/**
* Called every time there is a message on the GStreamer pipeline's bus.
*
* We are mostly interested in the new pixbug message.
* In that case, checks if the video recording or the intervalometer is enabled.
* If so, grabs an image if it's time to do so.
*/
void Pipeline::bus_message_cb(GstBus* /*bus*/, GstMessage *msg, gpointer user_data)
{
Pipeline *context = static_cast<Pipeline*>(user_data);
bool verbose = context->owner_->get_configuration()->get_verbose();
switch (GST_MESSAGE_TYPE (msg))
{
case GST_MESSAGE_ELEMENT:
{
const GValue *val;
GdkPixbuf *pixbuf = NULL;
/* only interested in element messages from our gdkpixbufsink */
if (msg->src != GST_OBJECT_CAST(context->gdkpixbufsink_))
break;
/* only interested in these two messages */
if (!gst_structure_has_name(msg->structure, "preroll-pixbuf") &&
!gst_structure_has_name(msg->structure, "pixbuf"))
{
break;
}
//g_print("pixbuf\n");
val = gst_structure_get_value(msg->structure, "pixbuf");
g_return_if_fail(val != NULL);
pixbuf = GDK_PIXBUF(g_value_dup_object(val));
if (context->get_record_all_frames() || context->get_intervalometer_is_on()) // if video grabbing is enabled
{
Clip *current_clip = context->owner_->get_current_clip();
unsigned long last_time_grabbed = current_clip->get_last_time_grabbed_image();
unsigned long now = timing::get_timestamp_now();
bool must_grab_now = false;
// VIDEO RECORDING:
if (context->get_record_all_frames())
{
//std::cout << "Video grabbing is on." << std::endl;
unsigned long time_between_frames = (unsigned long)(1.0f / float(current_clip->get_playhead_fps()) * timing::TIMESTAMP_PRECISION);
if (verbose)
std::cout << "now=" << now << " last_time_grabbed=" << last_time_grabbed << " time_between_frames" << time_between_frames << std::endl;
if ((now - last_time_grabbed) > time_between_frames)
{
must_grab_now = true;
}
} // not mutually exclusive - why not have both on?
// INTERVALOMETER:
if (context->get_intervalometer_is_on())
{
long time_between_intervalometer_ticks = long(current_clip->get_intervalometer_rate() * timing::TIMESTAMP_PRECISION);
long passed = (now - last_time_grabbed);
if (verbose)
std::cout << "time between intervalometer ticks: " << passed << "/" << time_between_intervalometer_ticks << std::endl;
if (passed > time_between_intervalometer_ticks)
{
if (verbose)
std::cout << "Interval has passed. Time to grab." << std::endl;
must_grab_now = true;
}
}
if (must_grab_now)
{
if (verbose)
std::cout << "Grabbing an image" << std::endl;
context->save_image_to_current_clip(pixbuf);
current_clip->set_last_time_grabbed_image(now);
}
}
g_object_unref(pixbuf);
break;
}
case GST_MESSAGE_ERROR:
{
GError *err = NULL;
gchar *dbg = NULL;
gst_message_parse_error(msg, &err, &dbg);
g_error("Error: %s\n%s\n", err->message, (dbg) ? dbg : "");
g_error_free(err);
g_free(dbg);
break;
}
default:
break;
}
}
static gboolean
gst_mms_start (GstBaseSrc * bsrc)
{
GstMMS *mms = GST_MMS (bsrc);
guint bandwidth_avail;
if (!mms->uri_name || *mms->uri_name == '\0')
goto no_uri;
if (mms->connection_speed)
bandwidth_avail = mms->connection_speed;
else
bandwidth_avail = G_MAXINT;
/* If we already have a connection, and the uri isn't changed, reuse it,
as connecting is expensive. */
if (mms->connection) {
if (!strcmp (mms->uri_name, mms->current_connection_uri_name)) {
GST_DEBUG_OBJECT (mms, "Reusing existing connection for %s",
mms->uri_name);
return TRUE;
} else {
mmsx_close (mms->connection);
g_free (mms->current_connection_uri_name);
mms->current_connection_uri_name = NULL;
}
}
/* FIXME: pass some sane arguments here */
GST_DEBUG_OBJECT (mms,
"Trying mms_connect (%s) with bandwidth constraint of %d bps",
mms->uri_name, bandwidth_avail);
mms->connection = mmsx_connect (NULL, NULL, mms->uri_name, bandwidth_avail);
if (mms->connection) {
/* Save the uri name so that it can be checked for connection reusing,
see above. */
mms->current_connection_uri_name = g_strdup (mms->uri_name);
GST_DEBUG_OBJECT (mms, "Connect successful");
return TRUE;
} else {
gchar *url, *location;
GST_ERROR_OBJECT (mms,
"Could not connect to this stream, redirecting to rtsp");
location = strstr (mms->uri_name, "://");
if (location == NULL || *location == '\0' || *(location + 3) == '\0')
goto no_uri;
url = g_strdup_printf ("rtsp://%s", location + 3);
gst_element_post_message (GST_ELEMENT_CAST (mms),
gst_message_new_element (GST_OBJECT_CAST (mms),
gst_structure_new ("redirect", "new-location", G_TYPE_STRING, url,
NULL)));
/* post an error message as well, so that applications that don't handle
* redirect messages get to see a proper error message */
GST_ELEMENT_ERROR (mms, RESOURCE, OPEN_READ,
("Could not connect to streaming server."),
("A redirect message was posted on the bus and should have been "
"handled by the application."));
return FALSE;
}
no_uri:
{
GST_ELEMENT_ERROR (mms, RESOURCE, OPEN_READ,
("No URI to open specified"), (NULL));
return FALSE;
}
}
static void
bus_message_cb (GstBus * bus, GstMessage * msg, AppInfo * info)
{
switch (GST_MESSAGE_TYPE (msg)) {
case GST_MESSAGE_ASYNC_DONE:{
/* only interested in async-done messages from the top-level pipeline */
if (msg->src != GST_OBJECT_CAST (info->pipe))
break;
if (!info->prerolled) {
/* make slider visible if it's not visible already */
gtk_widget_show (info->slider);
/* initial frame is often black, so seek to beginning plus a bit */
seek_to (info, 0.001);
info->prerolled = TRUE;
}
/* update position */
if (!gst_element_query_position (info->pipe, GST_FORMAT_TIME,
&info->cur_pos))
info->cur_pos = -1;
break;
}
case GST_MESSAGE_ELEMENT:{
const GValue *val;
GdkPixbuf *pixbuf = NULL;
const GstStructure *structure;
/* only interested in element messages from our gdkpixbufsink */
if (msg->src != GST_OBJECT_CAST (info->sink))
break;
/* only interested in these two messages */
if (!gst_message_has_name (msg, "preroll-pixbuf") &&
!gst_message_has_name (msg, "pixbuf")) {
break;
}
g_print ("pixbuf\n");
structure = gst_message_get_structure (msg);
val = gst_structure_get_value (structure, "pixbuf");
g_return_if_fail (val != NULL);
pixbuf = GDK_PIXBUF (g_value_dup_object (val));
gtk_image_set_from_pixbuf (GTK_IMAGE (info->img), pixbuf);
g_object_unref (pixbuf);
break;
}
case GST_MESSAGE_ERROR:{
GError *err = NULL;
gchar *dbg = NULL;
gst_message_parse_error (msg, &err, &dbg);
g_error ("Error: %s\n%s\n", err->message, (dbg) ? dbg : "");
g_clear_error (&err);
g_free (dbg);
break;
}
default:
break;
}
}
void PlaybinSession::busMessage(Message const& msg)
{
GstMessage* gm = msg.rawMessage();
if (gm == 0)
{ // Null message, query current position
GstFormat format = GST_FORMAT_TIME;
gint64 position = 0;
if (gst_element_query_position(d->playbin, &format, &position) == TRUE)
{
quint32 pos = position / 1000000;
if (pos != d->position)
{
d->position = pos;
emit positionChanged(d->position);
}
}
}
else if (GST_MESSAGE_SRC(gm) == GST_OBJECT_CAST(d->playbin))
{
switch (GST_MESSAGE_TYPE(gm))
{
case GST_MESSAGE_DURATION:
break;
case GST_MESSAGE_STATE_CHANGED:
{
GstState oldState;
GstState newState;
GstState pending;
gst_message_parse_state_changed(gm, &oldState, &newState, &pending);
switch (newState)
{
case GST_STATE_VOID_PENDING:
case GST_STATE_NULL:
case GST_STATE_READY:
case GST_STATE_PAUSED:
break;
case GST_STATE_PLAYING:
if (oldState == GST_STATE_PAUSED)
getStreamsInfo();
if (d->state != QtopiaMedia::Playing)
emit playerStateChanged(d->state = QtopiaMedia::Playing);
break;
}
}
break;
case GST_MESSAGE_EOS:
if (d->state != QtopiaMedia::Stopped)
emit playerStateChanged(d->state = QtopiaMedia::Stopped);
break;
case GST_MESSAGE_STREAM_STATUS:
case GST_MESSAGE_UNKNOWN:
case GST_MESSAGE_ERROR:
case GST_MESSAGE_WARNING:
case GST_MESSAGE_INFO:
case GST_MESSAGE_TAG:
case GST_MESSAGE_BUFFERING:
case GST_MESSAGE_STATE_DIRTY:
case GST_MESSAGE_STEP_DONE:
case GST_MESSAGE_CLOCK_PROVIDE:
case GST_MESSAGE_CLOCK_LOST:
case GST_MESSAGE_NEW_CLOCK:
case GST_MESSAGE_STRUCTURE_CHANGE:
case GST_MESSAGE_APPLICATION:
case GST_MESSAGE_ELEMENT:
case GST_MESSAGE_SEGMENT_START:
case GST_MESSAGE_SEGMENT_DONE:
case GST_MESSAGE_LATENCY:
case GST_MESSAGE_ANY:
break;
}
}
}
static void
gst_raw_parse_loop (GstElement * element)
{
GstRawParse *rp = GST_RAW_PARSE (element);
GstRawParseClass *rp_class = GST_RAW_PARSE_GET_CLASS (rp);
GstFlowReturn ret;
GstBuffer *buffer;
gint size;
if (!gst_raw_parse_set_src_caps (rp))
goto no_caps;
if (rp->close_segment) {
GST_DEBUG_OBJECT (rp, "sending close segment");
gst_pad_push_event (rp->srcpad, rp->close_segment);
rp->close_segment = NULL;
}
if (rp->start_segment) {
GST_DEBUG_OBJECT (rp, "sending start segment");
gst_pad_push_event (rp->srcpad, rp->start_segment);
rp->start_segment = NULL;
}
if (rp_class->multiple_frames_per_buffer && rp->framesize < 4096)
size = 4096 - (4096 % rp->framesize);
else
size = rp->framesize;
if (rp->segment.rate >= 0) {
if (rp->offset + size > rp->upstream_length) {
GstFormat fmt = GST_FORMAT_BYTES;
if (!gst_pad_query_peer_duration (rp->sinkpad, &fmt,
&rp->upstream_length)) {
GST_WARNING_OBJECT (rp,
"Could not get upstream duration, trying to pull frame by frame");
size = rp->framesize;
} else if (rp->upstream_length < rp->offset + rp->framesize) {
ret = GST_FLOW_UNEXPECTED;
goto pause;
} else if (rp->offset + size > rp->upstream_length) {
size = rp->upstream_length - rp->offset;
size -= size % rp->framesize;
}
}
} else {
if (rp->offset == 0) {
ret = GST_FLOW_UNEXPECTED;
goto pause;
} else if (rp->offset < size) {
size -= rp->offset;
}
rp->offset -= size;
}
ret = gst_pad_pull_range (rp->sinkpad, rp->offset, size, &buffer);
if (ret != GST_FLOW_OK) {
GST_DEBUG_OBJECT (rp, "pull_range (%" G_GINT64_FORMAT ", %u) "
"failed, flow: %s", rp->offset, size, gst_flow_get_name (ret));
buffer = NULL;
goto pause;
}
if (GST_BUFFER_SIZE (buffer) < size) {
GST_DEBUG_OBJECT (rp, "Short read at offset %" G_GINT64_FORMAT
", got only %u of %u bytes", rp->offset, GST_BUFFER_SIZE (buffer),
size);
if (size > rp->framesize) {
GST_BUFFER_SIZE (buffer) -= GST_BUFFER_SIZE (buffer) % rp->framesize;
} else {
gst_buffer_unref (buffer);
buffer = NULL;
ret = GST_FLOW_UNEXPECTED;
goto pause;
}
}
ret = gst_raw_parse_push_buffer (rp, buffer);
if (ret != GST_FLOW_OK)
goto pause;
return;
/* ERRORS */
no_caps:
{
GST_ERROR_OBJECT (rp, "could not negotiate caps");
ret = GST_FLOW_NOT_NEGOTIATED;
goto pause;
}
pause:
{
const gchar *reason = gst_flow_get_name (ret);
GST_LOG_OBJECT (rp, "pausing task, reason %s", reason);
gst_pad_pause_task (rp->sinkpad);
if (ret == GST_FLOW_UNEXPECTED) {
if (rp->segment.flags & GST_SEEK_FLAG_SEGMENT) {
GstClockTime stop;
GST_LOG_OBJECT (rp, "Sending segment done");
if ((stop = rp->segment.stop) == -1)
stop = rp->segment.duration;
gst_element_post_message (GST_ELEMENT_CAST (rp),
gst_message_new_segment_done (GST_OBJECT_CAST (rp),
rp->segment.format, stop));
} else {
GST_LOG_OBJECT (rp, "Sending EOS, at end of stream");
gst_pad_push_event (rp->srcpad, gst_event_new_eos ());
}
} else if (ret == GST_FLOW_NOT_LINKED || ret < GST_FLOW_UNEXPECTED) {
GST_ELEMENT_ERROR (rp, STREAM, FAILED,
("Internal data stream error."),
("stream stopped, reason %s", reason));
gst_pad_push_event (rp->srcpad, gst_event_new_eos ());
}
return;
}
}
static GstFlowReturn
gst_gdk_pixbuf_sink_handle_buffer (GstBaseSink * basesink, GstBuffer * buf,
const gchar * msg_name)
{
GstGdkPixbufSink *sink;
GdkPixbuf *pixbuf;
gboolean do_post;
sink = GST_GDK_PIXBUF_SINK (basesink);
pixbuf = gst_gdk_pixbuf_sink_get_pixbuf_from_buffer (sink, buf);
GST_OBJECT_LOCK (sink);
do_post = sink->post_messages;
if (sink->last_pixbuf)
g_object_unref (sink->last_pixbuf);
sink->last_pixbuf = pixbuf; /* take ownership */
GST_OBJECT_UNLOCK (sink);
if (G_UNLIKELY (pixbuf == NULL))
goto error;
if (do_post) {
GstStructure *s;
GstMessage *msg;
GstFormat format;
GstClockTime timestamp;
GstClockTime running_time, stream_time;
GstSegment *segment = &basesink->segment;
format = segment->format;
timestamp = GST_BUFFER_PTS (buf);
running_time = gst_segment_to_running_time (segment, format, timestamp);
stream_time = gst_segment_to_stream_time (segment, format, timestamp);
/* it's okay to keep using pixbuf here, we can be sure no one is going to
* unref or change sink->last_pixbuf before we return from this function.
