/* chain function
* this function does the actual processing
*/
static GstFlowReturn gst_throttle_chain(GstPad * pad, GstBuffer * buf)
{
GstThrottle * filter = GST_THROTTLE(GST_OBJECT_PARENT(pad));
if (filter->printOnly)
{
GstCaps * caps = gst_buffer_get_caps(buf);
gchar * capsStr = gst_caps_to_string(caps);
gst_caps_unref(caps);
GST_LOG_OBJECT(filter, "ts: %" GST_TIME_FORMAT " %sof type %s",
GST_TIME_ARGS(buf->timestamp),
GST_BUFFER_IS_DISCONT(buf) ? "and discontinuity " : "",
capsStr
);
g_free(capsStr);
GstFlowReturn ret = gst_pad_push(filter->srcpad, buf);
GST_TRACE_OBJECT(filter, "ts: %" GST_TIME_FORMAT " processed with status %d", GST_TIME_ARGS(buf->timestamp), ret);
return ret;
}
if (filter->clock == NULL)
{
return gst_pad_push(filter->srcpad, buf);
}
GstClockTime realTs = gst_clock_get_time(filter->clock);
if (filter->haveStartTime)
{
const char * discont = GST_BUFFER_IS_DISCONT(buf) ? " with discotinuity" : "";
GstClockTime expectedRealTs = filter->streamStartRealTime + buf->timestamp;
gboolean early = realTs < expectedRealTs;
if (early)
{
GstClockID * cid = gst_clock_new_single_shot_id(filter->clock, expectedRealTs);
GST_TRACE_OBJECT(filter, "ts: %" GST_TIME_FORMAT " %s, waiting for %ld ms", GST_TIME_ARGS(buf->timestamp), discont, (expectedRealTs - realTs)/1000000);
gst_clock_id_wait(cid, NULL);
gst_clock_id_unref(cid);
}
else
{
GST_TRACE_OBJECT(filter, "ts: %" GST_TIME_FORMAT " %s, pad on time", GST_TIME_ARGS(buf->timestamp), discont);
}
}
else
{
filter->streamStartRealTime = realTs - buf->timestamp;
filter->haveStartTime = TRUE;
}
return gst_pad_push(filter->srcpad, buf);
}
/**
* dxr3videosink_wait:
*
* Make the sink wait the specified amount of time.
*/
static void
dxr3videosink_wait (Dxr3VideoSink * sink, GstClockTime time)
{
GstClockID id;
GstClockTimeDiff jitter;
GstClockReturn ret;
GstClockTime current_time = gst_clock_get_time (sink->clock);
id = gst_clock_new_single_shot_id (sink->clock, current_time + time);
ret = gst_clock_id_wait (id, &jitter);
gst_clock_id_free (id);
}
static void
gst_dtls_connection_check_timeout_locked (GstDtlsConnection * self)
{
GstDtlsConnectionPrivate *priv;
struct timeval timeout;
gint64 end_time, wait_time;
g_return_if_fail (GST_IS_DTLS_CONNECTION (self));
priv = self->priv;
if (DTLSv1_get_timeout (priv->ssl, &timeout)) {
wait_time = timeout.tv_sec * G_USEC_PER_SEC + timeout.tv_usec;
GST_DEBUG_OBJECT (self, "waiting for %" G_GINT64_FORMAT " usec", wait_time);
if (wait_time) {
GstClock *system_clock = gst_system_clock_obtain ();
GstClockID clock_id;
#ifndef G_DISABLE_ASSERT
GstClockReturn clock_return;
#endif
end_time = gst_clock_get_time (system_clock) + wait_time * GST_USECOND;
clock_id = gst_clock_new_single_shot_id (system_clock, end_time);
#ifndef G_DISABLE_ASSERT
clock_return =
#else
(void)
#endif
gst_clock_id_wait_async (clock_id, schedule_timeout_handling,
g_object_ref (self), (GDestroyNotify) g_object_unref);
g_assert (clock_return == GST_CLOCK_OK);
gst_clock_id_unref (clock_id);
gst_object_unref (system_clock);
} else {
