/* This will be called "at the start of acquisition into each image buffer."
* If acquisition blocks because we don't copy buffers fast enough, the number
* of times this function is called will be less than the IMAQ cumulative
* buffer count. */
uInt32
gst_niimaqsrc_frame_start_callback (SESSION_ID sid, IMG_ERR err,
IMG_SIGNAL_TYPE signal_type, uInt32 signal_identifier, void *userdata)
{
GstNiImaqSrc *src = GST_NIIMAQSRC (userdata);
GstClockTime abstime;
static guint32 index = 0;
g_mutex_lock (&src->mutex);
/* time hasn't been read yet, this frame will be dropped */
if (src->times[index] != GST_CLOCK_TIME_NONE) {
g_mutex_unlock (&src->mutex);
return 1;
}
/* get clock time */
abstime = gst_clock_get_time (GST_ELEMENT_CLOCK (src));
src->times[index] = abstime;
if (G_UNLIKELY (src->start_time == NULL))
src->start_time = gst_date_time_new_now_utc ();
/* first frame, use as element base time */
if (src->base_time == GST_CLOCK_TIME_NONE)
src->base_time = abstime;
index = (index + 1) % src->bufsize;
g_mutex_unlock (&src->mutex);
/* return 1 to rearm the callback */
return 1;
}
static void
gst_cel_video_src_timestamp_buffer (GstCelVideoSrc * self, GstBuffer * buf)
{
GstClock *clock;
GstClockTime ts;
GST_OBJECT_LOCK (self);
if ((clock = GST_ELEMENT_CLOCK (self)) != NULL) {
ts = gst_clock_get_time (clock);
if (ts > GST_ELEMENT (self)->base_time)
ts -= GST_ELEMENT (self)->base_time;
else
ts = 0;
if (ts > self->duration)
ts -= self->duration;
else
ts = 0;
} else {
ts = GST_CLOCK_TIME_NONE;
}
GST_OBJECT_UNLOCK (self);
GST_BUFFER_OFFSET (buf) = self->offset;
GST_BUFFER_OFFSET_END (buf) = self->offset + 1;
GST_BUFFER_TIMESTAMP (buf) = ts;
GST_BUFFER_DURATION (buf) = self->duration;
if (self->offset == 0)
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
self->offset++;
}
static void
gst_cel_video_src_update_statistics (GstCelVideoSrc * self)
{
GstClock *clock;
GST_OBJECT_LOCK (self);
clock = GST_ELEMENT_CLOCK (self);
if (clock != NULL)
gst_object_ref (clock);
GST_OBJECT_UNLOCK (self);
if (clock != NULL) {
GstClockTime now = gst_clock_get_time (clock);
gst_object_unref (clock);
self->count++;
if (GST_CLOCK_TIME_IS_VALID (self->last_sampling)) {
if (now - self->last_sampling >= GST_SECOND) {
GST_OBJECT_LOCK (self);
self->fps = self->count;
GST_OBJECT_UNLOCK (self);
g_object_notify (G_OBJECT (self), "fps");
self->last_sampling = now;
self->count = 0;
}
} else {
self->last_sampling = now;
}
}
}
static void
gst_ks_video_src_update_statistics (GstKsVideoSrc * self)
{
GstKsVideoSrcPrivate *priv = GST_KS_VIDEO_SRC_GET_PRIVATE (self);
GstClock *clock;
GST_OBJECT_LOCK (self);
clock = GST_ELEMENT_CLOCK (self);
if (clock != NULL)
gst_object_ref (clock);
GST_OBJECT_UNLOCK (self);
if (clock != NULL) {
GstClockTime now = gst_clock_get_time (clock);
gst_object_unref (clock);
priv->count++;
if (GST_CLOCK_TIME_IS_VALID (priv->last_sampling)) {
if (now - priv->last_sampling >= GST_SECOND) {
GST_OBJECT_LOCK (self);
priv->fps = priv->count;
GST_OBJECT_UNLOCK (self);
g_object_notify (G_OBJECT (self), "fps");
priv->last_sampling = now;
priv->count = 0;
}
} else {
priv->last_sampling = now;
}
}
}
static GstClockTime
gst_mio_video_src_get_timestamp (GstMIOVideoSrc * self, CMSampleBufferRef sbuf)
{
GstClock *clock;
GstClockTime base_time;
GstClockTime timestamp;
GST_OBJECT_LOCK (self);
if ((clock = GST_ELEMENT_CLOCK (self)) != NULL) {
gst_object_ref (clock);
}
base_time = GST_ELEMENT_CAST (self)->base_time;
GST_OBJECT_UNLOCK (self);
if (G_UNLIKELY (clock == NULL))
goto no_clock;
timestamp = GST_CLOCK_TIME_NONE;
/*
* If the current clock is GstSystemClock, we know that it's using the
* CoreAudio/CoreVideo clock. As such we may use the timestamp attached
* to the CMSampleBuffer.
*/
if (G_TYPE_FROM_INSTANCE (clock) == GST_TYPE_SYSTEM_CLOCK) {
CFNumberRef number;
UInt64 ht;
number = self->ctx->cm->CMGetAttachment (sbuf,
*self->ctx->mio->kTundraSampleBufferAttachmentKey_HostTime, NULL);
if (number != NULL && CFNumberGetValue (number, kCFNumberSInt64Type, &ht)) {
timestamp = gst_util_uint64_scale_int (ht,
self->cv_ratio_n, self->cv_ratio_d);
}
}
if (!GST_CLOCK_TIME_IS_VALID (timestamp)) {
timestamp = gst_clock_get_time (clock);
}
if (timestamp > base_time)
timestamp -= base_time;
else
timestamp = 0;
gst_object_unref (clock);
return timestamp;
no_clock:
return GST_CLOCK_TIME_NONE;
}
/* Set buffer timing and offset data */
static void set_gstbuf_time_and_offset(GstAndroidVideoSource * p_src, GstBuffer * p_buf)
{
GstElement *p_element;
GstClock *p_clock;
GstClockTime now;
GstClockTime base_time;
GA_LOGTRACE("ENTER %s --xx--> thread(%ld)", __FUNCTION__, pthread_self());
p_element = GST_ELEMENT_CAST(p_src);
GST_OBJECT_LOCK(p_element);
p_clock = GST_ELEMENT_CLOCK(p_element);
if (p_clock) {
gst_object_ref(p_clock);
base_time = p_element->base_time;
GA_LOGTRACE("%s: base_time is: %llu", __FUNCTION__, base_time);
} else {
base_time = GST_CLOCK_TIME_NONE;
}
GST_OBJECT_UNLOCK(p_element);
if (p_clock) {
/* Wrap around is not considered a problem due to the clock being 64 bit (famous last words? :-) ) */
now = gst_clock_get_time(p_clock) - base_time;
GA_LOGTRACE("%s: gst_clock_get_time returns: %llu", __FUNCTION__, gst_clock_get_time(p_clock));
} else {
now = GST_CLOCK_TIME_NONE;
}
if (p_clock) {
gst_object_unref(p_clock);
p_clock = NULL;
}
GST_BUFFER_PTS(p_buf) = now;
GST_BUFFER_DTS(p_buf) = GST_CLOCK_TIME_NONE;
GST_BUFFER_DURATION(p_buf) = GST_CLOCK_TIME_NONE;
GST_BUFFER_OFFSET(p_buf) = GST_BUFFER_OFFSET_NONE;
GST_BUFFER_OFFSET_END(p_buf) = GST_BUFFER_OFFSET_NONE;
GA_LOGTRACE("%s: setting presentation timestamp (GstBuffer) to: %llu (%"GST_TIME_FORMAT")", __FUNCTION__, now, GST_TIME_ARGS(now));
GA_LOGTRACE("%s: m_prev_timestamp: %llu (%"GST_TIME_FORMAT")", __FUNCTION__, p_src->m_prev_timestamp, GST_TIME_ARGS(p_src->m_prev_timestamp));
GA_LOGTRACE("%s: timestamp diff: %llu (%"GST_TIME_FORMAT")", __FUNCTION__, now - p_src->m_prev_timestamp, GST_TIME_ARGS(now - p_src->m_prev_timestamp));
p_src->m_prev_timestamp = now;
GA_LOGTRACE("EXIT %s", __FUNCTION__);
return;
}
static gboolean
gst_ks_video_src_timestamp_buffer (GstKsVideoSrc * self, GstBuffer * buf,
GstClockTime presentation_time)
{
GstKsVideoSrcPrivate *priv = GST_KS_VIDEO_SRC_GET_PRIVATE (self);
