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authorTim-Philipp Müller <tim@centricular.com>2021-05-16 04:10:55 +0300
committerTim-Philipp Müller <tim@centricular.com>2021-05-16 14:28:53 +0300
commitcaa608e5c0c5829f67a4fbe82e0abc28b24aa150 (patch)
treef80afd92a6b4d09f98abcb70b334f45f05932078
parent2a710a484c6509e5bf08e1d70d3868f0d282c65a (diff)
docs: random: clean up outdated documents
Most of these are only of historical interest, and for that it's fine if they're maintained in the git history. They're confusing for anyone stumbling across them expecting documentation relating to current versions of GStreamer. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/824>
-rw-r--r--docs/random/API2
-rw-r--r--docs/random/LICENSE18
-rw-r--r--docs/random/TODO-pre-0.9170
-rw-r--r--docs/random/autoplug1206
-rw-r--r--docs/random/autoplug2289
-rw-r--r--docs/random/bbb/streamselection85
-rw-r--r--docs/random/bbb/subtitles105
-rw-r--r--docs/random/buffers19
-rw-r--r--docs/random/caps209
-rw-r--r--docs/random/caps.dia1132
-rw-r--r--docs/random/caps2231
-rw-r--r--docs/random/classes.diabin5601 -> 0 bytes
-rw-r--r--docs/random/ds/0.9-planning165
-rw-r--r--docs/random/ds/buffer_locking30
-rw-r--r--docs/random/ds/bufferpools57
-rw-r--r--docs/random/ds/categories50
-rw-r--r--docs/random/ds/element-checklist43
-rw-r--r--docs/random/ds/registry15
-rw-r--r--docs/random/ds/roadmap79
-rw-r--r--docs/random/ensonic/audiobaseclasses.txt18
-rw-r--r--docs/random/ensonic/dparams.txt14
-rw-r--r--docs/random/ensonic/draft-registry-change-hooks.txt56
-rw-r--r--docs/random/ensonic/dynlink.txt89
-rw-r--r--docs/random/ensonic/interfaces.txt85
-rw-r--r--docs/random/ensonic/lazycaps.txt72
-rw-r--r--docs/random/ensonic/logging.txt56
-rw-r--r--docs/random/ensonic/media-device-daemon.txt42
-rw-r--r--docs/random/ensonic/plugindocs.txt51
-rw-r--r--docs/random/ensonic/receipies.txt37
-rw-r--r--docs/random/eos209
-rw-r--r--docs/random/error59
-rw-r--r--docs/random/example108
-rw-r--r--docs/random/hierarchy153
-rw-r--r--docs/random/interfaces337
-rw-r--r--docs/random/metadata80
-rw-r--r--docs/random/mutability30
-rw-r--r--docs/random/negotiation286
-rw-r--r--docs/random/omega/EOS/chain-walkthrough19
-rw-r--r--docs/random/omega/IDEAS21
-rw-r--r--docs/random/omega/TODO-0.1.054
-rw-r--r--docs/random/omega/TYPE_FOURCC3
-rw-r--r--docs/random/omega/build/TODO7
-rw-r--r--docs/random/omega/caps2320
-rw-r--r--docs/random/omega/caps320
-rw-r--r--docs/random/omega/debug-commit30
-rw-r--r--docs/random/omega/eos.old52
-rw-r--r--docs/random/omega/filterfactory3
-rw-r--r--docs/random/omega/output_policies61
-rw-r--r--docs/random/omega/pad-negotiation40
-rw-r--r--docs/random/omega/padtemplates46
-rw-r--r--docs/random/omega/plan-generation87
-rw-r--r--docs/random/omega/sched-case20
-rw-r--r--docs/random/omega/sched-commit1103
-rw-r--r--docs/random/omega/sched/chains89
-rw-r--r--docs/random/omega/sched/walkthrough-7239
-rw-r--r--docs/random/omega/sched213
-rw-r--r--docs/random/omega/scheduling66
-rw-r--r--docs/random/omega/testing/Makefile3
-rw-r--r--docs/random/omega/testing/framework144
-rw-r--r--docs/random/omega/testing/gstobject.c314
-rw-r--r--docs/random/omega/testing/gstobject.txt95
-rw-r--r--docs/random/omega/type-properties77
-rw-r--r--docs/random/phonon-gst269
-rw-r--r--docs/random/plugins93
-rw-r--r--docs/random/plugins.dia941
-rw-r--r--docs/random/porting-to-0.11.txt5
-rw-r--r--docs/random/queue38
-rw-r--r--docs/random/richardb/syncmail3
-rw-r--r--docs/random/rtp44
-rw-r--r--docs/random/signal12
-rw-r--r--docs/random/sources324
-rw-r--r--docs/random/status-0.11-14-jun-2011.txt192
-rw-r--r--docs/random/styleguide18
-rw-r--r--docs/random/testing/syntax37
-rw-r--r--docs/random/thaytan/opengl20
-rw-r--r--docs/random/thaytan/video-overlays28
-rw-r--r--docs/random/thomasvs/0.10121
-rw-r--r--docs/random/thomasvs/0.4.0108
-rw-r--r--docs/random/thomasvs/TODO3
-rw-r--r--docs/random/thomasvs/docreview37
-rw-r--r--docs/random/thomasvs/features15
-rw-r--r--docs/random/thomasvs/guadec-416
-rw-r--r--docs/random/thomasvs/pthread75
-rw-r--r--docs/random/thomasvs/pwg39
-rw-r--r--docs/random/thomasvs/registry36
-rw-r--r--docs/random/types33
-rw-r--r--docs/random/types2282
-rw-r--r--docs/random/types3177
-rw-r--r--docs/random/use-cases-0.11.txt21
-rw-r--r--docs/random/usecases44
-rw-r--r--docs/random/vis-transform78
-rw-r--r--docs/random/wingo/porting-plugins-to-0.932
-rw-r--r--docs/random/wingo/threadsafe-properties76
-rw-r--r--docs/random/wingo/without-factories62
-rw-r--r--docs/random/zaheerm/dvb-interface.txt26
95 files changed, 0 insertions, 9718 deletions
diff --git a/docs/random/API b/docs/random/API
deleted file mode 100644
index 4962b3283f..0000000000
--- a/docs/random/API
+++ /dev/null
@@ -1,2 +0,0 @@
-* signals should use dashes in their names, not underscores, so ::notify
- works correctly
diff --git a/docs/random/LICENSE b/docs/random/LICENSE
deleted file mode 100644
index 696b1c1b22..0000000000
--- a/docs/random/LICENSE
+++ /dev/null
@@ -1,18 +0,0 @@
-/* GStreamer
- * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
- * Boston, MA 02110-1301, USA.
- */
diff --git a/docs/random/TODO-pre-0.9 b/docs/random/TODO-pre-0.9
deleted file mode 100644
index 248024aaa6..0000000000
--- a/docs/random/TODO-pre-0.9
+++ /dev/null
@@ -1,170 +0,0 @@
-TODO:
------
-
-short term core API stability
------------------------------
-
-Changes that probably impact the API, careful discussion (IRC) + design doc is required
-before changes are accepted.
-
-target release ! description
- !
- 0.4.1 ! expose and API to query the supported seek formats/flags on
- (done) ! pads, something like an extra arg to gst_pad_set_convert_function
- ! and gst_pad_set_event_function with some function to query the
- ! flags and formats. more ideas in docs/random/wtay/query_events
- ! (API: medium difficulty)
- !
- 0.4.1 ! add event for segment playback/looping and seeking (docs/random/wtay/segments)
- (done) ! (API: medium difficulty, plugins: HARD to very HARD)
- !
- ? ! add event to adjust rate (reverse playback, slow motion, frame skipping)
- ! (docs/random/wtay/rate_event)
- ! (API: medium difficulty, plugins: HARD to very HARD)
- !
- ? ! add method in the scheduler to set the entry point (frame stepping?)
- ! (docs/random/wtay/scheduler_entry)
- ! (API: moderatly EASY, scheduler implementation MEDIUM)
- !
- 0.6.0 ! create a design doc for a timecache implementation,
- (done) ! (docs/wtay/random/timecache)
- ! (API: MEDIUM, needs lots of discussion, plugin implementation MEDIUM to HARD)
- ! (done: implemented using GstIndex base class + subclasses)
- !
- ? ! implement a QoS event and a policy for handling the event.
- ! (API: kindof EASY, plugins MEDIUM to very HARD)
- !
- 0.4.1 ! implement user defined GstFormat values, make a format factory etc..
- (done) ! (API: MEDIUM, plugins MEDIUM)
- !
- ? ! strip the API to a bare bones minimal set of functions, leave the automatic
- ! stuff to the app instead of forcing a policy in the core.
- ! create a library with useful higher level function for people who don't want
- ! to deal with the lowlevel stuff.
- ! (HARD, need to negotiate with people :))
- !
- ? ! use GMarkup to load/save objects, remove dependency on libxml2
- ! (MEDIUM) breaks API/ABI compatibility
- !
-
-
-shortterm core feature enhancements
------------------------------------
-
- 0.4.1 ! Implement PAD_DISABLED. This requires simple checks in the scheduler so that
- ! it doesn't try to pull/push data from/to a disabled pad.
- ! When an element goes to the PAUSED state, all of its pads should be disabled.
- ! This should also work for ghostpads.
- ! (API: MEDIUM to moderatly HARD, requires some scheduler understanding)
-
-
-short term usability
---------------------
-
-Writing docs is NOT boring, you learn a lot, you get insight in stuff, you help a lot
-of people, hey! you might even find YOUR book on a shelf in a bookstore!!
-
-
- ? ! plugin writers guide
- ! (we have almost nothing, so any start is welcomed)
- ! (MEDIUM)
- !
- ? ! app writers guide needs to cover common tips and tricks and HOWTOs
- ! (MEDIUM)
-
-
-short to midterm policy definition
-----------------------------------
-
-Policy definition is closely related to a HOWTO but sometimes needs some thinking.
-
-
- ? ! document thread safety guidelines, what stuff needs locking in the app, what
- ! is done in the core.
- ! most of this stuff is in the heads of various people but needs to be written
- ! down so that people get more insights into the design and vision of GStreamer.
- ! (MEDIUM, some research and discussion needed)
- !
- ? ! a step by step guide to the implementation of various events in a plugin, what can you
- ! do, when is data passing forbidden etc..
- ! (MEDIUM, some research needed)
- !
- ? ! figure out a policy for the NEW_MEDIA event
- ! (MEDIUM to HARD)
- !
- ? ! figure out how to better handle clock resync
- ! (MEDIUM to HARD)
- !
-
-
-midterm feature enhancement
----------------------------
-
- 0.6.0 | Define and implement a syntax in gst_parse to handle filtered pad connections.
- (done) | (MEDIUM)
- |
- ? | Figure out a way to set properties on schedulers (and bins?) from gst_parse.
- | (MEDIUM)
- |
- ? | Make gst-inspect do inspection of plugins, schedulers, autopluggers, types.
- | An idea would be to make -inspect output an XML representation of the objects
- | and use XSLT to transform this into text, HTML, docbook, ...
- | (MEDIUM to EASY)
- |
-
-
-midterm (longterm) optimisations
---------------------------------
-
-These optimisations can be done without changing the existing API.
-
-
- (in progress) ! implement an optimal scheduler that uses chaining when possible
- ! (HARD, requires detailed knowledge of element scheduling)
- !
- ? ! alternatively optimisations to the current scheduler can be done such
- ! as: do nothing when the pipeline structure (or chain) has not changed
- ! (MEDIUM)
- !
- ? ! GstQueue is a little mess. implement a better queue (lockfree?), refactor
- ! queueing policy (max buffer, max time, levels etc..)
- ! (MEDIUM to farily EASY)
-
-
-longterm feature enhancements
------------------------------
-
-Various features that are not critical yet.
-
- ? ! factory aliases. map a generic name like "videosink" to and actual
- ! user configurable plugin (aasink, sdlsink, xvideosink, ...)
- ! (MEDIUM)
- !
- ? ! property proxy in compound elements. not sure if it's possible at all.
- ! what with elements with the same property?
- ! (MEDIUM, needs some thinking)
- !
- ? ! Make _pad_select work for muxers
- ! (MEDIUM to HARD)
-
-
-needs consensus
----------------
-
-Some stuff that needs to be figured out based on a pro/con comparison.
-
- ? ! can we decide on the fact that downstream events are traveling using the
- ! scheduler? do we need to reevaluate that design decision?
- ! (MEDIUM, needs pros vs cons document)
-
-
-benchmarks
-----------
-
-Benchmarks are always good to get acceptance in a wider comunity or to identify performance
-problems that need fixing.
-
- ? ! do a latency comparison with popular other frameworks, document GStreamer
- ! overhead.
- ! (MEDIUM to somewhat EASY)
- !
diff --git a/docs/random/autoplug1 b/docs/random/autoplug1
deleted file mode 100644
index f3f94dfec9..0000000000
--- a/docs/random/autoplug1
+++ /dev/null
@@ -1,206 +0,0 @@
-COMPLETELY OUTDATED
--------------------
-
-
-A little explanation of the first autoplugger in GStreamer:
-
-Autoplugging is implemented in the following places:
-
- gstpipeline.c : construction of the pipeline
- gstautoplug.c : selection of the elementfactories needed for autoplugging
-
-1) pipeline setup
------------------
-
-before any autoplugging will take place, a new GstPipeline has to be created.
-The autoplugger needs to have a src element and one or more sink elements. the
-autoplugger will try to find the elements needed to connect the src element
-to the sinks.
-
-using:
-
- gst_pipeline_add_src (GstPipeline *pipeline, GstElement *element);
-
-a source element is added to the pipeline. only one src element can be added
-for now.
-
-using:
-
- gst_pipeline_add_sink (GstPipeline *pipeline, GstElement *element);
-
-a sink element can be added to the pipeline.
-
-2) starting autoplug
---------------------
-
-when the pipeline has been set up as above, you will call
-
- gst_pipeline_autoplug (GstPipeline *pipeline);
-
-to start the autoplugger. this will be done in four phases
-
-ex. we are going to autoplug an mpeg1 system stream.
-
-2a) phase1: figure out the type (GstCaps) of the src element.
--------------------------------------------------------------
-
-the gsttypefind element is connected to the "src" pad of the source
-element. gst_bin_iterate is called in a loop until gsttypefind
-signals "have_type". the gst_bin_iterate is stopped and the GstCaps
-is retrieved from the gsttypefind element.
-
-gsttypefind is disconnected from the src element and removed from the
-bin.
-
-the GstCaps of the source element is called src_caps later on.
-
-ex. all typefind functions are tried and the one in mpeg1types will
- return a GstCaps:
-
- video/mpeg,
- "systemstream", GST_PROPS_BOOLEAN (TRUE),
- "mpegversion", GST_PROPS_INT (1),
- NULL
-
-
-2b) phase2: create lists of factories.
----------------------------------------
-
-for each sink:
-{
- sinkpad = take the first sinkpad of the sink (HACK)
- call
-
- list[i] = gst_autoplug_caps (src_caps, sinkpad->caps);
-
- I++;
-}
-
-gst_autoplug_caps will figure out (based on the padtemplates)
-which elementfactories are needed to connect src_caps to sinkpad->caps
-and will return them in a list.
-
-ex. we have two sinks with following caps:
-
- video/raw audio/raw
- "...." "...."
-
- gst_autoplug_caps will figure out that for the first sink the following
- elements are needed:
-
- mpeg1parse, mp1videoparse, mpeg_play
-
- for the second sink the following is needed:
-
- mpeg1parse, mp3parse, mpg123
-
- We now have two lists of elementfactories.
-
-2c) phase3: collect common elements from the lists.
----------------------------------------------------
-
-the rationale is that from the lists we have created in phase2, there
-must be some element that is a splitter and that it has to come first (HACK)
-We try to find that element by comparing the lists until an element differs.
-
-we add the common elements to the bin and run gst_pipeline_pads_autoplug. this
-function will loop over the pads of the previous element and the one we
-just added, and tries to connect src to sink if possible.