* The structure will take its own ref to the pixbuf. */
s = gst_structure_new (msg_name,
"pixbuf", GDK_TYPE_PIXBUF, pixbuf,
"pixel-aspect-ratio", GST_TYPE_FRACTION, sink->par_n, sink->par_d,
"timestamp", G_TYPE_UINT64, timestamp,
"stream-time", G_TYPE_UINT64, stream_time,
"running-time", G_TYPE_UINT64, running_time, NULL);
msg = gst_message_new_element (GST_OBJECT_CAST (sink), s);
gst_element_post_message (GST_ELEMENT_CAST (sink), msg);
}
g_object_notify (G_OBJECT (sink), "last-pixbuf");
return GST_FLOW_OK;
/* ERRORS */
error:
{
/* This shouldn't really happen */
GST_ELEMENT_ERROR (sink, LIBRARY, FAILED,
("Couldn't create pixbuf from RGB image."),
("Probably not enough free memory"));
return GST_FLOW_ERROR;
}
}
static GstStateChangeReturn
gst_base_audio_src_change_state (GstElement * element,
GstStateChange transition)
{
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
GST_DEBUG_OBJECT (src, "NULL->READY");
GST_OBJECT_LOCK (src);
if (src->ringbuffer == NULL) {
gst_audio_clock_reset (GST_AUDIO_CLOCK (src->clock), 0);
src->ringbuffer = gst_base_audio_src_create_ringbuffer (src);
}
GST_OBJECT_UNLOCK (src);
if (!gst_ring_buffer_open_device (src->ringbuffer))
goto open_failed;
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
GST_DEBUG_OBJECT (src, "READY->PAUSED");
src->next_sample = -1;
gst_ring_buffer_set_flushing (src->ringbuffer, FALSE);
gst_ring_buffer_may_start (src->ringbuffer, FALSE);
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
GST_DEBUG_OBJECT (src, "PAUSED->PLAYING");
gst_ring_buffer_may_start (src->ringbuffer, TRUE);
break;
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
GST_DEBUG_OBJECT (src, "PLAYING->PAUSED");
gst_ring_buffer_may_start (src->ringbuffer, FALSE);
gst_ring_buffer_pause (src->ringbuffer);
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
GST_DEBUG_OBJECT (src, "PAUSED->READY");
gst_ring_buffer_set_flushing (src->ringbuffer, TRUE);
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
GST_DEBUG_OBJECT (src, "PAUSED->READY");
gst_ring_buffer_release (src->ringbuffer);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
GST_DEBUG_OBJECT (src, "READY->NULL");
gst_ring_buffer_close_device (src->ringbuffer);
GST_OBJECT_LOCK (src);
gst_object_unparent (GST_OBJECT_CAST (src->ringbuffer));
src->ringbuffer = NULL;
GST_OBJECT_UNLOCK (src);
break;
default:
break;
}
return ret;
/* ERRORS */
open_failed:
{
/* subclass must post a meaningfull error message */
GST_DEBUG_OBJECT (src, "open failed");
return GST_STATE_CHANGE_FAILURE;
}
}
static GstPad *
gst_ghost_pad_new_full (const gchar * name, GstPadDirection dir,
GstPadTemplate * templ)
{
GstPad *ret;
GstPad *internal;
GstPadDirection otherdir;
g_return_val_if_fail (dir != GST_PAD_UNKNOWN, NULL);
/* OBJECT CREATION */
if (templ) {
ret = g_object_new (GST_TYPE_GHOST_PAD, "name", name,
"direction", dir, "template", templ, NULL);
} else {
ret = g_object_new (GST_TYPE_GHOST_PAD, "name", name,
"direction", dir, NULL);
}
/* Set directional padfunctions for ghostpad */
if (dir == GST_PAD_SINK) {
gst_pad_set_bufferalloc_function (ret,
GST_DEBUG_FUNCPTR (gst_proxy_pad_do_bufferalloc));
gst_pad_set_chain_function (ret,
GST_DEBUG_FUNCPTR (gst_proxy_pad_do_chain));
} else {
gst_pad_set_getrange_function (ret,
GST_DEBUG_FUNCPTR (gst_proxy_pad_do_getrange));
gst_pad_set_checkgetrange_function (ret,
GST_DEBUG_FUNCPTR (gst_proxy_pad_do_checkgetrange));
}
/* link/unlink functions */
gst_pad_set_link_function (ret, GST_DEBUG_FUNCPTR (gst_ghost_pad_do_link));
gst_pad_set_unlink_function (ret,
GST_DEBUG_FUNCPTR (gst_ghost_pad_do_unlink));
/* INTERNAL PAD, it always exists and is child of the ghostpad */
otherdir = (dir == GST_PAD_SRC) ? GST_PAD_SINK : GST_PAD_SRC;
if (templ) {
internal =
g_object_new (GST_TYPE_PROXY_PAD, "name", NULL,
"direction", otherdir, "template", templ, NULL);
} else {
internal =
g_object_new (GST_TYPE_PROXY_PAD, "name", NULL,
"direction", otherdir, NULL);
}
GST_PAD_UNSET_FLUSHING (internal);
/* Set directional padfunctions for internal pad */
if (dir == GST_PAD_SRC) {
gst_pad_set_bufferalloc_function (internal,
GST_DEBUG_FUNCPTR (gst_proxy_pad_do_bufferalloc));
gst_pad_set_chain_function (internal,
GST_DEBUG_FUNCPTR (gst_proxy_pad_do_chain));
} else {
gst_pad_set_getrange_function (internal,
GST_DEBUG_FUNCPTR (gst_proxy_pad_do_getrange));
gst_pad_set_checkgetrange_function (internal,
GST_DEBUG_FUNCPTR (gst_proxy_pad_do_checkgetrange));
}
GST_PROXY_LOCK (ret);
/* now make the ghostpad a parent of the internal pad */
if (!gst_object_set_parent (GST_OBJECT_CAST (internal),
GST_OBJECT_CAST (ret)))
goto parent_failed;
/* The ghostpad is the parent of the internal pad and is the only object that
* can have a refcount on the internal pad.
* At this point, the GstGhostPad has a refcount of 1, and the internal pad has
* a refcount of 1.
* When the refcount of the GstGhostPad drops to 0, the ghostpad will dispose
* it's refcount on the internal pad in the dispose method by un-parenting it.
* This is why we don't take extra refcounts in the assignments below
*/
GST_PROXY_PAD_INTERNAL (ret) = internal;
GST_PROXY_PAD_INTERNAL (internal) = ret;
/* could be more general here, iterating over all writable properties...
* taking the short road for now tho */
GST_GHOST_PAD_CAST (ret)->notify_id =
g_signal_connect (internal, "notify::caps", G_CALLBACK (on_int_notify),
ret);
/* call function to init values of the pad caps */
on_int_notify (internal, NULL, GST_GHOST_PAD_CAST (ret));
/* special activation functions for the internal pad */
gst_pad_set_activatepull_function (internal,
GST_DEBUG_FUNCPTR (gst_ghost_pad_internal_do_activate_pull));
gst_pad_set_activatepush_function (internal,
GST_DEBUG_FUNCPTR (gst_ghost_pad_internal_do_activate_push));
GST_PROXY_UNLOCK (ret);
return ret;
/* ERRORS */
parent_failed:
{
GST_WARNING_OBJECT (ret, "Could not set internal pad %s:%s",
GST_DEBUG_PAD_NAME (internal));
g_critical ("Could not set internal pad %s:%s",
GST_DEBUG_PAD_NAME (internal));
GST_PROXY_UNLOCK (ret);
gst_object_unref (ret);
gst_object_unref (internal);
return NULL;
}
}
* payloaded RTP packets. We simply ghost the pad here. */
static void
new_session_pad (GstElement * session, GstPad * pad, GstSDPDemux * demux)
{
gchar *name;
GstPadTemplate *template;
gint id, ssrc, pt;
GList *lstream;
GstSDPStream *stream;
gboolean all_added;
GST_DEBUG_OBJECT (demux, "got new session pad %" GST_PTR_FORMAT, pad);
GST_SDP_STREAM_LOCK (demux);
/* find stream */
name = gst_object_get_name (GST_OBJECT_CAST (pad));
if (sscanf (name, "recv_rtp_src_%d_%d_%d", &id, &ssrc, &pt) != 3)
goto unknown_stream;
GST_DEBUG_OBJECT (demux, "stream: %u, SSRC %d, PT %d", id, ssrc, pt);
stream =
find_stream (demux, GINT_TO_POINTER (id), (gpointer) find_stream_by_id);
if (stream == NULL)
goto unknown_stream;
/* no need for a timeout anymore now */
g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout", (guint64) 0, NULL);
/* create a new pad we will use to stream to */
template = gst_static_pad_template_get (&rtptemplate);
static gboolean
gst_type_find_element_activate (GstPad * pad)
{
GstTypeFindProbability probability = 0;
GstCaps *found_caps = NULL;
GstTypeFindElement *typefind;
typefind = GST_TYPE_FIND_ELEMENT (GST_OBJECT_PARENT (pad));
/* if we have force caps, use those */
if (typefind->force_caps) {
found_caps = gst_caps_ref (typefind->force_caps);
probability = GST_TYPE_FIND_MAXIMUM;
goto done;
}
/* 1. try to activate in pull mode. if not, switch to push and succeed.
2. try to pull type find.
3. deactivate pull mode.
4. src pad might have been activated push by the state change. deactivate.
5. if we didn't find any caps, try getting the uri extension by doing an uri
query.
6. if we didn't find any caps, fail.
7. emit have-type; maybe the app connected the source pad to something.
8. if the sink pad is activated, we are in pull mode. succeed.
otherwise activate both pads in push mode and succeed.
