
gst_element_set_state (pipeline, GST_STATE_PLAYING);
/* get the clock now. Since we never set the pipeline to PAUSED again, the
* clock will not change, even when we add new clock providers later. */
theclock = gst_element_get_clock (pipeline);
/* start our actions while we are in the mainloop so that we can catch errors
* and other messages. */
g_idle_add ((GSourceFunc) perform_step, GINT_TO_POINTER (0));
/* go to main loop */
g_main_loop_run (loop);
gst_element_set_state (pipeline, GST_STATE_NULL);
gst_object_unref (bus);
gst_object_unref (pipeline);
gst_object_unref (theclock);
return 0;
}
/**
* @brief Push GstBuffer
*/
static void
gst_tensor_reposink_render_buffer (GstTensorRepoSink * self, GstBuffer * buffer)
{
GstClockTime now = GST_CLOCK_TIME_NONE;
guint signal_rate;
gboolean notify = FALSE;
g_return_if_fail (GST_IS_TENSOR_REPOSINK (self));
signal_rate = self->signal_rate;
if (signal_rate) {
GstClock *clock;
GstClockTime render_time;
clock = gst_element_get_clock (GST_ELEMENT (self));
if (clock) {
now = gst_clock_get_time (clock);
render_time = (1000 / signal_rate) * GST_MSECOND + self->last_render_time;
if (!GST_CLOCK_TIME_IS_VALID (self->last_render_time) ||
GST_CLOCK_DIFF (now, render_time) <= 0) {
notify = TRUE;
}
gst_object_unref (clock);
}
} else {
notify = TRUE;
}
if (notify) {
self->last_render_time = now;
if (!gst_tensor_repo_set_buffer (self->myid, self->o_myid, buffer,
self->in_caps)) {
GST_ELEMENT_ERROR (self, RESOURCE, WRITE,
("Cannot Set buffer into repo [key: %d]", self->myid), NULL);
}
}
}
static GstStateChangeReturn
gst_alsasrc_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
GstAlsaSrc *alsa = GST_ALSA_SRC (element);
GstClock *clk;
switch (transition) {
/* show the compiler that we care */
case GST_STATE_CHANGE_NULL_TO_READY:
case GST_STATE_CHANGE_READY_TO_PAUSED:
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
case GST_STATE_CHANGE_PAUSED_TO_READY:
case GST_STATE_CHANGE_READY_TO_NULL:
case GST_STATE_CHANGE_NULL_TO_NULL:
case GST_STATE_CHANGE_READY_TO_READY:
case GST_STATE_CHANGE_PAUSED_TO_PAUSED:
case GST_STATE_CHANGE_PLAYING_TO_PLAYING:
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
alsa->driver_timestamps = FALSE;
clk = gst_element_get_clock (element);
if (clk != NULL) {
if (GST_IS_SYSTEM_CLOCK (clk)) {
gint clocktype;
g_object_get (clk, "clock-type", &clocktype, NULL);
if (clocktype == GST_CLOCK_TYPE_MONOTONIC) {
GST_INFO ("Using driver timestamps !");
alsa->driver_timestamps = TRUE;
}
}
gst_object_unref (clk);
}
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
return ret;
}
static gboolean
check_last_request_time (GstPad * pad)
{
GstClockTime *last, now;
GstClock *clock;
GstElement *element = gst_pad_get_parent_element (pad);
gboolean ret = FALSE;
if (element == NULL) {
GST_ERROR_OBJECT (pad, "Cannot get parent object to get clock");
return TRUE;
}
clock = gst_element_get_clock (element);
now = gst_clock_get_time (clock);
g_object_unref (clock);
g_object_unref (element);
GST_OBJECT_LOCK (pad);
last =
g_object_get_qdata (G_OBJECT (pad), last_key_frame_request_time_quark ());
if (last == NULL) {
last = g_slice_new (GstClockTime);
g_object_set_qdata_full (G_OBJECT (pad),
last_key_frame_request_time_quark (), last,
(GDestroyNotify) kms_utils_destroy_GstClockTime);
*last = now;
ret = TRUE;
} else if (((*last) + DEFAULT_KEYFRAME_DISPERSION) < now) {
ret = TRUE;
*last = now;
}
GST_OBJECT_UNLOCK (pad);
return ret;
}
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 GstStateChangeReturn
gst_decklink_video_sink_change_state (GstElement * element,
GstStateChange transition)
{
GstDecklinkVideoSink *self = GST_DECKLINK_VIDEO_SINK_CAST (element);
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
g_mutex_lock (&self->output->lock);
self->output->clock_start_time = GST_CLOCK_TIME_NONE;
self->output->clock_epoch += self->output->clock_last_time;
self->output->clock_last_time = 0;
self->output->clock_offset = 0;
g_mutex_unlock (&self->output->lock);
gst_element_post_message (element,
gst_message_new_clock_provide (GST_OBJECT_CAST (element),
self->output->clock, TRUE));
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:{
GstClock *clock, *audio_clock;
clock = gst_element_get_clock (GST_ELEMENT_CAST (self));
if (clock) {
audio_clock = gst_decklink_output_get_audio_clock (self->output);
if (clock && clock != self->output->clock && clock != audio_clock) {
gst_clock_set_master (self->output->clock, clock);
}
gst_object_unref (clock);
if (audio_clock)
gst_object_unref (audio_clock);
} else {
GST_ELEMENT_ERROR (self, STREAM, FAILED,
(NULL), ("Need a clock to go to PLAYING"));
ret = GST_STATE_CHANGE_FAILURE;
}
break;
}
default:
break;
}
if (ret == GST_STATE_CHANGE_FAILURE)
return ret;
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
if (ret == GST_STATE_CHANGE_FAILURE)
return ret;
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_element_post_message (element,
gst_message_new_clock_lost (GST_OBJECT_CAST (element),
self->output->clock));
gst_clock_set_master (self->output->clock, NULL);
// Reset calibration to make the clock reusable next time we use it
gst_clock_set_calibration (self->output->clock, 0, 0, 1, 1);
g_mutex_lock (&self->output->lock);
self->output->clock_start_time = GST_CLOCK_TIME_NONE;
self->output->clock_epoch += self->output->clock_last_time;
self->output->clock_last_time = 0;
self->output->clock_offset = 0;
g_mutex_unlock (&self->output->lock);
gst_decklink_video_sink_stop (self);
break;
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:{
if (gst_decklink_video_sink_stop_scheduled_playback (self) ==
GST_STATE_CHANGE_FAILURE)
ret = GST_STATE_CHANGE_FAILURE;
break;
}
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:{
g_mutex_lock (&self->output->lock);
if (self->output->start_scheduled_playback)
self->output->start_scheduled_playback (self->output->videosink);
g_mutex_unlock (&self->output->lock);
break;
}
default:
break;
}
return ret;
}
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 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 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_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;
}