if (self->priv->is_alive && !self->priv->timeout_pending) {
self->priv->timeout_pending = TRUE;
GST_TRACE_OBJECT (self, "Schedule timeout now");
g_thread_pool_push (self->priv->thread_pool, GINT_TO_POINTER (0xc0ffee),
NULL);
}
}
} else {
GST_DEBUG_OBJECT (self, "no timeout set");
}
}
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;
}
/**
* 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 void
gst_net_client_clock_finalize (GObject * object)
{
GstNetClientClock *self = GST_NET_CLIENT_CLOCK (object);
GList *l;
if (self->priv->synced_id)
g_signal_handler_disconnect (self->priv->internal_clock,
self->priv->synced_id);
self->priv->synced_id = 0;
G_LOCK (clocks_lock);
for (l = clocks; l; l = l->next) {
ClockCache *cache = l->data;
if (cache->clock == self->priv->internal_clock) {
cache->clocks = g_list_remove (cache->clocks, self);
if (cache->clocks) {
update_clock_cache (cache);
} else {
GstClock *sysclock = gst_system_clock_obtain ();
GstClockTime time = gst_clock_get_time (sysclock) + 60 * GST_SECOND;
cache->remove_id = gst_clock_new_single_shot_id (sysclock, time);
gst_clock_id_wait_async (cache->remove_id, remove_clock_cache, cache,
NULL);
gst_object_unref (sysclock);
}
break;
}
}
G_UNLOCK (clocks_lock);
g_free (self->priv->address);
self->priv->address = NULL;
if (self->priv->bus != NULL) {
gst_object_unref (self->priv->bus);
self->priv->bus = NULL;
}
G_OBJECT_CLASS (gst_net_client_clock_parent_class)->finalize (object);
}
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 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;
}
/*
* Class method: new(clock, time, interval=nil)
* clock: a Gst::Clock.
* time: a time period, in nanoseconds.
* interval: an interval period, in nanoseconds.
*
* Creates a new Gst::ClockEntry object based on the given Gst::Clock.
*
* Two types of Gst::ClockEntry objects can be created:
*
* * One-shot: if the interval is ommited or nil, the entry will trigger a single shot notification, at the requested time (in nanoseconds);
* * Periodic: if the interval is not nil, the timer entry will trigger a periodic notification, starting at time (in nanoseconds), and be fired with the given interval (also in nanoseconds).
*
* The timer will be issued after Gst::ClockEntry#wait or
* Gst::ClockEntry#wait_async.
*
* Returns: a new Gst::ClockEntry object.
*/
static VALUE
rg_initialize (int argc, VALUE * argv, VALUE self)
{
VALUE clock, time, interval;
GstClockID id;
rb_scan_args (argc, argv, "21", &clock, &time, &interval);
/*
* Single-shot
*/
if (NIL_P (interval))
id = gst_clock_new_single_shot_id (RVAL2GST_CLOCK (clock), NUM2ULL (time));
/*
* Periodic
*/
else
id = gst_clock_new_periodic_id (RVAL2GST_CLOCK (clock),
NUM2ULL (time), NUM2ULL (interval));
G_INITIALIZE (self, GST_CLOCK_ENTRY (id));
return Qnil;
}
static GstFlowReturn
gst_rtp_dtmf_src_create (GstBaseSrc * basesrc, guint64 offset,
guint length, GstBuffer ** buffer)
{
GstRTPDTMFSrcEvent *event;
GstRTPDTMFSrc *dtmfsrc;
GstClock *clock;
GstClockID *clockid;
GstClockReturn clockret;
GstMessage *message;
