/**
* gst_clock_set_master
* @clock: a #GstClock
* @master: a master #GstClock
*
* Set @master as the master clock for @clock. @clock will be automatically
* calibrated so that gst_clock_get_time() reports the same time as the
* master clock.
*
* A clock provider that slaves its clock to a master can get the current
* calibration values with gst_clock_get_calibration().
*
* @master can be %NULL in which case @clock will not be slaved anymore. It will
* however keep reporting its time adjusted with the last configured rate
* and time offsets.
*
* Returns: %TRUE if the clock is capable of being slaved to a master clock.
* Trying to set a master on a clock without the
* #GST_CLOCK_FLAG_CAN_SET_MASTER flag will make this function return %FALSE.
*
* MT safe.
*/
gboolean
gst_clock_set_master (GstClock * clock, GstClock * master)
{
GstClock **master_p;
g_return_val_if_fail (GST_IS_CLOCK (clock), FALSE);
g_return_val_if_fail (master != clock, FALSE);
GST_OBJECT_LOCK (clock);
/* we always allow setting the master to NULL */
if (master && !GST_OBJECT_FLAG_IS_SET (clock, GST_CLOCK_FLAG_CAN_SET_MASTER))
goto not_supported;
GST_CAT_DEBUG_OBJECT (GST_CAT_CLOCK, clock,
"slaving %p to master clock %p", clock, master);
master_p = &clock->master;
gst_object_replace ((GstObject **) master_p, (GstObject *) master);
GST_OBJECT_UNLOCK (clock);
GST_CLOCK_SLAVE_LOCK (clock);
if (clock->clockid) {
gst_clock_id_unschedule (clock->clockid);
gst_clock_id_unref (clock->clockid);
clock->clockid = NULL;
}
if (master) {
clock->filling = TRUE;
clock->time_index = 0;
/* use the master periodic id to schedule sampling and
* clock calibration. */
clock->clockid = gst_clock_new_periodic_id (master,
gst_clock_get_time (master), clock->timeout);
gst_clock_id_wait_async (clock->clockid,
(GstClockCallback) gst_clock_slave_callback, clock);
}
GST_CLOCK_SLAVE_UNLOCK (clock);
return TRUE;
/* ERRORS */
not_supported:
{
GST_CAT_DEBUG_OBJECT (GST_CAT_CLOCK, clock,
"cannot be slaved to a master clock");
GST_OBJECT_UNLOCK (clock);
return FALSE;
}
}
static gboolean
_stop_pad (GstAggregator * self, GstAggregatorPad * pad, gpointer unused_udata)
{
_aggpad_flush (pad, self);
PAD_LOCK_EVENT (pad);
/* remove the timeouts */
if (pad->priv->timeout_id) {
gst_clock_id_unschedule (pad->priv->timeout_id);
gst_clock_id_unref (pad->priv->timeout_id);
pad->priv->timeout_id = NULL;
}
PAD_UNLOCK_EVENT (pad);
return TRUE;
}
static gboolean gstreamill_monitor (GstClock *clock, GstClockTime time, GstClockID id, gpointer user_data)
{
GstClockID nextid;
GstClockReturn ret;
GstClockTime now;
Gstreamill *gstreamill;
GSList *list;
gstreamill = (Gstreamill *)user_data;
g_mutex_lock (&(gstreamill->job_list_mutex));
/* remove stoped job from job list */
clean_job_list (gstreamill);
/* stop? */
if (gstreamill->stop && g_slist_length (gstreamill->job_list) == 0) {
GST_ERROR ("streamill stopped");
exit (0);
}
/* check job stat */
if (!gstreamill->stop) {
list = gstreamill->job_list;
g_slist_foreach (list, job_check_func, gstreamill);
}
/* log rotate. */
if (gstreamill->daemon) {
log_rotate (gstreamill);
dvr_clean (gstreamill);
}
g_mutex_unlock (&(gstreamill->job_list_mutex));
/* register streamill monitor */
now = gst_clock_get_time (gstreamill->system_clock);
nextid = gst_clock_new_single_shot_id (gstreamill->system_clock, now + 2000 * GST_MSECOND);
ret = gst_clock_id_wait_async (nextid, gstreamill_monitor, gstreamill, NULL);
gst_clock_id_unref (nextid);
if (ret != GST_CLOCK_OK) {
GST_WARNING ("Register gstreamill monitor failure");
return FALSE;
}
return TRUE;
}
static void gst_imx_v4l2src_apply_focus_settings(GstImxV4l2VideoSrc *v4l2src,
gboolean activate)
{
int locks, range;
/* even when activating, first ensure that it is not running */
/* ensure that continuous autofocus is not running */
v4l2_s_ctrl(v4l2src, V4L2_CID_FOCUS_AUTO, 0);
/* ensure that single shot AF is not running */
v4l2_s_ctrl(v4l2src, V4L2_CID_AUTO_FOCUS_STOP, 0);
if (v4l2src->af_clock_id) {
gst_clock_id_unschedule(v4l2src->af_clock_id);
gst_clock_id_unref(v4l2src->af_clock_id);
v4l2src->af_clock_id = NULL;
}
/* ensure that focus is not locked */
if (v4l2_g_ctrl(v4l2src, V4L2_CID_3A_LOCK, &locks) == 0 && (locks & V4L2_LOCK_FOCUS))
v4l2_s_ctrl(v4l2src, V4L2_CID_3A_LOCK, locks & ~V4L2_LOCK_FOCUS);
if (activate) {
/* set focus range */
switch (v4l2src->focus_mode) {
case GST_PHOTOGRAPHY_FOCUS_MODE_AUTO:
range = V4L2_AUTO_FOCUS_RANGE_AUTO;
break;
case GST_PHOTOGRAPHY_FOCUS_MODE_MACRO:
range = V4L2_AUTO_FOCUS_RANGE_MACRO;
break;
case GST_PHOTOGRAPHY_FOCUS_MODE_INFINITY:
range = V4L2_AUTO_FOCUS_RANGE_INFINITY;
break;
default:
range = V4L2_AUTO_FOCUS_RANGE_NORMAL;
break;
}
v4l2_s_ctrl(v4l2src, V4L2_CID_AUTO_FOCUS_RANGE, range);
/* enable continuous autofocus if requested */
if (v4l2src->focus_mode == GST_PHOTOGRAPHY_FOCUS_MODE_CONTINUOUS_NORMAL)
v4l2_s_ctrl(v4l2src, V4L2_CID_FOCUS_AUTO, 1);
}
}
static gboolean
remove_clock_cache (GstClock * clock, GstClockTime time, GstClockID id,
gpointer user_data)
{
ClockCache *cache = user_data;
G_LOCK (clocks_lock);
if (!cache->clocks) {
gst_clock_id_unref (cache->remove_id);
gst_object_unref (cache->clock);
clocks = g_list_remove (clocks, cache);
g_free (cache);
}
G_UNLOCK (clocks_lock);
return TRUE;
}
static gboolean gst_imx_v4l2src_af_status_cb(GstClock *clock, GstClockTime time,
GstClockID id, gpointer user_data)
{
GstImxV4l2VideoSrc *v4l2src = GST_IMX_V4L2SRC(user_data);
g_mutex_lock(&v4l2src->af_mutex);
if (v4l2src->af_clock_id == id) {
gst_clock_id_unref(v4l2src->af_clock_id);
v4l2src->af_clock_id = NULL;
gst_imx_v4l2src_af_check_status(v4l2src);
}
g_mutex_unlock(&v4l2src->af_mutex);
return TRUE;
}
static void
gst_directsound_src_finalize (GObject * object)
{
GstDirectSoundSrc *dsoundsrc = GST_DIRECTSOUND_SRC (object);
g_mutex_clear (&dsoundsrc->dsound_lock);
gst_object_unref (dsoundsrc->system_clock);
if (dsoundsrc->read_wait_clock_id != NULL)
gst_clock_id_unref (dsoundsrc->read_wait_clock_id);
g_free (dsoundsrc->device_name);
g_free (dsoundsrc->device_id);
g_free (dsoundsrc->device_guid);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
/**
* gstreamill_start:
* @gstreamill: (in): gstreamill to be starting
*
* start gstreamill
*
* Returns: 0 on success.
