Use breaks for case branches instead of return 0. We don't expect these to
happen anyway. Thus have a warning before the final return to make it easier to
see when things go out of sync.
When closing rtspsrc the state change blocks until the polling in the
connection timeouts. This is because the second time we loop to read a
full message controllable is set to FALSE in the poll group, even though no
message is half read.
This can be avoided by not setting controllable to FALSE the poll group
unless we had begin to read a message.
Fixes#610916
Instead of writing only the xmp tag for the first found entry
that matches the gstreamer tag, look for all mappings to write
the tag to different schemas.
The rationale here is that some reader application might only
be interested on a particular schema tags, so we should try
to write as many tags for all schemas.
This can be used by sinks to take compressed formats, correctly payload
these in IEC 61937 frames and feed these to sinks that support
passthrough output over IEC 60958 (S/PDIF) or, in the case of MP3, over
Bluetooth.
Initial implementation includes AC3, E-AC3, MPEG-1, MPEG-2 (non-AAC),
and DTS (type-I/II/II) payloading. More formats can be added as needed.
API: gst_audio_iec61937_frame_size()
API: gst_audio_iec61937_payload()
https://bugzilla.gnome.org/show_bug.cgi?id=642730
This allows subclasses to provide a "payload" function to prepare
buffers for consumption. The immediate use for this is for sinks that
can handle compressed formats - parsers are directly connected to the
sink, and for formats such as AC3, DTS, and MPEG, IEC 61937 patyloading
might be used.
API: GstBaseAudioSinkClass:payload()
https://bugzilla.gnome.org/show_bug.cgi?id=642730
Adds support for pushing E-AC3 buffers and doing bytes-to-ms conversion
correctly. The assumption (as with other formats) is that something like
IEC 61937 payloading will be used. Correspondingly the ringbuffer spec
is populated so that the data rate is 4x normal AC3.
https://bugzilla.gnome.org/show_bug.cgi?id=642730
These are meant to be used for buffers containing AAC data. Nothing uses
this yet, but for now it serves to distinguish from GST_BUFTYPE_MPEG
which represents non-AAC MPEG audio.
API: GST_BUFTYPE_MPEG2_AAC
API: GST_BUFTYPE_MPEG4_AAC
Exif uses tags like image-vertical-ppi or image-horizontal-ppi which are
registered in gst_tag_register_musicbrainz_tags(), but neither GstExifReader
nor GstExifWriter register them.
https://bugzilla.gnome.org/show_bug.cgi?id=648459
libgstfft doesn't actually use any symbols from libgstreamer, so when
compiling with -Wl,--as-needed it won't even link to it, which can
cause failures with older versions of g-i that ignore the --pkg
arguments.
Should fix PPA build failure on Ubuntu Maverick
Remove the android/ top dir
Fixe the Makefile.am to be androgenized
To build gstreamer for android we are now using androgenizer which generates the
needed Android.mk files.
Androgenizer can be found here:
http://git.collabora.co.uk/?p=user/derek/androgenizer.git
1) We need to lock and get a strong ref to the parent, if still there.
2) If it has gone away, we need to handle that gracefully.
This is necessary in order to safely modify a running pipeline. Has been
observed when a streaming thread is doing a buffer_alloc() while an
application thread sends an event on a pad further downstream, and from
within a pad probe (holding STREAM_LOCK) carries out the pipeline plumbing
while the streaming thread has its buffer_alloc() in progress.
A race was observed between query() and setcaps() where the latter would
change the ringbuffer spec while the former was performing operations
based this data.
Observed a case where the src went to null-state during the query,
hence the spec pointer was no longer valid, and
gst_util_unit64_scale_int crashed (assertion `denom > 0´failed)
Add locking to make sure the ringbuffer can't disappear.
Given a large enough drift-tolerance, one could end up in a situation
where one would keep aligning the written buffers behind the current
read-segment position. The result for the reader would be complete
silence, possible preceded by very choppy audio.
By checking the available headroom, one can determine if there is
room to do alignment, or if one should resort to a resync instead to get
the pointers back on track.
Also refactor the alignment-logic out of the render function for cleaner
code.
This is the official, standardized way of embedding pictures
inside vorbiscomments now. Parsing code taken from flacparse
and slightly changed.
Fixes bug #635669.
Commit ba2e500bd9 ensured to provide
a running clock when EOS had finished rendering. However,
other measures are needed (and were in place before) to ensure a
running clock when EOS still needs rendering (i.e. waiting).
So, specifically, re-introduce eos_rendering removed in aforementioned commit,
this time as a public variable so subclasses can be aware of the situation.
Fixes (part of) #645961.
API: GstBaseAudioSink:eos_rendering
The GstTagXmpWriter interface is to be implemented on elements that
provide xmp serialization. It allows users to select which
xmp schemas should be used on serialization.