*/
/* 1 */
if (!gst_pad_check_pull_range (pad) || !gst_pad_activate_pull (pad, TRUE)) {
start_typefinding (typefind);
return gst_pad_activate_push (pad, TRUE);
}
/* 2 */
{
GstPad *peer;
peer = gst_pad_get_peer (pad);
if (peer) {
gint64 size;
GstFormat format = GST_FORMAT_BYTES;
if (!gst_pad_query_duration (peer, &format, &size)) {
GST_WARNING_OBJECT (typefind, "Could not query upstream length!");
gst_object_unref (peer);
return FALSE;
}
/* the size if 0, we cannot continue */
if (size == 0) {
/* keep message in sync with message in sink event handler */
GST_ELEMENT_ERROR (typefind, STREAM, TYPE_NOT_FOUND,
(_("Stream contains no data.")), ("Can't typefind empty stream"));
gst_object_unref (peer);
return FALSE;
}
found_caps = gst_type_find_helper_get_range (GST_OBJECT_CAST (peer),
(GstTypeFindHelperGetRangeFunction) (GST_PAD_GETRANGEFUNC (peer)),
(guint64) size, &probability);
gst_object_unref (peer);
}
}
/* 3 */
gst_pad_activate_pull (pad, FALSE);
/* 4 */
gst_pad_activate_push (typefind->src, FALSE);
/* 5 */
if (!found_caps || probability < typefind->min_probability) {
found_caps = gst_type_find_guess_by_extension (typefind, pad, &probability);
}
/* 6 */
if (!found_caps || probability < typefind->min_probability) {
GST_ELEMENT_ERROR (typefind, STREAM, TYPE_NOT_FOUND, (NULL), (NULL));
gst_caps_replace (&found_caps, NULL);
return FALSE;
}
done:
/* 7 */
g_signal_emit (typefind, gst_type_find_element_signals[HAVE_TYPE],
0, probability, found_caps);
gst_caps_unref (found_caps);
typefind->mode = MODE_NORMAL;
/* 8 */
if (gst_pad_is_active (pad))
return TRUE;
else {
gboolean ret;
ret = gst_pad_activate_push (typefind->src, TRUE);
ret &= gst_pad_activate_push (pad, TRUE);
return ret;
}
}
static GstFlowReturn
gst_udpsrc_create (GstPushSrc * psrc, GstBuffer ** buf)
{
GstUDPSrc *udpsrc;
GstBuffer *outbuf = NULL;
GSocketAddress *saddr = NULL;
gint flags = G_SOCKET_MSG_NONE;
gboolean try_again;
GError *err = NULL;
gssize res;
gsize offset;
udpsrc = GST_UDPSRC_CAST (psrc);
if (!gst_udpsrc_ensure_mem (udpsrc))
goto memory_alloc_error;
retry:
do {
gint64 timeout;
try_again = FALSE;
if (udpsrc->timeout)
timeout = udpsrc->timeout / 1000;
else
timeout = -1;
GST_LOG_OBJECT (udpsrc, "doing select, timeout %" G_GINT64_FORMAT, timeout);
if (!g_socket_condition_timed_wait (udpsrc->used_socket, G_IO_IN | G_IO_PRI,
timeout, udpsrc->cancellable, &err)) {
if (g_error_matches (err, G_IO_ERROR, G_IO_ERROR_BUSY)
|| g_error_matches (err, G_IO_ERROR, G_IO_ERROR_CANCELLED)) {
goto stopped;
} else if (g_error_matches (err, G_IO_ERROR, G_IO_ERROR_TIMED_OUT)) {
g_clear_error (&err);
/* timeout, post element message */
gst_element_post_message (GST_ELEMENT_CAST (udpsrc),
gst_message_new_element (GST_OBJECT_CAST (udpsrc),
gst_structure_new ("GstUDPSrcTimeout",
"timeout", G_TYPE_UINT64, udpsrc->timeout, NULL)));
} else {
goto select_error;
}
try_again = TRUE;
}
} while (G_UNLIKELY (try_again));
if (saddr != NULL) {
g_object_unref (saddr);
saddr = NULL;
}
res =
g_socket_receive_message (udpsrc->used_socket, &saddr, udpsrc->vec, 2,
NULL, NULL, &flags, udpsrc->cancellable, &err);
if (G_UNLIKELY (res < 0)) {
/* EHOSTUNREACH for a UDP socket means that a packet sent with udpsink
* generated a "port unreachable" ICMP response. We ignore that and try
* again. */
if (g_error_matches (err, G_IO_ERROR, G_IO_ERROR_HOST_UNREACHABLE)) {
g_clear_error (&err);
goto retry;
}
goto receive_error;
}
/* remember maximum packet size */
if (res > udpsrc->max_size)
udpsrc->max_size = res;
outbuf = gst_buffer_new ();
/* append first memory chunk to buffer */
gst_buffer_append_memory (outbuf, udpsrc->mem);
/* if the packet didn't fit into the first chunk, add second one as well */
if (res > udpsrc->map.size) {
gst_buffer_append_memory (outbuf, udpsrc->mem_max);
gst_memory_unmap (udpsrc->mem_max, &udpsrc->map_max);
udpsrc->vec[1].buffer = NULL;
udpsrc->vec[1].size = 0;
udpsrc->mem_max = NULL;
}
/* make sure we allocate a new chunk next time (we do this only here because
* we look at map.size to see if the second memory chunk is needed above) */
gst_memory_unmap (udpsrc->mem, &udpsrc->map);
udpsrc->vec[0].buffer = NULL;
udpsrc->vec[0].size = 0;
udpsrc->mem = NULL;
offset = udpsrc->skip_first_bytes;
if (G_UNLIKELY (offset > 0 && res < offset))
goto skip_error;
gst_buffer_resize (outbuf, offset, res - offset);
/* use buffer metadata so receivers can also track the address */
if (saddr) {
gst_buffer_add_net_address_meta (outbuf, saddr);
g_object_unref (saddr);
saddr = NULL;
}
GST_LOG_OBJECT (udpsrc, "read packet of %d bytes", (int) res);
*buf = GST_BUFFER_CAST (outbuf);
return GST_FLOW_OK;
/* ERRORS */
memory_alloc_error:
{
GST_ELEMENT_ERROR (udpsrc, RESOURCE, READ, (NULL),
("Failed to allocate or map memory"));
return GST_FLOW_ERROR;
}
select_error:
{
GST_ELEMENT_ERROR (udpsrc, RESOURCE, READ, (NULL),
("select error: %s", err->message));
g_clear_error (&err);
return GST_FLOW_ERROR;
}
stopped:
{
GST_DEBUG ("stop called");
g_clear_error (&err);
return GST_FLOW_FLUSHING;
}
receive_error:
{
if (g_error_matches (err, G_IO_ERROR, G_IO_ERROR_BUSY) ||
g_error_matches (err, G_IO_ERROR, G_IO_ERROR_CANCELLED)) {
g_clear_error (&err);
return GST_FLOW_FLUSHING;
} else {
GST_ELEMENT_ERROR (udpsrc, RESOURCE, READ, (NULL),
("receive error %" G_GSSIZE_FORMAT ": %s", res, err->message));
g_clear_error (&err);
return GST_FLOW_ERROR;
}
}
skip_error:
{
gst_buffer_unref (outbuf);
GST_ELEMENT_ERROR (udpsrc, STREAM, DECODE, (NULL),
("UDP buffer to small to skip header"));
return GST_FLOW_ERROR;
}
}
static GstCaps *
gst_type_find_guess_by_extension (GstTypeFindElement * typefind, GstPad * pad,
GstTypeFindProbability * probability)
{
GstQuery *query;
gchar *uri;
size_t len;
gint find;
GstCaps *caps;
query = gst_query_new_uri ();
/* try getting the caps with an uri query and from the extension */
if (!gst_pad_peer_query (pad, query))
goto peer_query_failed;
gst_query_parse_uri (query, &uri);
if (uri == NULL)
goto no_uri;
GST_DEBUG_OBJECT (typefind, "finding extension of %s", uri);
/* find the extension on the uri, this is everything after a '.' */
len = strlen (uri);
find = len - 1;
while (find >= 0) {
if (uri[find] == '.')
break;
find--;
}
if (find < 0)
goto no_extension;
GST_DEBUG_OBJECT (typefind, "found extension %s", &uri[find + 1]);
caps =
gst_type_find_helper_for_extension (GST_OBJECT_CAST (typefind),
&uri[find + 1]);
if (caps)
*probability = GST_TYPE_FIND_MAXIMUM;
gst_query_unref (query);
return caps;
/* ERRORS */
peer_query_failed:
{
GST_WARNING_OBJECT (typefind, "failed to query peer uri");
gst_query_unref (query);
return NULL;
}
no_uri:
{
GST_WARNING_OBJECT (typefind, "could not parse the peer uri");
gst_query_unref (query);
return NULL;
}
no_extension:
{
GST_WARNING_OBJECT (typefind, "could not find uri extension in %s", uri);
gst_query_unref (query);
return NULL;
}
}
static GstStateChangeReturn
gst_pulsesrc_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret;
GstPulseSrc *this = GST_PULSESRC_CAST (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
if (!(this->mainloop = pa_threaded_mainloop_new ()))
goto mainloop_failed;
if (pa_threaded_mainloop_start (this->mainloop) < 0) {
pa_threaded_mainloop_free (this->mainloop);
this->mainloop = NULL;
goto mainloop_start_failed;
}
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
gst_element_post_message (element,
gst_message_new_clock_provide (GST_OBJECT_CAST (element),
GST_AUDIO_BASE_SRC (this)->clock, TRUE));
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
/* uncork and start recording */
gst_pulsesrc_play (this);
break;
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
/* stop recording ASAP by corking */
pa_threaded_mainloop_lock (this->mainloop);
GST_DEBUG_OBJECT (this, "corking");
gst_pulsesrc_set_corked (this, TRUE, FALSE);
pa_threaded_mainloop_unlock (this->mainloop);
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
/* now make sure we get out of the _read method */
gst_pulsesrc_pause (this);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
if (this->mainloop)
pa_threaded_mainloop_stop (this->mainloop);
gst_pulsesrc_destroy_context (this);
if (this->mainloop) {
pa_threaded_mainloop_free (this->mainloop);
this->mainloop = NULL;
}
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
/* format_lost is reset in release() in baseaudiosink */
gst_element_post_message (element,
gst_message_new_clock_lost (GST_OBJECT_CAST (element),
GST_AUDIO_BASE_SRC (this)->clock));
break;
default:
break;
}
return ret;
/* ERRORS */
mainloop_failed:
{
GST_ELEMENT_ERROR (this, RESOURCE, FAILED,
("pa_threaded_mainloop_new() failed"), (NULL));
return GST_STATE_CHANGE_FAILURE;
}
mainloop_start_failed:
{
GST_ELEMENT_ERROR (this, RESOURCE, FAILED,
("pa_threaded_mainloop_start() failed"), (NULL));
return GST_STATE_CHANGE_FAILURE;
}
}
static void
gst_image_freeze_src_loop (GstPad * pad)
{
GstImageFreeze *self = GST_IMAGE_FREEZE (GST_PAD_PARENT (pad));
GstBuffer *buffer;
guint64 offset;
GstClockTime timestamp, timestamp_end;
guint64 cstart, cstop;
gboolean in_seg, eos;
GstFlowReturn flow_ret = GST_FLOW_OK;
g_mutex_lock (&self->lock);
if (!gst_pad_has_current_caps (self->srcpad)) {
GST_ERROR_OBJECT (pad, "Not negotiated yet");
flow_ret = GST_FLOW_NOT_NEGOTIATED;
g_mutex_unlock (&self->lock);
goto pause_task;
}
if (!self->buffer) {
GST_ERROR_OBJECT (pad, "Have no buffer yet");
flow_ret = GST_FLOW_ERROR;
g_mutex_unlock (&self->lock);
goto pause_task;
}
buffer = gst_buffer_ref (self->buffer);
buffer = gst_buffer_make_writable (buffer);
g_mutex_unlock (&self->lock);
if (self->need_segment) {
GstEvent *e;
GST_DEBUG_OBJECT (pad, "Pushing SEGMENT event: %" GST_SEGMENT_FORMAT,
&self->segment);
e = gst_event_new_segment (&self->segment);
g_mutex_lock (&self->lock);
if (self->segment.rate >= 0) {
self->offset =
gst_util_uint64_scale (self->segment.start, self->fps_n,
self->fps_d * GST_SECOND);
} else {
self->offset =
gst_util_uint64_scale (self->segment.stop, self->fps_n,
self->fps_d * GST_SECOND);
}
g_mutex_unlock (&self->lock);
self->need_segment = FALSE;
gst_pad_push_event (self->srcpad, e);
}
g_mutex_lock (&self->lock);
offset = self->offset;
if (self->fps_n != 0) {
timestamp =
gst_util_uint64_scale (offset, self->fps_d * GST_SECOND, self->fps_n);
timestamp_end =
gst_util_uint64_scale (offset + 1, self->fps_d * GST_SECOND,
self->fps_n);
} else {
timestamp = self->segment.start;
timestamp_end = GST_CLOCK_TIME_NONE;
}
eos = (self->fps_n == 0 && offset > 0) ||
(self->segment.rate >= 0 && self->segment.stop != -1
&& timestamp > self->segment.stop) || (self->segment.rate < 0
&& offset == 0) || (self->segment.rate < 0
&& self->segment.start != -1 && timestamp_end < self->segment.start);
if (self->fps_n == 0 && offset > 0)
in_seg = FALSE;
else
in_seg =
gst_segment_clip (&self->segment, GST_FORMAT_TIME, timestamp,
timestamp_end, &cstart, &cstop);
if (in_seg) {
self->segment.position = cstart;
if (self->segment.rate >= 0)
self->segment.position = cstop;
}
if (self->segment.rate >= 0)
self->offset++;
else
self->offset--;
g_mutex_unlock (&self->lock);
GST_DEBUG_OBJECT (pad, "Handling buffer with timestamp %" GST_TIME_FORMAT,
GST_TIME_ARGS (timestamp));
if (in_seg) {
GST_BUFFER_DTS (buffer) = GST_CLOCK_TIME_NONE;
GST_BUFFER_PTS (buffer) = cstart;
GST_BUFFER_DURATION (buffer) = cstop - cstart;
GST_BUFFER_OFFSET (buffer) = offset;
GST_BUFFER_OFFSET_END (buffer) = offset + 1;
flow_ret = gst_pad_push (self->srcpad, buffer);
GST_DEBUG_OBJECT (pad, "Pushing buffer resulted in %s",
gst_flow_get_name (flow_ret));
if (flow_ret != GST_FLOW_OK)
goto pause_task;
} else {
gst_buffer_unref (buffer);
}
if (eos) {
flow_ret = GST_FLOW_EOS;
goto pause_task;
}
return;
pause_task:
{
const gchar *reason = gst_flow_get_name (flow_ret);
GST_LOG_OBJECT (self, "pausing task, reason %s", reason);
gst_pad_pause_task (pad);
if (flow_ret == GST_FLOW_EOS) {
if ((self->segment.flags & GST_SEEK_FLAG_SEGMENT)) {
GstMessage *m;
GstEvent *e;
GST_DEBUG_OBJECT (pad, "Sending segment done at end of segment");
if (self->segment.rate >= 0) {
m = gst_message_new_segment_done (GST_OBJECT_CAST (self),
GST_FORMAT_TIME, self->segment.stop);
e = gst_event_new_segment_done (GST_FORMAT_TIME, self->segment.stop);
} else {
m = gst_message_new_segment_done (GST_OBJECT_CAST (self),
GST_FORMAT_TIME, self->segment.start);
