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 void
gst_rtp_jitter_buffer_flush_start (GstRtpJitterBuffer * jitterbuffer)
{
GstRtpJitterBufferPrivate *priv;
priv = jitterbuffer->priv;
JBUF_LOCK (priv);
/* mark ourselves as flushing */
priv->srcresult = GST_FLOW_WRONG_STATE;
GST_DEBUG_OBJECT (jitterbuffer, "Disabling pop on queue");
/* this unblocks any waiting pops on the src pad task */
JBUF_SIGNAL (priv);
/* unlock clock, we just unschedule, the entry will be released by the
* locking streaming thread. */
if (priv->clock_id)
gst_clock_id_unschedule (priv->clock_id);
JBUF_UNLOCK (priv);
}
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 void
gst_directsound_src_reset (GstAudioSrc * asrc)
{
GstDirectSoundSrc *dsoundsrc;
LPVOID pLockedBuffer = NULL;
DWORD dwSizeBuffer = 0;
GST_DEBUG_OBJECT (asrc, "reset directsoundsrc");
dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
GST_DSOUND_LOCK (dsoundsrc);
dsoundsrc->reset_while_sleeping = TRUE;
/* Interrupt read sleep if required */
if (dsoundsrc->read_wait_clock_id != NULL)
gst_clock_id_unschedule (dsoundsrc->read_wait_clock_id);
if (dsoundsrc->pDSBSecondary) {
/*stop capturing */
HRESULT hRes = IDirectSoundCaptureBuffer_Stop (dsoundsrc->pDSBSecondary);
/*reset position */
/* hRes = IDirectSoundCaptureBuffer_SetCurrentPosition (dsoundsrc->pDSBSecondary, 0); */
/*reset the buffer */
hRes = IDirectSoundCaptureBuffer_Lock (dsoundsrc->pDSBSecondary,
dsoundsrc->current_circular_offset, dsoundsrc->buffer_size,
pLockedBuffer, &dwSizeBuffer, NULL, NULL, 0L);
if (SUCCEEDED (hRes)) {
memset (pLockedBuffer, 0, dwSizeBuffer);
hRes =
IDirectSoundCaptureBuffer_Unlock (dsoundsrc->pDSBSecondary,
pLockedBuffer, dwSizeBuffer, NULL, 0);
}
dsoundsrc->current_circular_offset = 0;
}
GST_DSOUND_UNLOCK (dsoundsrc);
}
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 void
gst_live_adder_flush_start (GstLiveAdder * adder)
{
GST_DEBUG_OBJECT (adder, "Disabling pop on queue");
GST_OBJECT_LOCK (adder);
/* mark ourselves as flushing */
adder->srcresult = GST_FLOW_WRONG_STATE;
/* Empty the queue */
g_queue_foreach (adder->buffers, (GFunc) gst_mini_object_unref, NULL);
while (g_queue_pop_head (adder->buffers));
/* unlock clock, we just unschedule, the entry will be released by the
* locking streaming thread. */
if (adder->clock_id)
gst_clock_id_unschedule (adder->clock_id);
g_cond_broadcast (adder->not_empty_cond);
GST_OBJECT_UNLOCK (adder);
}
static gboolean
gst_rtp_dtmf_src_unlock (GstBaseSrc * src)
{
GstRTPDTMFSrc *dtmfsrc = GST_RTP_DTMF_SRC (src);
GstRTPDTMFSrcEvent *event = NULL;
GST_DEBUG_OBJECT (dtmfsrc, "Called unlock");
GST_OBJECT_LOCK (dtmfsrc);
dtmfsrc->paused = TRUE;
if (dtmfsrc->clockid) {
gst_clock_id_unschedule (dtmfsrc->clockid);
}
GST_OBJECT_UNLOCK (dtmfsrc);
GST_DEBUG_OBJECT (dtmfsrc, "Pushing the PAUSE_TASK event on unlock request");
event = g_slice_new0 (GstRTPDTMFSrcEvent);
event->event_type = RTP_DTMF_EVENT_TYPE_PAUSE_TASK;
g_async_queue_push (dtmfsrc->event_queue, event);
return TRUE;
}
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 void
fs_rtp_sub_stream_stop_no_rtcp_timeout_thread (FsRtpSubStream *self)
{
FS_RTP_SUB_STREAM_LOCK(self);
self->priv->next_no_rtcp_timeout = 0;
if (self->priv->no_rtcp_timeout_id)
gst_clock_id_unschedule (self->priv->no_rtcp_timeout_id);
if (self->priv->no_rtcp_timeout_thread == NULL)
{
FS_RTP_SUB_STREAM_UNLOCK(self);
return;
}
else
{
FS_RTP_SUB_STREAM_UNLOCK(self);
}
g_thread_join (self->priv->no_rtcp_timeout_thread);
FS_RTP_SUB_STREAM_LOCK(self);
self->priv->no_rtcp_timeout_thread = NULL;
FS_RTP_SUB_STREAM_UNLOCK(self);
}
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;
}
/*
* Method: unschedule
*
* Cancels an outstanding async notification request.