-
-If a connection between the two elements could not be made, a signal "new_pad"
-is connected to the element so that pad connection can occur later on when
-the pad is actually created.
-
-ex. when we compare the two lists we see that we have common element: mpeg1parse.
-
- we add this element to the bin and try to connect it to the previous element in
- the bin, the disksrc.
-
- we see that the src pad of the disksrc and the sinkpad of the mpeg1parse element
- can be connected because they are compatible. We have a pipeline like:
-
- ---------) (--------
- disksrc ! ! mpeg1parse
- src --- sink
- ---------) (--------
-
-
-2d) phase4: add remaining elements
-----------------------------------
-
-now we loop over all the list and try to add the remaining elements
-
-(HACK) we always use a new thread for the elements when there is a common
-element found.
-
-if a new thread is needed (either bacuase the previous element is a common
-element or the object flag of the next element is set to GST_SUGGEST_THREAD)
-we add a queue to the bin and we add a new thread. We add the elements to
-the bin and connect them using gst_pipeline_pads_autoplug.
-
-If we add a queue, we have to copy the caps of the sink element of the queue
-to the src pad of the queue (else they won't connect)
-
-we finally arrive at the sink element and we're done.
-
-ex.
-
- we have just found our mpeg1parse common element, so we start a thread.
- We add a queue to the bin and a new thread, we add the elements
- mp1videoparse and mpeg_play to the thread. We arrive at the videosink, we
- see that the SUGGEST_THREAD flag is set, we add a queue and a thread and
- add the videosink in the thread.
-
- the same procedure happens for the audio part. We are now left with the
- following pipeline:
-
- We will also have set a signal "new_pad" on the mpeg1parse element because
- the element mp1videoparse could not be connected to the element just yet.
-
- (------------------------------------) (----------
- !thread ! ! thread
- ! (-------------) (---------) ! ! (---------)
- ! !mp1videoparse! !mpeg_play! ! ! !videosink!
- videoqueue--sink src -- sink src -- queue --- sink !
- ---------) (-----------) ! (-------------) (---------) ! ! (---------)
- disksrc ! ! mpeg1parse! (------------------------------------) (-------------
- src --- sink !
- ---------) (-----------)
- queue----- same for audio
-
-
- then we play, create_plan happens, data is flowing and the "new_pad" signal is called
- from mpeg1parse, gst_pipeline_pad_autoplug is called and the connection between
- mpeg1parse and the videoqueue is made. same for audio.
-
- voila. smame procedure for mp3/vorbis/avi/qt/mpeg2 etc...
-
-
-Problems:
----------
-
-this is obviously a very naive solution. the creation of the elements actually happens
-beforehand. MPEG2, for one, fails because there are multiple possibilities to go
-from the mpeg demuxer to audio/raw (ac3, mp3)
-
-Also any intermedia elements like mixers (subtitles) are not possible because we
-assume that after the common elements, the streams to not converge anymore.
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
diff --git a/docs/random/autoplug2 b/docs/random/autoplug2
deleted file mode 100644
index f3374b693f..0000000000
--- a/docs/random/autoplug2
+++ /dev/null
@@ -1,289 +0,0 @@
-
-1) The Autoplugger API
-----------------------
-
-We'll first describe how to use the autoplugger. We will provide
-a use case: autoplug an mpeg1 system stream for audio/video playback.
-
-
-a) creating an autoplugger
---------------------------
-
-Before any autoplugging can be done, you'll have to create an
-autoplugger object. Autoplugger objects (autopluggers) are
-provided by plugins and are created with gst_autoplugfactor_make().
-
-GStreamer has provisions for two types of autopluggers:
-
- - regular autopluggers, which act as a complex element construction
- mechanism. They usually don't create threads and operate solely on
- GstCaps* for the source and destination. The complex elements
- created by regular autopluggers have src and sink pad compatible
- with the requested GstCaps*.
-
- - renderer autopluggers, which are designed to create a complex
- object that can be used to playback media. Renderer autoplugged
- complex elements have no src pads, only one sink pad.
-
-We'll create a renderer autoplugger like this:
-
-!
-! GstAutoplug *autoplug;
-!
-! autoplug = gst_autoplugfactory_make ("staticrender");
-!
-
-
-b) finding out the source media type.
--------------------------------------
-
-Before we can start the autoplugger, we have to find out the
-source media type. This can be done using the typefind functions
-provided by various plugins.
-
-We will create a little pipeline to detect the media type by connecting
-a disksrc element to a typefind element. The typefind element will
-repeatedly call all registered typefind functions with the buffer it
-receives on its sink pad. when a typefind function returns a non NULL
-GstCaps*, that caps is set to the sink pad of the typefind element and
-a signal is emitted to notify the app.
-
-Due to caps negotiation, the disksrc will have the detected GstCaps*
-set on its src pad.
-
-We typically use a function like below to detect the type of a media stream
-on an element (typically a disksrc). The function accepts a pipeline and the
-element inside the pipeline on which the typefind should be performed (passing
-a GstPad* is probably a better option FIXME).
-
-!
-! static GstCaps*
-! gst_play_typefind (GstBin *bin, GstElement *element)
-! {
-! GstElement *typefind;
-! GstCaps *caps = NULL;
-!
-! typefind = gst_elementfactory_make ("typefind", "typefind");
-! g_return_val_if_fail (typefind != NULL, FALSE);
-!
-! gst_pad_connect (gst_element_get_pad (element, "src"),
-! gst_element_get_pad (typefind, "sink"));
-!
-! gst_bin_add (bin, typefind);
-!
-! gst_element_set_state (GST_ELEMENT (bin), GST_STATE_PLAYING);
-!
-! // push a buffer... the have_type signal handler will set the found flag
-! gst_bin_iterate (bin);
-!
-! gst_element_set_state (GST_ELEMENT (bin), GST_STATE_NULL);
-!
-! caps = gst_pad_get_caps (gst_element_get_pad (element, "src"));
-!
-! gst_pad_disconnect (gst_element_get_pad (element, "src"),
-! gst_element_get_pad (typefind, "sink"));
-! gst_bin_remove (bin, typefind);
-! gst_object_unref (GST_OBJECT (typefind));
-!
-! return caps;
-! }
-!
-
-Also note that the disksrc was added to the pipeline before calling this
-typefind function.
-
-When the function returns a non-NULL pointer, the media type has been
-determined and autoplugging can commence.
-
-Assume that in our mpeg1 use case the above function returns a GstCaps*
-like:
-
-!
-! srccaps = GST_CAPS_NEW ("mpeg1system_typefind",
-! "video/mpeg",
-! "mpegversion", GST_PROPS_INT (1),
-! "systemstream", GST_PROPS_BOOLEAN (TRUE)
-! );
-!
-
-
-c) Performing the autoplugging
-------------------------------
-
-Since we use the renderer API, we have to create the output elements
-that are going to be used as the final sink elements.
-
-!
-! osssink = gst_elementfactory_make("osssink", "play_audio");
-! videosink = gst_elementfactory_make("xvideosink", "play_video");
-!
-
-We then create a complex element using the following code.
-
-!
-! new_element = gst_autoplug_to_renderers (autoplug,
-! srccaps,
-! videosink,
-! osssink,
-! NULL);
-!
-! if (!new_element) {
-! g_print ("could not autoplug, no suitable codecs found...\n");
-! exit (-1);
-! }
-!
-
-2) Autoplugging internals
--------------------------
-
-We will now describe the internals of the above gst_autoplug_to_renderers()
-function call. This code is implemented in a plugin found in:
-
- gst/autoplug/gststaticautoplugrender.c
-
-
-
-a) phase1: create lists of factories.
----------------------------------------
-
-The autoplugger will start with executing the following piece of
-code:
-
-!
-! i = 0;
-!
-! for each sink:
-! {
-! sinkpad = take the first sinkpad of the sink (HACK)
-!
-! list[i] = gst_autoplug_caps (srccaps, sinkpad->caps);
-!
-! i++;
-! }
-!
-
-gst_autoplug_caps will figure out (based on the padtemplates)
-which elementfactories are needed to connect srccaps to sinkpad->caps
-and will return them in a list.
-
-The element list is created by using a modified shortest path algorithm
-by Dijkstra (http://www.orie.cornell.edu/~or115/handouts/handout3/handout3.html).
-The nodes of the graph are the elementfactories and the weight of the
-arcs is based on the pad compatibility of the padtemplates of the
-elementfactory. For incompatible elementfactories, we use a weight of
-MAX_COST (999999) and for compatible padtemplates we use 1.
-
-ex. we have two sinks with following caps:
-
-!
-! video/raw audio/raw
-! "...." "...."
-!
-
-gst_autoplug_caps will figure out that for the first sink the following
-elements are needed:
-
-!
-! mpeg1parse, mp1videoparse, mpeg_play
-!
-
-for the second sink the following is needed:
-
-!
-! mpeg1parse, mad
-!
-
-Note that for the audio connection the element list "mpeg1parse, mp3parse,
-mpg123" would also connect the srccaps to the audiosink caps. Since the
-"mpeg1parse, mad" list is shorter, it it always preferred by the autoplugger.
-
-We now have two lists of elementfactories.
-
-
-b) phase2: collect common elements from the lists and add them to a bin.
-------------------------------------------------------------------------
-
-The rationale is that from the lists we have created in phase1, there
-must be some element that is a splitter and that it has to come first (HACK)
-We try to find that element by comparing the lists until an element differs.
-
-We start by creating a toplevel bin that is going to be our complex element.
-
-In our use-case we find that mpeg1parse is an element common to both lists,
-so we add it to the bin. We then try to find a good ghostpad for the resulting
-complex element. This is done by looping over the sink pads of the first common
-element and taking the pad that is compatible with the srcaps.
-
-We end up with a bin like this:
-!
-! (----------------------)
-! ! autoplug_bin !
-! ! !
-! ! (------------) !
-! ! ! mpeg1parse ! !
-! ! - sink ! !
-! ! / (------------) !
-! sink !
-! (----------------------)
-!
-
-
-c) phase3: add remaining elements
----------------------------------
-
-now we loop over all the list and try to add the remaining elements
-
-(HACK) we always use a new thread for the elements when there is a common
-element found.
-
-if a new thread is needed (either because the previous element is a common
-element or the object flag of the next element is set to GST_SUGGEST_THREAD)
-we add a queue to the bin and we add a new thread. We add the elements to
-the bin and connect them using gst_pipeline_pads_autoplug.
-
-we finally arrive at the sink element and we're done.
-
-ex.
-
- we have just found our mpeg1parse common element, so we start a thread.
- We add a queue to the bin and a new thread, we add the elements
- mp1videoparse and mpeg_play to the thread. We arrive at the videosink, we
- see that the SUGGEST_THREAD flag is set, we add a queue and a thread and
- add the videosink in the thread.
-
- the same procedure happens for the audio part. We are now left with the
- following pipeline:
-
- We will also have set a signal "new_pad" on the mpeg1parse element because
- the element mp1videoparse could not be connected to the element just yet.
-
- (---------------------------------------------------------------------------------------------)
- !autoplug_bin !
- ! !
- ! (----------------------------------------) (------------) !
- ! !thread ! ! thread ! !
- ! (-----) ! (-------------) (---------) (-----) ! ! (---------)! !
- ! !queue! ! !mp1videoparse! !mpeg_play! !queue! ! ! !videosink!! !
- ! sink src-sink src-sink src-sink src-sink !! !
- ! (-----------) (-----) ! (-------------) (---------) (-----) ! ! (---------)! !
- ! ! mpeg1parse! (----------------------------------------) (------------) !
- ! - sink ! !
- ! / (-----------) !
- sink (----------------------------------------) (------------) !
- ! !thread ! ! thread ! !
- ! (-----) ! (-------------) (-----) ! ! (---------)! !
- ! !queue! ! !mad ! !queue! ! ! !videosink!! !
- ! sink src-sink src ------------ sink src-sink !! !
- ! (-----) ! (-------------) (-----) ! ! (---------)! !
- ! (----------------------------------------) (------------) !
- (---------------------------------------------------------------------------------------------)
-
- The autoplugger will return the autoplug_bin. the app will then connect the
- disksrc to the sinkpad of the autoplugged bin.
-
- Then we play, create_plan happens, data is flowing and the "new_pad" signal is called
- from mpeg1parse, gst_pipeline_pad_autoplug is called and the connection between
- mpeg1parse and the videoqueue is made. same for audio.
-
- Et voila. same procedure for mp3/vorbis/avi/qt/mpeg2 etc...
-
diff --git a/docs/random/bbb/streamselection b/docs/random/bbb/streamselection
deleted file mode 100644
index 234782e263..0000000000
--- a/docs/random/bbb/streamselection
+++ /dev/null
@@ -1,85 +0,0 @@
-Stream selection
-=
-
-1. Problem
-URIs (that being either a media stream or a media stream plus subtitle) can
-contain multiple streams of a type (audio, subtitle). A user has to be given
-the option of selecting one of those streams (or none altogether).
-
-2. Implementation ideas
-Stream selection, in GStreamer, has to be integrated at the player plugging
-level, which is (in the case of Totem) playbin. Playbin offers a feature to
-'mute' a stream (which means that no processing is done on that stream
-altogether, saving the decoding step). Currently, playbin will select the
-first occurrence of a stream type and mute all others. A queue is connected
-(for pre-roll) to the active stream. What is missing here is a way to change
-the active stream.
-Playbin interface - one possible interface could simply consist of a bunch of
-GObject properties: 'textstream' and 'audiostream', both integer. The number
-of available streams can be retrieved using the 'stream-info' GObject property.
-Similar to the 'nstreams' property, we could add utility GObject properties
-for getting the number of available audio/text streams ('naudiostreams' and
-'ntextstreams'). Names of these streams (like language name or so) can be
-added as an additional GObject property to streaminfo. Some container
-formats contain such names internally. Alternatively, we could allow those
-to be user-settable as well (for .sub files).
-On a set of either of these properties, playbasebin would mute the old
-selected stream (if any), unmute the newly selected stream (if any) and
-replug the preroll queue. The queue itself is disabled as well if no new
-stream was linked. Alternatively, a switch-like element is used, which
-requires no replugging. Pad disabling/enabling is then enough. This also
-makes relinking less painful. The switch-like element needs to proxy the
-active pads' caps. However, since those caps are (in practice) always the
-same across streams, caps setting will (inside the core) immediately
-return success.
-The switch-like element simply works like this:
-=
-static void
-loop_func (GstElement * element)
-{
- GList *inpads;
- GstPad *pad;
- GstData *data;
-
- for (inpads = ..; inpads != NULL; inpads = inpads->next) {
- pad = inpads->data;
- if (!GST_PAD_IS_USABLE (pad))
- continue;
-
- /* you'd also want some way to synchronize the inputs... */
- data = gst_pad_pull (pad);
- if (is_active_pad (pad))
- gst_pad_push (srcpad, data);
- else
- gst_data_unref (data);
- }
-}
-=
-It'd require an active-stream property itself, which (when set) takes
-care of doing renegotiation and so on. Using internal pad linkage is
-extremely useful here, and requires little code in the switch-like
-element itself. Note that there is a slight bit of duplication in the
-playbin interface and the switch-like element interface, but that's "just
-the way it is".
-The implementation of the switch-like element could initially be local to
-playbin, until it has been cleaned up and confirmed to be useful to a
-wider audience. This allows a lot of experimenting with interfaces because
-we won't be forced to maintain a stable interface.
-The current 'switch' element (gst-plugins/gst/switch/) already does a few
-of those operations, but stream synchronization, re-negotiation on stream
-changes, internal pad linkage and some other things are completely missing.
-If we're gonna use this element, it'll need a large overhaul. The choice of
-writing a new element or using an existing element as basis, and also the
-choice of whether or not to make this element local to playbin, should be
-based on technical merits and cost/effect analysis and not on personal
-pride.