*/
gint gstreamill_start (Gstreamill *gstreamill)
{
GstClockID id;
GstClockTime t;
GstClockReturn ret;
/* regist gstreamill monitor */
t = gst_clock_get_time (gstreamill->system_clock) + 5000 * GST_MSECOND;
id = gst_clock_new_single_shot_id (gstreamill->system_clock, t);
ret = gst_clock_id_wait_async (id, gstreamill_monitor, gstreamill, NULL);
gst_clock_id_unref (id);
if (ret != GST_CLOCK_OK) {
GST_WARNING ("Regist gstreamill monitor failure");
return 1;
}
return 0;
}
static gpointer
no_rtcp_timeout_func (gpointer user_data)
{
FsRtpSubStream *self = FS_RTP_SUB_STREAM (user_data);
GstClock *sysclock = NULL;
GstClockID id;
gboolean emit = TRUE;
sysclock = gst_system_clock_obtain ();
if (sysclock == NULL)
goto no_sysclock;
FS_RTP_SUB_STREAM_LOCK(self);
id = self->priv->no_rtcp_timeout_id = gst_clock_new_single_shot_id (sysclock,
self->priv->next_no_rtcp_timeout);
FS_RTP_SUB_STREAM_UNLOCK(self);
gst_clock_id_wait (id, NULL);
FS_RTP_SUB_STREAM_LOCK(self);
gst_clock_id_unref (id);
self->priv->no_rtcp_timeout_id = NULL;
if (self->priv->next_no_rtcp_timeout == 0)
emit = FALSE;
FS_RTP_SUB_STREAM_UNLOCK(self);
gst_object_unref (sysclock);
if (emit)
g_signal_emit (self, signals[NO_RTCP_TIMEDOUT], 0);
return NULL;
no_sysclock:
{
fs_rtp_sub_stream_emit_error (self, FS_ERROR_INTERNAL,
"Could not get system clock",
"Could not get system clock");
return NULL;
}
}
/* 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;
}
static void
gst_clock_finalize (GObject * object)
{
GstClock *clock = GST_CLOCK (object);
GST_CLOCK_SLAVE_LOCK (clock);
if (clock->priv->clockid) {
gst_clock_id_unschedule (clock->priv->clockid);
gst_clock_id_unref (clock->priv->clockid);
clock->priv->clockid = NULL;
}
g_free (clock->priv->times);
clock->priv->times = NULL;
GST_CLOCK_SLAVE_UNLOCK (clock);
g_mutex_clear (&clock->priv->slave_lock);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static GstStateChangeReturn
gst_identity_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret;
GstIdentity *identity = GST_IDENTITY (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
GST_OBJECT_LOCK (identity);
if (identity->clock_id) {
GST_DEBUG_OBJECT (identity, "unlock clock wait");
gst_clock_id_unschedule (identity->clock_id);
gst_clock_id_unref (identity->clock_id);
identity->clock_id = NULL;
}
GST_OBJECT_UNLOCK (identity);
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_NULL:
break;
default:
break;
}
return ret;
}
static GstFlowReturn
gst_identity_do_sync (GstIdentity * identity, GstClockTime running_time)
{
GstFlowReturn ret = GST_FLOW_OK;
if (identity->sync &&
GST_BASE_TRANSFORM_CAST (identity)->segment.format == GST_FORMAT_TIME) {
GstClock *clock;
GST_OBJECT_LOCK (identity);
while (identity->blocked)
g_cond_wait (&identity->blocked_cond, GST_OBJECT_GET_LOCK (identity));
if ((clock = GST_ELEMENT (identity)->clock)) {
GstClockReturn cret;
GstClockTime timestamp;
timestamp = running_time + GST_ELEMENT (identity)->base_time +
identity->upstream_latency;
/* save id if we need to unlock */
identity->clock_id = gst_clock_new_single_shot_id (clock, timestamp);
GST_OBJECT_UNLOCK (identity);
cret = gst_clock_id_wait (identity->clock_id, NULL);
GST_OBJECT_LOCK (identity);
if (identity->clock_id) {
gst_clock_id_unref (identity->clock_id);
identity->clock_id = NULL;
}
if (cret == GST_CLOCK_UNSCHEDULED)
ret = GST_FLOW_EOS;
}
GST_OBJECT_UNLOCK (identity);
}
return ret;
}
void gst_imx_v4l2src_set_autofocus(GstPhotography *photo, gboolean on)
{
GstImxV4l2VideoSrc *v4l2src = GST_IMX_V4L2SRC(photo);
int locks;
g_mutex_lock(&v4l2src->af_mutex);
if (v4l2src->af_clock_id)
{
gst_clock_id_unschedule(v4l2src->af_clock_id);
gst_clock_id_unref(v4l2src->af_clock_id);
v4l2src->af_clock_id = NULL;
}
if (v4l2src->focus_mode == GST_PHOTOGRAPHY_FOCUS_MODE_CONTINUOUS_NORMAL)
{
if (v4l2_g_ctrl(v4l2src, V4L2_CID_3A_LOCK, &locks) == 0)
{
if (on && !(locks & V4L2_LOCK_FOCUS))
v4l2_s_ctrl(v4l2src, V4L2_CID_3A_LOCK, locks | V4L2_LOCK_FOCUS);
else if (!on && (locks & V4L2_LOCK_FOCUS))
v4l2_s_ctrl(v4l2src, V4L2_CID_3A_LOCK, locks & ~V4L2_LOCK_FOCUS);
}
}
else
{
if (on)
{
if (v4l2_s_ctrl(v4l2src, V4L2_CID_AUTO_FOCUS_START, 0) == 0)
gst_imx_v4l2src_af_check_status(v4l2src);
}
else
v4l2_s_ctrl(v4l2src, V4L2_CID_AUTO_FOCUS_STOP, 0);
}
g_mutex_unlock(&v4l2src->af_mutex);
}
gboolean
clear_sender (gpointer key, gpointer value, gpointer user_data)
{
FsRtpTfrc *self = FS_RTP_TFRC (user_data);
struct TrackedSource *src = value;
src->send_ts_base = 0;
src->send_ts_cycles = 0;
src->fb_last_ts = 0;
src->fb_ts_cycles = 0;
if (src->sender_id)
{
gst_clock_id_unschedule (src->sender_id);
gst_clock_id_unref (src->sender_id);
src->sender_id = 0;
}
if (src->sender)
tfrc_sender_free (src->sender);
src->sender = NULL;
if (src->idl)
{
tfrc_is_data_limited_free (src->idl);
src->idl = NULL;
}
if (self->last_src == src)
self->last_src = NULL;
if (src->receiver)
return FALSE;
else
return TRUE;
}
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;
}
gint
main (gint argc, gchar ** argv)
{
gint res = 1;
GstElement *src, *sink;
GstElement *bin;
GstController *ctrl;
GstInterpolationControlSource *csource1, *csource2;
GstClock *clock;
GstClockID clock_id;
GstClockReturn wait_ret;
GValue vol = { 0, };
gst_init (&argc, &argv);
gst_controller_init (&argc, &argv);
/* build pipeline */
bin = gst_pipeline_new ("pipeline");
clock = gst_pipeline_get_clock (GST_PIPELINE (bin));
src = gst_element_factory_make ("audiotestsrc", "gen_audio");
if (!src) {
GST_WARNING ("need audiotestsrc from gst-plugins-base");
goto Error;
}
sink = gst_element_factory_make ("autoaudiosink", "play_audio");
if (!sink) {
GST_WARNING ("need autoaudiosink from gst-plugins-base");
goto Error;
}
gst_bin_add_many (GST_BIN (bin), src, sink, NULL);