*
* Returns: self.
*/
static VALUE
rg_unschedule (VALUE self)
{
gst_clock_id_unschedule (RGST_CLOCK_ENTRY (self));
return self;
}
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);
}
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;
}
static void
gst_net_client_clock_constructed (GObject * object)
{
GstNetClientClock *self = GST_NET_CLIENT_CLOCK (object);
GstClock *internal_clock;
GstClockTime internal;
GList *l;
ClockCache *cache = NULL;
G_OBJECT_CLASS (gst_net_client_clock_parent_class)->constructed (object);
G_LOCK (clocks_lock);
for (l = clocks; l; l = l->next) {
ClockCache *tmp = l->data;
GstNetClientInternalClock *internal_clock =
GST_NET_CLIENT_INTERNAL_CLOCK (tmp->clock);
if (strcmp (internal_clock->address, self->priv->address) == 0 &&
internal_clock->port == self->priv->port) {
cache = tmp;
if (cache->remove_id) {
gst_clock_id_unschedule (cache->remove_id);
cache->remove_id = NULL;
}
break;
}
}
if (!cache) {
cache = g_new0 (ClockCache, 1);
cache->clock =
g_object_new (GST_TYPE_NET_CLIENT_INTERNAL_CLOCK, "address",
self->priv->address, "port", self->priv->port, "is-ntp",
self->priv->is_ntp, NULL);
clocks = g_list_prepend (clocks, cache);
}
cache->clocks = g_list_prepend (cache->clocks, self);
GST_OBJECT_LOCK (cache->clock);
if (gst_clock_is_synced (cache->clock))
gst_clock_set_synced (GST_CLOCK (self), TRUE);
self->priv->synced_id =
g_signal_connect (cache->clock, "synced",
G_CALLBACK (gst_net_client_clock_synced_cb), self);
GST_OBJECT_UNLOCK (cache->clock);
G_UNLOCK (clocks_lock);
self->priv->internal_clock = internal_clock = cache->clock;
/* gst_clock_get_time() values are guaranteed to be increasing. because no one
* has called get_time on this clock yet we are free to adjust to any value
* without worrying about worrying about MAX() issues with the clock's
* internal time.
*/
/* update our internal time so get_time() give something around base_time.
assume that the rate is 1 in the beginning. */
internal = gst_clock_get_internal_time (internal_clock);
gst_clock_set_calibration (internal_clock, internal,
self->priv->base_time, 1, 1);
{
GstClockTime now = gst_clock_get_time (internal_clock);
if (GST_CLOCK_DIFF (now, self->priv->base_time) > 0 ||
GST_CLOCK_DIFF (now, self->priv->base_time + GST_SECOND) < 0) {
g_warning ("unable to set the base time, expect sync problems!");
}
}
/* all systems go, cap'n */
}
static GstFlowReturn
gst_rtp_jitter_buffer_chain (GstPad * pad, GstBuffer * buffer)
{
GstRtpJitterBuffer *jitterbuffer;
GstRtpJitterBufferPrivate *priv;
guint16 seqnum;
GstFlowReturn ret = GST_FLOW_OK;
GstClockTime timestamp;
guint64 latency_ts;
gboolean tail;
jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
if (!gst_rtp_buffer_validate (buffer))
goto invalid_buffer;
priv = jitterbuffer->priv;
if (priv->last_pt != gst_rtp_buffer_get_payload_type (buffer)) {
GstCaps *caps;
priv->last_pt = gst_rtp_buffer_get_payload_type (buffer);
/* reset clock-rate so that we get a new one */
priv->clock_rate = -1;
/* Try to get the clock-rate from the caps first if we can. If there are no
* caps we must fire the signal to get the clock-rate. */
if ((caps = GST_BUFFER_CAPS (buffer))) {
gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
}
}
if (priv->clock_rate == -1) {
guint8 pt;
/* no clock rate given on the caps, try to get one with the signal */
pt = gst_rtp_buffer_get_payload_type (buffer);