-
-Notes:
-* seamless has the same switch-like element, but it's chain-based. Apart
-from scheduler considerations, this is a good idea, but limits its use
-(making either good docs and abuse-prevention [see multifilesrc] or
-private-to-playbin a prerequisite).
-* maybe text-* properties need to be renamed to subtitle-*.
-
-3. Written by
-Ronald S. Bultje <rbultje@ronald.bitfreak.net> - Jan. 2nd, 2005.
diff --git a/docs/random/bbb/subtitles b/docs/random/bbb/subtitles
deleted file mode 100644
index 6cbe204036..0000000000
--- a/docs/random/bbb/subtitles
+++ /dev/null
@@ -1,105 +0,0 @@
-Subtitles
-=========
-
-1. Problem
-GStreamer currently does not support subtitles.
-
-2. Proposed solution
- - Elements
- - Text-overlay
- - Autoplugging
- - Scheduling
- - Stream selection
-
-The first thing we'll need is subtitle awareness. I'll focus on AVI/MKV/OGM
-here, because I know how that works. The same methods apply to DVD subtitles
-as well. The matroska demuxer (and Ogg) will need subtitle awareness. For
-AVI, this is not needed. Secondly, we'll need subtitle stream parsers (for
-all popular subtitle formats), that can deal both with parsed streams (MKV,
-OGM) as well as .sub file chunks (AVI). Sample code is available in
-gst-sandbox/textoverlay/.
-
-Secondly, we'll need a textoverlay filter that can take text and video and
-blits text on video. We have several such elements (e.g. the cairo-based
-element) in gst-plugins already. Those might need some updates to work
-exactly as expected.
-
-Thirdly, playbin will need to handle all that. We expect subtitle streams
-to end up as subimages or plain text (or xhtml text). Note that playbin
-should also allow access to the unblitted subtitle as text (if available)
-for accessibility purposes.
-
-A problem popping up is that subtitles are no continuous streams. This is
-especially noticeable in the MKV/OGM case, because there the input of data
-depends on the other streams, so we'll only notice delays inside an element
-when we've received the next data chunk. There are two possible solutions:
-using timestamped filler events or using decoupled subtitle overlay elements
-(bins, probably). The first has as a difficulty that it only works well in
-the AVI/.sub case, where we will notice discontinuities before they become
-problematic. The second is more difficult to implement, but works for both
-cases.
-A) fillers
-Imagine that two subtitles come after each other, with 10 seconds of no-data
-in between. By parsing a .sub file, we would notice immediately and we could
-send a filler event (or empty data) with a timestamp and duration in between.
-B) decoupled
-Imagine this text element:
-------------------------------
-video ----- | actual element |out
-| / -----------------|
-text - - |
-------------------------------
-where the text pad is decoupled, like a queue. When no text data is available,
-the pad will have received no data, and the element will render no subtitles.
-The actual element can be a bin here, containing another subtitle rendering
-element. Disadvantage: it requires threading, and the element itself is (in
-concept) kinda gross. The element can be embedded in playbin to hide this
-fact (i.e. not be available outside the scope of playbin).
-Whichever solution we take, it'll require effort from the implementer.
-Scheduling (process, not implementation) knowledge is assumed.
-
-Stream selection is a problem that audio has, too. We'll need a solution for
-this at the playback bin level, e.g. playbin. By muting all unused streams
-and dynamically unmuting the selected stream, this is easily solved. Note
-that synchronization needs to be checked in this case. The solution is not
-hard, but someone has to do it.
-
-3. Written by
-Ronald S. Bultje <rbultje@ronald.bitfreak.net>, Dec. 25th, 2004
-
-
-Appendix A: random IRC addition
-<Company> intersting question: would it be a good idea to have a "max-buffer-length" property?
-<Company> that way demuxewrs would now how often they'd need to generate filler events
-<Company> s/now/know/
-<BBB> hm...
-<BBB> I don't think it's good to make that variable
-<Company> dunno
-<Company> (i'm btw always looking at this from the midi perspective, too)
-<Company> (because both subtitles and midi are basically the same in this regard)
-<BBB> and do you mean 'after the stream has advanced <time> and we didn't read a new subtitle in this mkv stream, we should send a filler'?
-<Company> yeah
-<BBB> it goes for avi with large init_delay values, too
-<Company> so you don't need to send fillers every frame
-<BBB> right
-<BBB> cant' we just set that to, for example, 1s?
-<BBB> it's fairly random, but still
-<Company> that's another option, too
-<Company> though you could write all file parsers with max-delay=MAXINT
-<Company> would make them a lot easier
-<BBB> it's true that queue size, for example, depends on this value
-<BBB> e.g. if you make this 5s and set queue size to 1s, it'll hang
-<Company> right
-<BBB> whereas if you set it to 1s and queue size to 5s, you waste space
-<BBB> :)
-<BBB> you ought to set it to max-delay * (n_streams + 1)
-<BBB> or so
-<BBB> or -1
-<BBB> I forgot
-<BBB> ohwell
-<Company> if you'd use filtercaps and queue sizes in your app, you could at least work around deadlocks
-<BBB> yeah
-<Company> though ideally it should just work of course...
-<BBB> good point...
-
-
diff --git a/docs/random/buffers b/docs/random/buffers
deleted file mode 100644
index 78ac8dca52..0000000000
--- a/docs/random/buffers
+++ /dev/null
@@ -1,19 +0,0 @@
-METADATA ON BUFFERS IS OUTDATED
--------------------------------
-
-
-Buffer mutability properties are the most important part of gst, and
-similarly are the most complex.
-
-The simple case is that a buffer is created, memory allocated, data put
-in it, and passed to the next filter. That filter reads the data, does
-something (like creating a new buffer and decoding into it), and
-unreferences the buffer. This causes the data to be freed and the buffer
-to be destroyed.
-
-A more complex case is when the filter modifies the data in place. It
-does so and simply passes on the buffer to the next element. This is just
-as easy to deal with.
-
-If the second filter adds metadata to the buffer, it just has to add the
-pointer to the list. The next element simply traverses the list and
diff --git a/docs/random/caps b/docs/random/caps
deleted file mode 100644
index f291b1fc12..0000000000
--- a/docs/random/caps
+++ /dev/null
@@ -1,209 +0,0 @@
-SOME OF THE FIRST IDEAS, PRETTY OUTDATED
-----------------------------------------
-
-
-During the course of a discussion on IRC, it turned out
-that there are many possible ways to handle the capabilities.
-
-A capability is basically a set of properties attached to a
-mimetype in order to more closely describe the mimetype.
-Capabilities are supposed to be attached to pads so that the
-autoplugging algorithm has more specific information to connect
-compatible pads.
-
-We present 3 possible implementation for the capabilities. we need
-to pick one of them.
-
-1. static capabilities
-----------------------
-
-When an element is created, it creates its pads like:
-
- mpg123->sinkpad = gst_pad_new ("sink", GST_PAD_SINK);
- gst_element_add_pad (GST_ELEMENT (mpg123), mpg123->sinkpad);
-
- mpg123->srcpad = gst_pad_new ("src", GST_PAD_SRC);
- gst_element_add_pad (GST_ELEMENT (mpg123), mpg123->srcpad);
-
-In the static capabilities case, it will attach a GstCaps* structure
-to the pad. The GstCaps structure in the above example might look like:
-
- static GstCapsFactory mpg123_sink_caps = {
- "audio/mp3",
- "layer", GST_CAPS_INT_RANGE (1, 3),
- "bitrate", GST_CAPS_INT_RANGE (8, 320),
- NULL
- };
-
-with
-
-mpg123sinkcaps = gst_caps_register (mpg123_sink_caps);
-
-the factory can be converted into a GstCaps* structure. The
-GstCaps* structure is attached to the pad with:
-
-gst_pad_add_caps (mpg123->sinkpad, mpg123sinkcaps);
-
-The GstElement would then have a sinkpad with the given
-mimetype (audio/mp3) and with the capabilitities of accepting
-mpeg layer 1 to 3 and a bitrate from 8 up to 320 Kbps.
-
-Likewise, the src pad could be set up in the same way. An
-example capability factory could look like:
-
- static GstCapsFactory mpg123_src_caps = {
- "audio/raw",
- "format", GST_CAPS_BITFIELD (...),
- "depth", GST_CAPS_INT (16),
- "rate", GST_CAPS_INT_RANGE (4000, 96000),
- "channels", GST_CAPS_INT_RANGE (1, 2),
- NULL
- };
-
-All GstElements would present their pads with the appropriate
-capabilities structure.
-
-The autoplugger would then proceed (once the source media type
-is known with a typefind function) in finding all the elements
-with compatible pads and connecting them into a pipeline.
-
-All elements of the complete pipeline could then be constructed
-with one single pass. No new elements should be added to the
-pipeline because we can figure out all the possibilities using the
-pad capabilities.
-
-We call this the static case because the capabilities of the pads
-are supposed to stay the same after creating the element.
-
-While the ability to completely setup the pipeline before actually
-starting playback is an advantage regarding performance, one obvious
-problem with this setup is that the static case may be too static in
-some cases. We can illustrate this with the following setup:
-
- ----------) (------------
- mpg123 ! ! audiosink
- src sink
- ! !
- ----------) (------------
-
-The mpg123 element has its src capabilities set up as mpg123_src_caps
-in the above example.
-
-The audio renderer has its capabilities set up with the following
-factory:
-
- static GstCapsFactory audio_sink_caps = {
- "audio/raw",
- "format", GST_CAPS_BITFIELD (...),
- "depth", GST_CAPS_INT (16),
- "rate", GST_CAPS_INT_RANGE (22000, 44000),
- "channels", GST_CAPS_INT_RANGE (1, 2),
- NULL
- };
-
-The static autoplugger has to be careful when connecting the mpg123
-element with the audiosink because it is theoretically possible that
-the mpg123 element outputs raw audio with a rate that cannot be
-handled by the audiosink (ex. 4000KHz). In the absence of another
-audiosink with more capabilities, the autoplugging of this simple
-pipeline will not be possible and would fail.
-
-the autoplugging algorithm would probably select another element to
-insert between the mpg123 element and the audiosink in order to handle
-the (uncommon) case of a rate conversion (audioscaler).
-
-It is clear that this static setup might even fail or work suboptimal
-for even the common case and should therefore be considered as too
-restrictive.
-
-
-2. dynamic capabilities
------------------------
-
-The idea of dynamic capabilities is that the capabilities are not set
-at element create time but rather while the pipeline is running.
-
-An element would still list its mime type using:
-
- gst_pad_add_type_id(mpg123->sinkpad, mp3type);
-
-The idea would then be that a rough draft of the pipeline would be
-built after the media type of the stream has been detected with the
-typefind functions. The rough draft would consist of laying out a
-global plan to reach the renderer(s). this plan would basically list
-the set of conversions that have to be performed. (mime-type to
-mime-type conversion).
-
-Elements that accept the src mime-type are tried by giving it a buffer.
-If the element accepts the buffer, it will set its capabilities for
-both the sink pad and the src pad. At that time other elements can be
-tried and added to the src pad, until we reach the renderer. As usual
-one has to be careful to add just the minimum amount of elements to
-reach the renderer. The global plan will help with that.
-
-Since we basically do not use the capabilities of the sink pad one has
-to question the need for sink pad capabilities in the first place.
-
-We might also have a hard time trying different elements until we find
-a compatible one that does not cause a dead end at some point.
-
-
-3. combined setup
------------------
-
-This combined setup will minimise the effort needed to try different
-elements encountered by option 2 while still allowing a more dynamic
-setup based on the actual media stream we are handling.
-
-The combined setup will list/add the sink capabilities at create time.
-It will only set the mime-type of its src pads.
-
-As with option2, a global plan will be built. At runtime the src pads
-will actually specify the capabilities they need for any element that
-wants to be connected to its source pads.
-
-In this case we specify the capabilities for all the sink pads of an
-element at create time. The capabilities of the src pads would only
-become available when data has been processed by the element.
-
-The autoplugger would then be able to choose an element that can handle
-the capability listed by the src pad.
-
-in our previous example:
-
- ----------) (------------
- mpg123 ! ! audiosink
- src sink
- ! !
- ----------) (------------
-
-the audiosink element would specify its sink pad capabilities at create
-time, while the mpg123 elements src pad would not yet have any capabilities
-set.
-
-When data is handled by the mpg123 element, a capability would be added to
-the mpg123 src pad. This capability might be:
-
- static GstCapsFactory mpg123_src_caps = {
- "audio/raw",
- "format", GST_CAPS_INT (S16),
- "depth", GST_CAPS_INT (16),
- "rate", GST_CAPS_INT (44000),
- "channels", GST_CAPS_INT (2),
- NULL
- };
-
-This capability would be compatible with the audiosinks sinkpad capabilities
-and the autoplugger would therefore be able to connect the two elements.
-
-While allowing a more flexible setup with option3, compared to option1, we
-introduce a slightly higher overhead when we need to dynamically connect
-elements. This overhead will not be as big as option2 because we don't
-have to 'try' elements.
-
-so:
-
- src caps: added at runtime to list the caps needed for an element that
- wants to connect to this pad.
- sink caps: the (static) capabilities that this sinkpad has.