GstClockTime duration;
GstClock *clock;
GstClockTime timestamp, base_time;
/* Don't timestamp muxed streams */
if (gst_ks_video_device_stream_is_muxed (priv->device)) {
duration = timestamp = GST_CLOCK_TIME_NONE;
goto timestamp;
}
duration = gst_ks_video_device_get_duration (priv->device);
GST_OBJECT_LOCK (self);
clock = GST_ELEMENT_CLOCK (self);
if (clock != NULL) {
gst_object_ref (clock);
base_time = GST_ELEMENT (self)->base_time;
} else {
timestamp = GST_CLOCK_TIME_NONE;
}
GST_OBJECT_UNLOCK (self);
if (clock != NULL) {
/* The time according to the current clock */
timestamp = gst_clock_get_time (clock) - base_time;
if (timestamp > duration)
timestamp -= duration;
else
timestamp = 0;
gst_object_unref (clock);
clock = NULL;
}
timestamp:
GST_BUFFER_PTS (buf) = timestamp;
GST_BUFFER_DTS (buf) = GST_CLOCK_TIME_NONE;
GST_BUFFER_DURATION (buf) = duration;
return TRUE;
}
static GstFlowReturn
gst_rpi_cam_src_create (GstPushSrc * parent, GstBuffer ** buf)
{
GstRpiCamSrc *src = GST_RPICAMSRC (parent);
GstFlowReturn ret;
GstClock *clock = NULL;
GstClockTime base_time;
if (!src->started) {
g_mutex_lock (&src->config_lock);
raspi_capture_update_config (src->capture_state, &src->capture_config, FALSE);
src->capture_config.change_flags = 0;
g_mutex_unlock (&src->config_lock);
if (!raspi_capture_start (src->capture_state))
return GST_FLOW_ERROR;
src->started = TRUE;
}
GST_OBJECT_LOCK (src);
if ((clock = GST_ELEMENT_CLOCK (src)) != NULL)
gst_object_ref (clock);
base_time = GST_ELEMENT_CAST (src)->base_time;
GST_OBJECT_UNLOCK (src);
g_mutex_lock (&src->config_lock);
if (src->capture_config.change_flags) {
raspi_capture_update_config (src->capture_state, &src->capture_config, TRUE);
src->capture_config.change_flags = 0;
}
g_mutex_unlock (&src->config_lock);
/* FIXME: Use custom allocator */
ret = raspi_capture_fill_buffer (src->capture_state, buf, clock, base_time);
if (*buf)
GST_LOG_OBJECT (src, "Made buffer of size %" G_GSIZE_FORMAT,
gst_buffer_get_size (*buf));
if (clock)
gst_object_unref (clock);
return ret;
}
static GstFlowReturn
gst_rfb_src_fill (GstPushSrc * psrc, GstBuffer * outbuf)
{
GstRfbSrc *src = GST_RFB_SRC (psrc);
RfbDecoder *decoder = src->decoder;
GstMapInfo info;
rfb_decoder_send_update_request (decoder, src->incremental_update,
decoder->offset_x, decoder->offset_y, decoder->rect_width,
decoder->rect_height);
while (decoder->state != NULL) {
if (!rfb_decoder_iterate (decoder)) {
if (decoder->error != NULL) {
GST_ELEMENT_ERROR (src, RESOURCE, READ,
("Error on VNC connection to host %s on port %d: %s",
src->host, src->port, decoder->error->message), (NULL));
} else {
GST_ELEMENT_ERROR (src, RESOURCE, READ,
("Error on setup VNC connection to host %s on port %d", src->host,
src->port), (NULL));
}
return GST_FLOW_ERROR;
}
}
if (!gst_buffer_map (outbuf, &info, GST_MAP_WRITE)) {
GST_ELEMENT_ERROR (src, RESOURCE, WRITE,
("Could not map the output frame"), (NULL));
return GST_FLOW_ERROR;
}
memcpy (info.data, decoder->frame, info.size);
GST_BUFFER_PTS (outbuf) =
gst_clock_get_time (GST_ELEMENT_CLOCK (src)) -
GST_ELEMENT_CAST (src)->base_time;
gst_buffer_unmap (outbuf, &info);
return GST_FLOW_OK;
}
static void
kms_send_data_new_code (KmsSendData * self, guint64 ts, gchar * type,
gchar * symbol)
{
GstClockTime running_time, base_time, now;
GstClock *clock;
GstBuffer *buffer;
gchar *buffer_data;
GstFlowReturn ret;
if ((clock = GST_ELEMENT_CLOCK (self->priv->appsrc)) == NULL) {
GST_ERROR_OBJECT (GST_ELEMENT (self), "no clock, we can't sync");
return;
}
buffer_data =
g_strdup_printf ("Code detected in time %lu, type %s, symbol %s", ts,
type, symbol);
buffer = gst_buffer_new_wrapped (buffer_data, strlen (buffer_data));
base_time = GST_ELEMENT_CAST (self->priv->appsrc)->base_time;
now = gst_clock_get_time (clock);
running_time = now - base_time;
/* Live sources always timestamp their buffers with the running_time of the */
/* pipeline. This is needed to be able to match the timestamps of different */
/* live sources in order to synchronize them. */
GST_BUFFER_PTS (buffer) = running_time;
g_signal_emit_by_name (self->priv->appsrc, "push-buffer", buffer, &ret);
if (ret != GST_FLOW_OK) {
/* something wrong */
GST_WARNING ("Could not send buffer");
}
gst_buffer_unref (buffer);
}
static void
kms_dummy_src_feed_data_channel (GstElement * appsrc, guint unused_size,
gpointer data)
{
KmsDummySrc *self = KMS_DUMMY_SRC (data);
GstClockTime running_time, base_time, now;
GstClock *clock;
GstBuffer *buffer;
gchar *buffer_data;
GstFlowReturn ret;
if ((clock = GST_ELEMENT_CLOCK (appsrc)) == NULL) {
GST_ERROR_OBJECT (GST_ELEMENT (data), "no clock, we can't sync");
return;
}
buffer_data = g_strdup_printf ("Test buffer %d",
g_atomic_int_add (&self->priv->data_index, 1));
buffer = gst_buffer_new_wrapped (buffer_data, strlen (buffer_data));
base_time = GST_ELEMENT_CAST (appsrc)->base_time;
now = gst_clock_get_time (clock);
running_time = now - base_time;
/* Live sources always timestamp their buffers with the running_time of the */
/* pipeline. This is needed to be able to match the timestamps of different */
/* live sources in order to synchronize them. */
GST_BUFFER_PTS (buffer) = running_time;
g_signal_emit_by_name (appsrc, "push-buffer", buffer, &ret);
if (ret != GST_FLOW_OK) {
/* something wrong */
GST_WARNING ("Could not send buffer");
}
gst_buffer_unref (buffer);
}
static void
gst_decklink_video_src_start_streams (GstElement * element)
{
GstDecklinkVideoSrc *self = GST_DECKLINK_VIDEO_SRC_CAST (element);
HRESULT res;
if (self->input->video_enabled && (!self->input->audiosrc
|| self->input->audio_enabled)
&& (GST_STATE (self) == GST_STATE_PLAYING
|| GST_STATE_PENDING (self) == GST_STATE_PLAYING)) {
GST_DEBUG_OBJECT (self, "Starting streams");
res = self->input->input->StartStreams ();
if (res != S_OK) {
GST_ELEMENT_ERROR (self, STREAM, FAILED,
(NULL), ("Failed to start streams: 0x%08x", res));
return;
}
self->input->started = TRUE;
self->input->clock_restart = TRUE;
// Need to unlock to get the clock time
g_mutex_unlock (&self->input->lock);
// Current times of internal and external clock when we go to
// playing. We need this to convert the pipeline running time
// to the running time of the hardware
//
// We can't use the normal base time for the external clock