if (!gst_element_link (src, sink)) {
GST_WARNING ("can't link elements");
goto Error;
}
/* square wave
g_object_set (G_OBJECT(src), "wave", 1, NULL);
*/
/* add a controller to the source */
if (!(ctrl = gst_controller_new (G_OBJECT (src), "freq", "volume", NULL))) {
GST_WARNING ("can't control source element");
goto Error;
}
csource1 = gst_interpolation_control_source_new ();
csource2 = gst_interpolation_control_source_new ();
gst_controller_set_control_source (ctrl, "volume",
GST_CONTROL_SOURCE (csource1));
gst_controller_set_control_source (ctrl, "freq",
GST_CONTROL_SOURCE (csource2));
/* Set interpolation mode */
gst_interpolation_control_source_set_interpolation_mode (csource1,
GST_INTERPOLATE_LINEAR);
gst_interpolation_control_source_set_interpolation_mode (csource2,
GST_INTERPOLATE_LINEAR);
/* set control values */
g_value_init (&vol, G_TYPE_DOUBLE);
g_value_set_double (&vol, 0.0);
gst_interpolation_control_source_set (csource1, 0 * GST_SECOND, &vol);
g_value_set_double (&vol, 1.0);
gst_interpolation_control_source_set (csource1, 5 * GST_SECOND, &vol);
g_object_unref (csource1);
g_value_set_double (&vol, 220.0);
gst_interpolation_control_source_set (csource2, 0 * GST_SECOND, &vol);
g_value_set_double (&vol, 3520.0);
gst_interpolation_control_source_set (csource2, 3 * GST_SECOND, &vol);
g_value_set_double (&vol, 440.0);
gst_interpolation_control_source_set (csource2, 6 * GST_SECOND, &vol);
g_object_unref (csource2);
clock_id =
gst_clock_new_single_shot_id (clock,
gst_clock_get_time (clock) + (7 * GST_SECOND));
/* run for 7 seconds */
if (gst_element_set_state (bin, GST_STATE_PLAYING)) {
if ((wait_ret = gst_clock_id_wait (clock_id, NULL)) != GST_CLOCK_OK) {
GST_WARNING ("clock_id_wait returned: %d", wait_ret);
}
gst_element_set_state (bin, GST_STATE_NULL);
}
/* cleanup */
g_object_unref (G_OBJECT (ctrl));
gst_clock_id_unref (clock_id);
gst_object_unref (G_OBJECT (clock));
gst_object_unref (G_OBJECT (bin));
res = 0;
Error:
return (res);
}
// get spectrum messages and delay them
static gboolean on_message(GstBus *bus, GstMessage *message, gpointer data)
{
base *base_object = data;
GstElement *spectrum = GST_ELEMENT(base_object->spectrum_element->obj);
gst_object_ref(spectrum);
GstElement *message_element = GST_ELEMENT(GST_MESSAGE_SRC(message));
gst_object_ref(message_element);
if (message_element == spectrum)
{
GstClockTime waittime = GST_CLOCK_TIME_NONE;
const GstStructure *message_structure = gst_message_get_structure(message);
// determine waittime
GstClockTime timestamp, duration;
if (
gst_structure_get_clock_time(message_structure, "running-time", ×tamp)
&& gst_structure_get_clock_time(message_structure, "duration", &duration)
)
{
/* wait for middle of buffer */
waittime = timestamp + duration/2;
}
else if (gst_structure_get_clock_time(message_structure, "endtime", ×tamp))
{
waittime = timestamp;
}
// delay message
if (GST_CLOCK_TIME_IS_VALID(waittime))
{
GstClockTime basetime = gst_element_get_base_time(spectrum);
GstClockID clock_id = gst_clock_new_single_shot_id(base_object->sync_clock, basetime+waittime);
spectrum_message *mess = g_malloc(sizeof(spectrum_message));
// set bands and threshold
g_object_get(message_element, "bands", &(mess->bands), "threshold", &(mess->threshold), NULL);
// set start and duration
GstClockTime streamtime, duration;
gst_structure_get_clock_time(message_structure, "stream-time", &streamtime);
gst_structure_get_clock_time(message_structure, "duration", &duration);
mess->start = (gfloat)streamtime / GST_SECOND;
mess->duration = (gfloat)duration / GST_SECOND;
// set rate
GstPad *sink = gst_element_get_static_pad(GST_ELEMENT(base_object->spectrum_element->obj), "sink");
GstCaps *caps = gst_pad_get_negotiated_caps(sink);
gst_object_unref(sink);
GstStructure *caps_structure = gst_caps_get_structure(caps, 0);
gst_structure_get_int(caps_structure, "rate", &(mess->rate));
gst_caps_unref(caps);
// set magnitudes
const GValue *list = gst_structure_get_value(message_structure, "magnitude");
PyGILState_STATE gstate = PyGILState_Ensure();
int i;
mess->magnitudes = PyList_New(mess->bands);
for (i=0; i < (mess->bands); i++)
{
const GValue *value = gst_value_list_get_value(list, i);
gfloat f = g_value_get_float(value);
PyList_SetItem(mess->magnitudes, i, Py_BuildValue("f", f));
}
PyGILState_Release(gstate);
// set gobj
GObject *gobj = (base_object->gobj).obj;
g_assert(gobj != NULL);
g_object_ref(gobj);
mess->gobj = gobj;
// delay message
gst_clock_id_wait_async(clock_id, delayed_spectrum_update, mess);
gst_clock_id_unref(clock_id);
}
}
gst_object_unref(spectrum);
gst_object_unref(message_element);
return TRUE;
}
/**
* This funcion will push out buffers on the source pad.
*
* For each pushed buffer, the seqnum is recorded, if the next buffer B has a
* different seqnum (missing packets before B), this function will wait for the
* missing packet to arrive up to the timestamp of buffer B.
*/
static void
gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer)
{
GstRtpJitterBufferPrivate *priv;
GstBuffer *outbuf;
GstFlowReturn result;
guint16 seqnum;
guint32 next_seqnum;
GstClockTime timestamp, out_time;
gboolean discont = FALSE;
gint gap;
priv = jitterbuffer->priv;
JBUF_LOCK_CHECK (priv, flushing);
again:
GST_DEBUG_OBJECT (jitterbuffer, "Peeking item");
while (TRUE) {
/* always wait if we are blocked */
if (!priv->blocked) {
/* if we have a packet, we can exit the loop and grab it */
if (rtp_jitter_buffer_num_packets (priv->jbuf) > 0)
break;
/* no packets but we are EOS, do eos logic */
if (priv->eos)
goto do_eos;
}
/* underrun, wait for packets or flushing now */
priv->waiting = TRUE;
JBUF_WAIT_CHECK (priv, flushing);
priv->waiting = FALSE;
}
/* peek a buffer, we're just looking at the timestamp and the sequence number.