gst_rtp_jitter_buffer_get_clock_rate (jitterbuffer, pt);
if (priv->clock_rate == -1)
goto not_negotiated;
}
/* take the timestamp of the buffer. This is the time when the packet was
* received and is used to calculate jitter and clock skew. We will adjust
* this timestamp with the smoothed value after processing it in the
* jitterbuffer. */
timestamp = GST_BUFFER_TIMESTAMP (buffer);
/* bring to running time */
timestamp = gst_segment_to_running_time (&priv->segment, GST_FORMAT_TIME,
timestamp);
seqnum = gst_rtp_buffer_get_seq (buffer);
GST_DEBUG_OBJECT (jitterbuffer,
"Received packet #%d at time %" GST_TIME_FORMAT, seqnum,
GST_TIME_ARGS (timestamp));
JBUF_LOCK_CHECK (priv, out_flushing);
/* don't accept more data on EOS */
if (priv->eos)
goto have_eos;
/* let's check if this buffer is too late, we can only accept packets with
* bigger seqnum than the one we last pushed. */
if (priv->last_popped_seqnum != -1) {
gint gap;
gap = gst_rtp_buffer_compare_seqnum (priv->last_popped_seqnum, seqnum);
if (gap <= 0) {
/* priv->last_popped_seqnum >= seqnum, this packet is too late or the
* sender might have been restarted with different seqnum. */
if (gap < -100) {
GST_DEBUG_OBJECT (jitterbuffer, "reset: buffer too old %d", gap);
priv->last_popped_seqnum = -1;
priv->next_seqnum = -1;
} else {
goto too_late;
}
} else {
/* priv->last_popped_seqnum < seqnum, this is a new packet */
if (gap > 3000) {
GST_DEBUG_OBJECT (jitterbuffer, "reset: too many dropped packets %d",
gap);
priv->last_popped_seqnum = -1;
priv->next_seqnum = -1;
}
}
}
/* let's drop oldest packet if the queue is already full and drop-on-latency
* is set. We can only do this when there actually is a latency. When no
* latency is set, we just pump it in the queue and let the other end push it
* out as fast as possible. */
if (priv->latency_ms && priv->drop_on_latency) {
latency_ts =
gst_util_uint64_scale_int (priv->latency_ms, priv->clock_rate, 1000);
if (rtp_jitter_buffer_get_ts_diff (priv->jbuf) >= latency_ts) {
GstBuffer *old_buf;
GST_DEBUG_OBJECT (jitterbuffer, "Queue full, dropping old packet #%d",
seqnum);
old_buf = rtp_jitter_buffer_pop (priv->jbuf);
gst_buffer_unref (old_buf);
}
}
/* now insert the packet into the queue in sorted order. This function returns
* FALSE if a packet with the same seqnum was already in the queue, meaning we
* have a duplicate. */
if (!rtp_jitter_buffer_insert (priv->jbuf, buffer, timestamp,
priv->clock_rate, &tail))
goto duplicate;
/* signal addition of new buffer when the _loop is waiting. */
if (priv->waiting)
JBUF_SIGNAL (priv);
/* let's unschedule and unblock any waiting buffers. We only want to do this
* when the tail buffer changed */
if (priv->clock_id && tail) {
GST_DEBUG_OBJECT (jitterbuffer,
"Unscheduling waiting buffer, new tail buffer");
gst_clock_id_unschedule (priv->clock_id);
}
GST_DEBUG_OBJECT (jitterbuffer, "Pushed packet #%d, now %d packets",
seqnum, rtp_jitter_buffer_num_packets (priv->jbuf));
finished:
JBUF_UNLOCK (priv);
gst_object_unref (jitterbuffer);
return ret;
/* ERRORS */
invalid_buffer:
{
/* this is not fatal but should be filtered earlier */
GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
("Received invalid RTP payload, dropping"));
gst_buffer_unref (buffer);
gst_object_unref (jitterbuffer);
return GST_FLOW_OK;
}
not_negotiated:
{
GST_WARNING_OBJECT (jitterbuffer, "No clock-rate in caps!");