GQueue messages = G_QUEUE_INIT;
dtmfsrc = GST_RTP_DTMF_SRC (basesrc);
do {
if (dtmfsrc->payload == NULL) {
GST_DEBUG_OBJECT (dtmfsrc, "popping");
event = g_async_queue_pop (dtmfsrc->event_queue);
GST_DEBUG_OBJECT (dtmfsrc, "popped %d", event->event_type);
switch (event->event_type) {
case RTP_DTMF_EVENT_TYPE_STOP:
GST_WARNING_OBJECT (dtmfsrc,
"Received a DTMF stop event when already stopped");
gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-dropped", event);
break;
case RTP_DTMF_EVENT_TYPE_START:
dtmfsrc->first_packet = TRUE;
dtmfsrc->last_packet = FALSE;
/* Set the redundancy on the first packet */
dtmfsrc->redundancy_count = dtmfsrc->packet_redundancy;
if (!gst_rtp_dtmf_prepare_timestamps (dtmfsrc))
goto no_clock;
g_queue_push_tail (&messages,
gst_dtmf_src_prepare_message (dtmfsrc, "dtmf-event-processed",
event));
dtmfsrc->payload = event->payload;
dtmfsrc->payload->duration =
dtmfsrc->ptime * dtmfsrc->clock_rate / 1000;
event->payload = NULL;
break;
case RTP_DTMF_EVENT_TYPE_PAUSE_TASK:
/*
* We're pushing it back because it has to stay in there until
* the task is really paused (and the queue will then be flushed
*/
GST_OBJECT_LOCK (dtmfsrc);
if (dtmfsrc->paused) {
g_async_queue_push (dtmfsrc->event_queue, event);
goto paused_locked;
}
GST_OBJECT_UNLOCK (dtmfsrc);
break;
}
gst_rtp_dtmf_src_event_free (event);
} else if (!dtmfsrc->first_packet && !dtmfsrc->last_packet &&
(dtmfsrc->timestamp - dtmfsrc->start_timestamp) / GST_MSECOND >=
MIN_PULSE_DURATION) {
GST_DEBUG_OBJECT (dtmfsrc, "try popping");
event = g_async_queue_try_pop (dtmfsrc->event_queue);
if (event != NULL) {
GST_DEBUG_OBJECT (dtmfsrc, "try popped %d", event->event_type);
switch (event->event_type) {
case RTP_DTMF_EVENT_TYPE_START:
GST_WARNING_OBJECT (dtmfsrc,
"Received two consecutive DTMF start events");
gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-dropped", event);
break;
case RTP_DTMF_EVENT_TYPE_STOP:
dtmfsrc->first_packet = FALSE;
dtmfsrc->last_packet = TRUE;
/* Set the redundancy on the last packet */
dtmfsrc->redundancy_count = dtmfsrc->packet_redundancy;
g_queue_push_tail (&messages,
gst_dtmf_src_prepare_message (dtmfsrc, "dtmf-event-processed",
event));
break;
case RTP_DTMF_EVENT_TYPE_PAUSE_TASK:
/*
* We're pushing it back because it has to stay in there until
* the task is really paused (and the queue will then be flushed)
*/
GST_DEBUG_OBJECT (dtmfsrc, "pushing pause_task...");
GST_OBJECT_LOCK (dtmfsrc);
if (dtmfsrc->paused) {
g_async_queue_push (dtmfsrc->event_queue, event);
goto paused_locked;
}
GST_OBJECT_UNLOCK (dtmfsrc);
break;
}
gst_rtp_dtmf_src_event_free (event);
}
}
} while (dtmfsrc->payload == NULL);
GST_DEBUG_OBJECT (dtmfsrc, "Processed events, now lets wait on the clock");
clock = gst_element_get_clock (GST_ELEMENT (basesrc));
if (!clock)
goto no_clock;
clockid = gst_clock_new_single_shot_id (clock, dtmfsrc->timestamp +
gst_element_get_base_time (GST_ELEMENT (dtmfsrc)));
gst_object_unref (clock);
GST_OBJECT_LOCK (dtmfsrc);
if (!dtmfsrc->paused) {
dtmfsrc->clockid = clockid;
GST_OBJECT_UNLOCK (dtmfsrc);