* If all is fine, we'll pop and push it. If the sequence number is wrong we
* wait on the timestamp. In the chain function we will unlock the wait when a
* new buffer is available. The peeked buffer is valid for as long as we hold
* the jitterbuffer lock. */
outbuf = rtp_jitter_buffer_peek (priv->jbuf);
/* get the seqnum and the next expected seqnum */
seqnum = gst_rtp_buffer_get_seq (outbuf);
next_seqnum = priv->next_seqnum;
/* get the timestamp, this is already corrected for clock skew by the
* jitterbuffer */
timestamp = GST_BUFFER_TIMESTAMP (outbuf);
GST_DEBUG_OBJECT (jitterbuffer,
"Peeked buffer #%d, expect #%d, timestamp %" GST_TIME_FORMAT
", now %d left", seqnum, next_seqnum, GST_TIME_ARGS (timestamp),
rtp_jitter_buffer_num_packets (priv->jbuf));
/* apply our timestamp offset to the incomming buffer, this will be our output
* timestamp. */
out_time = apply_offset (jitterbuffer, timestamp);
/* get the gap between this and the previous packet. If we don't know the
* previous packet seqnum assume no gap. */
if (next_seqnum != -1) {
gap = gst_rtp_buffer_compare_seqnum (next_seqnum, seqnum);
/* if we have a packet that we already pushed or considered dropped, pop it
* off and get the next packet */
if (gap < 0) {
GST_DEBUG_OBJECT (jitterbuffer, "Old packet #%d, next #%d dropping",
seqnum, next_seqnum);
outbuf = rtp_jitter_buffer_pop (priv->jbuf);
gst_buffer_unref (outbuf);
goto again;
}
} else {
GST_DEBUG_OBJECT (jitterbuffer, "no next seqnum known, first packet");
gap = -1;
}
/* If we don't know what the next seqnum should be (== -1) we have to wait
* because it might be possible that we are not receiving this buffer in-order,
* a buffer with a lower seqnum could arrive later and we want to push that
* earlier buffer before this buffer then.
* If we know the expected seqnum, we can compare it to the current seqnum to
* determine if we have missing a packet. If we have a missing packet (which
* must be before this packet) we can wait for it until the deadline for this
* packet expires. */
if (gap != 0 && out_time != -1) {
GstClockID id;
GstClockTime sync_time;
GstClockReturn ret;
GstClock *clock;
GstClockTime duration = GST_CLOCK_TIME_NONE;
if (gap > 0) {
/* we have a gap */
GST_WARNING_OBJECT (jitterbuffer,
"Sequence number GAP detected: expected %d instead of %d (%d missing)",
next_seqnum, seqnum, gap);
if (priv->last_out_time != -1) {
GST_DEBUG_OBJECT (jitterbuffer,
"out_time %" GST_TIME_FORMAT ", last %" GST_TIME_FORMAT,
GST_TIME_ARGS (out_time), GST_TIME_ARGS (priv->last_out_time));
/* interpolate between the current time and the last time based on
* number of packets we are missing, this is the estimated duration
* for the missing packet based on equidistant packet spacing. Also make
* sure we never go negative. */
if (out_time > priv->last_out_time)
duration = (out_time - priv->last_out_time) / (gap + 1);
else
goto lost;
GST_DEBUG_OBJECT (jitterbuffer, "duration %" GST_TIME_FORMAT,
GST_TIME_ARGS (duration));
/* add this duration to the timestamp of the last packet we pushed */
out_time = (priv->last_out_time + duration);
}
} else {
/* we don't know what the next_seqnum should be, wait for the last
* possible moment to push this buffer, maybe we get an earlier seqnum
* while we wait */
GST_DEBUG_OBJECT (jitterbuffer, "First buffer %d, do sync", seqnum);
}
GST_OBJECT_LOCK (jitterbuffer);
clock = GST_ELEMENT_CLOCK (jitterbuffer);
if (!clock) {
GST_OBJECT_UNLOCK (jitterbuffer);
/* let's just push if there is no clock */
goto push_buffer;
}
GST_DEBUG_OBJECT (jitterbuffer, "sync to timestamp %" GST_TIME_FORMAT,
GST_TIME_ARGS (out_time));
/* prepare for sync against clock */
sync_time = out_time + GST_ELEMENT_CAST (jitterbuffer)->base_time;
/* add latency, this includes our own latency and the peer latency. */
sync_time += (priv->latency_ms * GST_MSECOND);
sync_time += priv->peer_latency;
/* create an entry for the clock */
id = priv->clock_id = gst_clock_new_single_shot_id (clock, sync_time);
GST_OBJECT_UNLOCK (jitterbuffer);
/* release the lock so that the other end can push stuff or unlock */
JBUF_UNLOCK (priv);
ret = gst_clock_id_wait (id, NULL);
JBUF_LOCK (priv);
/* and free the entry */
gst_clock_id_unref (id);
priv->clock_id = NULL;
/* at this point, the clock could have been unlocked by a timeout, a new
* tail element was added to the queue or because we are shutting down. Check
* for shutdown first. */
if (priv->srcresult != GST_FLOW_OK)
goto flushing;
/* if we got unscheduled and we are not flushing, it's because a new tail
* element became available in the queue. Grab it and try to push or sync. */
if (ret == GST_CLOCK_UNSCHEDULED) {
GST_DEBUG_OBJECT (jitterbuffer,
"Wait got unscheduled, will retry to push with new buffer");
goto again;
}
lost:
/* we now timed out, this means we lost a packet or finished synchronizing
* on the first buffer. */
if (gap > 0) {
GstEvent *event;
/* we had a gap and thus we lost a packet. Create an event for this. */
GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d lost", next_seqnum);
priv->num_late++;
discont = TRUE;
if (priv->do_lost) {
/* create paket lost event */
event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM,
gst_structure_new ("GstRTPPacketLost",
"seqnum", G_TYPE_UINT, (guint) next_seqnum,
"timestamp", G_TYPE_UINT64, out_time,
"duration", G_TYPE_UINT64, duration, NULL));
gst_pad_push_event (priv->srcpad, event);
}
/* update our expected next packet */
priv->last_popped_seqnum = next_seqnum;
priv->last_out_time = out_time;
priv->next_seqnum = (next_seqnum + 1) & 0xffff;
/* look for next packet */
goto again;
}
/* there was no known gap,just the first packet, exit the loop and push */
GST_DEBUG_OBJECT (jitterbuffer, "First packet #%d synced", seqnum);
/* get new timestamp, latency might have changed */
out_time = apply_offset (jitterbuffer, timestamp);
}
push_buffer:
/* when we get here we are ready to pop and push the buffer */
outbuf = rtp_jitter_buffer_pop (priv->jbuf);
if (discont || priv->discont) {
/* set DISCONT flag when we missed a packet. */
outbuf = gst_buffer_make_metadata_writable (outbuf);
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
priv->discont = FALSE;
}
/* apply timestamp with offset to buffer now */
GST_BUFFER_TIMESTAMP (outbuf) = out_time;
/* now we are ready to push the buffer. Save the seqnum and release the lock
* so the other end can push stuff in the queue again. */
priv->last_popped_seqnum = seqnum;
priv->last_out_time = out_time;
priv->next_seqnum = (seqnum + 1) & 0xffff;
JBUF_UNLOCK (priv);
/* push buffer */
GST_DEBUG_OBJECT (jitterbuffer,
"Pushing buffer %d, timestamp %" GST_TIME_FORMAT, seqnum,
GST_TIME_ARGS (out_time));
result = gst_pad_push (priv->srcpad, outbuf);
if (result != GST_FLOW_OK)
goto pause;
return;
/* ERRORS */
do_eos:
{
/* store result, we are flushing now */
GST_DEBUG_OBJECT (jitterbuffer, "We are EOS, pushing EOS downstream");
priv->srcresult = GST_FLOW_UNEXPECTED;
gst_pad_pause_task (priv->srcpad);
gst_pad_push_event (priv->srcpad, gst_event_new_eos ());
JBUF_UNLOCK (priv);
return;
}
flushing:
{
GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
gst_pad_pause_task (priv->srcpad);
JBUF_UNLOCK (priv);
return;
}
pause:
{
const gchar *reason = gst_flow_get_name (result);
GST_DEBUG_OBJECT (jitterbuffer, "pausing task, reason %s", reason);
JBUF_LOCK (priv);
/* store result */
priv->srcresult = result;
/* we don't post errors or anything because upstream will do that for us
* when we pass the return value upstream. */
gst_pad_pause_task (priv->srcpad);
JBUF_UNLOCK (priv);
return;
}
}
static GstFlowReturn
gst_dx9screencapsrc_create (GstPushSrc * push_src, GstBuffer ** buf)
{
GstDX9ScreenCapSrc *src = GST_DX9SCREENCAPSRC (push_src);
GstBuffer *new_buf;
gint new_buf_size, i;
gint width, height, stride;
GstClock *clock;
GstClockTime buf_time, buf_dur;
D3DLOCKED_RECT locked_rect;
LPBYTE p_dst, p_src;
HRESULT hres;
GstMapInfo map;
guint64 frame_number;
if (G_UNLIKELY (!src->d3d9_device)) {
GST_ELEMENT_ERROR (src, CORE, NEGOTIATION, (NULL),
("format wasn't negotiated before create function"));
return GST_FLOW_NOT_NEGOTIATED;
}
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;
height = (src->src_rect.bottom - src->src_rect.top);
width = (src->src_rect.right - src->src_rect.left);
new_buf_size = width * 4 * height;
GST_LOG_OBJECT (src,
"creating buffer of %d bytes with %dx%d image",
new_buf_size, width, height);
/* Do screen capture and put it into buffer...