-
diff --git a/docs/random/caps.dia b/docs/random/caps.dia
deleted file mode 100644
index b511083c7a..0000000000
--- a/docs/random/caps.dia
+++ /dev/null
@@ -1,1132 +0,0 @@
-<?xml version="1.0"?>
-<diagram xmlns:dia="http://www.lysator.liu.se/~alla/dia/">
- <diagramdata>
- <attribute name="background">
- <color val="#ffffff"/>
- </attribute>
- <attribute name="paper">
- <composite type="paper">
- <attribute name="name">
- <string>#A4#</string>
- </attribute>
- <attribute name="tmargin">
- <real val="2.82"/>
- </attribute>
- <attribute name="bmargin">
- <real val="2.82"/>
- </attribute>
- <attribute name="lmargin">
- <real val="2.82"/>
- </attribute>
- <attribute name="rmargin">
- <real val="2.82"/>
- </attribute>
- <attribute name="is_portrait">
- <boolean val="true"/>
- </attribute>
- <attribute name="scaling">
- <real val="1"/>
- </attribute>
- <attribute name="fitto">
- <boolean val="false"/>
- </attribute>
- </composite>
- </attribute>
- <attribute name="grid">
- <composite type="grid">
- <attribute name="width_x">
- <real val="1"/>
- </attribute>
- <attribute name="width_y">
- <real val="1"/>
- </attribute>
- <attribute name="visible_x">
- <int val="1"/>
- </attribute>
- <attribute name="visible_y">
- <int val="1"/>
- </attribute>
- </composite>
- </attribute>
- <attribute name="guides">
- <composite type="guides">
- <attribute name="hguides"/>
- <attribute name="vguides"/>
- </composite>
- </attribute>
- </diagramdata>
- <layer name="Background" visible="true">
- <object type="UML - Class" version="0" id="O0">
- <attribute name="obj_pos">
- <point val="0.75,5.55"/>
- </attribute>
- <attribute name="obj_bb">
- <rectangle val="0.7,5.5;12.7504,9.6"/>
- </attribute>
- <attribute name="elem_corner">
- <point val="0.75,5.55"/>
- </attribute>
- <attribute name="elem_width">
- <real val="11.9504"/>
- </attribute>
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- </attribute>
- <attribute name="orth_points">
- <point val="16.0404,22.4"/>
- <point val="16.0404,18.25"/>
- <point val="20.4828,18.25"/>
- <point val="20.4828,16.65"/>
- </attribute>
- <attribute name="orth_orient">
- <enum val="1"/>
- <enum val="0"/>
- <enum val="1"/>
- </attribute>
- <attribute name="name">
- <string/>
- </attribute>
- <attribute name="direction">
- <enum val="0"/>
- </attribute>
- <attribute name="ends">
- <composite>
- <attribute name="role">
- <string/>
- </attribute>
- <attribute name="multiplicity">
- <string/>
- </attribute>
- <attribute name="arrow">
- <boolean val="false"/>
- </attribute>
- <attribute name="aggregate">
- <enum val="0"/>
- </attribute>
- </composite>
- <composite>
- <attribute name="role">
- <string/>
- </attribute>
- <attribute name="multiplicity">
- <string/>
- </attribute>
- <attribute name="arrow">
- <boolean val="false"/>
- </attribute>
- <attribute name="aggregate">
- <enum val="1"/>
- </attribute>
- </composite>
- </attribute>
- <connections>
- <connection handle="0" to="O6" connection="1"/>
- <connection handle="1" to="O3" connection="6"/>
- </connections>
- </object>
- <object type="UML - Association" version="0" id="O14">
- <attribute name="obj_pos">
- <point val="27.0208,22.55"/>
- </attribute>
- <attribute name="obj_bb">
- <rectangle val="19.0328,15.2;28.4708,24.15"/>
- </attribute>
- <attribute name="orth_points">
- <point val="27.0208,22.55"/>
- <point val="27.0208,18.25"/>
- <point val="20.4828,18.25"/>
- <point val="20.4828,16.65"/>
- </attribute>
- <attribute name="orth_orient">
- <enum val="1"/>
- <enum val="0"/>
- <enum val="1"/>
- </attribute>
- <attribute name="name">
- <string>#union#</string>
- </attribute>
- <attribute name="direction">
- <enum val="0"/>
- </attribute>
- <attribute name="ends">
- <composite>
- <attribute name="role">
- <string/>
- </attribute>
- <attribute name="multiplicity">
- <string/>
- </attribute>
- <attribute name="arrow">
- <boolean val="false"/>
- </attribute>
- <attribute name="aggregate">
- <enum val="0"/>
- </attribute>
- </composite>
- <composite>
- <attribute name="role">
- <string/>
- </attribute>
- <attribute name="multiplicity">
- <string/>
- </attribute>
- <attribute name="arrow">
- <boolean val="false"/>
- </attribute>
- <attribute name="aggregate">
- <enum val="1"/>
- </attribute>
- </composite>
- </attribute>
- <connections>
- <connection handle="0" to="O8" connection="1"/>
- <connection handle="1" to="O3" connection="6"/>
- </connections>
- </object>
- <object type="UML - Association" version="0" id="O15">
- <attribute name="obj_pos">
- <point val="31.5416,24.45"/>
- </attribute>
- <attribute name="obj_bb">
- <rectangle val="24.7656,12.7;36.3,25.9"/>
- </attribute>
- <attribute name="orth_points">
- <point val="31.5416,24.45"/>
- <point val="34.85,24.45"/>
- <point val="34.85,14.15"/>
- <point val="26.2156,14.15"/>
- </attribute>
- <attribute name="orth_orient">
- <enum val="0"/>
- <enum val="1"/>
- <enum val="0"/>
- </attribute>
- <attribute name="name">
- <string/>
- </attribute>
- <attribute name="direction">
- <enum val="0"/>
- </attribute>
- <attribute name="ends">
- <composite>
- <attribute name="role">
- <string/>
- </attribute>
- <attribute name="multiplicity">
- <string>#1#</string>
- </attribute>
- <attribute name="arrow">
- <boolean val="false"/>
- </attribute>
- <attribute name="aggregate">
- <enum val="2"/>
- </attribute>
- </composite>
- <composite>
- <attribute name="role">
- <string/>
- </attribute>
- <attribute name="multiplicity">
- <string>#*#</string>
- </attribute>
- <attribute name="arrow">
- <boolean val="false"/>
- </attribute>
- <attribute name="aggregate">
- <enum val="0"/>
- </attribute>
- </composite>
- </attribute>
- <connections>
- <connection handle="0" to="O8" connection="9"/>
- <connection handle="1" to="O3" connection="4"/>
- </connections>
- </object>
- </layer>
-</diagram>
diff --git a/docs/random/caps2 b/docs/random/caps2
deleted file mode 100644
index 3ca0a747fb..0000000000
--- a/docs/random/caps2
+++ /dev/null
@@ -1,231 +0,0 @@
-
-The new caps code uses the type name GstCaps2 and the function
-names gst_caps2_*(). Before the CAPS branch is merged, there
-will be a global change from caps2 to caps. Since GstCaps is
-no longer defined, it no longer compiles, thus highlighting
-exactly what needs to be changed in an element.
-
-
-
-Pad Templates:
-
-Old style:
-
- GST_PAD_TEMPLATE_FACTORY (fakesrc_src_factory,
- "src%d",
- GST_PAD_SRC,
- GST_PAD_REQUEST,
- GST_CAPS_ANY
- );
-
-New style:
-
- GstStaticPadTemplate fakesrc_src_template = GST_STATIC_PAD_TEMPLATE (
- "src%d",
- GST_PAD_SRC,
- GST_PAD_REQUEST,
- GST_STATIC_CAPS2_ANY
- );
-
-The old style defined a function called fakesrc_src_factory(), which,
-when called, returns a pad template. The new style defines a
-GstStaticPadTemplate, which can be converted to a GstPadTemplate
-by the function gst_static_pad_template_get(). The 4th argument
-is also different -- previously it would call the GST_CAPS_NEW()
-function. Now it is a GstStaticCaps.
-
-Not every pad template can be converted to a GstStaticPadTemplate,
-particularly those which create caps from another source at runtime,
-such as videotestsrc.
-
-Caps:
-
-Old style:
-
- GST_CAPS_NEW (
- "sinesrc_src",
- "audio/x-raw-int",
- "endianness", GST_PROPS_INT (G_BYTE_ORDER),
- "signed", GST_PROPS_BOOLEAN (TRUE),
- "width", GST_PROPS_INT (16),
- "depth", GST_PROPS_INT (16),
- "rate", GST_PROPS_INT_RANGE (8000, 48000),
- "channels", GST_PROPS_INT (1)
- )
-
-New style:
-
- GST_STATIC_CAPS2 ( "audio/x-raw-int, "
- "endianness = (int) " G_STRINGIFY (G_BYTE_ORDER) ", "
- "signed = (boolean) true, "
- "width = (int) 16, "
- "depth = (int) 16, "
- "rate = (int) [ 8000, 48000 ], "
- "channels = (int) 1"
- )
-
-The old style calls a function that creates a GstCaps. The new style
-stores a string in a GstStaticCaps2, and this string is converted to
-a caps in the function gst_static_caps2_get().
-
-Note that the old caps name is no longer used.
-
-Old style:
-
- caps = GST_CAPS_NEW ("videotestsrc_filter",
- "video/x-raw-rgb",
- "bpp", GST_PROPS_INT(format->bitspp),
- "endianness", GST_PROPS_INT(endianness),
- "depth", GST_PROPS_INT(format->depth),
- "red_mask", GST_PROPS_INT(format->red_mask),
- "green_mask", GST_PROPS_INT(format->green_mask),
- "blue_mask", GST_PROPS_INT(format->blue_mask));
-
-New style:
-
- caps = gst_caps2_new_simple("video/x-raw-rgb",
- "bpp", G_TYPE_INT, format->bitspp,
- "endianness", G_TYPE_INT, endianness,
- "depth", G_TYPE_INT, format->depth,
- "red_mask", G_TYPE_INT, format->red_mask,
- "green_mask", G_TYPE_INT, format->green_mask,
- "blue_mask", G_TYPE_INT, format->blue_mask);
-
-Not everything can be converted in this way, especially lists and
-ranges.
-
-
-IMPLEMENTATION
-
-Pad Capabilities (caps) are mathematical sets that represent all the
-possible stream types that a pad can use. These general sets are
-represented by unions of simpler sets known as caps structures. Each
-caps structure has a media type (e.g., "audio/mpeg") and a number of
-properties. Each property has a name and a GValue. In normal
-circumstances, the GValue will have the types int, boolean, string,
-fourcc, and double. Simple sets are constructed by using GValues
-that are lists of other GValues, or the special types that represent
-int ranges and double ranges.
-
-A "fixed" caps represents exactly one media format. This means that
-the caps is a union of exactly one caps structure, and each property
-in the caps structure is a simple type, i.e., no ranges or lists.
-
-There are two special caps values, "ANY" which represents the union
-of all stream types, and "EMPTY", which represents the set of no
-stream types. The ANY caps is often used on generic elements that
-handle any type of data (e.g., filesrc and filesink). The EMPTY
-caps is the return value of gst_caps_intersect(), when the two
-given caps do not intersect. In many cases, using EMPTY is invalid.
-
-
-CAPS NEGOTIATION
-
-Elements provide information to the core about what stream formats
-they understand in four ways: the caps in the pad templates, the
-caps returned by a pad's getcaps function, accepting/denying
-a given caps in the pad link function, and a new fixate function.
-
-The pad template caps should be the union of caps a pad supports
-in any potential situation. Simultaneously, these caps should be
-as specific as possible, since it is used to decide which elements
-to attempt for autoplugging, without having to load the element.
-The pad template caps are generally determined at compile time, but
-might be actually computed at run-time from other information.
-
-The getcaps() function returns the caps supported by a given pad,
-in the context of the element's state, its link to other elements,
-and the devices or files it has opened. These caps must be a
-subset of the pad template caps. In the NULL state with no links,
-the getcaps function should ideally return the same caps as the
-pad template. In rare circumstances, an object property can affect
-the caps returned by getcaps, but this is discouraged. For most
-filters, the caps returned by getcaps is directly affected by the
-allowed caps on other pads. For demuxers and decoders, the caps
-returned by the srcpad's getcaps function is directly related to
-the stream data. Again, getcaps should return the most specific
-caps it reasonably can, since this helps with autoplugging.
-
-The pad link function is the last step in negotiating caps. The
-core calls the pad link function with a fixed caps, meaning that
-the stream format is precisely defined, with the caps having one
-structure, with no fields that are ranges or lists.
-
-There is also a new pad function "fixate", which is used to help
-choose a fixed caps from a non-fixed caps. This is called in
-situations where normal negotiation cannot decide on a fixed caps.
-You should almost never implement a fixate function, please ask
-me if it is appropriate for your case. Fixate functions are called
-iteratively on the pads until a fixed caps is found. Fixate functions
-are called with a const caps, and should return a caps that is a
-strict subset of the given caps. That is, the function should
-create a caps that is "more fixed" than previously, but does not
-have to return fixed caps. If the fixate function can't provide
-more fixed caps, it should return NULL.
-
-
-
-Checklist for getcaps:
-
- - The getcaps function prototype no longer has the caps parameter.
- Remove it.
-
- - The returned caps is owned by the caller. Make sure you don't
- keep a pointer to the caps.
-
- - Make sure that the getcaps function can be called safely in each
- element state (NULL, READY, PAUSED, PLAYING), and for any element
- configuration (properties, links, devices/files opened or not,
- error state, etc.)
-
- - Make sure that the returned caps do not depend on the caps that
- indicate the stream type that the pad is currently using.
-
-Checklist for pad_link:
-
- - The pad link function prototypes uses a const GstCaps *.
-
- - Pad link functions are called with fixed caps. There's no need
- to check for this. This means that you can assume that the caps
- is not ANY or EMPTY, and that there is exactly one structure in
- the caps, and that all the fields in the structure are fixed.
-
- - Pad link functions are called with caps that are a subset of the
- most recent return value of the pad's getcaps function. Generally,
- the getcaps function was called immediately prior to calling the
- src_link function. For 0.8, you can assume that nothing has changed
- in your element that would cause a change to the return value of
- getcaps.
-
- - the return value GST_PAD_LINK_OK should be used when the caps are
- acceptable, and you've extracted all the necessary information from
- the caps and set the element's internal state appropriately.
-
- - the return value GST_PAD_LINK_REFUSED should be used when the caps
- are unacceptable for whatever reason.
-
- - the return value GST_PAD_LINK_DELAYED should be used when the
- element is in a state where it can't determine whether the caps
- are acceptable or not. This is often used if the element needs
- to open a device or process data before determining acceptable
- caps.
-
- - the pad_link function must not call gst_caps_try_set_caps() on
- the pad that was specified as a parameter.
-
- - the pad_link function may (and often should) call
- gst_caps_try_set_caps() on pads that are not specified as the
- pad parameter.
-
-Checklist for fixate:
-
- - Make sure you actually should be using a fixate function. Fixate
- functions are reasonable for non-fixed primary sources, such as
- videotestsrc, v4lsrc, and osssrc.
-
- - The user_data parameter is mainly used for user-provided fixate
- function. It should be ignored in element fixate functions.
-
-
-
-
diff --git a/docs/random/classes.dia b/docs/random/classes.dia
deleted file mode 100644
index de4aa3e05b..0000000000
--- a/docs/random/classes.dia
+++ /dev/null
Binary files differ
diff --git a/docs/random/ds/0.9-planning b/docs/random/ds/0.9-planning
deleted file mode 100644
index f9f60e9fc0..0000000000
--- a/docs/random/ds/0.9-planning
+++ /dev/null
@@ -1,165 +0,0 @@
-
-
-
-Scheduling:
-
- - remove loop/get/chain from GstElement and add a "iterate" method.
- The iterate method is called with the event (or events) that
- triggered it, performs some action, and resets the events (file
- descriptors becoming readable, semaphores, pads becoming readable
- or writable, or a time occurs).
-
- - Add GstLoopElement, GstChainElement, etc. for compatibility.
-
- - Remove existing state handling and create 2 states, "playing" and
- "stopped". "playing" means that the iterate() method of the
- element may be called, that is, the element is allowed to move
- buffers, negotiate, etc. "stopped" means that no gstreamer-ish
- things happen to an element, only gobject-ish. A separate
- reset() method will handle the difference between READY and NULL.
-
- - Add a flag "ready" to GstElement that is under the control of the
- element. If the element is ready to stream, it sets this flag,
- and the entire pipeline starts streaming. (This is basically
- the difference between PAUSED and PLAYING.) For example, osssink
- won't set the ready flag until the device is opened and there is
- a buffer available to write to the device.
-
- - Scheduling of elements and movement of buffers will be timed by
- clocks.
-
-
-
-Example #1:
-
- Pipeline: sinesrc ! osssink
-
- - The application creates the pipeline and sets it to "playing".
-
- - The clock is created and set to "paused".
-
- - sinesrc.iterate() decides to watch for the event "src pad
- negotiation" and sets the available caps on the pad.
-
- - osssink.iterate() opens device, determines available caps, and
- sets the available caps on the pad. Then it decides to wait for
- "sink pad negotiation".
-
- - The scheduler realizes that the two elements are waiting for
- negotiation, so it negotiates the link.
-
- - sinesrc.iterate() sets the "ready" flag (because it needs no more
- preparation to stream) and decides to watch for the event "src
- pad ready to accept buffer".
-
- - osssink.iterate() decides to watch for the event "sink pad has
- available buffer".
-
- - The scheduler realizes that sinesrc.srcpad is now ready, so it
- calls sinesrc.iterate()
-
- - sinesrc.iterate() creates a buffer and pushes it, and decides to
- wait for the same event.
-
- - The scheduler realizes that osssink.sinkpad now has a buffer, so
- it calls osssink.iterate().
-
- - osssink.iterate() is now ready to stream, so it sets the "ready"
- flag and waits for "time 0".
-
- - The pipeline is now completely ready, so the clock may be
- started. A signal is fired to let the application know this
- (and possibly change the default behavior).
-
- - The clock starts with the time 0. The scheduler realizes this,
- and decides to schedule osssink.
-
- - osssink.iterate() is called, and writes the buffer to the device.
- This starts the clock counting. (Actually, the buffer could be
- written by the clock code, since presumably the clock is related
- to osssink.) iterate() then waits for "sink pad has available
- buffer".
-
- We're now basically in streaming mode. A streaming cycle:
-
- - osssink.iterate() decides the audio output buffer is full enough,
- so it waits for "time X", where X is the time when the output
- buffer will be below some threshold.
-
- - osssink.iterate() waits for "sink pad has available buffer"
-
- - sinesrc.iterate() creates and pushes a buffer, then waits for
- "src pad ready".