// because we might go to PLAYING later than the pipeline
self->internal_base_time = gst_clock_get_internal_time (self->input->clock);
self->external_base_time =
gst_clock_get_internal_time (GST_ELEMENT_CLOCK (self));
g_mutex_lock (&self->input->lock);
} else {
GST_DEBUG_OBJECT (self, "Not starting streams yet");
}
}
static GstFlowReturn
gst_v4l2src_create (GstPushSrc * src, GstBuffer ** buf)
{
GstV4l2Src *v4l2src = GST_V4L2SRC (src);
GstV4l2Object *obj = v4l2src->v4l2object;
GstV4l2BufferPool *pool = GST_V4L2_BUFFER_POOL_CAST (obj->pool);
GstFlowReturn ret;
GstClock *clock;
GstClockTime abs_time, base_time, timestamp, duration;
GstClockTime delay;
do {
ret = GST_BASE_SRC_CLASS (parent_class)->alloc (GST_BASE_SRC (src), 0,
obj->info.size, buf);
if (G_UNLIKELY (ret != GST_FLOW_OK))
goto alloc_failed;
ret = gst_v4l2_buffer_pool_process (pool, buf);
} while (ret == GST_V4L2_FLOW_CORRUPTED_BUFFER);
if (G_UNLIKELY (ret != GST_FLOW_OK))
goto error;
timestamp = GST_BUFFER_TIMESTAMP (*buf);
duration = obj->duration;
/* timestamps, LOCK to get clock and base time. */
/* FIXME: element clock and base_time is rarely changing */
GST_OBJECT_LOCK (v4l2src);
if ((clock = GST_ELEMENT_CLOCK (v4l2src))) {
/* we have a clock, get base time and ref clock */
base_time = GST_ELEMENT (v4l2src)->base_time;
gst_object_ref (clock);
} else {
/* no clock, can't set timestamps */
base_time = GST_CLOCK_TIME_NONE;
}
GST_OBJECT_UNLOCK (v4l2src);
/* sample pipeline clock */
if (clock) {
abs_time = gst_clock_get_time (clock);
gst_object_unref (clock);
} else {
abs_time = GST_CLOCK_TIME_NONE;
}
if (timestamp != GST_CLOCK_TIME_NONE) {
struct timespec now;
GstClockTime gstnow;
/* v4l2 specs say to use the system time although many drivers switched to
* the more desirable monotonic time. We first try to use the monotonic time
* and see how that goes */
clock_gettime (CLOCK_MONOTONIC, &now);
gstnow = GST_TIMESPEC_TO_TIME (now);
if (gstnow < timestamp && (timestamp - gstnow) > (10 * GST_SECOND)) {
GTimeVal now;
/* very large diff, fall back to system time */
g_get_current_time (&now);
gstnow = GST_TIMEVAL_TO_TIME (now);
}
if (gstnow > timestamp) {
delay = gstnow - timestamp;
} else {
delay = 0;
}
GST_DEBUG_OBJECT (v4l2src, "ts: %" GST_TIME_FORMAT " now %" GST_TIME_FORMAT
" delay %" GST_TIME_FORMAT, GST_TIME_ARGS (timestamp),
GST_TIME_ARGS (gstnow), GST_TIME_ARGS (delay));
} else {
/* we assume 1 frame latency otherwise */
if (GST_CLOCK_TIME_IS_VALID (duration))
delay = duration;
else
delay = 0;
}
/* set buffer metadata */
GST_BUFFER_OFFSET (*buf) = v4l2src->offset++;
GST_BUFFER_OFFSET_END (*buf) = v4l2src->offset;
if (G_LIKELY (abs_time != GST_CLOCK_TIME_NONE)) {
/* the time now is the time of the clock minus the base time */
timestamp = abs_time - base_time;
/* adjust for delay in the device */
if (timestamp > delay)
timestamp -= delay;
else
timestamp = 0;
} else {
timestamp = GST_CLOCK_TIME_NONE;
}
/* activate settings for next frame */
if (GST_CLOCK_TIME_IS_VALID (duration)) {
v4l2src->ctrl_time += duration;
} else {
/* this is not very good (as it should be the next timestamp),
* still good enough for linear fades (as long as it is not -1)
*/
v4l2src->ctrl_time = timestamp;
}
gst_object_sync_values (GST_OBJECT (src), v4l2src->ctrl_time);
GST_INFO_OBJECT (src, "sync to %" GST_TIME_FORMAT " out ts %" GST_TIME_FORMAT,
GST_TIME_ARGS (v4l2src->ctrl_time), GST_TIME_ARGS (timestamp));
GST_BUFFER_TIMESTAMP (*buf) = timestamp;
GST_BUFFER_DURATION (*buf) = duration;
return ret;
/* ERROR */
alloc_failed:
{
if (ret != GST_FLOW_FLUSHING)
GST_ELEMENT_ERROR (src, RESOURCE, NO_SPACE_LEFT,
("Failed to allocate a buffer"), (NULL));
return ret;
}
error:
{
if (ret == GST_V4L2_FLOW_LAST_BUFFER) {
GST_ELEMENT_ERROR (src, RESOURCE, FAILED,
("Driver returned a buffer with no payload, this most likely "
"indicate a bug in the driver."), (NULL));
ret = GST_FLOW_ERROR;
} else {
GST_DEBUG_OBJECT (src, "error processing buffer %d (%s)", ret,
gst_flow_get_name (ret));
}
return ret;
}
}
static void
gst_vader_set_property(GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstVader *filter;
g_return_if_fail(GST_IS_VADER(object));
filter = GST_VADER(object);
switch (prop_id) {
case PROP_THRESHOLD:
filter->threshold_level = (gint)(g_value_get_double(value) * 32768.0);
break;
case PROP_AUTO_THRESHOLD:
/* We are going to muck around with things... */
g_static_rec_mutex_lock(&filter->mtx);
filter->auto_threshold = g_value_get_boolean(value);
/* Setting this to TRUE re-initializes auto calibration. */
if (filter->auto_threshold) {
/* We have to be in silence mode to calibrate. */
filter->silent_prev = filter->silent;
filter->silent = TRUE;
/* Do "artifical" sil-speech or speech-sil transitions. */
if (filter->silent != filter->silent_prev) {
gst_vader_transition(filter, gst_clock_get_time(GST_ELEMENT_CLOCK(filter)));
}
/* Reset counters and such. */
filter->threshold_level = -1;
memset(filter->window, 0, sizeof(*filter->window) * VADER_WINDOW);
filter->silence_mean = 0;
filter->silence_stddev = 0;
filter->silence_frames = 0;
}
g_static_rec_mutex_unlock(&filter->mtx);
break;
case PROP_SILENT:
/* We are going to muck around with things... */
g_static_rec_mutex_lock(&filter->mtx);
filter->silent_prev = filter->silent;
filter->silent = g_value_get_boolean(value);
/* Do "artifical" sil-speech or speech-sil transitions. */
if (filter->silent != filter->silent_prev) {
gst_vader_transition(filter, gst_clock_get_time(GST_ELEMENT_CLOCK(filter)));
/* Also flush the voting window so we don't go right back into speech. */
memset(filter->window, 0, sizeof(*filter->window) * VADER_WINDOW);
}
g_static_rec_mutex_unlock(&filter->mtx);
break;
case PROP_RUN_LENGTH:
filter->threshold_length = g_value_get_uint64(value);
break;
case PROP_PRE_LENGTH:
filter->pre_length = g_value_get_uint64(value);
break;
case PROP_DUMPDIR:
g_free(filter->dumpdir);
filter->dumpdir = g_strdup(g_value_get_string(value));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID(object, prop_id, pspec);
break;
}
}
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;
}
}
/* messages are treated as warnings bcz those code should not be checked in.
* and no error handling will supported for same manner.