* Aquire front buffer, and lock it
*/
hres =
IDirect3DDevice9_GetFrontBufferData (src->d3d9_device, 0, src->surface);
if (FAILED (hres)) {
GST_DEBUG_OBJECT (src, "DirectX::GetBackBuffer failed.");
return GST_FLOW_ERROR;
}
if (src->show_cursor) {
CURSORINFO ci;
ci.cbSize = sizeof (CURSORINFO);
GetCursorInfo (&ci);
if (ci.flags & CURSOR_SHOWING) {
ICONINFO ii;
HDC memDC;
GetIconInfo (ci.hCursor, &ii);
if (SUCCEEDED (IDirect3DSurface9_GetDC (src->surface, &memDC))) {
HCURSOR cursor = CopyImage (ci.hCursor, IMAGE_CURSOR, 0, 0,
LR_MONOCHROME | LR_DEFAULTSIZE);
DrawIcon (memDC,
ci.ptScreenPos.x - ii.xHotspot - src->monitor_info.rcMonitor.left,
ci.ptScreenPos.y - ii.yHotspot - src->monitor_info.rcMonitor.top,
cursor);
DestroyCursor (cursor);
IDirect3DSurface9_ReleaseDC (src->surface, memDC);
}
DeleteObject (ii.hbmColor);
DeleteObject (ii.hbmMask);
}
}
hres =
IDirect3DSurface9_LockRect (src->surface, &locked_rect, &(src->src_rect),
D3DLOCK_NO_DIRTY_UPDATE | D3DLOCK_NOSYSLOCK | D3DLOCK_READONLY);
if (FAILED (hres)) {
GST_DEBUG_OBJECT (src, "DirectX::LockRect failed.");
return GST_FLOW_ERROR;
}
new_buf = gst_buffer_new_and_alloc (new_buf_size);
gst_buffer_map (new_buf, &map, GST_MAP_WRITE);
p_dst = (LPBYTE) map.data;
p_src = (LPBYTE) locked_rect.pBits;
stride = width * 4;
for (i = 0; i < height; ++i) {
memcpy (p_dst, p_src, stride);
p_dst += stride;
p_src += locked_rect.Pitch;
}
gst_buffer_unmap (new_buf, &map);
/* Unlock copy of front buffer */
IDirect3DSurface9_UnlockRect (src->surface);
GST_BUFFER_TIMESTAMP (new_buf) = buf_time;
GST_BUFFER_DURATION (new_buf) = buf_dur;
if (clock != NULL)
gst_object_unref (clock);
*buf = new_buf;
return GST_FLOW_OK;
}
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;
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");
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;
gst_rtp_dtmf_prepare_timestamps (dtmfsrc);
/* Don't forget to get exclusive access to the stream */
gst_rtp_dtmf_src_set_stream_lock (dtmfsrc, TRUE);
dtmfsrc->payload = event->payload;
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");
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;
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));
#ifdef MAEMO_BROKEN
clockid = gst_clock_new_single_shot_id (clock, dtmfsrc->timestamp);
#else
clockid = gst_clock_new_single_shot_id (clock, dtmfsrc->timestamp +
gst_element_get_base_time (GST_ELEMENT (dtmfsrc)));
#endif
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;
}
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) {
/* Don't forget to release the stream lock */
gst_rtp_dtmf_src_set_stream_lock (dtmfsrc, FALSE);
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_WRONG_STATE;
}
}
static GstFlowReturn
_chain (GstPad * pad, GstObject * object, GstBuffer * buffer)
{
GstBuffer *actual_buf = buffer;
GstAggregator *self = GST_AGGREGATOR (object);
GstAggregatorPrivate *priv = self->priv;
GstAggregatorPad *aggpad = GST_AGGREGATOR_PAD (pad);
GstAggregatorClass *aggclass = GST_AGGREGATOR_GET_CLASS (object);
GstClockTime timeout = gst_aggregator_get_timeout (self);
GstClockTime now;
GST_DEBUG_OBJECT (aggpad, "Start chaining a buffer %" GST_PTR_FORMAT, buffer);
if (aggpad->priv->timeout_id) {
gst_clock_id_unschedule (aggpad->priv->timeout_id);
gst_clock_id_unref (aggpad->priv->timeout_id);
aggpad->priv->timeout_id = NULL;
}
g_atomic_int_set (&aggpad->unresponsive, FALSE);
PAD_STREAM_LOCK (aggpad);
if (g_atomic_int_get (&aggpad->priv->flushing) == TRUE)
goto flushing;
if (g_atomic_int_get (&aggpad->priv->pending_eos) == TRUE)
goto eos;
PAD_LOCK_EVENT (aggpad);
if (aggpad->buffer) {
GST_DEBUG_OBJECT (aggpad, "Waiting for buffer to be consumed");
PAD_WAIT_EVENT (aggpad);
}
PAD_UNLOCK_EVENT (aggpad);
if (g_atomic_int_get (&aggpad->priv->flushing) == TRUE)
goto flushing;
if (aggclass->clip) {
aggclass->clip (self, aggpad, buffer, &actual_buf);
}
PAD_LOCK_EVENT (aggpad);
if (aggpad->buffer)
gst_buffer_unref (aggpad->buffer);
aggpad->buffer = actual_buf;
PAD_UNLOCK_EVENT (aggpad);
PAD_STREAM_UNLOCK (aggpad);
QUEUE_PUSH (self);
if (GST_CLOCK_TIME_IS_VALID (timeout)) {
now = gst_clock_get_time (self->clock);
aggpad->priv->timeout_id =
gst_clock_new_single_shot_id (self->clock, now + timeout);
gst_clock_id_wait_async (aggpad->priv->timeout_id, _unresponsive_timeout,
gst_object_ref (aggpad), gst_object_unref);
}
GST_DEBUG_OBJECT (aggpad, "Done chaining");
return priv->flow_return;
flushing:
PAD_STREAM_UNLOCK (aggpad);
gst_buffer_unref (buffer);
GST_DEBUG_OBJECT (aggpad, "We are flushing");
return GST_FLOW_FLUSHING;
eos:
PAD_STREAM_UNLOCK (aggpad);
gst_buffer_unref (buffer);
GST_DEBUG_OBJECT (pad, "We are EOS already...");
return GST_FLOW_EOS;
}
static gboolean
gst_identity_sink_event (GstBaseTransform * trans, GstEvent * event)
{
GstIdentity *identity;
gboolean ret = TRUE;
identity = GST_IDENTITY (trans);
if (!identity->silent) {
const GstStructure *s;
const gchar *tstr;
gchar *sstr;
GST_OBJECT_LOCK (identity);
g_free (identity->last_message);
tstr = gst_event_type_get_name (GST_EVENT_TYPE (event));
if ((s = gst_event_get_structure (event)))
sstr = gst_structure_to_string (s);
else
sstr = g_strdup ("");
identity->last_message =