-
-
- Further ideas:
-
- - osssink can set a hard deadline time, which means that if it is
- not scheduled before that time, you'll get a skip. Skipping
- involves setting osssink to "not ready" and pauses the clock.
- Then the scheduler needs to go through the same process as above
- to start the clock.
-
- - As a shortcut, osssink can say "I need a buffer on the sinkpad
- at time X". This information can be passed upstream, and be used
- in filters -- filter.sinkpad says "I need a buffer at time X-N",
- where N is the latency of the filter.
-
-
-Example #2:
-
- Pipeline: osssrc ! osssink
-
- - The application creates the pipeline and sets it to "playing".
-
- - The clock is created and set to "paused".
-
- - negotiation happens roughly as in example #1, although osssrc
- additionally opens and prepares the device.
-
- - osssrc.iterate() sets the "ready" flag (because it needs no more
- preparation to stream) and waits for "time 0", since it presumably
- can't wait for the file descriptor (audio input hasn't been
- enabled on the device yet.)
-
- - osssink.iterate() decides to watch for the event "sink pad has
- available buffer".
-
- - The scheduler realizes the deadlock and (somehow) tells osssink
- that it can't pre-roll. (This needs more work) In other words,
- osssink can't be the clock master, but only a clock slave.
-
- - osssink.iterates() agrees to start at time SOME_LATENCY, sets the
- "ready" flag, and waits for a buffer on its sink pad.
-
- - The pipeline is now completely ready, so the clock may be
- started. A signal is fired to let the application know this
- (and possibly change the default behavior).
-
- - The clock starting causes two things to happen: osssrc starts
- the recording of data, and osssink starts the outputting of data.
- The data being output is a chunk of silence equal to SOME_LATENCY.
-
- - osssrc.iterate() is called for "time 0", does nothing, and waits
- on the file descriptor (via the scheduler, of course). All waiting
- on file descriptors should have an associated timeout.
-
- - osssrc.iterate() is called when the file descriptor is ready,
- reads a chunk of data, and pushes the buffer. It then waits for
- its file descriptor to be ready.
-
- - osssink.iterate() is called
-
-
-Evil:
-
- fakesrc ! tee ! fakesink tee0. ! never_accept_a_buffer_sink
-
- sinesrc ! osssink videotestsrc ! ximagesink
-
- fakesrc ! fakesink (pausing)
-
- sinesrc ! identity ! osssink
-
-
-
diff --git a/docs/random/ds/buffer_locking b/docs/random/ds/buffer_locking
deleted file mode 100644
index 8200a81e6c..0000000000
--- a/docs/random/ds/buffer_locking
+++ /dev/null
@@ -1,30 +0,0 @@
-
-
-Buffers should have readlocks and writelocks to enforce
-GST_BUFFER_DONTKEEP and relax the restriction that buffers with
-multiple refcounts are read-only.
-
-
-
-Example:
-
- videotestsrc ! ximagesink
-
- videotestsrc requests a buffer from its src pad
-
- ximagesink creates a buffer (refcount:1 readlock:0 writelock:0)
-
- videotestsrc writelocks it (refcount:1 readlock:0 writelock:1)
-
- videotestsrc writes to the buffer
-
- videotestsrc un-writelocks it (refcount:1 readlock:0 writelock:0)
-
- ximagesink readlocks it (refcount:1 readlock:1 writelock:0)
-
- ximagesink writes it to the screen
-
- ximagesink un-readlocks it (refcount:1 readlock:0 writelock:0)
-
-
-
diff --git a/docs/random/ds/bufferpools b/docs/random/ds/bufferpools
deleted file mode 100644
index b0f211bca5..0000000000
--- a/docs/random/ds/bufferpools
+++ /dev/null
@@ -1,57 +0,0 @@
-On Thu, Jan 08, 2004 at 04:10:00PM +0100, Julien MOUTTE wrote:
->
-> Hi David,
->
-> I'd like to implement bufferpools again in x[v]imagesink asap.. Could
-> you please point me to a template/doc on how to do that ?
->
-> The best for me would be a simple testcase showing how to use the buffer
-> free methods to replace bufferpools.
-
-
-x[v]imagesink should call gst_pad_set_bufferalloc_function() on
-their sink pads with a bufferalloc implementation. This bufferalloc
-function is to allocate buffers that _peers_ will send _to_ that pad.
-
-A trivial version of a bufferalloc function, i.e., one that just
-allocates normal buffers:
-
-static GstBuffer *
-gst_ximagesink_sink_bufferalloc (GstPad *pad, guint64 offset, guint
-size)
-{
- GstBuffer *buffer;
-
- buffer = gst_buffer_new_and_alloc (size);
- GST_DATA_FREE_FUNC (data) = gst_ximagesink_buffer_free;
- GST_BUFFER_POOL_PRIVATE (data) = ximagesink; /* whatever */
-
- return buffer;
-}
-
-static void
-gst_ximagesink_buffer_free (GstData *data)
-{
- g_free (GST_BUFFER_DATA (data));
-}
-
-The hard part is going through each element, and every time it
-allocates a buffer using gst_buffer_new_and_alloc () that is then
-sent to a sink pad, the call should be replaced with
-gst_pad_alloc_buffer (sinkpad, offset, size).
-
-
-
-dave...
-
-
-
--------------------------------------------------------
-This SF.net email is sponsored by: Perforce Software.
-Perforce is the Fast Software Configuration Management System offering
-advanced branching capabilities and atomic changes on 50+ platforms.
-Free Eval! http://www.perforce.com/perforce/loadprog.html
-_______________________________________________
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-gstreamer-devel@lists.sourceforge.net
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diff --git a/docs/random/ds/categories b/docs/random/ds/categories
deleted file mode 100644
index 2636f110c4..0000000000
--- a/docs/random/ds/categories
+++ /dev/null
@@ -1,50 +0,0 @@
-
-Element Categories:
--------------------
-
-
-Decoder:
-
-
-Encoder:
-
-
-Converter:
- A converter has one or more source pads and one or more sink pads.
- Converter pads may have different caps templates. Converters are
- expected
-
-
-Filter:
- A filter has one source and one sink pad. These pads have the same
- caps and (thus) the same caps template.
-
- Filters generally do not handle events.
-
- Filters may have interfaces.
-
- Filters are generally not autoplugged unless they have interfaces.
-
-
-Source:
- A source has one source pad and no sink pads.
-
- Sources usually handle events.
-
- Sources may have interfaces.
-
- Sources are not autoplugged.
-
-Sink:
- A sink has one sink pad and no source pads.
-
- Sources usually handle events.
-
- Sources may have interfaces.
-
- Sinks are not autoplugged.
-
-
-
-Converter/Colorspace
-
diff --git a/docs/random/ds/element-checklist b/docs/random/ds/element-checklist
deleted file mode 100644
index fcac4e45f5..0000000000
--- a/docs/random/ds/element-checklist
+++ /dev/null
@@ -1,43 +0,0 @@
-
-This is a list of things to check in elements.
-
-
-
-1. Add a "Element-Checklist-Version: 1" comment
-
-This comment is to indicate which items on this checklist have been
-checked, so that future checkers can skip things that you've already
-carefully checked. This number should obviously be updated whenever
-you do additional checks. Ideally, you should not do checks out of
-order, but using "1, also 4-6" is acceptable.
-
-
-2. Each pad should have a pad template
-
-Create and register a pad template to the element class.
-
-
-3. config.h
-
-Make sure that each .c file includes config.h (conditionally on
-HAVE_CONFIG_H). Make sure that each .h file does _not_ include
-config.h.
-
-4. src event handling
-
-Every element having multiple sink pads or having a specific need to handle
-src events in a different manner than the default event handler should do that
-correctly. Correctly means that :
-
-- It should not leak the event if it is handled (gst_event_unref)
-- It should call return gst_pad_event_default (pad, event) for all non handled
-localy events. Otherwise it breaks navigation, seeking, etc...
-- it should return a TRUE/FALSE boolean if the localy handled event was handled.
-
-
-
-
-other ideas:
-
-- plugins should avoid using gst_caps_to_string() in debug statement.
- They should use %"GST_PTR_FORMAT" instead. Check all usage for leaks.
diff --git a/docs/random/ds/registry b/docs/random/ds/registry
deleted file mode 100644
index ff06ef6ab5..0000000000
--- a/docs/random/ds/registry
+++ /dev/null
@@ -1,15 +0,0 @@
-
-Ideas for a new registry system:
-
-- each plugin has an associated registry XML file. Each xml file in
- a plugin directory is loaded to create the in-core registry.
-
-- needed for avidemux:
-
- <Company> plugin x create some registry information that says stuff like:
- <Company> "caps x are fourcc FOOO in avi"
- <Company> or "caps y may be contained in id3 tagged files"
- <Company> or "caps z is a valid format to put in an ogg container"
-
-
-
diff --git a/docs/random/ds/roadmap b/docs/random/ds/roadmap
deleted file mode 100644
index 4a54bb8084..0000000000
--- a/docs/random/ds/roadmap
+++ /dev/null
@@ -1,79 +0,0 @@
-
-After some discussion on IRC, I think it's time to propose a
-formal plan for the next 6 months or so to the list.
-
-I think most people are in agreement that we need a period of time
-in which we can focus on improving non-core bits (like elements,
-schedulers, autopluggers, applications, etc.) In the last cycle,
-there was a lot of time spent with a significant number of plugins
-broken, and it's realistic to assume that this could happen again
-in a 0.9 unstable series.
-
-What I propose is:
-
- - We continue to develop the 0.8.x series as HEAD, with the obvious
- requirement that all changes be ABI/API compatible.
-
- - API additions are encouraged, as long as they are well-thought-out.
-
- - Significant API additions should be developed on a separate branch
- (not HEAD) to test out any bugs.
-
-In mid-August (or so, in order to coordinate with GNOME-2.8), we have
-two options:
-
- - Continue with 0.8.x releases, obviously ABI compatible with 0.8.0.
-
- - or, remove deprecated functions, readjust the padding on structures,
- perhaps make a few additional ABI changes [1], and quickly go to
- 0.10.0.
-
-I prefer the latter, although we don't have to decide that until
-later. In either case, there should be no API changes that affect
-more than a bare minimum of elements or applications.
-
-A few things that we won't be able to do without a true unstable
-branch are:
-
- - using GstStructure for all GstEvents.
-
- - significant clock changes
-
- - significant scheduling changes
-
- - separation of headers into application and plugin headers
-
- - anything that requires modification of every plugin
-
-There are perils with having HEAD being the stable branch,
-specifically that bugs can creep in and accidentally cause regressions
-in releases. I'm hoping that the introduction of media regression
-testing and also the development of new testsuites will keep this
-to a minimum. Don't forget that accidental bugs that get into
-releases typically cause rude IRC conversations, which we really
-don't need. Please keep the bugs (and the rudeness) to a minimum.
-
-Also, keep in mind that we will have to live with 0.8's unfixable
-bugs for an entire year.[2]
-
-
-
-dave...
-
---
-[1] I'm thinking about making GstData a subclass of GTypeInstance or
- GObject.
-[2] But then, 0.6 was 1 year ago, and felt a lot more buggy when it
- was released.
-
-
-
--------------------------------------------------------
-This SF.Net email is sponsored by: IBM Linux Tutorials
-Free Linux tutorial presented by Daniel Robbins, President and CEO of
-GenToo technologies. Learn everything from fundamentals to system
-administration.http://ads.osdn.com/?ad_id=1470&alloc_id=3638&op=click
-_______________________________________________
-gstreamer-devel mailing list
-gstreamer-devel@lists.sourceforge.net
-https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
diff --git a/docs/random/ensonic/audiobaseclasses.txt b/docs/random/ensonic/audiobaseclasses.txt
deleted file mode 100644
index e31047fbde..0000000000
--- a/docs/random/ensonic/audiobaseclasses.txt
+++ /dev/null
@@ -1,18 +0,0 @@
-$Id$
-
-= audio base classes =
-We have gst-plugins-base/gst-libs/gst/audio with some audio helper and base
-classes.
-
-audiofilter: for src_caps=sink_caps
-basetransform: *
-
-= new stuff =
-
-
-= todo =
-* mv gstaudiofilterexample -> gst-template
-* make more elements using the baseclass
- * base: audiorate, audioresample, volume
- * good: audiopanorama
-
diff --git a/docs/random/ensonic/dparams.txt b/docs/random/ensonic/dparams.txt
deleted file mode 100644
index 27dc4b74c8..0000000000
--- a/docs/random/ensonic/dparams.txt
+++ /dev/null
@@ -1,14 +0,0 @@
-$Id$
-
-* controller changes
- * handling of G_TYPE_XXX is distributed over
- gst-controller.c::gst_controlled_property_set_interpolation_mode()
- gst-controller.c::gst_controlled_property_new
- gst-interpolation.c
- * it would be better to put handlers for each type into a struct
- and register it into a hashmap with the G_TYPE as the key
- * functions in gst-controller.c could try the key,
- if no handler is registered the try the type_parent and so on
-
-* implement quadric/cubic interpolation
-
diff --git a/docs/random/ensonic/draft-registry-change-hooks.txt b/docs/random/ensonic/draft-registry-change-hooks.txt
deleted file mode 100644
index 102d6cefbb..0000000000
--- a/docs/random/ensonic/draft-registry-change-hooks.txt
+++ /dev/null
@@ -1,56 +0,0 @@
-Registry Change Hooks
-----------------------
-
-This document proposes a mechanism to register on registry updates.
-
-Last edited: 2009-11-09 Stefan Kost
-
-
-Current Behaviour
------------------
-
-When new plugins are installed or some are removed, the next application that
-runs rebuiilds the registry. Any application that runs later on won't know about
-it.
-
-
-Problem
--------
-
-Some usecases need to cache gstreamer capabilities. They need to know when the
-cache needs to be rebuilt. We could either export the registry mtime as api or
-do the following.
-
-This gets more useful once we can have more plugin/element metadata in the
-registry (e.g. mime-types for demuxers and codecs, see
-http://www.rfc-editor.org/rfc/rfc4281.txt).
-
-Proposal
---------
-We add a ~/.gstreamer-0.10/registry-hooks directory. Applications can install
-scripts there. When the registry is rebuild, those scripts are run async in no
-specific order. There is no feedback channel (return values, stdout/err).
-
-
-Examples
----------
-
-gst-inspector
--------------
-it would install a script that simply deletes its cachefile.
-
-gupnp
------
-it would install a script that runs gst-inspect-0.10 --print-plugin-auto-install-info
-and post process it.
-
-totem,banshee,rythmbox
------------------------
-they could tune the "MimeType=" lists in their desktop files
-
-
-Open Items
-----------
-Should we have a gst-registry-hook script with --install/--remove/--list actions
-to manage hooks?
-
diff --git a/docs/random/ensonic/dynlink.txt b/docs/random/ensonic/dynlink.txt
deleted file mode 100644
index 007783b2c5..0000000000
--- a/docs/random/ensonic/dynlink.txt
+++ /dev/null
@@ -1,89 +0,0 @@
-$Id$
-
-Currently its only save to link/unlink elements/pad when pipeline is in READY.
-Belowe some thoughts. See http://bugzilla.gnome.org/show_bug.cgi?id=435487
-for patches.