*/
gboolean
__util_gst_pad_probe(GstPad *pad, GstBuffer *buffer, gpointer u_data)
{
gint flag = (gint) u_data;
GstElement* parent = NULL;
gboolean ret = TRUE;
/* show name as default */
parent = (GstElement*)gst_object_get_parent(GST_OBJECT(pad));
debug_warning("PAD PROBE : %s:%s\n", GST_ELEMENT_NAME(parent), GST_PAD_NAME(pad));
/* show time stamp */
if ( flag & MM_PROBE_TIMESTAMP )
{
debug_warning("ts : %u:%02u:%02u.%09u\n", GST_TIME_ARGS(GST_BUFFER_TIMESTAMP(buffer)));
}
/* show buffer size */
if ( flag & MM_PROBE_BUFFERSIZE )
{
debug_warning("buffer size : %ud\n", GST_BUFFER_SIZE(buffer));
}
/* show buffer duration */
if ( flag & MM_PROBE_BUFFER_DURATION )
{
debug_warning("dur : %lld\n", GST_BUFFER_DURATION(buffer));
}
/* show buffer caps */
if ( flag & MM_PROBE_CAPS )
{
debug_warning("caps : %s\n", gst_caps_to_string(GST_BUFFER_CAPS(buffer)));
}
/* drop buffer if flag is on */
if ( flag & MM_PROBE_DROP_BUFFER )
{
debug_warning("dropping\n");
ret = FALSE;
}
/* show clock time */
if ( flag & MM_PROBE_CLOCK_TIME )
{
GstClock* clock = NULL;
GstClockTime now = GST_CLOCK_TIME_NONE;
clock = GST_ELEMENT_CLOCK ( parent );
if ( clock )
{
now = gst_clock_get_time( clock );
debug_warning("clock time : %" GST_TIME_FORMAT "\n", GST_TIME_ARGS( now ));
}
}
if ( parent )
gst_object_unref(parent);
return ret;
}
static GstFlowReturn
gst_v4l2src_create (GstPushSrc * src, GstBuffer ** buf)
{
GstV4l2Src *v4l2src = GST_V4L2SRC (src);
GstV4l2Object *obj = v4l2src->v4l2object;
GstV4l2BufferPool *pool = GST_V4L2_BUFFER_POOL_CAST (obj->pool);
GstFlowReturn ret;
GstClock *clock;
GstClockTime abs_time, base_time, timestamp, duration;
GstClockTime delay;
GstMessage *qos_msg;
do {
ret = GST_BASE_SRC_CLASS (parent_class)->alloc (GST_BASE_SRC (src), 0,
obj->info.size, buf);
if (G_UNLIKELY (ret != GST_FLOW_OK))
goto alloc_failed;
ret = gst_v4l2_buffer_pool_process (pool, buf);
} while (ret == GST_V4L2_FLOW_CORRUPTED_BUFFER);
if (G_UNLIKELY (ret != GST_FLOW_OK))
goto error;
timestamp = GST_BUFFER_TIMESTAMP (*buf);
duration = obj->duration;
/* timestamps, LOCK to get clock and base time. */
/* FIXME: element clock and base_time is rarely changing */
GST_OBJECT_LOCK (v4l2src);
if ((clock = GST_ELEMENT_CLOCK (v4l2src))) {
/* we have a clock, get base time and ref clock */
base_time = GST_ELEMENT (v4l2src)->base_time;
gst_object_ref (clock);
} else {
/* no clock, can't set timestamps */
base_time = GST_CLOCK_TIME_NONE;
}
GST_OBJECT_UNLOCK (v4l2src);
/* sample pipeline clock */
if (clock) {
abs_time = gst_clock_get_time (clock);
gst_object_unref (clock);
} else {
abs_time = GST_CLOCK_TIME_NONE;
}
retry:
if (!v4l2src->has_bad_timestamp && timestamp != GST_CLOCK_TIME_NONE) {
struct timespec now;
GstClockTime gstnow;
/* v4l2 specs say to use the system time although many drivers switched to
* the more desirable monotonic time. We first try to use the monotonic time
* and see how that goes */
clock_gettime (CLOCK_MONOTONIC, &now);
gstnow = GST_TIMESPEC_TO_TIME (now);
if (timestamp > gstnow || (gstnow - timestamp) > (10 * GST_SECOND)) {
GTimeVal now;
/* very large diff, fall back to system time */
g_get_current_time (&now);
gstnow = GST_TIMEVAL_TO_TIME (now);
}
/* Detect buggy drivers here, and stop using their timestamp. Failing any
* of these condition would imply a very buggy driver:
* - Timestamp in the future
* - Timestamp is going backward compare to last seen timestamp
* - Timestamp is jumping forward for less then a frame duration
* - Delay is bigger then the actual timestamp
* */
if (timestamp > gstnow) {
GST_WARNING_OBJECT (v4l2src,
"Timestamp in the future detected, ignoring driver timestamps");
v4l2src->has_bad_timestamp = TRUE;
goto retry;
}
if (v4l2src->last_timestamp > timestamp) {
GST_WARNING_OBJECT (v4l2src,
"Timestamp going backward, ignoring driver timestamps");
v4l2src->has_bad_timestamp = TRUE;
goto retry;
}
delay = gstnow - timestamp;
if (delay > timestamp) {
GST_WARNING_OBJECT (v4l2src,
"Timestamp does not correlate with any clock, ignoring driver timestamps");
v4l2src->has_bad_timestamp = TRUE;
goto retry;
}
/* Save last timestamp for sanity checks */
v4l2src->last_timestamp = timestamp;
GST_DEBUG_OBJECT (v4l2src, "ts: %" GST_TIME_FORMAT " now %" GST_TIME_FORMAT
" delay %" GST_TIME_FORMAT, GST_TIME_ARGS (timestamp),
GST_TIME_ARGS (gstnow), GST_TIME_ARGS (delay));
} else {
/* we assume 1 frame latency otherwise */
if (GST_CLOCK_TIME_IS_VALID (duration))
delay = duration;
else
delay = 0;
}
/* set buffer metadata */
if (G_LIKELY (abs_time != GST_CLOCK_TIME_NONE)) {
/* the time now is the time of the clock minus the base time */
timestamp = abs_time - base_time;
/* adjust for delay in the device */
if (timestamp > delay)
timestamp -= delay;
else
timestamp = 0;
} else {
timestamp = GST_CLOCK_TIME_NONE;
}
/* activate settings for next frame */
if (GST_CLOCK_TIME_IS_VALID (duration)) {
v4l2src->ctrl_time += duration;
} else {
/* this is not very good (as it should be the next timestamp),
* still good enough for linear fades (as long as it is not -1)
*/
v4l2src->ctrl_time = timestamp;
}
gst_object_sync_values (GST_OBJECT (src), v4l2src->ctrl_time);
GST_INFO_OBJECT (src, "sync to %" GST_TIME_FORMAT " out ts %" GST_TIME_FORMAT,
GST_TIME_ARGS (v4l2src->ctrl_time), GST_TIME_ARGS (timestamp));
/* use generated offset values only if there are not already valid ones
* set by the v4l2 device */
if (!GST_BUFFER_OFFSET_IS_VALID (*buf) || !GST_BUFFER_OFFSET_END_IS_VALID (*buf)) {
GST_BUFFER_OFFSET (*buf) = v4l2src->offset++;
GST_BUFFER_OFFSET_END (*buf) = v4l2src->offset;
} else {
/* adjust raw v4l2 device sequence, will restart at null in case of renegotiation
* (streamoff/streamon) */
GST_BUFFER_OFFSET (*buf) += v4l2src->renegotiation_adjust;
GST_BUFFER_OFFSET_END (*buf) += v4l2src->renegotiation_adjust;
/* check for frame loss with given (from v4l2 device) buffer offset */
if ((v4l2src->offset != 0) && (GST_BUFFER_OFFSET (*buf) != (v4l2src->offset + 1))) {
guint64 lost_frame_count = GST_BUFFER_OFFSET (*buf) - v4l2src->offset - 1;
GST_WARNING_OBJECT (v4l2src,
"lost frames detected: count = %" G_GUINT64_FORMAT " - ts: %" GST_TIME_FORMAT,
lost_frame_count, GST_TIME_ARGS (timestamp));
qos_msg = gst_message_new_qos (GST_OBJECT_CAST (v4l2src), TRUE,
GST_CLOCK_TIME_NONE, GST_CLOCK_TIME_NONE, timestamp,
GST_CLOCK_TIME_IS_VALID (duration) ? lost_frame_count * duration : GST_CLOCK_TIME_NONE);
gst_element_post_message (GST_ELEMENT_CAST (v4l2src), qos_msg);
}
v4l2src->offset = GST_BUFFER_OFFSET (*buf);
}
GST_BUFFER_TIMESTAMP (*buf) = timestamp;
GST_BUFFER_DURATION (*buf) = duration;
return ret;
/* ERROR */
alloc_failed:
{
if (ret != GST_FLOW_FLUSHING)
GST_ELEMENT_ERROR (src, RESOURCE, NO_SPACE_LEFT,
("Failed to allocate a buffer"), (NULL));
return ret;
}
error:
{
if (ret == GST_V4L2_FLOW_LAST_BUFFER) {
GST_ELEMENT_ERROR (src, RESOURCE, FAILED,
("Driver returned a buffer with no payload, this most likely "
"indicate a bug in the driver."), (NULL));
ret = GST_FLOW_ERROR;
} else {
GST_DEBUG_OBJECT (src, "error processing buffer %d (%s)", ret,
gst_flow_get_name (ret));
}
return ret;
}
}
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;
}
static gboolean
gst_ks_video_src_timestamp_buffer (GstKsVideoSrc * self, GstBuffer * buf,
GstClockTime presentation_time)
{
GstKsVideoSrcPrivate *priv = GST_KS_VIDEO_SRC_GET_PRIVATE (self);
GstClockTime duration;
GstClock *clock;
GstClockTime timestamp;
duration = gst_ks_video_device_get_duration (priv->device);
GST_OBJECT_LOCK (self);
clock = GST_ELEMENT_CLOCK (self);
if (clock != NULL) {
gst_object_ref (clock);
timestamp = GST_ELEMENT (self)->base_time;
if (GST_CLOCK_TIME_IS_VALID (presentation_time)) {
if (presentation_time > GST_ELEMENT (self)->base_time)
presentation_time -= GST_ELEMENT (self)->base_time;
else
presentation_time = 0;
}
} else {
timestamp = GST_CLOCK_TIME_NONE;
}
GST_OBJECT_UNLOCK (self);
if (clock != NULL) {
/* The time according to the current clock */
timestamp = gst_clock_get_time (clock) - timestamp;
if (timestamp > duration)
timestamp -= duration;
else
timestamp = 0;
if (GST_CLOCK_TIME_IS_VALID (presentation_time)) {
/*
* We don't use this for anything yet, need to ponder how to deal
* with pins that use an internal clock and timestamp from 0.