e = gst_event_new_segment_done (GST_FORMAT_TIME, self->segment.start);
}
gst_element_post_message (GST_ELEMENT_CAST (self), m);
gst_pad_push_event (self->srcpad, e);
} else {
GST_DEBUG_OBJECT (pad, "Sending EOS at end of segment");
gst_pad_push_event (self->srcpad, gst_event_new_eos ());
}
} else if (flow_ret == GST_FLOW_NOT_LINKED || flow_ret < GST_FLOW_EOS) {
GST_ELEMENT_ERROR (self, STREAM, FAILED,
("Internal data stream error."),
("stream stopped, reason %s", reason));
gst_pad_push_event (self->srcpad, gst_event_new_eos ());
}
return;
}
}
static void
gst_wavpack_parse_loop (GstElement * element)
{
GstWavpackParse *parse = GST_WAVPACK_PARSE (element);
GstFlowReturn flow_ret;
WavpackHeader header = { {0,}, 0, };
GstBuffer *buf = NULL;
flow_ret = gst_wavpack_parse_resync_loop (parse, &header);
if (flow_ret != GST_FLOW_OK)
goto pause;
GST_LOG_OBJECT (parse, "Read header at offset %" G_GINT64_FORMAT
": chunk size = %u+8", parse->current_offset, header.ckSize);
buf = gst_wavpack_parse_pull_buffer (parse, parse->current_offset,
header.ckSize + 8, &flow_ret);
if (flow_ret != GST_FLOW_OK)
goto pause;
if (parse->srcpad == NULL) {
if (!gst_wavpack_parse_create_src_pad (parse, buf, &header)) {
GST_ERROR_OBJECT (parse, "Failed to create src pad");
flow_ret = GST_FLOW_ERROR;
goto pause;
}
}
if (header.flags & INITIAL_BLOCK)
gst_wavpack_parse_index_append_entry (parse, parse->current_offset,
header.block_index, header.block_samples);
flow_ret = gst_wavpack_parse_push_buffer (parse, buf, &header);
if (flow_ret != GST_FLOW_OK)
goto pause;
return;
pause:
{
const gchar *reason = gst_flow_get_name (flow_ret);
GST_LOG_OBJECT (parse, "pausing task, reason %s", reason);
gst_pad_pause_task (parse->sinkpad);
if (GST_FLOW_IS_FATAL (flow_ret) || flow_ret == GST_FLOW_NOT_LINKED) {
if (flow_ret == GST_FLOW_UNEXPECTED && parse->srcpad) {
if (parse->segment.flags & GST_SEEK_FLAG_SEGMENT) {
GstClockTime stop;
GST_LOG_OBJECT (parse, "Sending segment done");
if ((stop = parse->segment.stop) == -1)
stop = parse->segment.duration;
gst_element_post_message (GST_ELEMENT_CAST (parse),
gst_message_new_segment_done (GST_OBJECT_CAST (parse),
parse->segment.format, stop));
} else {
GST_LOG_OBJECT (parse, "Sending EOS, at end of stream");
gst_pad_push_event (parse->srcpad, gst_event_new_eos ());
}
} else {
GST_ELEMENT_ERROR (parse, STREAM, FAILED,
(_("Internal data stream error.")),
("stream stopped, reason %s", reason));
if (parse->srcpad)
gst_pad_push_event (parse->srcpad, gst_event_new_eos ());
}
}
return;
}
}
static GstFlowReturn gst_nanomsgsrc_create(GstPushSrc *psrc, GstBuffer **buf)
{
gboolean try_again, flushing;
GstNanomsgSrc *nanomsgsrc = GST_NANOMSGSRC_CAST(psrc);
struct pollfd fds[2];
g_assert(nanomsgsrc->main_fd >= 0);
/* Fetch the current state of the flushing flag
* this can be modified in the unlock() function by
* another thread, so synchronize the access */
LOCK_SRC_MUTEX(nanomsgsrc);
flushing = nanomsgsrc->flushing;
UNLOCK_SRC_MUTEX(nanomsgsrc);
if (flushing)
return GST_FLOW_FLUSHING;
/* Set up structures for the poll() call below */
fds[0].fd = nanomsgsrc->ctrl_fds[0];
fds[0].events = POLLIN;
fds[0].revents = 0;
fds[1].fd = nanomsgsrc->watch_fd;
fds[1].events = POLLIN;
fds[1].revents = 0;
do
{
int ret;
try_again = FALSE;
ret = poll(fds, 2, (nanomsgsrc->timeout == 0) ? -1 : nanomsgsrc->timeout);
if (ret == -1)
{
/* poll() failed - exit loop and report error */
GST_ERROR_OBJECT(nanomsgsrc, "poll() error: %s", strerror(errno));
return GST_FLOW_ERROR;
}
else if (ret == 0)
{
/* Timeout was reached; post message and wait again for data */
GST_LOG_OBJECT(nanomsgsrc, "timeout reached - trying again");
try_again = TRUE;
gst_element_post_message(
GST_ELEMENT_CAST(nanomsgsrc),
gst_message_new_element(
GST_OBJECT_CAST(nanomsgsrc),
gst_structure_new(
"GstNanomsgSrcTimeout",
"timeout", G_TYPE_UINT64, nanomsgsrc->timeout,
NULL
)
)
);
}
else
{
/* At least one FD got data to read */
if (fds[0].revents & POLLIN)
{
/* Control pipe read end received a wakeup call
* Sometimes, unlock_stop() is called so quickly after unlock()
* that this loop isn't reached before the unlock_stop() call,
* which means that sometimes this wakeup call should actually
* be ignored, because the flushing period is over by this point.
* For this reason, when a wakeup call is received, the code
* checks if flushing is still actually required, and only if
* this is the case, the function exits with GST_FLOW_FLUSHING */
char dummy_buf[1024];
if (read(fds[0].fd, dummy_buf, sizeof(dummy_buf)) == -1)
GST_ERROR_OBJECT(nanomsgsrc, "error while receiving wakeup call over control pipe: %s", strerror(errno));
LOCK_SRC_MUTEX(nanomsgsrc);
flushing = nanomsgsrc->flushing;
UNLOCK_SRC_MUTEX(nanomsgsrc);
if (flushing)
return GST_FLOW_FLUSHING;
else
{
/* Set try_again to TRUE in case no flushing is
* actually needed and no data arrived yet */
try_again = TRUE;
}
}
/* NOTE: no "else if". Sometimes, while the unlock() call mentioned
* above is finished (instantly followed by the unlock_stop() call
* sometimes), data is immediately sent to the SP socket, which means
* that there is data to read. */
if (fds[1].revents & POLLIN)
{
GstBuffer *outbuf;
char *msg = NULL;
GstFlowReturn flow_ret;
int msg_size;
/* Get the message from the SP socket. NN_MSG means the returned
* msg pointer refers to a buffer allocated internally by nanomsg
* with the appropriate size. This is considered more efficient,
* since nanomsg may be using zerocopy mechanisms here (and not
* actually be allocating anything). Refer to the nn_recv()
* reference for details.
* Using nn_recv() with the NN_DONTWAIT flag, as an extra error check.
* If nn_send() fails because it cannot write anything, then something
* is wrong inside nanomsg, since we already waited with poll() for
* writing to be possible. Without the NN_DONTWAIT we wouldn't get an
* error, and block forever instead.*/
if ((msg_size = nn_recv(nanomsgsrc->main_fd, &msg, NN_MSG, NN_DONTWAIT)) < 0)
{
GST_ERROR_OBJECT(nanomsgsrc, "error while trying to receive data: %s", strerror(errno));
return GST_FLOW_ERROR;
}
/* Create a GstBuffer with the appropriate size */
if ((flow_ret = GST_BASE_SRC_CLASS(gst_nanomsgsrc_parent_class)->alloc(GST_BASE_SRC_CAST(psrc), -1, msg_size, &outbuf)) != GST_FLOW_OK)
{
GST_DEBUG_OBJECT(nanomsgsrc, "could not allocate new output buffer");
nn_freemsg(msg);
return flow_ret;
}
/* Copy the message's bytes over to the GstBuffer */
gst_buffer_fill(outbuf, 0, msg, msg_size);
/* Signal nanomsg that the message buffer isn't needed anymore */
nn_freemsg(msg);
/* Data *was* actually received, so make sure the loop is exited
* (try_again may have been set to TRUE after the control pipe
* received a wakeup call even though no flushing was required) */
try_again = FALSE;
/* Done */
*buf = outbuf;
}
}
}
while (G_UNLIKELY(try_again));
return GST_FLOW_OK;
}
static gboolean
gst_wavpack_parse_handle_seek_event (GstWavpackParse * wvparse,
GstEvent * event)
{
GstSeekFlags seek_flags;
GstSeekType start_type;
GstSeekType stop_type;
GstSegment segment;
GstFormat format;
gboolean only_update;
gboolean flush, ret;
gdouble speed;
gint64 stop;
gint64 start; /* sample we want to seek to */
gint64 byte_offset; /* byte offset the chunk we seek to starts at */
gint64 chunk_start; /* first sample in chunk we seek to */
guint rate;
gint64 last_stop;
if (wvparse->adapter) {
GST_DEBUG_OBJECT (wvparse, "seeking in streaming mode not implemented yet");
return FALSE;
}
gst_event_parse_seek (event, &speed, &format, &seek_flags, &start_type,
&start, &stop_type, &stop);
if (format != GST_FORMAT_DEFAULT && format != GST_FORMAT_TIME) {
GST_DEBUG ("seeking is only supported in TIME or DEFAULT format");
return FALSE;
}
if (speed < 0.0) {
GST_DEBUG ("only forward playback supported, rate %f not allowed", speed);
return FALSE;
}
GST_OBJECT_LOCK (wvparse);
rate = wvparse->samplerate;
if (rate == 0) {
GST_OBJECT_UNLOCK (wvparse);
GST_DEBUG ("haven't read header yet");
return FALSE;
}
/* figure out the last position we need to play. If it's configured (stop !=
* -1), use that, else we play until the total duration of the file */
if (stop == -1)
stop = wvparse->segment.duration;
/* convert from time to samples if necessary */
if (format == GST_FORMAT_TIME) {
if (start_type != GST_SEEK_TYPE_NONE)
start = gst_util_uint64_scale_int (start, rate, GST_SECOND);
if (stop_type != GST_SEEK_TYPE_NONE)
stop = gst_util_uint64_scale_int (stop, rate, GST_SECOND);
}
if (start < 0) {
GST_OBJECT_UNLOCK (wvparse);
GST_DEBUG_OBJECT (wvparse, "Invalid start sample %" G_GINT64_FORMAT, start);
return FALSE;
}
flush = ((seek_flags & GST_SEEK_FLAG_FLUSH) != 0);
/* operate on segment copy until we know the seek worked */
segment = wvparse->segment;
gst_segment_set_seek (&segment, speed, GST_FORMAT_DEFAULT,
seek_flags, start_type, start, stop_type, stop, &only_update);
#if 0
if (only_update) {
wvparse->segment = segment;
gst_wavpack_parse_send_newsegment (wvparse, TRUE);
goto done;
}
#endif
gst_pad_push_event (wvparse->sinkpad, gst_event_new_flush_start ());
if (flush) {
gst_pad_push_event (wvparse->srcpad, gst_event_new_flush_start ());
} else {
gst_pad_pause_task (wvparse->sinkpad);
}
GST_PAD_STREAM_LOCK (wvparse->sinkpad);
/* Save current position */
last_stop = wvparse->segment.last_stop;
gst_pad_push_event (wvparse->sinkpad, gst_event_new_flush_stop ());
if (flush) {
gst_pad_push_event (wvparse->srcpad, gst_event_new_flush_stop ());
}
GST_DEBUG_OBJECT (wvparse, "Performing seek to %" GST_TIME_FORMAT " sample %"
G_GINT64_FORMAT, GST_TIME_ARGS (segment.start * GST_SECOND / rate),
start);
ret = gst_wavpack_parse_scan_to_find_sample (wvparse, segment.start,
&byte_offset, &chunk_start);
if (ret) {
GST_DEBUG_OBJECT (wvparse, "new offset: %" G_GINT64_FORMAT, byte_offset);
wvparse->current_offset = byte_offset;
/* we want to send a newsegment event with the actual seek position
* as start, even though our first buffer might start before the
* configured segment. We leave it up to the decoder or sink to crop
* the output buffers accordingly */
wvparse->segment = segment;
wvparse->segment.last_stop = chunk_start;
wvparse->need_newsegment = TRUE;
wvparse->discont = (last_stop != chunk_start) ? TRUE : FALSE;
/* if we're doing a segment seek, post a SEGMENT_START message */
if (wvparse->segment.flags & GST_SEEK_FLAG_SEGMENT) {
gst_element_post_message (GST_ELEMENT_CAST (wvparse),
gst_message_new_segment_start (GST_OBJECT_CAST (wvparse),
wvparse->segment.format, wvparse->segment.last_stop));
}
} else {
GST_DEBUG_OBJECT (wvparse, "seek failed: don't know where to seek to");
}
GST_PAD_STREAM_UNLOCK (wvparse->sinkpad);
GST_OBJECT_UNLOCK (wvparse);
gst_pad_start_task (wvparse->sinkpad,
(GstTaskFunction) gst_wavpack_parse_loop, wvparse);
return ret;
}
EXPORT_C
#endif
gchar *
gst_object_get_path_string (GstObject * object)
{
GSList *parentage;
GSList *parents;
void *parent;
gchar *prevpath, *path;
gchar *component;
gchar *separator;
/* ref object before adding to list */
gst_object_ref (object);
parentage = g_slist_prepend (NULL, object);
path = g_strdup ("");
/* first walk the object hierarchy to build a list of the parents,
* be carefull here with refcounting. */
do {
if (GST_IS_OBJECT (object)) {
parent = gst_object_get_parent (object);
/* add parents to list, refcount remains increased while
* we handle the object */
if (parent)
parentage = g_slist_prepend (parentage, parent);
} else {
break;
}
object = parent;
} while (object != NULL);
/* then walk the parent list and print them out. we need to
* decrease the refcounting on each element after we handled
* it. */
for (parents = parentage; parents; parents = g_slist_next (parents)) {
if (GST_IS_OBJECT (parents->data)) {
GstObject *item = GST_OBJECT_CAST (parents->data);
GstObjectClass *oclass = GST_OBJECT_GET_CLASS (item);
component = gst_object_get_name (item);
separator = oclass->path_string_separator;
/* and unref now */
gst_object_unref (item);
} else {
component = g_strdup_printf ("%p", parents->data);
separator = "/";
}
prevpath = path;
path = g_strjoin (separator, prevpath, component, NULL);
g_free (prevpath);
g_free (component);
}
g_slist_free (parentage);
return path;
}
/* this internal thread does nothing else but write samples to the audio device.