g_strdup_printf ("event ******* (%s:%s) E (type: %s (%d), %s) %p",
GST_DEBUG_PAD_NAME (trans->sinkpad), tstr, GST_EVENT_TYPE (event),
sstr, event);
g_free (sstr);
GST_OBJECT_UNLOCK (identity);
gst_identity_notify_last_message (identity);
}
if (identity->single_segment && (GST_EVENT_TYPE (event) == GST_EVENT_SEGMENT)) {
if (trans->have_segment == FALSE) {
GstEvent *news;
GstSegment segment;
gst_event_copy_segment (event, &segment);
gst_event_copy_segment (event, &trans->segment);
trans->have_segment = TRUE;
/* This is the first segment, send out a (0, -1) segment */
gst_segment_init (&segment, segment.format);
news = gst_event_new_segment (&segment);
gst_pad_event_default (trans->sinkpad, GST_OBJECT_CAST (trans), news);
}
}
/* Reset previous timestamp, duration and offsets on NEWSEGMENT
* to prevent false warnings when checking for perfect streams */
if (GST_EVENT_TYPE (event) == GST_EVENT_SEGMENT) {
identity->prev_timestamp = identity->prev_duration = GST_CLOCK_TIME_NONE;
identity->prev_offset = identity->prev_offset_end = GST_BUFFER_OFFSET_NONE;
}
if (identity->single_segment && GST_EVENT_TYPE (event) == GST_EVENT_SEGMENT) {
/* eat up segments */
gst_event_unref (event);
ret = TRUE;
} else {
if (GST_EVENT_TYPE (event) == GST_EVENT_FLUSH_START) {
GST_OBJECT_LOCK (identity);
if (identity->clock_id) {
GST_DEBUG_OBJECT (identity, "unlock clock wait");
gst_clock_id_unschedule (identity->clock_id);
gst_clock_id_unref (identity->clock_id);
identity->clock_id = NULL;
}
GST_OBJECT_UNLOCK (identity);
}
ret = GST_BASE_TRANSFORM_CLASS (parent_class)->sink_event (trans, event);
}
return ret;
}
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
gst_identity_transform_ip (GstBaseTransform * trans, GstBuffer * buf)
{
GstFlowReturn ret = GST_FLOW_OK;
GstIdentity *identity = GST_IDENTITY (trans);
GstClockTime runtimestamp = G_GINT64_CONSTANT (0);
gsize size;
size = gst_buffer_get_size (buf);
if (identity->check_imperfect_timestamp)
gst_identity_check_imperfect_timestamp (identity, buf);
if (identity->check_imperfect_offset)
gst_identity_check_imperfect_offset (identity, buf);
/* update prev values */
identity->prev_timestamp = GST_BUFFER_TIMESTAMP (buf);
identity->prev_duration = GST_BUFFER_DURATION (buf);
identity->prev_offset_end = GST_BUFFER_OFFSET_END (buf);
identity->prev_offset = GST_BUFFER_OFFSET (buf);
if (identity->error_after >= 0) {
identity->error_after--;
if (identity->error_after == 0)
goto error_after;
}
if (identity->drop_probability > 0.0) {
if ((gfloat) (1.0 * rand () / (RAND_MAX)) < identity->drop_probability)
goto dropped;
}
if (identity->dump) {
GstMapInfo info;
gst_buffer_map (buf, &info, GST_MAP_READ);
gst_util_dump_mem (info.data, info.size);
gst_buffer_unmap (buf, &info);
}
if (!identity->silent) {
gst_identity_update_last_message_for_buffer (identity, "chain", buf, size);
}
if (identity->datarate > 0) {
GstClockTime time = gst_util_uint64_scale_int (identity->offset,
GST_SECOND, identity->datarate);
GST_BUFFER_TIMESTAMP (buf) = time;
GST_BUFFER_DURATION (buf) = size * GST_SECOND / identity->datarate;
}
if (identity->signal_handoffs)
g_signal_emit (identity, gst_identity_signals[SIGNAL_HANDOFF], 0, buf);
if (trans->segment.format == GST_FORMAT_TIME)
runtimestamp = gst_segment_to_running_time (&trans->segment,
GST_FORMAT_TIME, GST_BUFFER_TIMESTAMP (buf));
if ((identity->sync) && (trans->segment.format == GST_FORMAT_TIME)) {
GstClock *clock;
GST_OBJECT_LOCK (identity);
if ((clock = GST_ELEMENT (identity)->clock)) {
GstClockReturn cret;
GstClockTime timestamp;
timestamp = runtimestamp + GST_ELEMENT (identity)->base_time;
/* save id if we need to unlock */
identity->clock_id = gst_clock_new_single_shot_id (clock, timestamp);
GST_OBJECT_UNLOCK (identity);
cret = gst_clock_id_wait (identity->clock_id, NULL);
GST_OBJECT_LOCK (identity);
if (identity->clock_id) {
gst_clock_id_unref (identity->clock_id);
identity->clock_id = NULL;
}
if (cret == GST_CLOCK_UNSCHEDULED)
ret = GST_FLOW_EOS;
}
GST_OBJECT_UNLOCK (identity);
}
identity->offset += size;
if (identity->sleep_time && ret == GST_FLOW_OK)
g_usleep (identity->sleep_time);
if (identity->single_segment && (trans->segment.format == GST_FORMAT_TIME)
&& (ret == GST_FLOW_OK)) {
GST_BUFFER_TIMESTAMP (buf) = runtimestamp;
GST_BUFFER_OFFSET (buf) = GST_CLOCK_TIME_NONE;
GST_BUFFER_OFFSET_END (buf) = GST_CLOCK_TIME_NONE;
}
return ret;
/* ERRORS */
error_after:
{
GST_ELEMENT_ERROR (identity, CORE, FAILED,
(_("Failed after iterations as requested.")), (NULL));
return GST_FLOW_ERROR;
}
dropped:
{
if (!identity->silent) {
gst_identity_update_last_message_for_buffer (identity, "dropping", buf,
size);
}
/* return DROPPED to basetransform. */
return GST_BASE_TRANSFORM_FLOW_DROPPED;
}
}
static GstFlowReturn
gst_identity_transform_ip (GstBaseTransform * trans, GstBuffer * buf)
{
GstFlowReturn ret = GST_FLOW_OK;
GstIdentity *identity = GST_IDENTITY (trans);
GstClockTime runtimestamp = G_GINT64_CONSTANT (0);
if (identity->check_perfect)
gst_identity_check_perfect (identity, buf);
if (identity->check_imperfect_timestamp)
gst_identity_check_imperfect_timestamp (identity, buf);
if (identity->check_imperfect_offset)