gst_buffer_unref (buffer);
gst_object_unref (jitterbuffer);
return GST_FLOW_OK;
}
out_flushing:
{
ret = priv->srcresult;
GST_DEBUG_OBJECT (jitterbuffer, "flushing %s", gst_flow_get_name (ret));
gst_buffer_unref (buffer);
goto finished;
}
have_eos:
{
ret = GST_FLOW_UNEXPECTED;
GST_WARNING_OBJECT (jitterbuffer, "we are EOS, refusing buffer");
gst_buffer_unref (buffer);
goto finished;
}
too_late:
{
GST_WARNING_OBJECT (jitterbuffer, "Packet #%d too late as #%d was already"
" popped, dropping", seqnum, priv->last_popped_seqnum);
priv->num_late++;
gst_buffer_unref (buffer);
goto finished;
}
duplicate:
{
GST_WARNING_OBJECT (jitterbuffer, "Duplicate packet #%d detected, dropping",
seqnum);
priv->num_duplicates++;
gst_buffer_unref (buffer);
goto finished;
}
}
static GstFlowReturn
gst_live_live_adder_chain (GstPad * pad, GstBuffer * buffer)
{
GstLiveAdder *adder = GST_LIVE_ADDER (gst_pad_get_parent_element (pad));
GstLiveAdderPadPrivate *padprivate = NULL;
GstFlowReturn ret = GST_FLOW_OK;
GList *item = NULL;
GstClockTime skip = 0;
gint64 drift = 0; /* Positive if new buffer after old buffer */
GST_OBJECT_LOCK (adder);
ret = adder->srcresult;
GST_DEBUG ("Incoming buffer time:%" GST_TIME_FORMAT " duration:%"
GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)));
if (ret != GST_FLOW_OK) {
GST_DEBUG_OBJECT (adder, "Passing non-ok result from src: %s",
gst_flow_get_name (ret));
gst_buffer_unref (buffer);
goto out;
}
padprivate = gst_pad_get_element_private (pad);
if (!padprivate) {
ret = GST_FLOW_NOT_LINKED;
gst_buffer_unref (buffer);
goto out;
}
if (padprivate->eos) {
GST_DEBUG_OBJECT (adder, "Received buffer after EOS");
ret = GST_FLOW_UNEXPECTED;
gst_buffer_unref (buffer);
goto out;
}
if (!GST_BUFFER_TIMESTAMP_IS_VALID (buffer))
goto invalid_timestamp;
if (padprivate->segment.format == GST_FORMAT_UNDEFINED) {
GST_WARNING_OBJECT (adder, "No new-segment received,"
" initializing segment with time 0..-1");
gst_segment_init (&padprivate->segment, GST_FORMAT_TIME);
gst_segment_set_newsegment (&padprivate->segment,
FALSE, 1.0, GST_FORMAT_TIME, 0, -1, 0);
}
if (padprivate->segment.format != GST_FORMAT_TIME)
goto invalid_segment;
buffer = gst_buffer_make_metadata_writable (buffer);
drift = GST_BUFFER_TIMESTAMP (buffer) - padprivate->expected_timestamp;
/* Just see if we receive invalid timestamp/durations */
if (GST_CLOCK_TIME_IS_VALID (padprivate->expected_timestamp) &&
!GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DISCONT) &&
(drift != 0)) {
GST_LOG_OBJECT (adder,
"Timestamp discontinuity without the DISCONT flag set"
" (expected %" GST_TIME_FORMAT ", got %" GST_TIME_FORMAT
" drift:%" G_GINT64_FORMAT "ms)",
GST_TIME_ARGS (padprivate->expected_timestamp),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)), drift / GST_MSECOND);
/* We accept drifts of 10ms */
if (ABS (drift) < (10 * GST_MSECOND)) {
GST_DEBUG ("Correcting minor drift");
GST_BUFFER_TIMESTAMP (buffer) = padprivate->expected_timestamp;
}
}
/* If there is no duration, lets set one */
if (!GST_BUFFER_DURATION_IS_VALID (buffer)) {
GST_BUFFER_DURATION (buffer) =
gst_audio_duration_from_pad_buffer (pad, buffer);
padprivate->expected_timestamp = GST_CLOCK_TIME_NONE;
} else {
padprivate->expected_timestamp = GST_BUFFER_TIMESTAMP (buffer) +
GST_BUFFER_DURATION (buffer);
}
/*
* Lets clip the buffer to the segment (so we don't have to worry about
* cliping afterwards).