* It will write each segment in the ringbuffer and will update the play
* pointer.
* The start/stop methods control the thread.
*/
static void
audioringbuffer_thread_func (GstRingBuffer * buf)
{
GstAudioSink *sink;
GstAudioSinkClass *csink;
GstAudioRingBuffer *abuf = GST_AUDIORING_BUFFER_CAST (buf);
WriteFunc writefunc;
GstMessage *message;
GValue val = { 0 };
sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf));
csink = GST_AUDIO_SINK_GET_CLASS (sink);
GST_DEBUG_OBJECT (sink, "enter thread");
GST_OBJECT_LOCK (abuf);
GST_DEBUG_OBJECT (sink, "signal wait");
GST_AUDIORING_BUFFER_SIGNAL (buf);
GST_OBJECT_UNLOCK (abuf);
writefunc = csink->write;
if (writefunc == NULL)
goto no_function;
g_value_init (&val, G_TYPE_POINTER);
g_value_set_pointer (&val, sink->thread);
message = gst_message_new_stream_status (GST_OBJECT_CAST (buf),
GST_STREAM_STATUS_TYPE_ENTER, GST_ELEMENT_CAST (sink));
gst_message_set_stream_status_object (message, &val);
GST_DEBUG_OBJECT (sink, "posting ENTER stream status");
gst_element_post_message (GST_ELEMENT_CAST (sink), message);
while (TRUE) {
gint left, len;
guint8 *readptr;
gint readseg;
/* buffer must be started */
if (gst_ring_buffer_prepare_read (buf, &readseg, &readptr, &len)) {
gint written;
left = len;
do {
written = writefunc (sink, readptr, left);
GST_LOG_OBJECT (sink, "transfered %d bytes of %d from segment %d",
written, left, readseg);
if (written < 0 || written > left) {
/* might not be critical, it e.g. happens when aborting playback */
GST_WARNING_OBJECT (sink,
"error writing data in %s (reason: %s), skipping segment (left: %d, written: %d)",
GST_DEBUG_FUNCPTR_NAME (writefunc),
(errno > 1 ? g_strerror (errno) : "unknown"), left, written);
break;
}
left -= written;
readptr += written;
} while (left > 0);
/* clear written samples */
gst_ring_buffer_clear (buf, readseg);
/* we wrote one segment */
gst_ring_buffer_advance (buf, 1);
} else {
GST_OBJECT_LOCK (abuf);
if (!abuf->running)
goto stop_running;
GST_DEBUG_OBJECT (sink, "signal wait");
GST_AUDIORING_BUFFER_SIGNAL (buf);
GST_DEBUG_OBJECT (sink, "wait for action");
#ifndef GSTREAMER_LITE
GST_AUDIORING_BUFFER_WAIT (buf);
#else // GSTREAMER_LITE
// In same cases we may have condition when we waiting here for ring buffer to start,
// while ring buffer is started and data is available. So, lets use wait with timeout
// and recheck if we good to go. wait_segment() will start ring buffer when data is available.
{
GTimeVal timeout;
g_get_current_time(&timeout);
g_time_val_add(&timeout, 100000); // 100 millisecond
GST_AUDIORING_BUFFER_TIMED_WAIT (buf, &timeout);
}
#endif // GSTREAMER_LITE
GST_DEBUG_OBJECT (sink, "got signal");
if (!abuf->running)
goto stop_running;
GST_DEBUG_OBJECT (sink, "continue running");
GST_OBJECT_UNLOCK (abuf);
}
}
/* Will never be reached */
g_assert_not_reached ();
return;
/* ERROR */
no_function:
{
GST_DEBUG_OBJECT (sink, "no write function, exit thread");
return;
}
stop_running:
{
GST_OBJECT_UNLOCK (abuf);
GST_DEBUG_OBJECT (sink, "stop running, exit thread");
message = gst_message_new_stream_status (GST_OBJECT_CAST (buf),
GST_STREAM_STATUS_TYPE_LEAVE, GST_ELEMENT_CAST (sink));
gst_message_set_stream_status_object (message, &val);
GST_DEBUG_OBJECT (sink, "posting LEAVE stream status");
gst_element_post_message (GST_ELEMENT_CAST (sink), message);
return;
}
}
GstStateChangeReturn
gst_sandboxed_decodebin_change_state (GstElement *element,
GstStateChange state_change)
{
GError *error = NULL;
GstSandboxedDecodebinPrivate *priv;
GstSandboxedDecodebin *self = GST_SANDBOXED_DECODEBIN (element);
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
priv = GST_SANDBOXED_DECODEBIN_GET_PRIVATE (self);
switch (state_change) {
case GST_STATE_CHANGE_NULL_TO_READY:
/* TODO: set up file monitoring */
/* spawn subprocess */
priv->subprocess_stdin = start_decoder (priv->shm_audio_socket_path,
priv->shm_video_socket_path,
&error);
if (priv->subprocess_stdin == -1) {
GST_WARNING_OBJECT (element,
"Could not spawn subprocess: %s", error->message);
ret = GST_STATE_CHANGE_FAILURE;
}
/* set the right fd to fdsink */
g_object_set (priv->fdsink, "fd", priv->subprocess_stdin, NULL);
GST_DEBUG_OBJECT (element, "Waiting for shm sockets to be available\n");
while (!self->priv->subprocess_ready) {
/* does that count as acceptable code? The alternatives are another
* thread to monitor file creation or sleep() */
g_main_context_iteration (NULL, TRUE);
}
GST_DEBUG_OBJECT (element, "Done waiting\n");
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
GST_DEBUG_OBJECT (element, "Going to PAUSED");
break;
#if 0
case GST_STATE_CHANGE_READY_TO_PAUSED:
/* TODO: check subprocess is ready with the help of file monitoring, if
* not, do the change asynchronously */
{
GstMessage *message;
ret = GST_STATE_CHANGE_ASYNC;
message = gst_message_new_async_start (GST_OBJECT_CAST (element), FALSE);
parent_class->handle_message (GST_BIN (element), message);
/* Yeah, this is a hack */
g_timeout_add (2000, do_async_done, self);
}
break;
#endif
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
GST_DEBUG_OBJECT (element, "Going to PLAYING");
break;
default:
break;
}
if (ret != GST_STATE_CHANGE_FAILURE) {
GstStateChangeReturn bret;
bret = GST_ELEMENT_CLASS (parent_class)->change_state (element, state_change);
if (bret == GST_STATE_CHANGE_FAILURE) {
GST_WARNING_OBJECT (element, "parent change_state failed!");
ret = bret;
}
}
if (ret != GST_STATE_CHANGE_FAILURE) {
GstStateChangeReturn fdret;
gint fd;
switch (state_change) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
GST_DEBUG_OBJECT (element, "Trying to set fdsink to PLAYING");
fdret = gst_element_set_state (priv->fdsink, GST_STATE_PLAYING);
GST_DEBUG_OBJECT (element, "Returned: %s",
gst_element_state_change_return_get_name (fdret));
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
g_object_get (priv->audiosrc,
"shm-area-name", &priv->audio_shm_area_name, NULL);
g_object_get (priv->videosrc,
"shm-area-name", &priv->video_shm_area_name, NULL);
GST_DEBUG_OBJECT (element, "Name of audio/video shm areas: "
"\"%s\" and \"%s\"\n",
priv->audio_shm_area_name,
priv->video_shm_area_name);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
/* Closing the fd sounds like a polite thing to do now*/
g_object_get (priv->fdsink, "fd", &fd, NULL);
close (fd);
/* Unlinking the stuff the decoder could not unlink because it doesn't
* have the necessary privileges */
GST_DEBUG_OBJECT (element, "Trying to unlink %s and %s",
priv->shm_video_socket_path,
priv->shm_audio_socket_path);
if (-1 == g_unlink (priv->shm_video_socket_path))
GST_WARNING_OBJECT (element, "Could not unlink %s: %m", priv->shm_video_socket_path);
if (-1 == g_unlink (priv->shm_audio_socket_path))
GST_WARNING_OBJECT (element, "Could not unlink %s: %m", priv->shm_audio_socket_path);
GST_DEBUG_OBJECT (element,
"Trying to unlink audio/video shm areas: \"%s\" and \"%s\"\n",
priv->audio_shm_area_name,
priv->video_shm_area_name);
if (-1 == shm_unlink (priv->audio_shm_area_name))
GST_WARNING_OBJECT (element, "Could not unlink shm area %s: %m",
priv->audio_shm_area_name);
if (-1 == shm_unlink (priv->video_shm_area_name))
GST_WARNING_OBJECT (element, "Could not unlink shm area %s: %m",
priv->video_shm_area_name);
/* FIXME: where do we free all these strings? */
break;
default:
break;
}
}
return ret;
}
static void
gst_qt_moov_recover_run (void *data)
{
FILE *moovrec = NULL;
FILE *mdatinput = NULL;
FILE *output = NULL;
MdatRecovFile *mdat_recov = NULL;
MoovRecovFile *moov_recov = NULL;
GstQTMoovRecover *qtmr = GST_QT_MOOV_RECOVER_CAST (data);
GError *err = NULL;
GST_LOG_OBJECT (qtmr, "Starting task");
GST_DEBUG_OBJECT (qtmr, "Validating properties");
GST_OBJECT_LOCK (qtmr);
/* validate properties */
if (qtmr->broken_input == NULL) {
GST_OBJECT_UNLOCK (qtmr);
GST_ELEMENT_ERROR (qtmr, RESOURCE, SETTINGS,
("Please set broken-input property"), (NULL));
goto end;
}
if (qtmr->recovery_input == NULL) {
GST_OBJECT_UNLOCK (qtmr);
GST_ELEMENT_ERROR (qtmr, RESOURCE, SETTINGS,
("Please set recovery-input property"), (NULL));
goto end;
}
if (qtmr->fixed_output == NULL) {
GST_OBJECT_UNLOCK (qtmr);
GST_ELEMENT_ERROR (qtmr, RESOURCE, SETTINGS,
("Please set fixed-output property"), (NULL));
goto end;
}
GST_DEBUG_OBJECT (qtmr, "Opening input/output files");
/* open files */
moovrec = g_fopen (qtmr->recovery_input, "rb");
if (moovrec == NULL) {
GST_OBJECT_UNLOCK (qtmr);
GST_ELEMENT_ERROR (qtmr, RESOURCE, OPEN_READ,
("Failed to open recovery-input file"), (NULL));
goto end;
}
mdatinput = g_fopen (qtmr->broken_input, "rb");
if (mdatinput == NULL) {
GST_OBJECT_UNLOCK (qtmr);
GST_ELEMENT_ERROR (qtmr, RESOURCE, OPEN_READ,
("Failed to open broken-input file"), (NULL));
goto end;
}
output = g_fopen (qtmr->fixed_output, "wb+");
if (output == NULL) {
GST_OBJECT_UNLOCK (qtmr);
GST_ELEMENT_ERROR (qtmr, RESOURCE, OPEN_READ_WRITE,
("Failed to open fixed-output file"), (NULL));
goto end;
}
GST_OBJECT_UNLOCK (qtmr);
GST_DEBUG_OBJECT (qtmr, "Parsing input files");
/* now create our structures */
mdat_recov = mdat_recov_file_create (mdatinput, qtmr->faststart_mode, &err);
mdatinput = NULL;
if (mdat_recov == NULL) {
GST_ELEMENT_ERROR (qtmr, RESOURCE, FAILED,
("Broken file could not be parsed correctly"), (NULL));
goto end;
}
moov_recov = moov_recov_file_create (moovrec, &err);
moovrec = NULL;
if (moov_recov == NULL) {