gst_identity_check_imperfect_offset (identity, buf);
/* update prev values */
identity->prev_timestamp = GST_BUFFER_TIMESTAMP (buf);
identity->prev_duration = GST_BUFFER_DURATION (buf);
identity->prev_offset_end = GST_BUFFER_OFFSET_END (buf);
identity->prev_offset = GST_BUFFER_OFFSET (buf);
if (identity->error_after >= 0) {
identity->error_after--;
if (identity->error_after == 0) {
GST_ELEMENT_ERROR (identity, CORE, FAILED,
(_("Failed after iterations as requested.")), (NULL));
return GST_FLOW_ERROR;
}
}
if (identity->drop_probability > 0.0) {
if ((gfloat) (1.0 * rand () / (RAND_MAX)) < identity->drop_probability) {
if (!identity->silent) {
GST_OBJECT_LOCK (identity);
g_free (identity->last_message);
identity->last_message =
g_strdup_printf
("dropping ******* (%s:%s)i (%d bytes, timestamp: %"
GST_TIME_FORMAT ", duration: %" GST_TIME_FORMAT ", offset: %"
G_GINT64_FORMAT ", offset_end: % " G_GINT64_FORMAT
", flags: %d) %p", GST_DEBUG_PAD_NAME (trans->sinkpad),
GST_BUFFER_SIZE (buf), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)), GST_BUFFER_OFFSET (buf),
GST_BUFFER_OFFSET_END (buf), GST_BUFFER_FLAGS (buf), buf);
GST_OBJECT_UNLOCK (identity);
g_object_notify (G_OBJECT (identity), "last-message");
}
/* return DROPPED to basetransform. */
return GST_BASE_TRANSFORM_FLOW_DROPPED;
}
}
if (identity->dump) {
gst_util_dump_mem (GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
}
if (!identity->silent) {
GST_OBJECT_LOCK (identity);
g_free (identity->last_message);
identity->last_message =
g_strdup_printf ("chain ******* (%s:%s)i (%d bytes, timestamp: %"
GST_TIME_FORMAT ", duration: %" GST_TIME_FORMAT ", offset: %"
G_GINT64_FORMAT ", offset_end: % " G_GINT64_FORMAT ", flags: %d) %p",
GST_DEBUG_PAD_NAME (trans->sinkpad), GST_BUFFER_SIZE (buf),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)),
GST_BUFFER_OFFSET (buf), GST_BUFFER_OFFSET_END (buf),
GST_BUFFER_FLAGS (buf), buf);
GST_OBJECT_UNLOCK (identity);
g_object_notify (G_OBJECT (identity), "last-message");
}
if (identity->datarate > 0) {
GstClockTime time = gst_util_uint64_scale_int (identity->offset,
GST_SECOND, identity->datarate);
GST_BUFFER_TIMESTAMP (buf) = time;
GST_BUFFER_DURATION (buf) =
GST_BUFFER_SIZE (buf) * GST_SECOND / identity->datarate;
}
if (identity->signal_handoffs)
g_signal_emit (G_OBJECT (identity), gst_identity_signals[SIGNAL_HANDOFF], 0,
buf);
if (trans->segment.format == GST_FORMAT_TIME)
runtimestamp = gst_segment_to_running_time (&trans->segment,
GST_FORMAT_TIME, GST_BUFFER_TIMESTAMP (buf));
if ((identity->sync) && (trans->segment.format == GST_FORMAT_TIME)) {
GstClock *clock;
GST_OBJECT_LOCK (identity);
if ((clock = GST_ELEMENT (identity)->clock)) {
GstClockReturn cret;
GstClockTime timestamp;
timestamp = runtimestamp + GST_ELEMENT (identity)->base_time;
/* save id if we need to unlock */
/* FIXME: actually unlock this somewhere in the state changes */
identity->clock_id = gst_clock_new_single_shot_id (clock, timestamp);
GST_OBJECT_UNLOCK (identity);
cret = gst_clock_id_wait (identity->clock_id, NULL);
GST_OBJECT_LOCK (identity);
if (identity->clock_id) {
gst_clock_id_unref (identity->clock_id);
identity->clock_id = NULL;
}
if (cret == GST_CLOCK_UNSCHEDULED)
ret = GST_FLOW_UNEXPECTED;
}
GST_OBJECT_UNLOCK (identity);
}
identity->offset += GST_BUFFER_SIZE (buf);
if (identity->sleep_time && ret == GST_FLOW_OK)
g_usleep (identity->sleep_time);
if (identity->single_segment && (trans->segment.format == GST_FORMAT_TIME)
&& (ret == GST_FLOW_OK)) {
GST_BUFFER_TIMESTAMP (buf) = runtimestamp;
GST_BUFFER_OFFSET (buf) = GST_CLOCK_TIME_NONE;
GST_BUFFER_OFFSET_END (buf) = GST_CLOCK_TIME_NONE;
}
return ret;
}
/*
return number of readed bytes */
static guint
gst_directsound_src_read (GstAudioSrc * asrc, gpointer data, guint length,
GstClockTime * timestamp)
{
GstDirectSoundSrc *dsoundsrc;
guint64 sleep_time_ms, sleep_until;
GstClockID clock_id;
HRESULT hRes; /* Result for windows functions */
DWORD dwCurrentCaptureCursor = 0;
DWORD dwBufferSize = 0;
LPVOID pLockedBuffer1 = NULL;
LPVOID pLockedBuffer2 = NULL;
DWORD dwSizeBuffer1 = 0;
DWORD dwSizeBuffer2 = 0;
DWORD dwStatus = 0;
GST_DEBUG_OBJECT (asrc, "reading directsoundsrc");
dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
GST_DSOUND_LOCK (dsoundsrc);
/* Get current buffer status */
hRes = IDirectSoundCaptureBuffer_GetStatus (dsoundsrc->pDSBSecondary,
&dwStatus);
if (FAILED (hRes)) {
GST_DSOUND_UNLOCK (dsoundsrc);
return -1;
}
/* Starting capturing if not already */
if (!(dwStatus & DSCBSTATUS_CAPTURING)) {
hRes = IDirectSoundCaptureBuffer_Start (dsoundsrc->pDSBSecondary,
DSCBSTART_LOOPING);
GST_INFO_OBJECT (asrc, "capture started");
}
/* Loop till the source has produced bytes equal to or greater than @length.
*
* DirectSound has a notification-based API that uses Windows CreateEvent()
* + WaitForSingleObject(), but it is completely useless for live streams.
*
* 1. You must schedule all events before starting capture
* 2. The events are all fired exactly once
* 3. You cannot schedule new events while a capture is running
* 4. You cannot stop/schedule/start either
*
* This means you cannot use the API while doing live looped capture and we
* must resort to this.