*/
GstClockTimeDiff diff = GST_CLOCK_DIFF (presentation_time, timestamp);
GST_DEBUG_OBJECT (self, "diff between gst and driver timestamp: %"
G_GINT64_FORMAT, diff);
}
gst_object_unref (clock);
clock = NULL;
/* Unless it's the first frame, align the current timestamp on a multiple
* of duration since the previous */
if (GST_CLOCK_TIME_IS_VALID (priv->prev_ts)) {
GstClockTime delta;
guint delta_remainder, delta_offset;
/* REVISIT: I've seen this happen with the GstSystemClock on Windows,
* scary... */
if (timestamp < priv->prev_ts) {
GST_INFO_OBJECT (self, "clock is ticking backwards");
return FALSE;
}
/* Round to a duration boundary */
delta = timestamp - priv->prev_ts;
delta_remainder = delta % duration;
if (delta_remainder < duration / 3)
timestamp -= delta_remainder;
else
timestamp += duration - delta_remainder;
/* How many frames are we off then? */
delta = timestamp - priv->prev_ts;
delta_offset = delta / duration;
if (delta_offset == 1) /* perfect */
GST_BUFFER_FLAG_UNSET (buf, GST_BUFFER_FLAG_DISCONT);
else if (delta_offset > 1) {
guint lost = delta_offset - 1;
GST_INFO_OBJECT (self, "lost %d frame%s, setting discont flag",
lost, (lost > 1) ? "s" : "");
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
} else if (delta_offset == 0) { /* overproduction, skip this frame */
GST_INFO_OBJECT (self, "skipping frame");
return FALSE;
}
priv->offset += delta_offset;
}
priv->prev_ts = timestamp;
}
GST_BUFFER_OFFSET (buf) = priv->offset;
GST_BUFFER_OFFSET_END (buf) = GST_BUFFER_OFFSET (buf) + 1;
GST_BUFFER_TIMESTAMP (buf) = timestamp;
GST_BUFFER_DURATION (buf) = duration;
return TRUE;
}
static GstStateChangeReturn
gst_decklink_video_sink_change_state (GstElement * element,
GstStateChange transition)
{
GstDecklinkVideoSink *self = GST_DECKLINK_VIDEO_SINK_CAST (element);
GstStateChangeReturn ret;
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
g_mutex_lock (&self->output->lock);
self->output->clock_start_time = GST_CLOCK_TIME_NONE;
self->output->clock_last_time = 0;
self->output->clock_offset = 0;
g_mutex_unlock (&self->output->lock);
gst_element_post_message (element,
gst_message_new_clock_provide (GST_OBJECT_CAST (element),
self->output->clock, TRUE));
self->last_render_time = GST_CLOCK_TIME_NONE;
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:{
GstClock *clock, *audio_clock;
clock = gst_element_get_clock (GST_ELEMENT_CAST (self));
audio_clock = gst_decklink_output_get_audio_clock (self->output);
if (clock && clock != self->output->clock && clock != audio_clock) {
gst_clock_set_master (self->output->clock, clock);
}
if (clock)
gst_object_unref (clock);
if (audio_clock)
gst_object_unref (audio_clock);
break;
}
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
if (ret == GST_STATE_CHANGE_FAILURE)
return ret;
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_element_post_message (element,
gst_message_new_clock_lost (GST_OBJECT_CAST (element),
self->output->clock));
gst_clock_set_master (self->output->clock, NULL);
g_mutex_lock (&self->output->lock);
self->output->clock_start_time = GST_CLOCK_TIME_NONE;
self->output->clock_last_time = 0;
self->output->clock_offset = 0;
g_mutex_unlock (&self->output->lock);
break;
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:{
GstClockTime start_time;
HRESULT res;
// FIXME: start time is the same for the complete pipeline,
// but what we need here is the start time of this element!