GST_ELEMENT_ERROR (qtmr, RESOURCE, FAILED,
("Recovery file could not be parsed correctly"), (NULL));
goto end;
}
/* now parse the buffers data from moovrec */
if (!moov_recov_parse_buffers (moov_recov, mdat_recov, &err)) {
goto end;
}
GST_DEBUG_OBJECT (qtmr, "Writing fixed file to output");
if (!moov_recov_write_file (moov_recov, mdat_recov, output, &err)) {
goto end;
}
/* here means success */
GST_DEBUG_OBJECT (qtmr, "Finished successfully, posting EOS");
gst_element_post_message (GST_ELEMENT_CAST (qtmr),
gst_message_new_eos (GST_OBJECT_CAST (qtmr)));
end:
GST_LOG_OBJECT (qtmr, "Finalizing task");
if (err) {
GST_ELEMENT_ERROR (qtmr, RESOURCE, FAILED, ("%s", err->message), (NULL));
g_error_free (err);
}
if (moov_recov)
moov_recov_file_free (moov_recov);
if (moovrec)
fclose (moovrec);
if (mdat_recov)
mdat_recov_file_free (mdat_recov);
if (mdatinput)
fclose (mdatinput);
if (output)
fclose (output);
GST_LOG_OBJECT (qtmr, "Leaving task");
gst_task_stop (qtmr->task);
}
gint
main (gint argc, gchar ** argv)
{
GstElement *pipeline;
GstElement *shapewipe;
GstControlSource *cs;
GMainLoop *loop;
GstBus *bus;
gchar *pipeline_string;
gfloat border = 0.05;
if (argc < 2) {
g_print ("Usage: shapewipe mask.png <border>\n");
return -1;
}
gst_init (&argc, &argv);
if (argc > 2) {
border = atof (argv[2]);
}
pipeline_string =
g_strdup_printf
("videotestsrc ! video/x-raw,format=(string)AYUV,width=640,height=480 ! shapewipe name=shape border=%f ! videomixer name=mixer ! videoconvert ! autovideosink filesrc location=%s ! typefind ! decodebin2 ! videoconvert ! videoscale ! queue ! shape.mask_sink videotestsrc pattern=snow ! video/x-raw,format=(string)AYUV,width=640,height=480 ! queue ! mixer.",
border, argv[1]);
pipeline = gst_parse_launch (pipeline_string, NULL);
g_free (pipeline_string);
if (pipeline == NULL) {
g_print ("Failed to create pipeline\n");
return -2;
}
shapewipe = gst_bin_get_by_name (GST_BIN (pipeline), "shape");
cs = gst_lfo_control_source_new ();
gst_object_add_control_binding (GST_OBJECT_CAST (shapewipe),
gst_direct_control_binding_new (GST_OBJECT_CAST (shapewipe), "position",
cs));
g_object_set (cs,
"amplitude", 0.5,
"offset", 0.5, "frequency", 0.25, "timeshift", 500 * GST_MSECOND, NULL);
g_object_unref (cs);
loop = g_main_loop_new (NULL, FALSE);
bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
gst_bus_add_signal_watch (bus);
g_signal_connect (G_OBJECT (bus), "message", G_CALLBACK (on_message), loop);
gst_object_unref (GST_OBJECT (bus));
if (gst_element_set_state (pipeline,
GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE) {
g_error ("Failed to go into PLAYING state");
return -4;
}
g_main_loop_run (loop);
gst_element_set_state (pipeline, GST_STATE_NULL);
g_main_loop_unref (loop);
gst_object_unref (G_OBJECT (pipeline));
return 0;
}
static void
gst_type_find_element_loop (GstPad * pad)
{
GstTypeFindElement *typefind;
GstFlowReturn ret = GST_FLOW_OK;
typefind = GST_TYPE_FIND_ELEMENT (GST_PAD_PARENT (pad));
if (typefind->need_stream_start) {
gchar *stream_id;
GstEvent *event;
stream_id = gst_pad_create_stream_id (typefind->src,
GST_ELEMENT_CAST (typefind), NULL);
GST_DEBUG_OBJECT (typefind, "Pushing STREAM_START");
event = gst_event_new_stream_start (stream_id);
gst_event_set_group_id (event, gst_util_group_id_next ());
gst_pad_push_event (typefind->src, event);
typefind->need_stream_start = FALSE;
g_free (stream_id);
}
if (typefind->mode == MODE_TYPEFIND) {
GstPad *peer = NULL;
GstCaps *found_caps = NULL;
GstTypeFindProbability probability = GST_TYPE_FIND_NONE;
GST_DEBUG_OBJECT (typefind, "find type in pull mode");
GST_OBJECT_LOCK (typefind);
if (typefind->force_caps) {
found_caps = gst_caps_ref (typefind->force_caps);
probability = GST_TYPE_FIND_MAXIMUM;
}
GST_OBJECT_UNLOCK (typefind);
if (!found_caps) {
peer = gst_pad_get_peer (pad);
if (peer) {
gint64 size;
gchar *ext;
if (!gst_pad_query_duration (peer, GST_FORMAT_BYTES, &size)) {
GST_WARNING_OBJECT (typefind, "Could not query upstream length!");
gst_object_unref (peer);
ret = GST_FLOW_ERROR;
goto pause;
}
/* the size if 0, we cannot continue */
if (size == 0) {
/* keep message in sync with message in sink event handler */
GST_ELEMENT_ERROR (typefind, STREAM, TYPE_NOT_FOUND,
(_("Stream contains no data.")), ("Can't typefind empty stream"));
gst_object_unref (peer);
ret = GST_FLOW_ERROR;
goto pause;
}
ext = gst_type_find_get_extension (typefind, pad);
found_caps =
gst_type_find_helper_get_range (GST_OBJECT_CAST (peer),
GST_OBJECT_PARENT (peer),
(GstTypeFindHelperGetRangeFunction) (GST_PAD_GETRANGEFUNC (peer)),
(guint64) size, ext, &probability);
g_free (ext);
GST_DEBUG ("Found caps %" GST_PTR_FORMAT, found_caps);
gst_object_unref (peer);
}
}
if (!found_caps || probability < typefind->min_probability) {
GST_DEBUG ("Trying to guess using extension");
gst_caps_replace (&found_caps, NULL);
found_caps =
gst_type_find_guess_by_extension (typefind, pad, &probability);
}
if (!found_caps || probability < typefind->min_probability) {
GST_ELEMENT_ERROR (typefind, STREAM, TYPE_NOT_FOUND, (NULL), (NULL));
gst_caps_replace (&found_caps, NULL);
ret = GST_FLOW_ERROR;
goto pause;
}
GST_DEBUG ("Emiting found caps %" GST_PTR_FORMAT, found_caps);
gst_type_find_element_emit_have_type (typefind, probability, found_caps);
typefind->mode = MODE_NORMAL;
gst_caps_unref (found_caps);
} else if (typefind->mode == MODE_NORMAL) {
GstBuffer *outbuf = NULL;
if (typefind->need_segment) {
typefind->need_segment = FALSE;
gst_pad_push_event (typefind->src,
gst_event_new_segment (&typefind->segment));
}
/* Pull 4k blocks and send downstream */
ret = gst_pad_pull_range (typefind->sink, typefind->offset, 4096, &outbuf);
if (ret != GST_FLOW_OK)
goto pause;
typefind->offset += gst_buffer_get_size (outbuf);
ret = gst_pad_push (typefind->src, outbuf);
if (ret != GST_FLOW_OK)
goto pause;
} else {
/* Error out */
ret = GST_FLOW_ERROR;
goto pause;
}
return;
pause:
{
const gchar *reason = gst_flow_get_name (ret);
gboolean push_eos = FALSE;
GST_LOG_OBJECT (typefind, "pausing task, reason %s", reason);
gst_pad_pause_task (typefind->sink);
if (ret == GST_FLOW_EOS) {
/* perform EOS logic */
if (typefind->segment.flags & GST_SEGMENT_FLAG_SEGMENT) {
gint64 stop;
/* for segment playback we need to post when (in stream time)
* we stopped, this is either stop (when set) or the duration. */
if ((stop = typefind->segment.stop) == -1)
stop = typefind->offset;
GST_LOG_OBJECT (typefind, "Sending segment done, at end of segment");
gst_element_post_message (GST_ELEMENT (typefind),
gst_message_new_segment_done (GST_OBJECT (typefind),
GST_FORMAT_BYTES, stop));
gst_pad_push_event (typefind->src,
gst_event_new_segment_done (GST_FORMAT_BYTES, stop));
} else {
push_eos = TRUE;
}
} else if (ret == GST_FLOW_NOT_LINKED || ret < GST_FLOW_EOS) {
/* for fatal errors we post an error message */
GST_ELEMENT_FLOW_ERROR (typefind, ret);
push_eos = TRUE;
}
if (push_eos) {
/* send EOS, and prevent hanging if no streams yet */
GST_LOG_OBJECT (typefind, "Sending EOS, at end of stream");
gst_pad_push_event (typefind->src, gst_event_new_eos ());
}
return;
}
}
/**
* gst_ghost_pad_construct:
* @gpad: the newly allocated ghost pad
*
* Finish initialization of a newly allocated ghost pad.
*
* This function is most useful in language bindings and when subclassing
* #GstGhostPad; plugin and application developers normally will not call this
* function. Call this function directly after a call to g_object_new
* (GST_TYPE_GHOST_PAD, "direction", @dir, ..., NULL).
*
* Returns: %TRUE if the construction succeeds, %FALSE otherwise.
*/
gboolean
gst_ghost_pad_construct (GstGhostPad * gpad)
{
GstPadDirection dir, otherdir;
GstPadTemplate *templ;
GstPad *pad, *internal;
g_return_val_if_fail (GST_IS_GHOST_PAD (gpad), FALSE);
g_return_val_if_fail (GST_GHOST_PAD_PRIVATE (gpad)->constructed == FALSE,
FALSE);
g_object_get (gpad, "direction", &dir, "template", &templ, NULL);
g_return_val_if_fail (dir != GST_PAD_UNKNOWN, FALSE);
pad = GST_PAD (gpad);
/* Set directional padfunctions for ghostpad */
if (dir == GST_PAD_SINK) {
gst_pad_set_chain_function (pad, gst_proxy_pad_chain_default);
gst_pad_set_chain_list_function (pad, gst_proxy_pad_chain_list_default);
} else {
gst_pad_set_getrange_function (pad, gst_proxy_pad_getrange_default);
}
/* INTERNAL PAD, it always exists and is child of the ghostpad */
otherdir = (dir == GST_PAD_SRC) ? GST_PAD_SINK : GST_PAD_SRC;
if (templ) {
internal =
g_object_new (GST_TYPE_PROXY_PAD, "name", NULL,
"direction", otherdir, "template", templ, NULL);
/* release ref obtained via g_object_get */
gst_object_unref (templ);
} else {
internal =
g_object_new (GST_TYPE_PROXY_PAD, "name", NULL,
"direction", otherdir, NULL);
}
GST_PAD_UNSET_FLUSHING (internal);
/* Set directional padfunctions for internal pad */
if (dir == GST_PAD_SRC) {
gst_pad_set_chain_function (internal, gst_proxy_pad_chain_default);
gst_pad_set_chain_list_function (internal,
gst_proxy_pad_chain_list_default);
} else {
gst_pad_set_getrange_function (internal, gst_proxy_pad_getrange_default);
}
GST_OBJECT_LOCK (pad);
/* now make the ghostpad a parent of the internal pad */
if (!gst_object_set_parent (GST_OBJECT_CAST (internal),
GST_OBJECT_CAST (pad)))
goto parent_failed;
/* The ghostpad is the parent of the internal pad and is the only object that
* can have a refcount on the internal pad.