API: GstTagXmpWriter
https://bugzilla.gnome.org/show_bug.cgi?id=645167
This fixes a regression that an assertion would happen if
gst_video_get_component_offset would be called with width or
height as 0.
Calling it with 0 is fine if the format isn't yuv and this
was already being used in some other places of video.c
Subtitle streams being parse can cause the pipeline to wait indefinitely
to PREROLL. This makes subtitle streams got to PAUSED even if no data is
available. This should not be a cause for concern as we don't expect to
get much data for subtitle streams other than language tags from the
container.
https://bugzilla.gnome.org/show_bug.cgi?id=632291
Maps to GST_BUFFER_FLAG_MEDIA4. The purpose is to explicitly indicate
whether a telecined buffer is progressive or not without having to make
assumptions based on previous buffers.
This makes sure we maintain a ref on the discoverer object while the
async timeout callback is alive to prevent a potential crash if the
object is freed while the callback is pending.
https://bugzilla.gnome.org/show_bug.cgi?id=641706
We want to make sure the discoverer object passed to the various
callbacks doesn't become invalid if a callback is pending and the object
is free'd in the mean time.
https://bugzilla.gnome.org/show_bug.cgi?id=641706
Otherwise, having 2 tagdemux in a row followed by an element operating in
pull mode will make the second tagdemux implictly eat the first tagdemux'
tag event(s).
Fixes (part of) #641047.
... as that is the specification and fixes compilation on Cygwin:
gstxmptaag.c: In function 'read_one_tag':
gstxmptag.c:1015: error: array subscript has type 'char'
Variable was being written to and could cause crashes
if multiple elements were parsing xmp at the same time.
Moving it to local scope solves the problem.
This makes sure we do not touch the stream taglist once the pipeline has
been prerolled. Adding of stream tags happens in the pad event probe
which runs in a different thread from discoverer stream processing, so
modifying the tag list while discoverer might be processing it can
sometimes cause a crash.
https://bugzilla.gnome.org/show_bug.cgi?id=639778
This avoids a race where the timeout callback is scheduled to run but we
get sufficient information to finish discovery before actually getting
around to executing the callback. See the documentation of
g_source_is_destroyed() for more details.
https://bugzilla.gnome.org/show_bug.cgi?id=639730
This ensures that everything is properly cleaned up before the
GstDiscoverer object is freed. Specifically, it makes sure that we've
removed the async timeout callback before freeing the object to avoid a
potential crash later on.
https://bugzilla.gnome.org/show_bug.cgi?id=639755
Use LC_MESSAGES rather than LC_ALL. Save/load description as untranslated string
when using an English language locale. Strip locale information to the language,
so we don't save keys like description[fr_FR.UTF-8]=...
https://bugzilla.gnome.org/show_bug.cgi?id=638860
Makes things work again properly in uninstalled setups (and
presumably in installed setups where GStreamer is installed
into a non-standard prefix). Requires fixes from core git.
https://bugzilla.gnome.org/show_bug.cgi?id=639039
Need to pass libgstreamer-0.10 explicitly to linker, since we're
calling gst_init(), which in turn is needed because the encoding
target get_type() function calls gst_value_register().
https://bugzilla.gnome.org/show_bug.cgi?id=639039
Make sure to use the PKG_CONFIG_PATH set at configure time instead of
just relying on an env-var set one. This makes sure both g-ir-compiler
and g-ir-scanner use the same PKG_CONFIG_PATH for determining include
paths etc.
Observed a case where the sink went to null-state during the query,
hence the ringbuffer-pointer was NULL, causing a crash.
Moving the ringbuffer-check code until after the query, and hold the
lock during the check and while using the spec-values. It should not matter
to the query wether the ringbuffer is present or not, and it actually
gets a time bit more time to get the ringbuffer set up in this case!
Fixes#635231
When we have an invalid running-time (because we clipped, for example) use the
RTP base time for timestamping instead of generating wrong RTP timestamps.
with i686-apple-darwin10-gcc-4.2.1:
encoding-profile.h:134: warning: type qualifiers ignored on function return type
encoding-profile.c:240: warning: type qualifiers ignored on function return type
gstencodebin.c: In function 'next_unused_stream_profile':
gstencodebin.c:454: warning: format '%d' expects type 'int', but argument 8 has type 'GType'
gstencodebin.c:464: warning: format '%d' expects type 'int', but argument 8 has type 'GType'
gst_discoverer_discover_uri() expects the caller to unref the returned
GstDiscovererInfo object. The corresponding gtk-doc annotation was not
updated to reflect this.
We want to send the keealive message a little earlier than the timeout value
specifies. Scale this based on the value of the timeout instead of just assuming
5 seconds.
Because we should act before the rtsp server does a timeout, we
reduce the timeout-time with 5 seconds, this should be safe to always
keep te rtsp connection alive.
https://bugzilla.gnome.org/show_bug.cgi?id=633455
Use GstDiscoverer{Audio,Video}Info in getters like
gst_discoverer_{audio,video}_info_get_*(). This avoids the casts in the macros,
help language bindings and is more correct.