* This should also guarantee us that we'll have valid timestamps and
* durations afterwards
*/
buffer = gst_audio_buffer_clip (buffer, &padprivate->segment, adder->rate,
adder->bps);
/* buffer can be NULL if it's completely outside of the segment */
if (!buffer) {
GST_DEBUG ("Buffer completely outside of configured segment, dropping it");
goto out;
}
/*
* Make sure all incoming buffers share the same timestamping
*/
GST_BUFFER_TIMESTAMP (buffer) =
gst_segment_to_running_time (&padprivate->segment,
padprivate->segment.format, GST_BUFFER_TIMESTAMP (buffer));
if (GST_CLOCK_TIME_IS_VALID (adder->next_timestamp) &&
GST_BUFFER_TIMESTAMP (buffer) < adder->next_timestamp) {
if (GST_BUFFER_TIMESTAMP (buffer) + GST_BUFFER_DURATION (buffer) <
adder->next_timestamp) {
GST_DEBUG_OBJECT (adder, "Buffer is late, dropping (ts: %" GST_TIME_FORMAT
" duration: %" GST_TIME_FORMAT ")",
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)));
gst_buffer_unref (buffer);
goto out;
} else {
skip = adder->next_timestamp - GST_BUFFER_TIMESTAMP (buffer);
GST_DEBUG_OBJECT (adder, "Buffer is partially late, skipping %"
GST_TIME_FORMAT, GST_TIME_ARGS (skip));
}
}
/* If our new buffer's head is higher than the queue's head, lets wake up,
* we may not have to wait for as long
*/
if (adder->clock_id &&
g_queue_peek_head (adder->buffers) != NULL &&
GST_BUFFER_TIMESTAMP (buffer) + skip <
GST_BUFFER_TIMESTAMP (g_queue_peek_head (adder->buffers)))
gst_clock_id_unschedule (adder->clock_id);
for (item = g_queue_peek_head_link (adder->buffers);
item; item = g_list_next (item)) {
GstBuffer *oldbuffer = item->data;
GstClockTime old_skip = 0;
GstClockTime mix_duration = 0;
GstClockTime mix_start = 0;
GstClockTime mix_end = 0;
/* We haven't reached our place yet */
if (GST_BUFFER_TIMESTAMP (buffer) + skip >=
GST_BUFFER_TIMESTAMP (oldbuffer) + GST_BUFFER_DURATION (oldbuffer))
continue;
/* We're past our place, lets insert ouselves here */
if (GST_BUFFER_TIMESTAMP (buffer) + GST_BUFFER_DURATION (buffer) <=
GST_BUFFER_TIMESTAMP (oldbuffer))
break;
/* if we reach this spot, we have overlap, so we must mix */
/* First make a subbuffer with the non-overlapping part */
if (GST_BUFFER_TIMESTAMP (buffer) + skip < GST_BUFFER_TIMESTAMP (oldbuffer)) {
GstBuffer *subbuffer = NULL;
GstClockTime subbuffer_duration = GST_BUFFER_TIMESTAMP (oldbuffer) -
(GST_BUFFER_TIMESTAMP (buffer) + skip);
subbuffer = gst_buffer_create_sub (buffer,
gst_live_adder_length_from_duration (adder, skip),
gst_live_adder_length_from_duration (adder, subbuffer_duration));
GST_BUFFER_TIMESTAMP (subbuffer) = GST_BUFFER_TIMESTAMP (buffer) + skip;
GST_BUFFER_DURATION (subbuffer) = subbuffer_duration;
skip += subbuffer_duration;
g_queue_insert_before (adder->buffers, item, subbuffer);
}
/* Now we are on the overlapping part */
oldbuffer = gst_buffer_make_writable (oldbuffer);
item->data = oldbuffer;
old_skip = GST_BUFFER_TIMESTAMP (buffer) + skip -
GST_BUFFER_TIMESTAMP (oldbuffer);
mix_start = GST_BUFFER_TIMESTAMP (oldbuffer) + old_skip;
if (GST_BUFFER_TIMESTAMP (buffer) + GST_BUFFER_DURATION (buffer) <
GST_BUFFER_TIMESTAMP (oldbuffer) + GST_BUFFER_DURATION (oldbuffer))
mix_end = GST_BUFFER_TIMESTAMP (buffer) + GST_BUFFER_DURATION (buffer);
else
mix_end = GST_BUFFER_TIMESTAMP (oldbuffer) +
GST_BUFFER_DURATION (oldbuffer);
mix_duration = mix_end - mix_start;