*
* However, this is almost as efficient as event-based capture since it's ok
* to consistently overwait by a fixed amount; the extra bytes will just end
* up being used in the next call, and the extra latency will be constant. */
while (TRUE) {
hRes =
IDirectSoundCaptureBuffer_GetCurrentPosition (dsoundsrc->pDSBSecondary,
&dwCurrentCaptureCursor, NULL);
if (FAILED (hRes)) {
GST_DSOUND_UNLOCK (dsoundsrc);
return -1;
}
/* calculate the size of the buffer that's been captured while accounting
* for wrap-arounds */
if (dwCurrentCaptureCursor < dsoundsrc->current_circular_offset) {
dwBufferSize = dsoundsrc->buffer_size -
(dsoundsrc->current_circular_offset - dwCurrentCaptureCursor);
} else {
dwBufferSize =
dwCurrentCaptureCursor - dsoundsrc->current_circular_offset;
}
if (dwBufferSize >= length) {
/* Yay, we got all the data we need */
break;
} else {
GST_DEBUG_OBJECT (asrc, "not enough data, got %lu (want at least %u)",
dwBufferSize, length);
/* If we didn't get enough data, sleep for a proportionate time */
sleep_time_ms = gst_util_uint64_scale (dsoundsrc->latency_time,
length - dwBufferSize, length * 1000);
/* Make sure we don't run in a tight loop unnecessarily */
sleep_time_ms = MAX (sleep_time_ms, 10);
/* Sleep using gst_clock_id_wait() so that we can be interrupted */
sleep_until = gst_clock_get_time (dsoundsrc->system_clock) +
sleep_time_ms * GST_MSECOND;
/* Setup the clock id wait */
if (G_UNLIKELY (dsoundsrc->read_wait_clock_id == NULL ||
gst_clock_single_shot_id_reinit (dsoundsrc->system_clock,
dsoundsrc->read_wait_clock_id, sleep_until) == FALSE)) {
if (dsoundsrc->read_wait_clock_id != NULL)
gst_clock_id_unref (dsoundsrc->read_wait_clock_id);
dsoundsrc->read_wait_clock_id =
gst_clock_new_single_shot_id (dsoundsrc->system_clock, sleep_until);
}
clock_id = dsoundsrc->read_wait_clock_id;
dsoundsrc->reset_while_sleeping = FALSE;
GST_DEBUG_OBJECT (asrc, "waiting %" G_GUINT64_FORMAT "ms for more data",
sleep_time_ms);
GST_DSOUND_UNLOCK (dsoundsrc);
gst_clock_id_wait (clock_id, NULL);
GST_DSOUND_LOCK (dsoundsrc);
if (dsoundsrc->reset_while_sleeping == TRUE) {
GST_DEBUG_OBJECT (asrc, "reset while sleeping, cancelled read");
GST_DSOUND_UNLOCK (dsoundsrc);
return -1;
}
}
}
GST_DEBUG_OBJECT (asrc, "Got enough data: %lu bytes (wanted at least %u)",
dwBufferSize, length);
/* Lock the buffer and read only the first @length bytes. Keep the rest in
* the capture buffer for the next read. */
hRes = IDirectSoundCaptureBuffer_Lock (dsoundsrc->pDSBSecondary,
dsoundsrc->current_circular_offset,
length,
&pLockedBuffer1, &dwSizeBuffer1, &pLockedBuffer2, &dwSizeBuffer2, 0L);
/* NOTE: We now assume that dwSizeBuffer1 + dwSizeBuffer2 == length since the
* API is supposed to guarantee that */
/* Copy buffer data to another buffer */
if (hRes == DS_OK) {
memcpy (data, pLockedBuffer1, dwSizeBuffer1);
}
/* ...and if something is in another buffer */
if (pLockedBuffer2 != NULL) {
memcpy (((guchar *) data + dwSizeBuffer1), pLockedBuffer2, dwSizeBuffer2);
}
dsoundsrc->current_circular_offset += dwSizeBuffer1 + dwSizeBuffer2;
dsoundsrc->current_circular_offset %= dsoundsrc->buffer_size;
IDirectSoundCaptureBuffer_Unlock (dsoundsrc->pDSBSecondary,
pLockedBuffer1, dwSizeBuffer1, pLockedBuffer2, dwSizeBuffer2);
GST_DSOUND_UNLOCK (dsoundsrc);
/* We always read exactly @length data */
return length;
}
static GstFlowReturn
gst_wasapi_src_create (GstPushSrc * src, GstBuffer ** buf)
{
GstWasapiSrc *self = GST_WASAPI_SRC (src);
GstFlowReturn ret = GST_FLOW_OK;
GstClock *clock;
GstClockTime timestamp, duration = self->period_time;
HRESULT hr;
gint16 *samples = NULL;
guint32 nsamples_read = 0, nsamples;
DWORD flags = 0;
guint64 devpos;
GST_OBJECT_LOCK (self);
clock = GST_ELEMENT_CLOCK (self);
if (clock != NULL)
gst_object_ref (clock);
GST_OBJECT_UNLOCK (self);
if (clock != NULL && GST_CLOCK_TIME_IS_VALID (self->next_time)) {
GstClockID id;
id = gst_clock_new_single_shot_id (clock, self->next_time);
gst_clock_id_wait (id, NULL);
gst_clock_id_unref (id);
}
do {
hr = IAudioCaptureClient_GetBuffer (self->capture_client,
(BYTE **) & samples, &nsamples_read, &flags, &devpos, NULL);
}
while (hr == AUDCLNT_S_BUFFER_EMPTY);
if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioCaptureClient::GetBuffer () failed: %s",
gst_wasapi_util_hresult_to_string (hr));
ret = GST_FLOW_ERROR;
goto beach;
}
if (flags != 0) {
GST_WARNING_OBJECT (self, "devpos %" G_GUINT64_FORMAT ": flags=0x%08x",
devpos, flags);
}
/* FIXME: Why do we get 1024 sometimes and not a multiple of
* samples_per_buffer? Shouldn't WASAPI provide a DISCONT
* flag if we read too slow?
*/
nsamples = nsamples_read;
g_assert (nsamples >= self->samples_per_buffer);
if (nsamples > self->samples_per_buffer) {
GST_WARNING_OBJECT (self,
"devpos %" G_GUINT64_FORMAT ": got %d samples, expected %d, clipping!",
devpos, nsamples, self->samples_per_buffer);
nsamples = self->samples_per_buffer;
}
if (clock == NULL || clock == self->clock) {
timestamp =
gst_util_uint64_scale (devpos, GST_SECOND, self->client_clock_freq);
} else {
GstClockTime base_time;
timestamp = gst_clock_get_time (clock);
base_time = GST_ELEMENT_CAST (self)->base_time;
if (timestamp > base_time)
timestamp -= base_time;
else
timestamp = 0;
if (timestamp > duration)
timestamp -= duration;
else
timestamp = 0;
}
ret = gst_pad_alloc_buffer_and_set_caps (GST_BASE_SRC_PAD (self),
devpos,
nsamples * sizeof (gint16), GST_PAD_CAPS (GST_BASE_SRC_PAD (self)), buf);
if (ret == GST_FLOW_OK) {
guint i;
gint16 *dst;
GST_BUFFER_OFFSET_END (*buf) = devpos + self->samples_per_buffer;
GST_BUFFER_TIMESTAMP (*buf) = timestamp;
GST_BUFFER_DURATION (*buf) = duration;
dst = (gint16 *) GST_BUFFER_DATA (*buf);
for (i = 0; i < nsamples; i++) {
*dst = *samples;
samples += 2;
dst++;
}
}
hr = IAudioCaptureClient_ReleaseBuffer (self->capture_client, nsamples_read);
if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioCaptureClient::ReleaseBuffer () failed: %s",
gst_wasapi_util_hresult_to_string (hr));
ret = GST_FLOW_ERROR;
goto beach;
}
beach:
if (clock != NULL)
gst_object_unref (clock);
return ret;
}
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 GstStateChangeReturn
gst_identity_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret;
GstIdentity *identity = GST_IDENTITY (element);
gboolean no_preroll = FALSE;
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
GST_OBJECT_LOCK (identity);
identity->blocked = TRUE;
GST_OBJECT_UNLOCK (identity);
if (identity->sync)
no_preroll = TRUE;
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
GST_OBJECT_LOCK (identity);
identity->blocked = FALSE;
g_cond_broadcast (&identity->blocked_cond);
GST_OBJECT_UNLOCK (identity);
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
GST_OBJECT_LOCK (identity);
if (identity->clock_id) {
GST_DEBUG_OBJECT (identity, "unlock clock wait");
gst_clock_id_unschedule (identity->clock_id);
gst_clock_id_unref (identity->clock_id);
identity->clock_id = NULL;
}
identity->blocked = FALSE;
g_cond_broadcast (&identity->blocked_cond);
GST_OBJECT_UNLOCK (identity);
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
GST_OBJECT_LOCK (identity);
identity->upstream_latency = 0;
identity->blocked = TRUE;
GST_OBJECT_UNLOCK (identity);
if (identity->sync)
no_preroll = TRUE;
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_NULL:
break;
default:
break;
}
if (no_preroll && ret == GST_STATE_CHANGE_SUCCESS)
ret = GST_STATE_CHANGE_NO_PREROLL;
return ret;
}