Force regeneration of marshal.[ch] files after prefix changes in
Makefile.am, to avoid build errors for those of us who don't
habitually make clean first.
Adds a tag to inform what mode was used by a camera to calculate
the picture capturing exposure
Also adds mapping to exif and tests
API: GST_TAG_CAPTURING_METERING_MODE
https://bugzilla.gnome.org/show_bug.cgi?id=631773
Adds new tag for tagging sharpness processing used
when capturing an image. Also maps it in the exif
tags.
Tests included.
API: GST_TAG_CAPTURING_SHARPNESS
https://bugzilla.gnome.org/show_bug.cgi?id=631773
There's no reason to make the marshaller public API. Don't install
pbutils-marshal.h header file and use prefix that makes sure the
symbol doesn't get exported.
So run-time bindings can introspect the names correctly (we abuse this
field as description field only in elements, not for public API
(where the description belongs into the gtk-doc chunk).
https://bugzilla.gnome.org/show_bug.cgi?id=629746
Add a new function called gst_rtp_buffer_list_from_buffer() that takes
a GstBuffer containing a RTP packets and spits out a GstBufferList
containing two buffers, one with the header and the other with the payload.
RFC 5285 describes a generic method to add multiple header extensions to RTP packets.
These functions parse these headers and return them, both for the one-byte header and the
two bytes headers.
This adds code to translate the profile_and_level indication from the
MPEG-4 video (ISO/IEC 14496-2) headers to a string profile/level. The
mappings are taken from the spec and Wireshark's code, and might need to
be expanded on.
https://bugzilla.gnome.org/show_bug.cgi?id=617314
API: gst_codec_utils_mpeg4video_get_profile()
API: gst_codec_utils_mpeg4video_get_level()
API: gst_codec_utils_mpeg4video_caps_set_level_and_profile()
This adds code to parse the first few bytes of H.264 sequence parameter
set in order to extract the profile and level as const strings. This
code was originally in both qtdemux and matroskademux.
https://bugzilla.gnome.org/show_bug.cgi?id=617314
API: gst_codec_utils_h264_get_level()
API: gst_codec_utils_h264_get_profile()
API: gst_codec_utils_h264_caps_set_level_and_profile()
This moves AAC profile detection to pbutils, and uses this in
typefindfunctions. This will also be used in qtdemux.
https://bugzilla.gnome.org/show_bug.cgi?id=617314
API: gst_codec_utils_aac_get_profile()
API: codec_utils_aac_caps_set_level_and_profile()
This allows us to add generic codec-specific functionality, like
extracting profile/level data from headers, without having to duplicate
code across demuxers and typefindfunctions.
As a starting point, this moves over AAC level extraction code from
typefindfunctions, so it can be reused in qtdemux, etc.
https://bugzilla.gnome.org/show_bug.cgi?id=617314
API: gst_codec_utils_aac_get_sample_rate_from_index()
API: gst_codec_utils_aac_get_level()
From gstinfo.h:
/* do not use this function, use the GST_DEBUG_CATEGORY_INIT macro */
GstDebugCategory *_gst_debug_category_new (const gchar * name,
And more importantly:
#pragma GCC poison _gst_debug_category_new
So this commit fixes --disable-gst-debug builds.
Make appsrc not set caps on buffers when its own caps is NULL.
This avoids calling make_metadata_writable on all buffers and
prevents losing buffer caps in case we are not replacing it
with something meaningful.
https://bugzilla.gnome.org/show_bug.cgi?id=630353
While the doc parser allows for certain variation, it is a good idea to not
use random characters here and there, but try to stick to the little markup
syntax there is.
Binding generators apparently need this as they can't really know
that the callback is guaranteed to be called exactly once and that
the user_data can be freed at the end of it.
And deprecate the gulong versions. This is to support platforms
where sizeof(unsigned long) < sizeof(void *). Fixes#627565.
API: Add gst_x_overlay_set_window_handle()
API: Deprecate: gst_x_overlay_set_xwindow_id()
API: Add gst_x_overlay_got_window_handle()
API: Deprecate: gst_x_overlay_got_xwindow_id()
API: Add GstXOverlay::set_window_handle()
API: Deprecate: GstXOverlay::set_xwindow_id()
There will always be only a single output buffer and if the
target caps have a different framerate than the input there
will be a negotiation error during conversion.
Add methods to convert between uri and sdpmessages, loosly based on
http://tools.ietf.org/html/draft-fujikawa-sdp-url-01
API: GstSDPMessage::gst_sdp_message_parse_uri
API: GstSDPMessage::gst_sdp_message_as_uri
Don't take an extra ref on the sink and source because that creates a reference
cycle. Instead, use the invalidate method of the clock when the sink and source
are freed. This way, we don't call into the time function anymore after the
objects are disposed.