* At this point, the GstGhostPad has a refcount of 1, and the internal pad has
* a refcount of 1.
* When the refcount of the GstGhostPad drops to 0, the ghostpad will dispose
* its refcount on the internal pad in the dispose method by un-parenting it.
* This is why we don't take extra refcounts in the assignments below
*/
GST_PROXY_PAD_INTERNAL (pad) = internal;
GST_PROXY_PAD_INTERNAL (internal) = pad;
/* special activation functions for the internal pad */
gst_pad_set_activatemode_function (internal,
gst_ghost_pad_internal_activate_mode_default);
GST_OBJECT_UNLOCK (pad);
GST_GHOST_PAD_PRIVATE (gpad)->constructed = TRUE;
return TRUE;
/* ERRORS */
parent_failed:
{
GST_WARNING_OBJECT (gpad, "Could not set internal pad %s:%s",
GST_DEBUG_PAD_NAME (internal));
g_critical ("Could not set internal pad %s:%s",
GST_DEBUG_PAD_NAME (internal));
GST_OBJECT_UNLOCK (pad);
gst_object_unref (internal);
return FALSE;
}
}
static gboolean
gst_raw_parse_handle_seek_pull (GstRawParse * rp, GstEvent * event)
{
gdouble rate;
GstFormat format;
GstSeekFlags flags;
GstSeekType start_type, stop_type;
gint64 start, stop;
gint64 last_stop;
gboolean ret = FALSE;
gboolean flush;
GstSegment seeksegment;
if (event) {
gst_event_parse_seek (event, &rate, &format, &flags, &start_type, &start,
&stop_type, &stop);
/* convert input offsets to time */
ret = gst_raw_parse_convert (rp, format, start, GST_FORMAT_TIME, &start);
ret &= gst_raw_parse_convert (rp, format, stop, GST_FORMAT_TIME, &stop);
if (!ret)
goto convert_failed;
GST_DEBUG_OBJECT (rp, "converted start - stop to time");
format = GST_FORMAT_TIME;
gst_event_unref (event);
} else {
format = GST_FORMAT_TIME;
flags = 0;
}
flush = ((flags & GST_SEEK_FLAG_FLUSH) != 0);
/* start flushing up and downstream so that the loop function pauses and we
* can acquire the STREAM_LOCK. */
if (flush) {
GST_LOG_OBJECT (rp, "flushing");
gst_pad_push_event (rp->sinkpad, gst_event_new_flush_start ());
gst_pad_push_event (rp->srcpad, gst_event_new_flush_start ());
} else {
GST_LOG_OBJECT (rp, "pause task");
gst_pad_pause_task (rp->sinkpad);
}
GST_PAD_STREAM_LOCK (rp->sinkpad);
memcpy (&seeksegment, &rp->segment, sizeof (GstSegment));
if (event) {
/* configure the seek values */
gst_segment_set_seek (&seeksegment, rate, format, flags,
start_type, start, stop_type, stop, NULL);
}
/* get the desired position */
last_stop = seeksegment.last_stop;
GST_LOG_OBJECT (rp, "seeking to %" GST_TIME_FORMAT,
GST_TIME_ARGS (last_stop));
/* convert the desired position to bytes */
ret =
gst_raw_parse_convert (rp, format, last_stop, GST_FORMAT_BYTES,
&last_stop);
/* prepare for streaming */
if (flush) {
GST_LOG_OBJECT (rp, "stop flush");
gst_pad_push_event (rp->sinkpad, gst_event_new_flush_stop ());
gst_pad_push_event (rp->srcpad, gst_event_new_flush_stop ());
} else if (ret && rp->running) {
/* we are running the current segment and doing a non-flushing seek,
* close the segment first based on the last_stop. */
GST_DEBUG_OBJECT (rp, "prepare close segment %" G_GINT64_FORMAT
" to %" G_GINT64_FORMAT, rp->segment.start, rp->segment.last_stop);
/* queue the segment for sending in the stream thread */
if (rp->close_segment)
gst_event_unref (rp->close_segment);
rp->close_segment =
gst_event_new_new_segment_full (TRUE,
rp->segment.rate, rp->segment.applied_rate, rp->segment.format,
rp->segment.start, rp->segment.last_stop, rp->segment.time);
}
if (ret) {
/* seek done */
/* Seek on a frame boundary */
last_stop -= last_stop % rp->framesize;
rp->offset = last_stop;
rp->n_frames = last_stop / rp->framesize;
GST_LOG_OBJECT (rp, "seeking to bytes %" G_GINT64_FORMAT, last_stop);
memcpy (&rp->segment, &seeksegment, sizeof (GstSegment));
if (rp->segment.flags & GST_SEEK_FLAG_SEGMENT) {
gst_element_post_message (GST_ELEMENT_CAST (rp),
gst_message_new_segment_start (GST_OBJECT_CAST (rp),
rp->segment.format, rp->segment.last_stop));
}
/* for deriving a stop position for the playback segment from the seek
* segment, we must take the duration when the stop is not set */
if ((stop = rp->segment.stop) == -1)
stop = rp->segment.duration;
GST_DEBUG_OBJECT (rp, "preparing newsegment from %" G_GINT64_FORMAT
" to %" G_GINT64_FORMAT, rp->segment.start, stop);
/* now replace the old segment so that we send it in the stream thread the
* next time it is scheduled. */
if (rp->start_segment)
gst_event_unref (rp->start_segment);
if (rp->segment.rate >= 0.0) {
/* forward, we send data from last_stop to stop */
rp->start_segment =
gst_event_new_new_segment_full (FALSE,
rp->segment.rate, rp->segment.applied_rate, rp->segment.format,
rp->segment.last_stop, stop, rp->segment.time);
} else {
/* reverse, we send data from last_stop to start */
rp->start_segment =
gst_event_new_new_segment_full (FALSE,
rp->segment.rate, rp->segment.applied_rate, rp->segment.format,
rp->segment.start, rp->segment.last_stop, rp->segment.time);
}
}
rp->discont = TRUE;
GST_LOG_OBJECT (rp, "start streaming");
rp->running = TRUE;
gst_pad_start_task (rp->sinkpad, (GstTaskFunction) gst_raw_parse_loop, rp);
GST_PAD_STREAM_UNLOCK (rp->sinkpad);
return ret;
/* ERRORS */
convert_failed:
{
GST_DEBUG_OBJECT (rp, "Seek failed: couldn't convert to byte positions");
return FALSE;
}
}
static gboolean
gst_hls_demux_change_playlist (GstHLSDemux * demux, guint max_bitrate,
gboolean * changed)
{
GList *previous_variant, *current_variant;
gint old_bandwidth, new_bandwidth;
GstAdaptiveDemux *adaptive_demux = GST_ADAPTIVE_DEMUX_CAST (demux);
GstAdaptiveDemuxStream *stream;
g_return_val_if_fail (adaptive_demux->streams != NULL, FALSE);
stream = adaptive_demux->streams->data;
previous_variant = demux->client->main->current_variant;
current_variant = gst_m3u8_client_get_playlist_for_bitrate (demux->client,
max_bitrate);
GST_M3U8_CLIENT_LOCK (demux->client);
retry_failover_protection:
old_bandwidth = GST_M3U8 (previous_variant->data)->bandwidth;
new_bandwidth = GST_M3U8 (current_variant->data)->bandwidth;
/* Don't do anything else if the playlist is the same */
if (new_bandwidth == old_bandwidth) {
GST_M3U8_CLIENT_UNLOCK (demux->client);
return TRUE;
}
demux->client->main->current_variant = current_variant;
GST_M3U8_CLIENT_UNLOCK (demux->client);
gst_m3u8_client_set_current (demux->client, current_variant->data);
GST_INFO_OBJECT (demux, "Client was on %dbps, max allowed is %dbps, switching"
" to bitrate %dbps", old_bandwidth, max_bitrate, new_bandwidth);
stream->discont = TRUE;
if (gst_hls_demux_update_playlist (demux, FALSE, NULL)) {
gchar *uri;
gchar *main_uri;
uri = gst_m3u8_client_get_current_uri (demux->client);
main_uri = gst_m3u8_client_get_uri (demux->client);
gst_element_post_message (GST_ELEMENT_CAST (demux),
gst_message_new_element (GST_OBJECT_CAST (demux),
gst_structure_new (GST_ADAPTIVE_DEMUX_STATISTICS_MESSAGE_NAME,
"manifest-uri", G_TYPE_STRING,
main_uri, "uri", G_TYPE_STRING,
uri, "bitrate", G_TYPE_INT, new_bandwidth, NULL)));
g_free (uri);
g_free (main_uri);
if (changed)
*changed = TRUE;
} else {
GList *failover = NULL;
GST_INFO_OBJECT (demux, "Unable to update playlist. Switching back");
GST_M3U8_CLIENT_LOCK (demux->client);
failover = g_list_previous (current_variant);
if (failover && new_bandwidth == GST_M3U8 (failover->data)->bandwidth) {
current_variant = failover;
goto retry_failover_protection;
}
demux->client->main->current_variant = previous_variant;
GST_M3U8_CLIENT_UNLOCK (demux->client);
gst_m3u8_client_set_current (demux->client, previous_variant->data);
/* Try a lower bitrate (or stop if we just tried the lowest) */
if (GST_M3U8 (previous_variant->data)->iframe && new_bandwidth ==
GST_M3U8 (g_list_first (demux->client->main->iframe_lists)->data)->
bandwidth)
return FALSE;
else if (!GST_M3U8 (previous_variant->data)->iframe && new_bandwidth ==
GST_M3U8 (g_list_first (demux->client->main->lists)->data)->bandwidth)
return FALSE;
else
return gst_hls_demux_change_playlist (demux, new_bandwidth - 1, changed);
}
/* Force typefinding since we might have changed media type */
demux->do_typefind = TRUE;
return TRUE;
}
static GstStateChangeReturn
gst_decklink_video_sink_change_state (GstElement * element,
GstStateChange transition)
{
GstDecklinkVideoSink *self = GST_DECKLINK_VIDEO_SINK_CAST (element);
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
g_mutex_lock (&self->output->lock);
self->output->clock_start_time = GST_CLOCK_TIME_NONE;
self->output->clock_epoch += self->output->clock_last_time;
self->output->clock_last_time = 0;
self->output->clock_offset = 0;
g_mutex_unlock (&self->output->lock);
gst_element_post_message (element,
gst_message_new_clock_provide (GST_OBJECT_CAST (element),
self->output->clock, TRUE));
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:{
GstClock *clock, *audio_clock;
clock = gst_element_get_clock (GST_ELEMENT_CAST (self));
if (clock) {
audio_clock = gst_decklink_output_get_audio_clock (self->output);
if (clock && clock != self->output->clock && clock != audio_clock) {
gst_clock_set_master (self->output->clock, clock);
}
gst_object_unref (clock);
if (audio_clock)
gst_object_unref (audio_clock);
} else {
GST_ELEMENT_ERROR (self, STREAM, FAILED,
(NULL), ("Need a clock to go to PLAYING"));
ret = GST_STATE_CHANGE_FAILURE;
}
break;
}
default:
break;
}
if (ret == GST_STATE_CHANGE_FAILURE)
return ret;
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
if (ret == GST_STATE_CHANGE_FAILURE)
return ret;
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_element_post_message (element,
gst_message_new_clock_lost (GST_OBJECT_CAST (element),
self->output->clock));
gst_clock_set_master (self->output->clock, NULL);
// Reset calibration to make the clock reusable next time we use it
gst_clock_set_calibration (self->output->clock, 0, 0, 1, 1);
g_mutex_lock (&self->output->lock);
self->output->clock_start_time = GST_CLOCK_TIME_NONE;
self->output->clock_epoch += self->output->clock_last_time;
self->output->clock_last_time = 0;
self->output->clock_offset = 0;
g_mutex_unlock (&self->output->lock);
gst_decklink_video_sink_stop (self);
break;
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:{
if (gst_decklink_video_sink_stop_scheduled_playback (self) ==
GST_STATE_CHANGE_FAILURE)
ret = GST_STATE_CHANGE_FAILURE;
break;
}
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:{
g_mutex_lock (&self->output->lock);
if (self->output->start_scheduled_playback)
self->output->start_scheduled_playback (self->output->videosink);
g_mutex_unlock (&self->output->lock);
break;
}
default:
break;
}
return ret;
}
static gboolean
gst_identity_sink_event (GstBaseTransform * trans, GstEvent * event)
{
GstIdentity *identity;
gboolean ret = TRUE;
identity = GST_IDENTITY (trans);