start_time = gst_element_get_base_time (element);
if (start_time != GST_CLOCK_TIME_NONE)
start_time = gst_clock_get_time (GST_ELEMENT_CLOCK (self)) - start_time;
// FIXME: This will probably not work
if (start_time == GST_CLOCK_TIME_NONE)
start_time = 0;
convert_to_internal_clock (self, &start_time, NULL);
// The start time is now the running time when we stopped
// playback
GST_DEBUG_OBJECT (self,
"Stopping scheduled playback at %" GST_TIME_FORMAT,
GST_TIME_ARGS (start_time));
g_mutex_lock (&self->output->lock);
self->output->started = FALSE;
g_mutex_unlock (&self->output->lock);
res =
self->output->output->StopScheduledPlayback (start_time, 0,
GST_SECOND);
if (res != S_OK) {
GST_ELEMENT_ERROR (self, STREAM, FAILED,
(NULL), ("Failed to stop scheduled playback: 0x%08x", res));
ret = GST_STATE_CHANGE_FAILURE;
}
self->internal_base_time = GST_CLOCK_TIME_NONE;
self->external_base_time = GST_CLOCK_TIME_NONE;
break;
}
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:{
g_mutex_lock (&self->output->lock);
if (self->output->start_scheduled_playback)
self->output->start_scheduled_playback (self->output->videosink);
g_mutex_unlock (&self->output->lock);
break;
}
default:
break;
}
return ret;
}
static 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 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);
}
static void
gst_frame_store_task (GstPad *pad)
{
GstFrameStore *fs;
GstBuffer *buffer;
GstEvent *event = NULL;
fs = GST_FRAME_STORE (gst_pad_get_parent (pad));
GST_DEBUG("task");
g_mutex_lock (fs->lock);
while(1) {
if (fs->stepping == FALSE || (fs->frame_number != fs->pushed_frame_number)) {
buffer = gst_frame_store_get_frame (fs, fs->frame_number);
}
if (buffer) break;
g_cond_wait (fs->cond, fs->lock);
}
if (fs->need_newsegment) {
GstClock *clock;
GstClockTime now;
GstClockTime stream_time;
clock = GST_ELEMENT_CLOCK (fs);
if (clock == NULL) {
now = 0;
stream_time = 0;
} else {
now = gst_clock_get_time (GST_ELEMENT_CLOCK (fs));
stream_time = now - GST_ELEMENT(fs)->base_time;
}
GST_ERROR("now %lld buffer %lld stream_time %lld",
now, GST_BUFFER_TIMESTAMP(buffer), stream_time);
stream_time = GST_SECOND*10;
event = gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_TIME,
GST_BUFFER_TIMESTAMP(buffer), -1, stream_time);
fs->need_newsegment = FALSE;
}
if (fs->stepping) {
buffer = gst_buffer_make_metadata_writable (buffer);
GST_BUFFER_TIMESTAMP(buffer) = -1;
GST_BUFFER_DURATION(buffer) = -1;
}
fs->pushed_frame_number = fs->frame_number;
if (!fs->stepping) {
fs->frame_number++;
}
if (fs->frame_number + 1 >= fs->range_offset + fs->range_size) {
gst_frame_store_advance (fs);
}
g_mutex_unlock (fs->lock);
if (event) {
gst_pad_push_event (fs->srcpad, event);
}
gst_pad_push (fs->srcpad, buffer);
GST_DEBUG("task done");
gst_object_unref (fs);
}
static GstFlowReturn
gst_base_audio_src_create (GstBaseSrc * bsrc, guint64 offset, guint length,
GstBuffer ** outbuf)
{
GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (bsrc);
GstBuffer *buf;
guchar *data;
guint samples, total_samples;
guint64 sample;
gint bps;
GstRingBuffer *ringbuffer;
GstRingBufferSpec *spec;
guint read;
GstClockTime timestamp, duration;
GstClock *clock;
ringbuffer = src->ringbuffer;
spec = &ringbuffer->spec;
if (G_UNLIKELY (!gst_ring_buffer_is_acquired (ringbuffer)))
goto wrong_state;
bps = spec->bytes_per_sample;
if ((length == 0 && bsrc->blocksize == 0) || length == -1)
/* no length given, use the default segment size */
length = spec->segsize;
else
/* make sure we round down to an integral number of samples */
length -= length % bps;
/* figure out the offset in the ringbuffer */
if (G_UNLIKELY (offset != -1)) {
sample = offset / bps;
/* if a specific offset was given it must be the next sequential
* offset we expect or we fail for now. */
if (src->next_sample != -1 && sample != src->next_sample)
goto wrong_offset;
} else {
/* calculate the sequentially next sample we need to read. This can jump and
* create a DISCONT. */
sample = gst_base_audio_src_get_offset (src);
}
GST_DEBUG_OBJECT (src, "reading from sample %" G_GUINT64_FORMAT, sample);
/* get the number of samples to read */
total_samples = samples = length / bps;
/* FIXME, using a bufferpool would be nice here */
buf = gst_buffer_new_and_alloc (length);
data = GST_BUFFER_DATA (buf);
do {
read = gst_ring_buffer_read (ringbuffer, sample, data, samples);
GST_DEBUG_OBJECT (src, "read %u of %u", read, samples);
/* if we read all, we're done */
if (read == samples)
break;
/* else something interrupted us and we wait for playing again. */
GST_DEBUG_OBJECT (src, "wait playing");
if (gst_base_src_wait_playing (bsrc) != GST_FLOW_OK)
goto stopped;
GST_DEBUG_OBJECT (src, "continue playing");
/* read next samples */
sample += read;
samples -= read;
data += read * bps;
} while (TRUE);
/* mark discontinuity if needed */
if (G_UNLIKELY (sample != src->next_sample) && src->next_sample != -1) {
GST_WARNING_OBJECT (src,
"create DISCONT of %" G_GUINT64_FORMAT " samples at sample %"
G_GUINT64_FORMAT, sample - src->next_sample, sample);
GST_ELEMENT_WARNING (src, CORE, CLOCK,
(_("Can't record audio fast enough")),
("dropped %" G_GUINT64_FORMAT " samples", sample - src->next_sample));
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
}
src->next_sample = sample + samples;
/* get the normal timestamp to get the duration. */
timestamp = gst_util_uint64_scale_int (sample, GST_SECOND, spec->rate);
duration = gst_util_uint64_scale_int (src->next_sample, GST_SECOND,
spec->rate) - timestamp;
GST_OBJECT_LOCK (src);
if (!(clock = GST_ELEMENT_CLOCK (src)))
goto no_sync;
if (clock != src->clock) {
/* we are slaved, check how to handle this */
switch (src->priv->slave_method) {
case GST_BASE_AUDIO_SRC_SLAVE_RESAMPLE:
/* not implemented, use skew algorithm. This algorithm should
* work on the readout pointer and produces more or less samples based
* on the clock drift */
case GST_BASE_AUDIO_SRC_SLAVE_SKEW:
{
GstClockTime running_time;
GstClockTime base_time;
GstClockTime current_time;
guint64 running_time_sample;
gint running_time_segment;
gint current_segment;
gint segment_skew;
gint sps;
/* samples per segment */
sps = ringbuffer->samples_per_seg;
/* get the current time */
current_time = gst_clock_get_time (clock);
/* get the basetime */
base_time = GST_ELEMENT_CAST (src)->base_time;
/* get the running_time */
running_time = current_time - base_time;
/* the running_time converted to a sample (relative to the ringbuffer) */
running_time_sample =
gst_util_uint64_scale_int (running_time, spec->rate, GST_SECOND);
/* the segmentnr corrensponding to running_time, round down */
running_time_segment = running_time_sample / sps;
/* the segment currently read from the ringbuffer */
current_segment = sample / sps;
/* the skew we have between running_time and the ringbuffertime */
segment_skew = running_time_segment - current_segment;
GST_DEBUG_OBJECT (bsrc, "\n running_time = %" GST_TIME_FORMAT
"\n timestamp = %" GST_TIME_FORMAT
"\n running_time_segment = %d"
"\n current_segment = %d"
"\n segment_skew = %d",
GST_TIME_ARGS (running_time),
GST_TIME_ARGS (timestamp),
running_time_segment, current_segment, segment_skew);
/* Resync the ringbuffer if:
* 1. We get one segment into the future.
* This is clearly a lie, because we can't
* possibly have a buffer with timestamp 1 at
* time 0. (unless it has time-travelled...)
*
* 2. We are more than the length of the ringbuffer behind.
* The length of the ringbuffer then gets to dictate
* the threshold for what is concidered "too late"
*
* 3. If this is our first buffer.
* We know that we should catch up to running_time
* the first time we are ran.
*/
if ((segment_skew < 0) ||
(segment_skew >= ringbuffer->spec.segtotal) ||
(current_segment == 0)) {
gint segments_written;
gint first_segment;
gint last_segment;
gint new_last_segment;
gint segment_diff;
gint new_first_segment;
guint64 new_sample;
/* we are going to say that the last segment was captured at the current time
(running_time), minus one segment of creation-latency in the ringbuffer.