clockret = gst_clock_id_wait (clockid, NULL);
GST_OBJECT_LOCK (dtmfsrc);
if (dtmfsrc->paused)
clockret = GST_CLOCK_UNSCHEDULED;
} else {
clockret = GST_CLOCK_UNSCHEDULED;
}
gst_clock_id_unref (clockid);
dtmfsrc->clockid = NULL;
GST_OBJECT_UNLOCK (dtmfsrc);
while ((message = g_queue_pop_head (&messages)) != NULL)
gst_element_post_message (GST_ELEMENT (dtmfsrc), message);
if (clockret == GST_CLOCK_UNSCHEDULED) {
goto paused;
}
send_last:
if (dtmfsrc->dirty)
if (!gst_rtp_dtmf_src_negotiate (basesrc))
return GST_FLOW_NOT_NEGOTIATED;
/* create buffer to hold the payload */
*buffer = gst_rtp_dtmf_src_create_next_rtp_packet (dtmfsrc);
if (dtmfsrc->redundancy_count)
dtmfsrc->redundancy_count--;
/* Only the very first one has a marker */
dtmfsrc->first_packet = FALSE;
/* This is the end of the event */
if (dtmfsrc->last_packet == TRUE && dtmfsrc->redundancy_count == 0) {
g_slice_free (GstRTPDTMFPayload, dtmfsrc->payload);
dtmfsrc->payload = NULL;
dtmfsrc->last_packet = FALSE;
}
return GST_FLOW_OK;
paused_locked:
GST_OBJECT_UNLOCK (dtmfsrc);
paused:
if (dtmfsrc->payload) {
dtmfsrc->first_packet = FALSE;
dtmfsrc->last_packet = TRUE;
/* Set the redundanc on the last packet */
dtmfsrc->redundancy_count = dtmfsrc->packet_redundancy;
goto send_last;
} else {
return GST_FLOW_FLUSHING;
}
no_clock:
GST_ELEMENT_ERROR (dtmfsrc, STREAM, MUX, ("No available clock"),
("No available clock"));
gst_pad_pause_task (GST_BASE_SRC_PAD (dtmfsrc));
return GST_FLOW_ERROR;
}
static void
gst_live_adder_loop (gpointer data)
{
GstLiveAdder *adder = GST_LIVE_ADDER (data);
GstClockTime buffer_timestamp = 0;
GstClockTime sync_time = 0;
GstClock *clock = NULL;
GstClockID id = NULL;
GstClockReturn ret;
GstBuffer *buffer = NULL;
GstFlowReturn result;
GstEvent *newseg_event = NULL;
GST_OBJECT_LOCK (adder);
again:
for (;;) {
if (adder->srcresult != GST_FLOW_OK)
goto flushing;
if (!g_queue_is_empty (adder->buffers))
break;
if (check_eos_locked (adder))
goto eos;
g_cond_wait (adder->not_empty_cond, GST_OBJECT_GET_LOCK (adder));
}
buffer_timestamp = GST_BUFFER_TIMESTAMP (g_queue_peek_head (adder->buffers));
clock = GST_ELEMENT_CLOCK (adder);
/* If we have no clock, then we can't do anything.. error */
if (!clock) {
if (adder->playing)
goto no_clock;
else
goto push_buffer;
}
GST_DEBUG_OBJECT (adder, "sync to timestamp %" GST_TIME_FORMAT,
GST_TIME_ARGS (buffer_timestamp));
sync_time = buffer_timestamp + GST_ELEMENT_CAST (adder)->base_time;
/* add latency, this includes our own latency and the peer latency. */
sync_time += adder->latency_ms * GST_MSECOND;
sync_time += adder->peer_latency;
/* create an entry for the clock */
id = adder->clock_id = gst_clock_new_single_shot_id (clock, sync_time);
GST_OBJECT_UNLOCK (adder);
ret = gst_clock_id_wait (id, NULL);
GST_OBJECT_LOCK (adder);
/* and free the entry */
gst_clock_id_unref (id);
adder->clock_id = NULL;
/* at this point, the clock could have been unlocked by a timeout, a new
* head element was added to the queue or because we are shutting down. Check
* for shutdown first. */