-
-= current api =
-
-gboolean gst_element_link (GstElement *src, GstElement *dest);
-void gst_element_unlink (GstElement *src, GstElement *dest);
-
-gst_element_link_many, gst_element_unlink_many, gst_element_link_filtered,
-gst_element_link_pads, gst_element_unlink_pads, gst_element_link_pads_filtered
-
-GstPadLinkReturn gst_pad_link (GstPad *srcpad, GstPad *sinkpad);
-gboolean gst_pad_unlink (GstPad *srcpad, GstPad *sinkpad);
-
-= use cases =
-
-== inserting an element ==
-
-* we have: e1 ! e4
-* we want: e1 ! e2 ! e3 ! e4
-* we want: e1 ! e2 ! e4
-
-gst_element_insert_linked(e1, e2, e3, e4); // e2 == e3 allowed
-gst_pads_insert_link (e1.src, e2.sink, e3.src, e4.sink);
- disconnect e1.src, e1.src.peer
- disconnect e4.sink, e4.sink.peer
- connect e1.src, e2.sink
- connect e3.src, e4.sink
-
-== removing an element ==
-
-* we have: e1 ! e2 ! e3
-* we want: e1 ! e3
-
-gst_element_remove_linked(e2);
-gst_pads_remove_link (e1.src, e3.sink);
- disconnect e1.src, e1.src.peer
- disconnect e3.sink, e3.sink.peer
- connect e1.src, e3.sink
-
-== swapping out an elelment ==
-
-* we have: e1 ! e2 ! e6
-* we have: e1 ! e2 ! e3 ! e6
-* we want: e1 ! e4 ! e5 ! e6
-* we want: e1 ! e3 ! e6
-
-gst_element_swap_linked(e1, e4, e5, e6);
-gst_pads_insert_link (e1.src, e4.sink, e5.src, e6.sink);
- disconnect e1.src, e1.src.peer (=e2.sink)
- disconnect e6.sink, e6.sink.peer
- connect e1.src, e4.sink
- connect e5.src, e6.sink
-
-= thoughts =
-* I don't think we need api for pads
-* Should current api check for the state?
-* do we want to swapp multiple elements at once
-
-== events ==
-* tee and adder need special treatment
- * both would need to cache an accumulated segment
- * tee
- * would also cache tags
- * when linkfunc is called, it can send out the segment and the tags
- * when all pads got unlinked it could clear the segment
- * adder
- * when linkfunc gets called it sends a seek-event
-
-= ideas =
-== dynlinkpoint ==
-* use cases
- * its meant to be used with one side disconnected to allow to connect elements
- at runtime
- * it can be used in a pipeline to remove/insert elements at runtime
-* element with 1 source- and 1 sinkpad
-* when both connected it passes data through
-* if src is not connected it drops received buffers
-* if sink is not connected
- * it does not push
- * it creates silence on pull
-* events
- * it caches events
- * down: newsegment, tags, buffersize
- * up: seek (needs to be modified)
- * when other-pad get connected it pushes events depending on direction
diff --git a/docs/random/ensonic/interfaces.txt b/docs/random/ensonic/interfaces.txt
deleted file mode 100644
index cda4c90c65..0000000000
--- a/docs/random/ensonic/interfaces.txt
+++ /dev/null
@@ -1,85 +0,0 @@
-$Id$
-
-* existing gstreamer interfaces
- ./gstreamer/gst/gsttaginterface.c
- ./gstreamer/gst/gstinterface.c
- ./gstreamer/gst/gsturi.c
- ./gst-plugins/gst-libs/gst/propertyprobe/propertyprobe.c
- ./gst-plugins/gst-libs/gst/mixer/mixer.c
- ./gst-plugins/gst-libs/gst/tuner/tuner.c
- ./gst-plugins/gst-libs/gst/xoverlay/xoverlay.c
- ./gst-plugins/gst-libs/gst/colorbalance/colorbalance.c
- ./gst-plugins/gst-libs/gst/navigation/navigation.c
-* new general interfaces
- * GST_TYPE_UI_HINT
- - add hints to generate 'good' looking interfaces to elements
- - API:
- GList *get_group_list();
- struct ui_hint_group {
- gchar *label;
- gachr *role;
- GList *entries;
- }
- struct ui_hint_group_entry {
- enum UiHintGroupEntryType type={PARAM,DPARAM};
- gchar *name;
- }
- roles {
- /* graphics */
- "color_selection/rgb",
- "color_selection/hsv",
- "aspect_ratio",
- /* audio */
- "envelope/adsr",
- }
- - features
- - grouping of parameters, each group has:
- - a label: giving a title to the group
- - a role:
- - this can give the UI a hint about the purpose of the controls
- - this only makes sense, if we don't make this a thousand templates
- - a list of dparams or properties
- - question
- - should this be aware of instruments (voice-groups)
- - no, instruments should auto-generate those
- * GST_TYPE_QUALITY_VS_SPEED
- - get the name of a property that can be used to switch between
- - a fast version for e.g. realtime usage
- - a slower version with higher precision that can be used for off-line
- rendering
-* new interfaces for audio applications
- * GST_TYPE_MULTI_VOICE
- - control interface for elements that support multiple voices (in one output-pad)
- - API:
- gulong number_of_voices;
- void add_voice();
- void remove_last_voice();
- gulong get_number_of_voices();
- - features
- - plugin will initially have one voice and that one can not be deleted
- * GST_TYPE_MUSIC_GENERATOR
- - add hints so that application can use a element as an instrument
- - API:
- // param types
- DParam *get_note_dparam();
- GList *get_trigger_dparams();
- // -- or
- DParamType get_dparam_type(DParam *);
- dparamtype = { NOTE, TRIGGER, OTHER }
- // voices
- char *get_number_of_voices_property();
- GList *get_global_param_names();
- GList *get_voice_param_names();
- - features
- - find out which params to use to play notes/trigger sounds
- - these params will not appear in a control-ui
- - notes involve a key to frequency translation
- - find out if the element has a number_of_voices property
- - if yes, we can find out about the max by looking at the gparamspec
- - setting the property, adds/removes voices
- - if the element supports it, it needs to:
- - register voice-dparams as e.g. note_XXX, where XXX is the voice-number
- - run the voice-loop in the chain/loop function
- each voice processes the same input, if at all
- the outputs of all voices are mixed together
-
diff --git a/docs/random/ensonic/lazycaps.txt b/docs/random/ensonic/lazycaps.txt
deleted file mode 100644
index b8761bfc28..0000000000
--- a/docs/random/ensonic/lazycaps.txt
+++ /dev/null
@@ -1,72 +0,0 @@
-Lazy Caps
-=========
-
-The idea is to have caps where we evaluate the structures as needed. We can have
-these variants of lazy caps:
-
-- construction (e.g. from_string)
- - need user_data (the string/string-array)
- - need a free_func for the user_data
- - need a get_structure_func to fill structure slots as needed
- - we can more easily make this iterator based too
-- operations (e.g. intersect) (see [1]).
- - need user_data (the iterator)
- - need a free_func for the user_data
- - need a get_structure_func to fill structure slots as needed
-
-We should add PERFORMANCE category logging to methods that cause lazy caps to be
-fully evaluated. To get maximum speed benefit we need to inspect places that
-loop over structures of a caps and turn them into while() loops where possible.
-
-We can use GST_CAPS_FLAGS_LAZY to indicate that caps are not fully constructed.
-Once caps are fully evaluated, we can remove the flag (and call the free_func).
-
-Lazy caps might need a lock to protect multiple threads requesting a structure
-(get_structure_func).
-
-Accessors
----------
-guint gst_caps_get_size (const GstCaps *caps);
-- needs to fully evaluate the caps if iterator based :/
-
-GstStructure *gst_caps_get_structure (const GstCaps *caps, guint index);- needs to fully evaluate the caps if iterator based :/
-- needs to evaluate all entries up to index if iterator based
-- needs to only evaluate requested index if e.g. parse based
-
-gchar *gst_caps_to_string (const GstCaps * caps)
-- needs to fully evaluate the caps
-- it is used in debugging and serialisation
-
-Construction
-------------
-
-Manipulation
-------------
-void gst_caps_append (GstCaps *caps1, GstCaps *caps2);
-- use an iterator
-
-void gst_caps_merge (GstCaps *caps1, GstCaps *caps2);
-- use an iterator
-
-void gst_caps_append_structure (GstCaps *caps, GstStructure *structure);
-void gst_caps_remove_structure (GstCaps *caps, guint idx);
-void gst_caps_merge_structure (GstCaps *caps,
-- fully evaluate caps?
-
-GstCaps * gst_caps_copy_nth (const GstCaps *caps, guint nth);
-- eval structure and copy
-
-void gst_caps_truncate (GstCaps *caps);
-- eval first structure as needed and remove GST_CAPS_FLAGS_LAZY + call free_func
-
-void gst_caps_set_value (GstCaps *caps, const char *field, const GValue *value);
-void gst_caps_set_simple (GstCaps *caps, const char *field, ...);
-void gst_caps_set_simple_valist (GstCaps *caps, const char *field, va_list varargs);
-- fully evaluate caps
-
-
-Operations
-----------
-
-
-[1] https://bugzilla.gnome.org/show_bug.cgi?id=618853
diff --git a/docs/random/ensonic/logging.txt b/docs/random/ensonic/logging.txt
deleted file mode 100644
index d3ab9176cd..0000000000
--- a/docs/random/ensonic/logging.txt
+++ /dev/null
@@ -1,56 +0,0 @@
-$Id$
-
-rethink log format. current format:
-* is not easy to parse/process by commandline tools
-* cannot be easily diffed (timestamps, pid)
-
-gst_debug_log_default() is default gst-log handler.
-try new via:
- gst_debug_remove_log_function(gst_debug_log_default)
- gst_debug_add_log_function(func,data)
-
-== reorder fields ==
-
-format of default handler is:
- DEBUG (0x8134bc0 - 0:00:00.848191000) GST_QOS( 3340) gstbasesink.c(1431):gst_base_sink_do_render_stats:<xvimagesink0> avg_render: 0:00:00.009044000
-log_level_name --^^^^^ | | | | | | | | | | | | |
-thread-id --------------^^^^^^^^^ | | | | | | | | | | |
-timestamp --------------------------^^^^^^^^^^^^^^^^^ | | | | | | | | |
-log_category -------------------------------------------------------^^^^^^^ | | | | | | |
-process-id -----------------------------------------------------------------^^^^^ | | | | |
-file:line -------------------------------------------------------------------------^^^^^^^^^^^^^^^^^^^ | | |
-function ----------------------------------------------------------------------------------------------^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
-message -----------------------------------------------------------------------------------------------------------------------------^^^^^^^^^^^^^^^
-
-new proposed format:
- 0:00:00.848191000 3340 0x8134bc0 DEBUG GST_QOS gstbasesink.c:1431:gst_base_sink_do_render_stats:<xvimagesink0> avg_render: 0:00:00.009044000
-timestamp -------^^^^^^^^^^^^^^^^^ | | | | | | | | | | | | |
-process-id ------------------------^^^^^ | | | | | | | | | | |
-thread-id -------------------------------^^^^^^^^^ | | | | | | | | |
-log_level_name ------------------------------------^^^^^ | | | | | | |
-log_category ----------------------------------------------------------^^^^^^^ | | | | |
-file:line ---------------------------------------------------------------------^^^^^^^^^^^^^^^^^^ | | |
-function -----------------------------------------------------------------------------------------^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
-message ------------------------------------------------------------------------------------------------------------------------^^^^^^^^^^^^^^^
-
-something like:
- cut -c35- | sed -e "s/0x[0-9a-f]\{7,8\}//g"
-should make logs easily 'diffable'.
-
-== color code level ==
-
-I suggedt to color-code loglevels if COLOR is allowed:
-ERROR : red
-WARNING : yellow
-INFO : green
-DEBUG : turkis
-LOG : gray
-NONE : gray
-
-== log sources ==
-
-What about differentating between log sources:
- core
- elements
- application
-
diff --git a/docs/random/ensonic/media-device-daemon.txt b/docs/random/ensonic/media-device-daemon.txt
deleted file mode 100644
index 81717f960e..0000000000
--- a/docs/random/ensonic/media-device-daemon.txt
+++ /dev/null
@@ -1,42 +0,0 @@
-$Id$
-
-components
-================================================================================
-
-- daemon process
- - is a gstreamer application
- - open physical sink, src elements
- - prepends an adder to sinks
- - appends an tee to sources
- - listens to dbus, to get notified by virtual-endpoints of init/finalize
- (the dbus notify, would also be useful for gst-editor to hook on running
- apps)
-
-- 4 new elements
- - virtual-audiosink, virtual-videosink
- virtual-audiosrc, virtual-videosrc
- - virtual sinks establish a connection to the daemon
- - they link to request_pads of the adder/tee elements
- - on init and finalize they send a dbus-message
-
-- gui app
- - lists instances as mixing-desk like channelstrips
- - channelstrips would contain
- - audio
- - volume, panorama, 3-band eq
- - video
- - brightness, contrast, alpha-level
- - user can
- - add insert-fx
- - route channel to targets, where targets can be real sinks or more
- virtual-sinks (sub-groups)
- - virtual sinks need queues to decouple application processes
-
-- interfaces
- - expose child-elements via child-proxy
- - then e.g. the applications volume-control could directly access the
- channelstrip
- - state-control (play, pause/mute)
- - it would be useful if one app could pause/mute others
- - think of a voip-client, if there is an incoming call, if pauses your
- media-player, or mutes the monitoring of your recording app
diff --git a/docs/random/ensonic/plugindocs.txt b/docs/random/ensonic/plugindocs.txt
deleted file mode 100644
index 987fc68118..0000000000
--- a/docs/random/ensonic/plugindocs.txt
+++ /dev/null
@@ -1,51 +0,0 @@
-Plugin Docs
-===========
-
-In contrast to normal gtk-doc workflow modules will include
-common/gtk-doc-plugins.mak. This provides an additional 'make udpate' target.
-
-the "make update" target calls
-1.) common/gstdoc-scangobj --type-init-func="gst_init(NULL,NULL)" --module=$(DOC_MODULE) --source=$(PACKAGE)
- common/scangobj-merge.py $(DOC_MODULE)
-
-the "tmpl" target is modified to run this extra step after "gtkdoc-mktmpl":
-2.) common/mangle-tmpl.py $(srcdir)/inspect tmpl
-
-and the "sgml" target is modified to run this step before "gtkdoc-mkdb":
-3.) for a in $(srcdir)/inspect/*.xml; do \
- xsltproc --stringparam module $(MODULE) $(top_srcdir)/common/plugins.xsl $$a > xml/`basename $$a`;
-done
-
-Details
-=======
-
-1a.) gstdoc-scangobj
--------------------
-- get types from registry instead of .types file
-- outputs inspect/plugin-<pluginname>.xml files
-- outputs signal and args files
-
-1b.) scangobj-merge.py
----------------------
-- parse, update and write .signals and .args files
-
-2.) mangle-tmpl.py
-------------------
-- read data from inspect/plugin-<pluginname>.xml
-- insert/overwrite "Short Description" and "Long Description" in tmpl/
-- the "Long Description" contains a <xi:include> for xml/element-<name>-details.xml
-
-3.) common/plugins.xsl
-----------------------
-- creates xml/plugin-<name>.xml and xml/element-<name>-details.xml
-
-
-TODO:
------
-- scangobj-merge.py -> gstdoc-scangobj
-- need a way to skip the tmpl step
- - gtkdoc-mkdb needs a way to insert extra content
- - maybe patch generated xml/*.xml files
- - could common/plugins.xsl patch the files?
- <refsect1 id="gstreamer-plugins-capsfilter.description" role="desc">
-
diff --git a/docs/random/ensonic/receipies.txt b/docs/random/ensonic/receipies.txt
deleted file mode 100644
index 0d2a0a3d7f..0000000000
--- a/docs/random/ensonic/receipies.txt
+++ /dev/null
@@ -1,37 +0,0 @@
-$Id$
-
-= receipies =
-
-The idea is to collect some recommendations for common, but not so trivial
-tasks. docs/design/part-block.txt has something like that already. Ideally these
-would go to the application developer manual and there would be sample code.
-
-== initial seeking ==
-=== question ===
-How to I configure the initial playback segment?
-
-=== idea ===
-1) set pipeline to PAUSED
-2) send seek event
-3) set pipeline to PLAYING
-
-=== problems ===
-1) would preroll the pipeline only, to flush it when the seek comes
-
-
-== async state changes ==
-=== question ===
-what to do when gst_element_set_state() returns ASYNC?
-
-=== idea ===
-1) listen to the STATE_CHANGED message on the bus
-2) trigger next action
-
-=== problems ===
-This scatters logic over multiple functions (callbacks).