adder->func (GST_BUFFER_DATA (oldbuffer) +
gst_live_adder_length_from_duration (adder, old_skip),
GST_BUFFER_DATA (buffer) +
gst_live_adder_length_from_duration (adder, skip),
gst_live_adder_length_from_duration (adder, mix_duration));
skip += mix_duration;
}
g_cond_broadcast (adder->not_empty_cond);
if (skip == GST_BUFFER_DURATION (buffer)) {
gst_buffer_unref (buffer);
} else {
if (skip) {
GstClockTime subbuffer_duration = GST_BUFFER_DURATION (buffer) - skip;
GstClockTime subbuffer_ts = GST_BUFFER_TIMESTAMP (buffer) + skip;
GstBuffer *new_buffer = gst_buffer_create_sub (buffer,
gst_live_adder_length_from_duration (adder, skip),
gst_live_adder_length_from_duration (adder, subbuffer_duration));
gst_buffer_unref (buffer);
buffer = new_buffer;
GST_BUFFER_TIMESTAMP (buffer) = subbuffer_ts;
GST_BUFFER_DURATION (buffer) = subbuffer_duration;
}
if (item)
g_queue_insert_before (adder->buffers, item, buffer);
else
g_queue_push_tail (adder->buffers, buffer);
}
out:
GST_OBJECT_UNLOCK (adder);
gst_object_unref (adder);
return ret;
invalid_timestamp:
GST_OBJECT_UNLOCK (adder);
gst_buffer_unref (buffer);
GST_ELEMENT_ERROR (adder, STREAM, FAILED,
("Buffer without a valid timestamp received"),
("Invalid timestamp received on buffer"));
return GST_FLOW_ERROR;
invalid_segment:
{
const gchar *format = gst_format_get_name (padprivate->segment.format);
GST_OBJECT_UNLOCK (adder);
gst_buffer_unref (buffer);
GST_ELEMENT_ERROR (adder, STREAM, FAILED,
("This element only supports TIME segments, received other type"),
("Received a segment of type %s, only support time segment", format));
return GST_FLOW_ERROR;
}
}
static GstStateChangeReturn
gst_mim_enc_change_state (GstElement * element, GstStateChange transition)
{
GstMimEnc *mimenc = GST_MIM_ENC (element);
GstStateChangeReturn ret;
gboolean paused_mode;
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
case GST_STATE_CHANGE_READY_TO_PAUSED:
GST_OBJECT_LOCK (mimenc);
gst_segment_init (&mimenc->segment, GST_FORMAT_UNDEFINED);
mimenc->last_buffer = GST_CLOCK_TIME_NONE;
GST_OBJECT_UNLOCK (mimenc);
break;
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
GST_OBJECT_LOCK (mimenc);
if (mimenc->clock_id)
gst_clock_id_unschedule (mimenc->clock_id);
mimenc->stop_paused_mode = TRUE;
GST_OBJECT_UNLOCK (mimenc);
gst_pad_pause_task (mimenc->srcpad);
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (gst_mim_enc_parent_class)->change_state (element,
transition);
if (ret == GST_STATE_CHANGE_FAILURE)
return ret;
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
GST_OBJECT_LOCK (mimenc);
mimenc->stop_paused_mode = FALSE;
paused_mode = mimenc->paused_mode;
if (paused_mode) {
if (!GST_ELEMENT_CLOCK (mimenc)) {
GST_OBJECT_UNLOCK (mimenc);
GST_ELEMENT_ERROR (mimenc, RESOURCE, FAILED,
("Using paused-mode requires a clock, but no clock was provided"
" to the element"), (NULL));
return GST_STATE_CHANGE_FAILURE;
}
if (mimenc->last_buffer == GST_CLOCK_TIME_NONE)
mimenc->last_buffer = gst_clock_get_time (GST_ELEMENT_CLOCK (mimenc))
- GST_ELEMENT_CAST (mimenc)->base_time;
}
GST_OBJECT_UNLOCK (mimenc);
if (paused_mode) {
if (!gst_pad_start_task (mimenc->srcpad, paused_mode_task, mimenc,
NULL)) {
ret = GST_STATE_CHANGE_FAILURE;
GST_ERROR_OBJECT (mimenc, "Can not start task");
}
}
break;
case GST_STATE_CHANGE_READY_TO_NULL:
gst_mim_enc_reset (mimenc);
break;
default:
break;
}
return ret;
}