if (adder->srcresult != GST_FLOW_OK)
goto flushing;
if (ret == GST_CLOCK_UNSCHEDULED) {
GST_DEBUG_OBJECT (adder,
"Wait got unscheduled, will retry to push with new buffer");
goto again;
}
if (ret != GST_CLOCK_OK && ret != GST_CLOCK_EARLY)
goto clock_error;
push_buffer:
buffer = g_queue_pop_head (adder->buffers);
if (!buffer)
goto again;
/*
* We make sure the timestamps are exactly contiguous
* If its only small skew (due to rounding errors), we correct it
* silently. Otherwise we put the discont flag
*/
if (GST_CLOCK_TIME_IS_VALID (adder->next_timestamp) &&
GST_BUFFER_TIMESTAMP (buffer) != adder->next_timestamp) {
GstClockTimeDiff diff = GST_CLOCK_DIFF (GST_BUFFER_TIMESTAMP (buffer),
adder->next_timestamp);
if (diff < 0)
diff = -diff;
if (diff < GST_SECOND / adder->rate) {
GST_BUFFER_TIMESTAMP (buffer) = adder->next_timestamp;
GST_DEBUG_OBJECT (adder, "Correcting slight skew");
GST_BUFFER_FLAG_UNSET (buffer, GST_BUFFER_FLAG_DISCONT);
} else {
GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DISCONT);
GST_DEBUG_OBJECT (adder, "Expected buffer at %" GST_TIME_FORMAT
", but is at %" GST_TIME_FORMAT ", setting discont",
GST_TIME_ARGS (adder->next_timestamp),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
}
} else {
GST_BUFFER_FLAG_UNSET (buffer, GST_BUFFER_FLAG_DISCONT);
}
GST_BUFFER_OFFSET (buffer) = GST_BUFFER_OFFSET_NONE;
GST_BUFFER_OFFSET_END (buffer) = GST_BUFFER_OFFSET_NONE;
if (GST_BUFFER_DURATION_IS_VALID (buffer))
adder->next_timestamp = GST_BUFFER_TIMESTAMP (buffer) +
GST_BUFFER_DURATION (buffer);
else
adder->next_timestamp = GST_CLOCK_TIME_NONE;
if (adder->segment_pending) {
/*
* We set the start at 0, because we re-timestamps to the running time
*/
newseg_event = gst_event_new_new_segment_full (FALSE, 1.0, 1.0,
GST_FORMAT_TIME, 0, -1, 0);
adder->segment_pending = FALSE;
}
GST_OBJECT_UNLOCK (adder);
if (newseg_event)
gst_pad_push_event (adder->srcpad, newseg_event);
GST_LOG_OBJECT (adder, "About to push buffer time:%" GST_TIME_FORMAT
" duration:%" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)));
result = gst_pad_push (adder->srcpad, buffer);
if (result != GST_FLOW_OK)
goto pause;
return;
flushing:
{
GST_DEBUG_OBJECT (adder, "we are flushing");
gst_pad_pause_task (adder->srcpad);
GST_OBJECT_UNLOCK (adder);
return;
}
clock_error:
{
gst_pad_pause_task (adder->srcpad);
GST_OBJECT_UNLOCK (adder);
GST_ELEMENT_ERROR (adder, STREAM, MUX, ("Error with the clock"),
("Error with the clock: %d", ret));
GST_ERROR_OBJECT (adder, "Error with the clock: %d", ret);
return;
}
no_clock:
{
gst_pad_pause_task (adder->srcpad);
GST_OBJECT_UNLOCK (adder);
GST_ELEMENT_ERROR (adder, STREAM, MUX, ("No available clock"),
("No available clock"));
GST_ERROR_OBJECT (adder, "No available clock");
return;
}
pause:
{
GST_DEBUG_OBJECT (adder, "pausing task, reason %s",
gst_flow_get_name (result));
GST_OBJECT_LOCK (adder);
/* store result */
adder->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 (adder->srcpad);
GST_OBJECT_UNLOCK (adder);
return;
}
eos:
{
/* store result, we are flushing now */