-
-
-== topic ==
-=== question ===
-=== idea ===
-=== problems ===
diff --git a/docs/random/eos b/docs/random/eos
deleted file mode 100644
index c928d264a2..0000000000
--- a/docs/random/eos
+++ /dev/null
@@ -1,209 +0,0 @@
-OUTDATED
---------
-
-
-case 1)
-
- (--------) (--------) (--------)
- ! fakesrc! !identity! !fakesink!
- ! src ----- sink src ---- sink !
- (--------) (--------) (--------)
-
-
-
- fakesrc detects the end of stream. It just sent the last buffer
- and sets the srcpad to EOS with gst_pad_eos ().
-
- gst_pad_eos() will notify the parent about the plugins attempt to
- signal eos. the parent adds the element to its possible EOS
- providers.
-
- gst_pad_eos() will by default propagate to identy and to fakesink.
- none of these plugins override the default behaviour so gst_pad_eos
- returns TRUE and fakesrc signals EOS with the value TRUE.
-
- The parent looks in the list of EOS providers and finds the faksrc
- element that is now signaling EOS. all EOS providers are now in EOS
- and so the bin fires EOS.
-
-
-
-case 2)
- (---------------)
- !thread !
- (--------) (--------) (--------) ! (--------)!
- ! fakesrc! !identity! ! queue ! ! !fakesink!!
- ! src ----- sink src ---- sink src ---- sink !!
- (--------) (--------) (--------) ! (--------)!
- (---------------)
-
-
- fakesrc detects the end of stream. It just sent the last buffer
- and sets the srcpad to EOS with gst_pad_eos ().
-
- gst_pad_eos() will notify the parent about the plugins attempt to
- signal eos. the parent adds the element to its possible EOS
- providers.
-
- gst_pad_eos() will by default propagate to identy and to queue.
- queue overrides the eos handler and returns false on the eos
- request. fakesrc signals EOS with a value of false and the parent
- bin removes the EOS provider from its list.
-
- after the queue has sent out the last buffer, its calls eos on its
- src pad. queue is added to the top level bin as an eos provider and
- the default eos handler signals EOS with a value of TRUE to the parent.
-
- the parent sees that all the eos providers are in eos now and signals
- EOS.
-
-
-case 3)
- (---------------)
- !thread !
- (--------) (--------) (--------) ! (--------)!
- ! fakesrc! ! tee ! ! queue1 ! ! !fakesink!!
- ! src ----- sink src ---- sink src ---- sink !!
- (--------) ! ! (--------) ! (--------)!
- ! ! (---------------)
- ! !
- ! ! (---------------)
- ! ! !thread !
- ! ! (--------) ! (--------)!
- ! ! ! queue2 ! ! !fakesink!!
- ! src ---- sink src ---- sink !!
- ! ! (--------) ! (--------)!
- (--------) (---------------)
-
-
- fakesrc detects the end of stream. It just sent the last buffer
- and sets the srcpad to EOS with gst_pad_eos ().
-
- the eos handler returns false because both queues return false on the
- eos request. the parent removes fakesrc as an EOS provider.
-
- queue1 and queue2 were responsible for the EOS delay and so they get
- added to the bin as possible EOS providers.
-
- after the queues have sent out their last buffer, they calls eos on their
- src pads.
- the parent already has the two queues in the EOS provider list so they don't
- get added twice.
- the two queues perform gst_pad_eos () on their pads when the queue is empty,
- the parent removes the EOS providers from its list, when the list is empty,
- the parent fires EOS.
-
-
-case 4)
-
- (---------------)
- !thread !
- (--------) (----------) (--------) ! (--------)!
- ! fakesrc! !mpeg1parse! ! queue1 ! ! !fakesink!!
- ! src -- sink src ---- sink src ---- sink !!
- (--------) ! ! (--------) ! (--------)!
- ! ! (---------------)
- ! !
- ! ! (---------------)
- ! ! !thread !
- ! ! (--------) ! (--------)!
- ! ! ! queue2 ! ! !fakesink!!
- ! src ---- sink src ---- sink !!
- ! ! (--------) ! (--------)!
- (----------) (---------------)
-
-
- this case differs from case3 in that one of the queues can be empty
- while the other isn't. we assume queue1 is empty while queue2 isn't yet.
-
- fakesrc detects the end of stream. It just sent the last buffer
- and sets the srcpad to EOS with gst_pad_eos ().
-
- the eos handler returns false because queue2 returns false on the
- eos request. the parent removes fakesrc as an EOS provider.
-
- queue2 was responsible for the EOS delay and so it gets added to the bin
- as a possible EOS provider.
-
- after the queue2 has sent its last buffer, it performs gst_pad_eos on its
- src pad.
- the parent already has the queue2 in the list of EOS providers so it does not
- get added twice.
- queue2 finally fires the EOS signal and the parent removes the EOS provider
- from its list, when the list is empty, the parent fires EOS.
-
-
-
-case 5)
-
- (--------) (--------) (--------)
- ! disksrc! ! mad ! !filesink!
- ! src ----- sink src ---- sink !
- (--------) (--------) (--------)
-
-
- disksrc detects the end of stream. It just sent the last buffer
- and sets the srcpad to EOS with gst_pad_eos ().
-
- the eos handler returns false because mad returns false on the
- eos request. the parent removes mad as an EOS provider.
-
- mad was responsible for the EOS delay and so it gets added to the bin
- as a possible EOS provider.
-
- After mad has sent its last buffer, it performs gst_pad_eos on its
- src pad.
- the parent already has mad in the list of EOS providers so it does not
- get added twice.
- mad finally fires the EOS signal. This time, filesink returns false on
- the eos request. the parent removes mad as an EOS provider.
-
- filesink was responsible for the EOS delay and gets added to the bin
- as a possible EOS provider.
- When filesink has written all of it's data and closed the output file,
- it fires EOS.
- The parent already has filesink in the list of EOS providers so it does
- not get added twice.
- The parent removes the EOS provider
- from its list, and since the list is empty, the parent fires EOS.
-
-case 6)
-
- (--------) (--------) (--------)
- !disksrc1! ! mad1 ! ! mixer !
- ! src ----- sink src ---- sink1 ! (--------)
- (--------) (--------) ! ! !filesink!
- ! src ---- sink !
- (--------) (--------) ! ! (--------)
- !disksrc2! ! mad2 ! ! !
- ! src ----- sink src ---- sink2 !
- (--------) (--------) (--------)
-
- In this case, we want to make sure the pipeline keeps running after one
- of the two sources reaches eos. Suppose in this case that disksrc1 will
- reach eos first.
-
- disksrc1 detects the end of stream. It sets eos, mad1 will return false,
- and mad1 will be responsible for eos. When mad1 had sent out the last
- buffer, it sends out eos.
-
- The mixer intercepts eos and returns false. mad1 is removed from the
- eos providers and mixer is added.
-
- (At this point, the mixer might choose to disconnect mad1->src and
- mixer->sink1 pads, since it's received eos on mad1->src)
-
- mixer will not send out eos since it hasn't received eos from
- mad2->src.
-
- After a while, disksrc2 will detect end of stream, and eos will finally
- propagate to mixer. mixer might disconnect mad->src2, and after
- realizing all of it's sources have reached eos, it sends out the final
- buffer and fires EOS.
-
- At this point, filesink will return false, mixer will be removed as an
- eos provider, and filesink will write out it's final buffer and close
- the file on disk. At this point, it fires eos, and since it's the last
- eos provider, the parent can fire eos.
-
-
diff --git a/docs/random/error b/docs/random/error
deleted file mode 100644
index bccb239309..0000000000
--- a/docs/random/error
+++ /dev/null
@@ -1,59 +0,0 @@
-Some notes on the error handling introduced after 0.7.3
-
-- there are four domains for errors:
- - CORE: core GStreamer errors
- - LIBRARY: supporting library errors
- - RESOURCE: errors accessing resources (files, network, memory, ...)
- - STREAM: errors in the data stream
-- Each error in GStreamer or plug-ins can be put under one of these four.
-- the enum is called Gst(Domain)Error and has GST_(DOMAIN)_ERROR_(CODE) members
- (see GError API docs for rationale)
-- each of the enums starts with _FAILED at 1 (with FAILED being the generic
- error if none of the others applies)
-- second in the enum is TOO_LAZY, which allows us to keep track of those errors
- that we haven't categorized yet. This means they really should either move
- to another one, or a new one ought to be created.
-- each enum ends with NUM_ERRORS
-
-- elements call gst_element_error to signal an error:
- gst_element_error (element, domain, code, message, debug);
- With :
- - element being the one signalling the error
- - domain one of CORE, LIBRARY, RESOURCE, STREAM
- - code one of the suffixes in the Gst(Domain)Error enum
- - message is either NULL (to signal the standard error message for the
- given domain and code), or a printf-like set with the format being
- marked for translation.
- The string should start with a capital and end in a period so it forms
- a complete sentence.
- The string can/should be shown to the user of an application.
- - debug is either NULL or a non-translated debug string, which can be used
- to diagnose errors after they've happened.
- The string can be shown to the user when he asks for additional debug info.
- A useful macro is GST_ERROR_SYSTEM, which prints out the system error
- that explains the failure by using g_strerror.
-
-- Some example calls:
-
- gst_element_error (src, RESOURCE, OPEN_READ,
- (_("Could not open file \"%s\" for reading"), src->filename),
- GST_ERROR_SYSTEM);
-
- The message is specified since we have more information:
- - the resource is a file
- - we know the file name
-
- gst_element_error (element, CORE, NEGOTIATION, NULL, NULL);
-
- This is a simple negotiation error. The default message will be
- signaled, telling the user that GStreamer had an internal error.
-
- gst_element_error (ebml, RESOURCE, READ, NULL,
- ("Read error at position %llu (0x%llx)",
- pos, pos));
-
- The plugin asked to read on the underlying resource (using bytestream),
- but failed. The user will get a generic read error. The debug info
- will contain the exact position in the stream at which the read error
- occurred.
-
diff --git a/docs/random/example b/docs/random/example
deleted file mode 100644
index 87bf619b83..0000000000
--- a/docs/random/example
+++ /dev/null
@@ -1,108 +0,0 @@
-OUTDATED, methods have different names now
-------------------------------------------
-
-
-Here's a pipeline that does audio/video MPEG streams with a queue on
-either side of each decompressor, for a total of 5 threads (read/split,
-decode audio, decode video, play audio, play video):
-
-NOTES: mpegsplit is the intelligence in this pipeline, providing an IDL
-that allows one to connect things to a GUI.
-
-Pipeline(mpegplay)
- Thread(reader)
- Element(:disk_async_src) [StreamerAsyncSrc]
- OutPad(disk1)
- Element(:mpegsplit)
- InPad(disk1)
- OutPad(audio1)
- OutPad(video1)
-
- Queue(audioQ1)
- InPad(audio1)
- OutPad(audio2)
- Thread(audiodecode)
- Element(:mpeg_audio_decode) [StreamerVideoFilter]
- InPad(audio2)
- OutPad(audio3)
- Queue(audioQ2)
- InPad(audio3)
- OutPad(audio4)
- Thread(audioplay)
- Element(:audio_play) [StreamerAudioSink]
- InPad(audio4)
-
- Queue(videoQ1)
- InPad(video1)
- OutPad(video2)
- Thread(videodecode)
- Element(:mpeg_video_decode) [StreamerVideoFilter]
- InPad(video2)
- OutPad(video3)
- Queue(videoQ2)
- InPad(video3)
- OutPad(video4)
- Thread(videoplay)
- Element(:video_play) [StreamerVideoSink]
- InPad(video4)
-
-
-A simpler pipeline that just does MPEG videos:
-
-Pipeline(mpegplay)
- Thread(reader)
- Element(:disk_async_src) [GstAsyncSrc]
- OutPad(disk1)
- Element(:mpeg_control)
- InPad(disk1)
- OutPad(video1)
- Element(:mpeg_video_decode) [GstVideoFilter]
- InPad(video1)
- InPad(video2)
- Queue(queue)
- InPad(video2)
- OutPad(video3)
- Thread(play)
- Element(:video_play) [GstVideoSink]
- InPad(video3)
-
-The code for the above looks roughly like:
-
-/* all the objects we're worried about */
-GstPipeline *mpegplay;
-GstThread *reader;
-GstSrc *disk_src;
-GstControl *mpeg_control;
-GstFilter *mpeg_video_decode;
-GstQueue *queue;
-GstThread *play;
-GstSink *video_play;
-
-/*** first we create all of the objects ***/
-
-mpegplay = gst_pipeline_new();
-reader = gst_thread_new();
-disk_src = gst_async_disk_src_new("filename.mpg");
-mpeg_control = gst_mpeg_control_new();
-mpeg_video_decode = gst_mpeg_video_decode_new();
-queue = gst_queue_new();
-play = gst_thread_new();
-video_play = gst_video_sink_new();
-
-
-/*** now we start to create the pipeline ***/
-
-/* first set up the reader thread */
-gst_bin_add(reader,disk_src);
-gst_object_connect(disk_src,"out",mpeg_control,"in");
-gst_object_connect(mpeg_control,"out",mpeg_audio_decode,"in");
-gst_bin_ghost_pad(reader,mpeg_audio_decode,"out");
-
-/* then set up the player thread */
-gst_bin_add(play,audio_play);
-gst_bin_ghost_pad(play,audio_play,"in");
-
-/* now plug them all into the main pipeline */
-gst_bin_add(mp3play,reader);
-gst_object_connect(reader,"out",queue,"in");
-gst_object_connect(queue,"out",play,"in");
diff --git a/docs/random/hierarchy b/docs/random/hierarchy
deleted file mode 100644
index a07f8b8674..0000000000
--- a/docs/random/hierarchy
+++ /dev/null
@@ -1,153 +0,0 @@
-OUTDATED
---------
-
-
-Face it, the plugins/ directory hierarchy is crap. We want to propose a
-better layout for it now. Some things to consider:
-
- - Elements have a klass member in the factory that is used to
- denote the functional type of the element. For example, the
- mp3 encoder has a klass of Codec/Encoder/Audio
-
- - The plugins can be grouped together by the media type they
- operate on or by the way they work (decoder/encoder)
-
-In GStreamer all plugins are technically filters, the only way they
-can be considered sources or sinks (input/output) elements is
-by the absence of src/sink pads. At first sight the source/filter/
-sink distinction is quite useless because most of the plugins
-will go into the filters category anyway.
-
-We don't want to make the hierarchy too deep, yet provide a
-clean way to ask for a mp3 decoder element..
-
-Anyway this is a rough proposal to fire off the discussions...
-
-Wim
-
-Source
- Disk
- disksrc
- fdsrc
- multifilesrc
- Network
- HTTPsrc
- RTPsrc
- CDDA
- cdparanoia
- XMMS
- ..
- DVD
- dvdsrc
- Audio
- ASLA
- OSS
- Capture
- v4lsrc
- firewire
-
-Demuxer
- AVI
- MPEG1
- MPEG2
- QT
-
-Muxer
- AVI
- MPEG1
- QT
-
-Aggregator
-
-Tee
- gsttee
-
-Connection
- queue
- CORBA
-
-Parser
- MPEG1
- MPEG2
- AC3
-
-Mixer
- Audio
- Merge
- Video
- Subtitles
- Merge
-
-Filters
- Audio
- ladspa
- resample
- Video
- colorspace
-
-Effect
- Audio
- stereo
- volume
- delay
- chorus
- Video
- median
- smooth
- XMMS
-
-Decoder
- MPEG1
- MPEG2
- MP3
- mpg123
- xing
- win32
- AU
- WAV
- JPEG
- AC3
- ac3dec
- RTJPEG
- vorbis
-
-Encoder
- MPEG1
- MPEG2
- MP3
- lame
- mpegaudio
- win32
- JPEG
- AU
- WAV
- RTJPEG
- Vorbis
-
-Visualisation
- Video
- histogram
- Audio
- smoothwave
- spectrum
- synaesthesia
- vumeter
- XMMS
-
-Sink
- Disk
- filesink
- multifilesink
- Network
- ICECASTsink
- FTPsink
- RTPsink
- XMMS
- ESD
- Video
- videosink
- SDLsink
- Audio
- OSSsink
- ALSAsink
-
diff --git a/docs/random/interfaces b/docs/random/interfaces
deleted file mode 100644
index 5bd9997401..0000000000
--- a/docs/random/interfaces
+++ /dev/null
@@ -1,337 +0,0 @@
-INTERFACES & ELEMENTS
----------------------
-
-1) Introduction
-===============
-Interfaces are descriptions on how to handle an object, without actually
-implementing the object. This allows for multiple objects to be instantiated
-based on this interface. Each of them can then be handled equally by an
-application.