This can be thought of as: The segment arrived in the ringbuffer at time X, and
that means it was created at time X - (one segment). */
new_last_segment = running_time_segment - 1;
/* for better readablity */
first_segment = current_segment;
/* get the amount of segments written from the device by now */
segments_written = g_atomic_int_get (&ringbuffer->segdone);
/* subtract the base to segments_written to get the number of the
last written segment in the ringbuffer (one segment written = segment 0) */
last_segment = segments_written - ringbuffer->segbase - 1;
/* we see how many segments the ringbuffer was timeshifted */
segment_diff = new_last_segment - last_segment;
/* we move the first segment an equal amount */
new_first_segment = first_segment + segment_diff;
/* and we also move the segmentbase the same amount */
ringbuffer->segbase -= segment_diff;
/* we calculate the new sample value */
new_sample = ((guint64) new_first_segment) * sps;
/* and get the relative time to this -> our new timestamp */
timestamp =
gst_util_uint64_scale_int (new_sample, GST_SECOND, spec->rate);
/* we update the next sample accordingly */
src->next_sample = new_sample + samples;
GST_DEBUG_OBJECT (bsrc,
"Timeshifted the ringbuffer with %d segments: "
"Updating the timestamp to %" GST_TIME_FORMAT ", "
"and src->next_sample to %" G_GUINT64_FORMAT, segment_diff,
GST_TIME_ARGS (timestamp), src->next_sample);
}
break;
}
case GST_BASE_AUDIO_SRC_SLAVE_RETIMESTAMP:
{
GstClockTime base_time, latency;
/* We are slaved to another clock, take running time of the pipeline clock and
* timestamp against it. Somebody else in the pipeline should figure out the
* clock drift. We keep the duration we calculated above. */
timestamp = gst_clock_get_time (clock);
base_time = GST_ELEMENT_CAST (src)->base_time;
if (timestamp > base_time)
timestamp -= base_time;
else
timestamp = 0;
/* subtract latency */
latency =
gst_util_uint64_scale_int (total_samples, GST_SECOND, spec->rate);
if (timestamp > latency)
timestamp -= latency;
else
timestamp = 0;
}
case GST_BASE_AUDIO_SRC_SLAVE_NONE:
break;
}
} else {
GstClockTime base_time;
/* we are not slaved, subtract base_time */
base_time = GST_ELEMENT_CAST (src)->base_time;
if (timestamp > base_time)
timestamp -= base_time;
else
timestamp = 0;
}
no_sync:
GST_OBJECT_UNLOCK (src);
GST_BUFFER_TIMESTAMP (buf) = timestamp;
GST_BUFFER_DURATION (buf) = duration;
GST_BUFFER_OFFSET (buf) = sample;
GST_BUFFER_OFFSET_END (buf) = sample + samples;
*outbuf = buf;
return GST_FLOW_OK;
/* ERRORS */
wrong_state:
{
GST_DEBUG_OBJECT (src, "ringbuffer in wrong state");
return GST_FLOW_WRONG_STATE;
}
wrong_offset:
{
GST_ELEMENT_ERROR (src, RESOURCE, SEEK,
(NULL), ("resource can only be operated on sequentially but offset %"
G_GUINT64_FORMAT " was given", offset));
return GST_FLOW_ERROR;
}
stopped:
{
gst_buffer_unref (buf);
GST_DEBUG_OBJECT (src, "ringbuffer stopped");
return GST_FLOW_WRONG_STATE;
}
}
static GstFlowReturn
create(GstPushSrc *base, GstBuffer **buf)
{
GstTICaptureSrc *src = (GstTICaptureSrc *)base;
Buffer_Handle hDstBuf;
GstBuffer *outBuf;
gint ret = GST_FLOW_OK;
BufferGfx_Attrs gfxAttrs = BufferGfx_Attrs_DEFAULT;
Int32 width, height;
GST_LOG("create begin");
/* create capture device */
if (src->hCapture == NULL) {
/* set framerate based on video standard */
switch(dmai_video_std(src->video_standard)) {
case VideoStd_D1_NTSC:
gst_value_set_fraction(&src->framerate,30000,1001);
break;
case VideoStd_D1_PAL:
gst_value_set_fraction(&src->framerate,25,1);
break;
default:
gst_value_set_fraction(&src->framerate,30,1);
break;
}
/* set width & height based on video standard */
src->cAttrs.videoStd = dmai_video_std(src->video_standard);
VideoStd_getResolution(src->cAttrs.videoStd, &width, &height);
width = 720;
height = 576;
GST_WARNING("force video size to %dx%d", src->width, src->height);
src->width = width;
src->height = height;
gfxAttrs.dim.height = src->height;
gfxAttrs.dim.width = src->width;
src->cAttrs.captureDimension = &gfxAttrs.dim;
if (!capture_create(src))
return GST_FLOW_ERROR;
}
/* Get buffer from driver */
if (Capture_get(src->hCapture, &hDstBuf)) {
GST_ELEMENT_ERROR(src, RESOURCE, FAILED,
("Failed to allocate buffer\n"), (NULL));
return GST_FLOW_ERROR;
}
/* Create a DMAI transport buffer object to carry a DMAI buffer to
* the source pad. The transport buffer knows how to release the
* buffer for re-use in this element when the source pad calls
* gst_buffer_unref().
*/
outBuf = gst_tidmaibuffertransport_new(hDstBuf, src->hBufTab, capture_buffer_finalize, (void*)src);
gst_buffer_set_data(outBuf, GST_BUFFER_DATA(outBuf), Buffer_getSize(hDstBuf));
*buf = outBuf;
/* set buffer metadata */
if (G_LIKELY (ret == GST_FLOW_OK && *buf)) {
GstClock *clock;
GstClockTime timestamp;
GST_BUFFER_OFFSET (*buf) = src->offset++;
GST_BUFFER_OFFSET_END (*buf) = src->offset;
/* timestamps, LOCK to get clock and base time. */
GST_OBJECT_LOCK (src);
if ((clock = GST_ELEMENT_CLOCK (src))) {
/* we have a clock, get base time and ref clock */
timestamp = GST_ELEMENT (src)->base_time;
gst_object_ref (clock);
} else {
/* no clock, can't set timestamps */
timestamp = GST_CLOCK_TIME_NONE;
}
GST_OBJECT_UNLOCK (src);
if (G_LIKELY (clock)) {
/* the time now is the time of the clock minus the base time */
timestamp = gst_clock_get_time (clock) - timestamp;
gst_object_unref (clock);
/* if we have a framerate adjust timestamp for frame latency */
if (GST_CLOCK_TIME_IS_VALID (src->duration)) {
if (timestamp > src->duration)
timestamp -= src->duration;
else
timestamp = 0;
}
}
/* FIXME: use the timestamp from the buffer itself! */
GST_BUFFER_TIMESTAMP (*buf) = timestamp;
GST_BUFFER_DURATION (*buf) = src->duration;
}
/* Create caps for buffer */
GstCaps *mycaps;
GstStructure *structure;
mycaps = gst_caps_new_empty();
if (src->cAttrs.colorSpace == ColorSpace_UYVY) {
structure = gst_structure_new( "video/x-raw-yuv",
"format", GST_TYPE_FOURCC, GST_MAKE_FOURCC('U', 'Y', 'V', 'Y'),
"framerate", GST_TYPE_FRACTION,
gst_value_get_fraction_numerator(&src->framerate),
gst_value_get_fraction_denominator(&src->framerate),
"width", G_TYPE_INT, src->width,
"height", G_TYPE_INT, src->height,
(gchar*) NULL);
}
else if(src->cAttrs.colorSpace == ColorSpace_YUV420PSEMI) {
structure = gst_structure_new( "video/x-raw-yuv",
"format", GST_TYPE_FOURCC, GST_MAKE_FOURCC('N', 'V', '1', '2'),
"framerate", GST_TYPE_FRACTION,
gst_value_get_fraction_numerator(&src->framerate),
gst_value_get_fraction_denominator(&src->framerate),
"width", G_TYPE_INT, src->width,
"height", G_TYPE_INT, src->height,
(gchar*) NULL);
}
else {
GST_ERROR("unsupported fourcc\n");
return FALSE;
}
gst_caps_append_structure(mycaps, gst_structure_copy (structure));
gst_structure_free(structure);
gst_buffer_set_caps(*buf, mycaps);
gst_caps_unref(mycaps);
{
static int fn;
fn++;
GST_INFO("capture frame %d", fn);
}
GST_LOG("create end");
return GST_FLOW_OK;
}
GstFlowReturn
gst_euresys_fill (GstPushSrc * src, GstBuffer * buf)
{
GstEuresys *euresys = GST_EURESYS (src);
MCSTATUS status = 0;
MCSIGNALINFO siginfo;
MCHANDLE hSurface;
int *pImage;
INT32 timeCode;
INT64 timeStamp;
int newsize;
int dropped_frame_count;
GstMapInfo minfo;
/* Start acquisition */
if (!euresys->acq_started) {
status =
McSetParamInt (euresys->hChannel, MC_ChannelState,
MC_ChannelState_ACTIVE);
if (status != MC_OK) {