GST_DEBUG_OBJECT (adder, "We are EOS, pushing EOS downstream");
adder->srcresult = GST_FLOW_UNEXPECTED;
gst_pad_pause_task (adder->srcpad);
GST_OBJECT_UNLOCK (adder);
gst_pad_push_event (adder->srcpad, gst_event_new_eos ());
return;
}
}
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_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;
}
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;
}
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);
}
/**
* 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_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_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
_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;
}
// 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;
}
static void
paused_mode_task (gpointer data)
{
GstMimEnc *mimenc = GST_MIM_ENC (data);
GstClockTime now;
GstClockTimeDiff diff;
GstFlowReturn ret;
GST_OBJECT_LOCK (mimenc);
if (!GST_ELEMENT_CLOCK (mimenc)) {
GST_OBJECT_UNLOCK (mimenc);
GST_ERROR_OBJECT (mimenc, "Element has no clock");
gst_pad_pause_task (mimenc->srcpad);
return;
}
if (mimenc->stop_paused_mode) {
GST_OBJECT_UNLOCK (mimenc);
goto stop_task;
}
now = gst_clock_get_time (GST_ELEMENT_CLOCK (mimenc));
diff = now - GST_ELEMENT_CAST (mimenc)->base_time - mimenc->last_buffer;
if (diff < 0)
diff = 0;
if (diff > 3.95 * GST_SECOND) {
GstBuffer *buffer;
GstMapInfo out_map;
buffer = gst_buffer_new_and_alloc (TCP_HEADER_SIZE);
gst_buffer_map (buffer, &out_map, GST_MAP_WRITE);
GST_BUFFER_TIMESTAMP (buffer) = mimenc->last_buffer + PAUSED_MODE_INTERVAL;
gst_mim_enc_create_tcp_header (mimenc, out_map.data, 0,
GST_BUFFER_TIMESTAMP (buffer), FALSE, TRUE);
gst_buffer_unmap (buffer, &out_map);
mimenc->last_buffer += PAUSED_MODE_INTERVAL;
GST_OBJECT_UNLOCK (mimenc);
GST_LOG_OBJECT (mimenc, "Haven't had an incoming buffer in 4 seconds,"
" sending out a pause frame");
ret = gst_pad_push (mimenc->srcpad, buffer);
if (ret < 0) {
GST_WARNING_OBJECT (mimenc, "Error pushing paused header: %s",
gst_flow_get_name (ret));
goto stop_task;
}
} else {
GstClockTime next_stop;
GstClockID id;
next_stop = now + (PAUSED_MODE_INTERVAL - MIN (diff, PAUSED_MODE_INTERVAL));
id = gst_clock_new_single_shot_id (GST_ELEMENT_CLOCK (mimenc), next_stop);
if (mimenc->stop_paused_mode) {
GST_OBJECT_UNLOCK (mimenc);
goto stop_task;
}
mimenc->clock_id = id;
GST_OBJECT_UNLOCK (mimenc);
gst_clock_id_wait (id, NULL);
GST_OBJECT_LOCK (mimenc);
mimenc->clock_id = NULL;
GST_OBJECT_UNLOCK (mimenc);
gst_clock_id_unref (id);
}
return;
stop_task:
gst_pad_pause_task (mimenc->srcpad);
}
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 void gst_imx_v4l2src_af_check_status(GstImxV4l2VideoSrc *v4l2src)
{
int status;
gboolean send_message;
GstPhotographyFocusStatus message_status;
gboolean schedule_recheck;
if (v4l2_g_ctrl(v4l2src, V4L2_CID_AUTO_FOCUS_STATUS, &status) < 0)
goto none;
switch (status)
{
case V4L2_AUTO_FOCUS_STATUS_IDLE:
default:
none:
send_message = TRUE;
message_status = GST_PHOTOGRAPHY_FOCUS_STATUS_NONE;
schedule_recheck = FALSE;