if (!identity->silent) {
const GstStructure *s;
const gchar *tstr;
gchar *sstr;
GST_OBJECT_LOCK (identity);
g_free (identity->last_message);
tstr = gst_event_type_get_name (GST_EVENT_TYPE (event));
if ((s = gst_event_get_structure (event)))
sstr = gst_structure_to_string (s);
else
sstr = g_strdup ("");
identity->last_message =
g_strdup_printf ("event ******* (%s:%s) E (type: %s (%d), %s) %p",
GST_DEBUG_PAD_NAME (trans->sinkpad), tstr, GST_EVENT_TYPE (event),
sstr, event);
g_free (sstr);
GST_OBJECT_UNLOCK (identity);
gst_identity_notify_last_message (identity);
}
if (identity->single_segment && (GST_EVENT_TYPE (event) == GST_EVENT_SEGMENT)) {
if (!trans->have_segment) {
GstEvent *news;
GstSegment segment;
gst_event_copy_segment (event, &segment);
gst_event_copy_segment (event, &trans->segment);
trans->have_segment = TRUE;
/* This is the first segment, send out a (0, -1) segment */
gst_segment_init (&segment, segment.format);
news = gst_event_new_segment (&segment);
gst_pad_event_default (trans->sinkpad, GST_OBJECT_CAST (trans), news);
} else {
/* need to track segment for proper running time */
gst_event_copy_segment (event, &trans->segment);
}
}
if (GST_EVENT_TYPE (event) == GST_EVENT_GAP &&
trans->have_segment && trans->segment.format == GST_FORMAT_TIME) {
GstClockTime start, dur;
gst_event_parse_gap (event, &start, &dur);
if (GST_CLOCK_TIME_IS_VALID (start)) {
start = gst_segment_to_running_time (&trans->segment,
GST_FORMAT_TIME, start);
gst_identity_do_sync (identity, start);
/* also transform GAP timestamp similar to buffer timestamps */
if (identity->single_segment) {
gst_event_unref (event);
event = gst_event_new_gap (start, dur);
}
}
}
/* Reset previous timestamp, duration and offsets on SEGMENT
* to prevent false warnings when checking for perfect streams */
if (GST_EVENT_TYPE (event) == GST_EVENT_SEGMENT) {
identity->prev_timestamp = identity->prev_duration = GST_CLOCK_TIME_NONE;
identity->prev_offset = identity->prev_offset_end = GST_BUFFER_OFFSET_NONE;
}
if (identity->single_segment && GST_EVENT_TYPE (event) == GST_EVENT_SEGMENT) {
/* eat up segments */
gst_event_unref (event);
ret = TRUE;
} else {
if (GST_EVENT_TYPE (event) == GST_EVENT_FLUSH_START) {
GST_OBJECT_LOCK (identity);
if (identity->clock_id) {
GST_DEBUG_OBJECT (identity, "unlock clock wait");
gst_clock_id_unschedule (identity->clock_id);
gst_clock_id_unref (identity->clock_id);
identity->clock_id = NULL;
}
GST_OBJECT_UNLOCK (identity);
}
ret = GST_BASE_TRANSFORM_CLASS (parent_class)->sink_event (trans, event);
}
return ret;
}
/* Note: lots of this code here is also in the videotestsrc.c unit test */
void test_rgb_to_rgb()
{
const struct
{
const gchar *pattern_name;
gint pattern_enum;
guint8 r_expected;
guint8 g_expected;
guint8 b_expected;
} test_patterns[] = {
{
"white", 3, 0xff, 0xff, 0xff}, {
"red", 4, 0xff, 0x00, 0x00}, {
"green", 5, 0x00, 0xff, 0x00}, {
"blue", 6, 0x00, 0x00, 0xff}, {
"black", 2, 0x00, 0x00, 0x00}
};
GstElement *pipeline, *src, *filter1, *csp, *filter2, *sink;
const GstCaps *template_caps;
GstBuffer *buf = NULL;
GstPad *srcpad;
GList *conversions, *l;
gint p;
/* test check function */
fail_unless (right_shift_colour (0x00ff0000, 0x11223344) == 0x22);
pipeline = gst_pipeline_new ("pipeline");
src = gst_check_setup_element ("videotestsrc");
filter1 = gst_check_setup_element ("capsfilter");
csp = gst_check_setup_element ("ffmpegcolorspace");
filter2 = gst_element_factory_make ("capsfilter", "to_filter");
sink = gst_check_setup_element ("fakesink");
gst_bin_add_many (GST_BIN (pipeline), src, filter1, csp, filter2, sink, NULL);
fail_unless (gst_element_link (src, filter1));
fail_unless (gst_element_link (filter1, csp));
fail_unless (gst_element_link (csp, filter2));
fail_unless (gst_element_link (filter2, sink));
srcpad = gst_element_get_pad (src, "src");
template_caps = gst_pad_get_pad_template_caps (srcpad);
gst_object_unref (srcpad);
g_object_set (sink, "signal-handoffs", TRUE, NULL);
g_signal_connect (sink, "preroll-handoff", G_CALLBACK (got_buf_cb), &buf);
GST_LOG ("videotestsrc src template caps: %" GST_PTR_FORMAT, template_caps);
conversions = create_rgb_conversions ();
for (l = conversions; l != NULL; l = l->next) {
RGBConversion *conv = (RGBConversion *) l->data;
/* does videotestsrc support the from_caps? */
if (!gst_caps_is_subset (conv->from_caps, template_caps)) {
GST_DEBUG ("videotestsrc doesn't support from_caps %" GST_PTR_FORMAT,
conv->from_caps);
continue;
}
/* caps are supported, let's run some tests then ... */
for (p = 0; p < G_N_ELEMENTS (test_patterns); ++p) {
GstStateChangeReturn state_ret;
RGBFormat *from = &conv->from_fmt;
RGBFormat *to = &conv->to_fmt;
/* trick compiler into thinking from is used, might throw warning
* otherwise if the debugging system is disabled */
fail_unless (from != NULL);
gst_element_set_state (pipeline, GST_STATE_NULL);
g_object_set (src, "pattern", test_patterns[p].pattern_enum, NULL);
GST_INFO ("%5s %u/%u %08x %08x %08x %08x %u => "
"%5s %u/%u %08x %08x %08x %08x %u, pattern=%s",
from->nick, from->bpp, from->depth, from->red_mask,
from->green_mask, from->blue_mask, from->alpha_mask,
from->endianness, to->nick, to->bpp, to->depth, to->red_mask,
to->green_mask, to->blue_mask, to->alpha_mask, to->endianness,
test_patterns[p].pattern_name);
/* now get videotestsrc to produce a buffer with the given caps */
g_object_set (filter1, "caps", conv->from_caps, NULL);
/* ... and force ffmpegcolorspace to convert to our target caps */
g_object_set (filter2, "caps", conv->to_caps, NULL);
state_ret = gst_element_set_state (pipeline, GST_STATE_PAUSED);
if (state_ret == GST_STATE_CHANGE_FAILURE) {
GstMessage *msg;
GError *err = NULL;
msg = gst_bus_poll (GST_ELEMENT_BUS (pipeline), GST_MESSAGE_ERROR, 0);
fail_if (msg == NULL, "expected ERROR message on the bus");
fail_unless (GST_MESSAGE_TYPE (msg) == GST_MESSAGE_ERROR);
gst_message_parse_error (msg, &err, NULL);
fail_unless (err != NULL);
if (msg->src == GST_OBJECT_CAST (src) &&
err->code == GST_STREAM_ERROR_FORMAT) {
GST_DEBUG ("ffmpegcolorspace does not support this conversion");
gst_message_unref (msg);
g_error_free (err);
continue;
}
fail_unless (state_ret != GST_STATE_CHANGE_FAILURE,
"pipeline _set_state() to PAUSED failed: %s", err->message);
}
state_ret = gst_element_get_state (pipeline, NULL, NULL, GST_CLOCK_TIME_NONE);
fail_unless (state_ret == GST_STATE_CHANGE_SUCCESS,
"pipeline failed going to PAUSED state");
state_ret = gst_element_set_state (pipeline, GST_STATE_NULL);
fail_unless (state_ret == GST_STATE_CHANGE_SUCCESS);
fail_unless (buf != NULL);
/* check buffer caps */
{
GstStructure *s;
gint v;
fail_unless (GST_BUFFER_CAPS (buf) != NULL);
s = gst_caps_get_structure (GST_BUFFER_CAPS (buf), 0);
fail_unless (gst_structure_get_int (s, "bpp", &v));
fail_unless_equals_int (v, to->bpp);
fail_unless (gst_structure_get_int (s, "depth", &v));
fail_unless_equals_int (v, to->depth);
fail_unless (gst_structure_get_int (s, "red_mask", &v));
fail_unless_equals_int (v, to->red_mask);
fail_unless (gst_structure_get_int (s, "green_mask", &v));
fail_unless_equals_int (v, to->green_mask);
fail_unless (gst_structure_get_int (s, "blue_mask", &v));
fail_unless_equals_int (v, to->blue_mask);
/* there mustn't be an alpha_mask if there's no alpha component */
if (to->depth == 32) {
fail_unless (gst_structure_get_int (s, "alpha_mask", &v));
fail_unless_equals_int (v, to->alpha_mask);
} else {
fail_unless (gst_structure_get_value (s, "alpha_mask") == NULL);
}
}
/* now check the top-left pixel */
check_rgb_buf (GST_BUFFER_DATA (buf), to->red_mask,
to->green_mask, to->blue_mask, to->alpha_mask,
test_patterns[p].r_expected, test_patterns[p].g_expected,
test_patterns[p].b_expected, to->endianness, to->bpp, to->depth);
gst_buffer_unref (buf);
buf = NULL;
}
}
g_list_foreach (conversions, (GFunc) rgb_conversion_free, NULL);
g_list_free (conversions);
gst_element_set_state (pipeline, GST_STATE_NULL);
gst_object_unref (pipeline);
std_log(LOG_FILENAME_LINE, "Test Successful");
create_xml(0);
}
static GstFlowReturn
gst_fd_src_create (GstPushSrc * psrc, GstBuffer ** outbuf)
{
GstFdSrc *src;
GstBuffer *buf;
gssize readbytes;
guint blocksize;
GstClockTime timeout;
#ifndef HAVE_WIN32
gboolean try_again;
gint retval;
#endif
src = GST_FD_SRC (psrc);
if (src->timeout > 0) {
timeout = src->timeout * GST_USECOND;
} else {
timeout = GST_CLOCK_TIME_NONE;
}
#ifndef HAVE_WIN32
do {
try_again = FALSE;
GST_LOG_OBJECT (src, "doing poll, timeout %" GST_TIME_FORMAT,
GST_TIME_ARGS (src->timeout));
retval = gst_poll_wait (src->fdset, timeout);
GST_LOG_OBJECT (src, "poll returned %d", retval);
if (G_UNLIKELY (retval == -1)) {
if (errno == EINTR || errno == EAGAIN) {
/* retry if interrupted */
try_again = TRUE;
} else if (errno == EBUSY) {
goto stopped;
} else {
goto poll_error;
}
} else if (G_UNLIKELY (retval == 0)) {
try_again = TRUE;
/* timeout, post element message */
gst_element_post_message (GST_ELEMENT_CAST (src),
gst_message_new_element (GST_OBJECT_CAST (src),
gst_structure_new ("GstFdSrcTimeout",
"timeout", G_TYPE_UINT64, src->timeout, NULL)));
}
} while (G_UNLIKELY (try_again)); /* retry if interrupted or timeout */
#endif
blocksize = GST_BASE_SRC (src)->blocksize;
/* create the buffer */
buf = gst_buffer_try_new_and_alloc (blocksize);
if (G_UNLIKELY (buf == NULL)) {
GST_ERROR_OBJECT (src, "Failed to allocate %u bytes", blocksize);
return GST_FLOW_ERROR;
}
do {
readbytes = read (src->fd, GST_BUFFER_DATA (buf), blocksize);
GST_LOG_OBJECT (src, "read %" G_GSSIZE_FORMAT, readbytes);
} while (readbytes == -1 && errno == EINTR); /* retry if interrupted */
if (readbytes < 0)
goto read_error;
if (readbytes == 0)
goto eos;
GST_BUFFER_OFFSET (buf) = src->curoffset;
GST_BUFFER_SIZE (buf) = readbytes;
GST_BUFFER_TIMESTAMP (buf) = GST_CLOCK_TIME_NONE;
src->curoffset += readbytes;
GST_LOG_OBJECT (psrc, "Read buffer of size %" G_GSSIZE_FORMAT, readbytes);
/* we're done, return the buffer */
*outbuf = buf;
return GST_FLOW_OK;
/* ERRORS */
#ifndef HAVE_WIN32
poll_error:
{
GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
("poll on file descriptor: %s.", g_strerror (errno)));
GST_DEBUG_OBJECT (psrc, "Error during poll");
return GST_FLOW_ERROR;
}
stopped:
{
GST_DEBUG_OBJECT (psrc, "Poll stopped");
return GST_FLOW_WRONG_STATE;
}
#endif
eos:
{
GST_DEBUG_OBJECT (psrc, "Read 0 bytes. EOS.");
gst_buffer_unref (buf);
return GST_FLOW_UNEXPECTED;
}
read_error:
{
GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
("read on file descriptor: %s.", g_strerror (errno)));
GST_DEBUG_OBJECT (psrc, "Error reading from fd");
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
}
}