GST_ELEMENT_ERROR (euresys, RESOURCE, FAILED,
(("Failed to set channel state to ACTIVE.")), (NULL));
return GST_FLOW_ERROR;
}
euresys->acq_started = TRUE;
}
/* Wait for next surface (frame) */
while (TRUE) {
/* Wait up to 5000 msecs for a signal */
status = McWaitSignal (euresys->hChannel, MC_SIG_ANY, 5000, &siginfo);
if (status == MC_TIMEOUT) {
GST_ELEMENT_ERROR (src, RESOURCE, FAILED,
(("Timeout waiting for signal.")), (("Timeout waiting for signal.")));
return GST_FLOW_ERROR;
} else if (siginfo.Signal == MC_SIG_ACQUISITION_FAILURE) {
GST_ELEMENT_ERROR (src, RESOURCE, FAILED,
(("Acquisition failure due to timeout.")), (NULL));
return GST_FLOW_ERROR;
} else if (siginfo.Signal == MC_SIG_SURFACE_PROCESSING) {
break;
} else {
continue;
}
}
/* Get pointer to image data and other info */
hSurface = (MCHANDLE) siginfo.SignalInfo;
/* "number of bytes actually written into the surface" */
status = McGetParamInt (hSurface, MC_FillCount, &newsize);
/* "Internal numbering of surface during acquisition sequence" (zero-based) */
status |= McGetParamInt (hSurface, MC_TimeCode, &timeCode);
/* "number of microseconds elapsed since midnight (00:00:00),
* January 1, 1970, coordinated universal time (UTC), according
* to the system clock when the surface is filled" */
status |= McGetParamInt64 (hSurface, MC_TimeStamp_us, &timeStamp);
status |= McGetParamPtr (hSurface, MC_SurfaceAddr, (PVOID *) & pImage);
if (G_UNLIKELY (status != MC_OK)) {
GST_ELEMENT_ERROR (euresys, RESOURCE, FAILED,
(("Failed to read surface parameter.")), (NULL));
return GST_FLOW_ERROR;
}
GST_INFO ("Got surface #%05d", timeCode);
dropped_frame_count = timeCode - (euresys->last_time_code + 1);
if (dropped_frame_count != 0) {
euresys->dropped_frame_count += dropped_frame_count;
GST_WARNING ("Dropped %d frames (%d total)", dropped_frame_count,
euresys->dropped_frame_count);
/* TODO: emit message here about dropped frames */
}
euresys->last_time_code = timeCode;
/* Copy image to buffer from surface */
gst_buffer_map (buf, &minfo, GST_MAP_WRITE);
/* TODO: fix strides? */
g_assert (minfo.size == newsize);
memcpy (minfo.data, pImage, newsize);
gst_buffer_unmap (buf, &minfo);
/* TODO: set buffer timestamp based on MC_TimeStamp_us */
GST_BUFFER_TIMESTAMP (buf) =
gst_clock_get_time (GST_ELEMENT_CLOCK (src)) -
GST_ELEMENT_CAST (src)->base_time;
/* Done processing surface, release control */
McSetParamInt (hSurface, MC_SurfaceState, MC_SurfaceState_FREE);
return GST_FLOW_OK;
}
static gboolean
gst_ks_video_src_open_device (GstKsVideoSrc * self)
{
GstKsVideoSrcPrivate *priv = GST_KS_VIDEO_SRC_GET_PRIVATE (self);
GstKsVideoDevice *device = NULL;
GList *devices, *cur;
g_assert (priv->device == NULL);
devices = ks_enumerate_devices (&KSCATEGORY_VIDEO);
if (devices == NULL)
goto error_no_devices;
devices = ks_video_device_list_sort_cameras_first (devices);
for (cur = devices; cur != NULL; cur = cur->next) {
KsDeviceEntry *entry = cur->data;
GST_DEBUG_OBJECT (self, "device %d: name='%s' path='%s'",
entry->index, entry->name, entry->path);
}
for (cur = devices; cur != NULL && device == NULL; cur = cur->next) {
KsDeviceEntry *entry = cur->data;
gboolean match;
if (priv->device_path != NULL) {
match = g_strcasecmp (entry->path, priv->device_path) == 0;
} else if (priv->device_name != NULL) {
match = g_strcasecmp (entry->name, priv->device_name) == 0;
} else if (priv->device_index >= 0) {
match = entry->index == priv->device_index;
} else {
match = TRUE; /* pick the first entry */
}
if (match) {
priv->ksclock = g_object_new (GST_TYPE_KS_CLOCK, NULL);
if (priv->ksclock != NULL && gst_ks_clock_open (priv->ksclock)) {
GstClock *clock = GST_ELEMENT_CLOCK (self);
if (clock != NULL)
gst_ks_clock_provide_master_clock (priv->ksclock, clock);
} else {
GST_WARNING_OBJECT (self, "failed to create/open KsClock");
g_object_unref (priv->ksclock);
priv->ksclock = NULL;
}
device = gst_ks_video_device_new (entry->path, priv->ksclock,
gst_ks_video_src_alloc_buffer, self);
}
ks_device_entry_free (entry);
}
g_list_free (devices);
if (device == NULL)
goto error_no_match;
if (!gst_ks_video_device_open (device))
goto error_open;
priv->device = device;
return TRUE;
/* ERRORS */
error_no_devices:
{
GST_ELEMENT_ERROR (self, RESOURCE, NOT_FOUND,
("No video capture devices found"), (NULL));
return FALSE;
}
error_no_match:
{
if (priv->device_path != NULL) {
GST_ELEMENT_ERROR (self, RESOURCE, NOT_FOUND,
("Specified video capture device with path '%s' not found",
priv->device_path), (NULL));
} else if (priv->device_name != NULL) {
GST_ELEMENT_ERROR (self, RESOURCE, NOT_FOUND,
("Specified video capture device with name '%s' not found",
priv->device_name), (NULL));
} else {
GST_ELEMENT_ERROR (self, RESOURCE, NOT_FOUND,
("Specified video capture device with index %d not found",
priv->device_index), (NULL));
}
return FALSE;
}
error_open:
{
GST_ELEMENT_ERROR (self, RESOURCE, OPEN_READ,
("Failed to open device"), (NULL));
g_object_unref (device);
return FALSE;
}
}
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 void
gst_decklink_video_sink_start_scheduled_playback (GstElement * element)
{
GstDecklinkVideoSink *self = GST_DECKLINK_VIDEO_SINK_CAST (element);
GstClockTime start_time;
HRESULT res;
bool active;
if (self->output->video_enabled && (!self->output->audiosink
|| self->output->audio_enabled)
&& (GST_STATE (self) == GST_STATE_PLAYING
|| GST_STATE_PENDING (self) == GST_STATE_PLAYING)) {
// Need to unlock to get the clock time
g_mutex_unlock (&self->output->lock);
// FIXME: start time is the same for the complete pipeline,
// but what we need here is the start time of this element!
start_time = gst_element_get_base_time (element);
if (start_time != GST_CLOCK_TIME_NONE)
start_time = gst_clock_get_time (GST_ELEMENT_CLOCK (self)) - start_time;
// FIXME: This will probably not work
if (start_time == GST_CLOCK_TIME_NONE)
start_time = 0;
// Current times of internal and external clock when we go to
// playing. We need this to convert the pipeline running time
// to the running time of the hardware
//
// We can't use the normal base time for the external clock
// because we might go to PLAYING later than the pipeline
self->internal_base_time =
gst_clock_get_internal_time (self->output->clock);
self->external_base_time =
gst_clock_get_internal_time (GST_ELEMENT_CLOCK (self));
convert_to_internal_clock (self, &start_time, NULL);
g_mutex_lock (&self->output->lock);
// Check if someone else started in the meantime
if (self->output->started)
return;
active = false;
self->output->output->IsScheduledPlaybackRunning (&active);
if (active) {
GST_DEBUG_OBJECT (self, "Stopping scheduled playback");
self->output->started = FALSE;
res = self->output->output->StopScheduledPlayback (0, 0, 0);
if (res != S_OK) {
GST_ELEMENT_ERROR (self, STREAM, FAILED,
(NULL), ("Failed to stop scheduled playback: 0x%08x", res));
return;
}
}
GST_DEBUG_OBJECT (self,
"Starting scheduled playback at %" GST_TIME_FORMAT,
GST_TIME_ARGS (start_time));
res =
self->output->output->StartScheduledPlayback (start_time,
GST_SECOND, 1.0);
if (res != S_OK) {
GST_ELEMENT_ERROR (self, STREAM, FAILED,
(NULL), ("Failed to start scheduled playback: 0x%08x", res));
return;
}
self->output->started = TRUE;
self->output->clock_restart = TRUE;
} else {
GST_DEBUG_OBJECT (self, "Not starting scheduled playback yet");
}
}
/**
* 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;
}
}