break;
case V4L2_AUTO_FOCUS_STATUS_BUSY:
send_message = FALSE;
schedule_recheck = TRUE;
break;
case V4L2_AUTO_FOCUS_STATUS_REACHED:
send_message = TRUE;
message_status = GST_PHOTOGRAPHY_FOCUS_STATUS_SUCCESS;
schedule_recheck = FALSE;
break;
case V4L2_AUTO_FOCUS_STATUS_FAILED:
send_message = TRUE;
message_status = GST_PHOTOGRAPHY_FOCUS_STATUS_FAIL;
schedule_recheck = FALSE;
break;
}
if (send_message)
{
GstStructure *s;
GstMessage *m;
s = gst_structure_new(GST_PHOTOGRAPHY_AUTOFOCUS_DONE,
"status", G_TYPE_INT, message_status,
NULL);
m = gst_message_new_custom(GST_MESSAGE_ELEMENT,
GST_OBJECT(v4l2src), s);
if (!gst_element_post_message(GST_ELEMENT(v4l2src), m))
GST_ERROR_OBJECT(v4l2src, "failed to post message");
}
if (schedule_recheck)
{
GstClock *c;
GstClockTime t;
c = gst_system_clock_obtain();
t = gst_clock_get_time(c) + 50 * GST_MSECOND;
v4l2src->af_clock_id = gst_clock_new_single_shot_id(c, t);
gst_object_unref(c);
if (gst_clock_id_wait_async(v4l2src->af_clock_id,
gst_imx_v4l2src_af_status_cb,
v4l2src, NULL) != GST_CLOCK_OK)
GST_ERROR_OBJECT(v4l2src, "failed to schedule recheck");
}
}
/*
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_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 void
paused_mode_task (gpointer data)
{
GstMimEnc *mimenc = GST_MIMENC (data);
GstClockTime now;
GstClockTimeDiff diff;
GstFlowReturn ret;
if (!GST_ELEMENT_CLOCK (mimenc)) {
GST_ERROR_OBJECT (mimenc, "Element has no clock");
gst_pad_pause_task (mimenc->srcpad);
return;
}
GST_OBJECT_LOCK (mimenc);
if (mimenc->stop_paused_mode) {
GST_OBJECT_UNLOCK (mimenc);
goto stop_task;
}
now = gst_clock_get_time (GST_ELEMENT_CLOCK (mimenc));
diff = now - GST_ELEMENT_CAST (mimenc)->base_time - mimenc->last_buffer;
if (diff < 0)
diff = 0;
if (diff > 3.95 * GST_SECOND) {
GstBuffer *buffer = gst_mimenc_create_tcp_header (mimenc, 0,
mimenc->last_buffer + 4 * GST_SECOND, FALSE, TRUE);
GstEvent *event = NULL;
mimenc->last_buffer += 4 * GST_SECOND;
if (mimenc->need_newsegment) {
event = gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_TIME, 0, -1, 0);
mimenc->need_newsegment = FALSE;
}
GST_OBJECT_UNLOCK (mimenc);
GST_LOG_OBJECT (mimenc, "Haven't had an incoming buffer in 4 seconds,"
" sending out a pause frame");
if (event) {
if (!gst_pad_push_event (mimenc->srcpad, event))
GST_WARNING_OBJECT (mimenc, "Failed to push NEWSEGMENT event");
}
ret = gst_pad_push (mimenc->srcpad, buffer);
if (ret < 0) {
GST_WARNING_OBJECT (mimenc, "Error pushing paused header: %s",
gst_flow_get_name (ret));
goto stop_task;
}
} else {
GstClockTime next_stop;
GstClockID id;
next_stop = now + (4 * GST_SECOND - MIN (diff, 4 * GST_SECOND));
id = gst_clock_new_single_shot_id (GST_ELEMENT_CLOCK (mimenc), next_stop);
if (mimenc->stop_paused_mode) {
GST_OBJECT_UNLOCK (mimenc);
goto stop_task;
}
mimenc->clock_id = id;
GST_OBJECT_UNLOCK (mimenc);
gst_clock_id_wait (id, NULL);
GST_OBJECT_LOCK (mimenc);
mimenc->clock_id = NULL;
GST_OBJECT_UNLOCK (mimenc);
gst_clock_id_unref (id);
}
return;
stop_task:
gst_pad_pause_task (mimenc->srcpad);
}