-Glib, apparently (unchecked), has a way of creating interfaces, probably
-by means of a class struct without actually defining the object. The object,
-then, does not define a class and these two add up. Benjamin knows more
-about interfaces, I didn't study interfaces & glib too deeply, yet. I know
-them just from Java.
-Interfaces are cool! It allows for some sort of random element creation
-without needing to link to the implementation. This is similar to how
-GStreamer currently handles media plugins. GStreamer itself could be seen
-as an interface too, in that respect.
-
-2) So why do we need interfaces?
-================================
-Because GStreamer doesn't handle it all. GStreamer in itself is a media
-framework for streams of data from one element to the next. There's lots
-of things that's media-related, but not handled in this description.
-Several examples will probably clarify this: think of the Xvideo output
-plugin. We can create an overlay here (Xv-based), and we currently control
-this X-connection using glib properties. However, what property name is
-associated with what control? And does it work the same as v4lsrc's
-overlay image control?
-The same goes for a mixer, for image control, audio control, and probably
-a lot more. The general idea is simple: *this needs to be documented*.
-But properties aren't all - they simply cannot do this all. Some things
-cannot be described in a simple one-argument property thing. Of course,
-we could give a pointer to a struct as argument, but that's merely a hack
-and requires both plugin and app to know the ABI of the struct. This
-kills the whole idea of making the plugin independent of the app.
-In short: we want interfaces for this.
-
-3) How to integrate an interface in GStreamer
-=============================================
-Let us start with some starting point: an interface is associated
-with an element. It is a feature exported by that specific element,
-not by a pipeline or anything more complex. Pipelines are already
-handled just fine by GStreamer (or you wouldn't be reading all
-this).
-Obviously, a pipeline can be a fallback for an interface. Imagine
-that we're looking for an audio sink that exposes a mixer, but our
-fakesink audio output doesn't ("I wonder why"). We could then create
-a pipeline with the volume element in it to "fake" a mixer. Ideally,
-the volume element would implement a mixer interface itself.
-
-How are we going to do that in programmatic way? We currently use
-properties. Their huge advantage is that we do not need to care
-about adding new functions or whatever. Their disadvantage is that
-they're limited to one argument. Anything more complex requires
-app/plugin knowledge about the shared data, and that defeats the
-point of them: to have no dependency on each other. This could be
-solved partially by using action signals, but that makes the whole
-picture quite complex (since you use multiple methods for doing one
-simple thing). Also, they are quite slow compared to functions
-because of the table lookups. In short: it'd work, but I'm not in
-facour of it...
-OK, so an element exposes interfaces. This allows us to think of
-the idea of embedding interfaces (dynamically, of course) in the
-GstElement object. Think of an object being able to register an
-indefinite number of interfaces per object instance, and a client
-application could then enumerate interfaces and instantiate one.
-Glib gives us GInterface for this purpose. The disadvantage of
-this is that it's on a per-class basis, not a per-instance basis.
-This is a problem in case of elements where it depends on several
-properties whether it supports an interface or not. This can be
-solved by simply making one generic virtual function "supported ()"
-in a generic derived object of GInterface (GstInterface?).
-
-GstInterface is then a generic thing that is inherited by specific
-interfaces (see examples). Obviously, the client will need to know
-about the ABI/API of this struct, but that'll happen either way.
-Surely, there needs to binary linkage, but I don't consider that a
-bad thing. It does improve performance compared to action signals!
-
-So an element contains interfaces. But where are these interfaces
-described? And who creates them? I suggest that we do that just as
-we handle gstvideo and gstaudio right now (these libs do *nothing*
-useful currently, so this would make them a lot more interesting).
-These interfaces inherit from GstInterface. The functions that
-are needed, can be provided through a class object. The element is
-then responsible for storing variables and so on. gstvideo/audio
-provides wrapper functions for the class functions. That's also how
-glib suggest us to use GInterfaces.
-
-Plugin and application then handle and retrieve interfaces as
-documented in the glib documentation, which is available at:
-http://www.gnome.org/~mathieu/gobject/main.html
-
-So the most important part left is to document the interfaces
-and make sure all elements exporting them work equally. For this,
-I'll give two examples.
-
-4) Examples
-===========
-
-/* This small extra virtual function is here to provide an
- * interface functionality on a per-instance basis rather
- * than a per-class basis, which is the case for glib.
- */
-typedef struct _GstInterfaceClass {
- GTypeInterface parent;
-
- /* virtual functions */
- gboolean (* supported) (GstInterface *iface);
-} GstInterfaceClass;
-
-There would probably be a convenience function that checks
-a specific interface's implementation (glib allows for this)
-and checks for ->supported () to be set and to return TRUE:
-
-gboolean
-gst_element_implements_interface (GstElement *element,
- GType iface_type)
-{
- if (G_TYPE_CHECK_INSTANCE_TYPE (G_OBJECT (element),
- type)) {
- GstInterface *iface;
- GstInterfaceClass *ifclass;
-
- iface = G_TYPE_CHECK_INSTANCE_CAST (G_OBJECT (element),
- type, GstInterface)
- ifclass = GST_INTERFACE_GET_CLASS (iface);
-
- if (ifclass->supported != NULL &&
- ifclass->supported (iface) == TRUE) {
- return TRUE;
- }
- }
-
- return FALSE;
-}
-
-Let's now add some functions so we can abuse this in case/check
-functions.
-
-GstInterface *
-gst_interface_cast (gpointer from,
- GType type)
-{
- GstInterface *iface;
-
- /* check cast, give warning+fail if it's invalid */
- if (!(iface = G_TYPE_CHECK_INSTANCE_CAST (G_OBJECT (element),
- type, GstInterface))) {
- return NULL;
- }
-
- /* if we're an element, take care that this interface
- * is actually implemented */
- if (GST_IS_ELEMENT (from)) {
- gboolean interface_is_implemented =
- gst_element_implements_interface (GST_ELEMENT (from),
- type);
- g_return_val_if_fail (interface_is_implemented == TRUE, NULL);
- }
-
- return iface;
-}
-
-gboolean
-gst_interface_check (gpointer from,
- GType type)
-{
- GstInterface *iface;
-
- /* check cast, return FALSE if it fails, don't give a warning... */
- if (!G_TYPE_CHECK_INSTANCE_CAST (from, type,
- GstInterface)) {
- return FALSE;
- }
-
- iface = G_TYPE_CHECK_INSTANCE_CAST (G_OBJECT (element),
- type, GstInterface);
-
- /* now, if we're an element (or derivative), is this thing
- * actually implemented for real? */
- if (GST_IS_ELEMENT (from)) {
- if (!gst_element_implements_interface (GST_ELEMENT (from),
- type)) {
- return FALSE;
- }
- }
-
- return TRUE;
-}
-
-#define GST_INTERFACE_CHECK_INSTANCE_CAST(obj, type, cast_t) \
- ((cast_t *) gst_interface_cast ((obj), (type))
-#define GST_INTERFACE_CHECK_INSTANCE_TYPE(obj, type) \
- (gst_interface_check ((obj), (type))
-
-We could use this in the GST_IS_... () macros. For example, the
-macros GST_IS_MIXER () and GST_MIXER () would then look like this:
-
-/* Note that this is a non-standard macro, and with a reason! */
-#define GST_MIXER(obj) \
- (GST_INTERFACE_CHECK_INSTANCE_CAST ((obj), \
- GST_TYPE_MIXER,
- GstMixer))
-#define GST_IS_MIXER(obj) \
- (GST_INTERFACE_CHECK_INSTANCE_TYPE ((obj), \
- GST_TYPE_MIXER))
-
-So the application would just tread it with the known macro, and
-everything would look extremely simple to the end user.
-
-4a) mixer
----------
-A mixer is a way of controlling volume and input/output channels.
-This doesn't mean that you control which channel is the subwoofer,
-all that is supposed to be done automatically. It is really meant
-as a way of representing system-level volumes and such. It could
-also be used to turn on/off certain outputs or inputs.
-As you've noticed, I'm not only talking about output, but also
-input. Indeed, I want both osssrc *and* osssink to export the
-same mixer interface! Or at least a very similar one. Volume
-control works the same for both. You could say that osssrc should
-enumerate the input channels (such as microphone, line-in). Of
-course, osssink should not. Or maybe it should, not sure... Maybe,
-we'd need a parent osselement which implements all mixer channels.
-And alsa* would surely implement the same interface.
-
-/* This is confusing naming... (i.e. FIXME)
- * A channel is referred to both as the number of simultaneous
- * sound streams the input can handle as well as the in-/output
- * itself
- */
-
-#define GST_MIXER_CHANNEL_INPUT (1<<0)
-#define GST_MIXER_CHANNEL_OUTPUT (1<<1)
-#define GST_MIXER_CHANNEL_MUTE (1<<2)
-#define GST_MIXER_CHANNEL_RECORD (1<<3)
-
-typedef struct _GstMixerChannel {
- gchar *label;
- gint current_num_channels,
- max_num_channels,
- flags;
-} GstMixerChannel;
-
-typedef struct _GstMixerClass {
- GTypeInterface klass;
-
- /* virtual functions */
- GList * (* list_channels) (GstMixer *mixer);
-
- void (* set_volume) (GstMixer *mixer,
- GstMixerChannel *channel,
- gint *volumes);
- void (* get_volume) (GstMixer *mixer,
- GstMixerChannel *channel,
- gint *volumes);
-
- void (* set_mute) (GstMixer *mixer,
- GstMixerChannel *channel,
- gboolean mute);
- void (* set_record) (GstMixer *mixer,
- GstMixerChannel *channel,
- gboolean record);
-} GstMixerClass;
-
-libgstmixer.la/so provides wrapper functions for each of the
-class' virtual functions. Possibly also some macros for
-GST_MIXER_CHANNEL_HAS_FLAG () or _get_channel ().
-
-The rest is done automatically, as described in the already-
-mentioned glib documentation for GInterface. This includes
-things like the base_init () function of the GstMixerClass,
-which fills all the virtual functions for the mixer, and the
-actual function implementations. The mixer, basically, operates
-as an element on its own. It gets the file descriptor from
-the interface->element (every oss... is a osscommon, etc.).
-
-4b) overlay
------------
-Overlay is used in both in- and output, too. Think of v4lsrc,
-v4l2src, v4lmjpegsrc, xvideosink - all overlays. But where do
-we position the overlay window? Control of this can be done at
-various levels: locational control (over the server, asynchronous)
-or XID control (but that makes you depend on X and limits the
-ability to broaden it over to non-X elements such as fbsink).
-
-However, simplicity *is* an issue here. Do we really care about
-overlay? In the end, users will have to link against either FB
-or X anyway, so we might want to create separate interfaces for
-both. On the other hand, we want to be general too... This is a
-decision that we need to make as early as possible in this process.
-For now, I propose making X- and FB-based interfaces.
-
-Let's assume that we take X as a basis. Then, overlay becomes as
-simple as one function. Possible extendible by providing inputs
-(like in the mixer) and norms, although that only applies to
-input-to-analog, not to-digital... Others simply return NULL.
-
-typedef struct _GstOverlayClass {
- GTypeInterface klass;
-
- /* virtual functions */
- void (* set_xwindowid) (GstOverlay *overlay,
- XID xid);
-} GstOverlayClass;
-
-That's all! It would look similar for FB & co.
-
-4c) user input
---------------
-And yes, user input could be an interface too. Even better, it
-should definitely be. And wasn't this one of our key issues for
-0.8.0?
-
-No code here. Go implement it, lazy ass!
-
-General ways of thinking: input can come from a plugin, or from
-the application (we don't have modules for joystick input et all).
-However, plugins handling input (such as dvdsrc) need to be able
-to handle each. So we get both input-to-application as well as
-input-from-application APIs.
-
-5) Status of this document
-==========================
-The interfaces are implemented, more (for metadata, framebuffer-
-overlay, video balancing (brightness), user input etc. are all
-pending.
-
-6) Copyright and blabla
-=======================
-(c) Ronald Bultje, 2003 <rbultje@ronald.bitfreak.net> under the
-terms of the GNU Free Documentation License. See http://www.gnu.org/
-for details.
-
-And no, I'm not for hire. ;).
diff --git a/docs/random/metadata b/docs/random/metadata
deleted file mode 100644
index d98655587e..0000000000
--- a/docs/random/metadata
+++ /dev/null
@@ -1,80 +0,0 @@
-OUTDATED
---------
-
-The point of the metadata is to provide some context for each buffer. In
-the case of audio data, for instance, it would provide the samplerate, bit
-depth, and channel count.
-
-The trick is that there may be multiple types of metadata ganged onto a
-single buffer. This is why they're going to be a GList. This does mean
-extra overhead in all cases, but I think it's minimal. The GList type
-uses a chunk allocater so we're not wasting too much memory or time when
-adding to the list.
-
-The trick is dealing with these structs as they pass through a pipeline,
-since they have potentially different mutability properties. For
-instance, if you've got a mp3 decoder connected to a tee, which sends the
-buffers off to both the decoder and a spectrum analyzer (and then a
-visualization element). The preferred setup would be where every time a
-audio/raw metadata comes down the pipe (indicating a potential change in
-audio format), the audiosink and spectrum would just save off pointers.
-
-So when exactly does this metadata go away (deallocated)? Well, that
-means metadata has to be refcounted. But that gets rather hairy. OK, in
-the simple case you create a metadata struct, it comes with refcount set
-to 1. You pass it through, it stays one, eventually someone drops the
-last reference on the buffer it's tied to, you free the metadata too.
-Easy. What if you tee? You could go through and for every metadata in
-the buffer, increment the refcount by the same as the buffer. So in the
-above case (tee'd), the audiosink and spectrum would get the buffer with a
-refcount of 2, and it'd have a metadata with refcount 2. Do they ref it
-each themselves, then unref the buffer? Or do they remove the metadata?
-Removing the metadata would require a buffer CoW, which would suck, so
-yes, they'd just ref the metadata.
-
-But.... what if they're all in different threads? Then we're off into
-the magical world of mutexes. Everything with a refcount in a threaded
-world must be mutexed, else you can do atomic increment and atomic
-dec&test. Can this be done from C easily? Perhaps it needs to be found
-from kernel includes via autoconf?
-
-
-
-
-The goal in designing the way metadata will be defined and used is to keep
-it as simple as possible. The basis for accomplishing this is the fact
-that in order to actually use (rather than just pass) the metadata, you
-have to know what the fields are, which means you have to have compiled in
-support for that metadata at build time. Therefore, if you're using
-metadata, you must have build-time access to the necessary include file
-that defines it.
-
-So, given that you've got an include file, it would be nice if the whole
-thing could be contained there. This would limit the need to be linked
-against something, or have load-time requirements as to that has to be
-loaded before you are.
-
-Given that really all metadata is is a region of memory of a given size
-with a certain signature, this isn't all that hard. First you lay out the
-struct that defines the metadata. Then you set up #defines that expand to
-the size of the struct in question, as well as the four-cc code that
-defines the type.
-
-The work is done by a few #defines, a la the #defines used in all Gtk
-objects. The first is a NEW() method that allocates the memory for the
-metadata and fills in all the normal fields (type, size, utility
-functions). Because of the way it's defined (as a #define, no less),
-you'll have to invoke it as META_NEW(meta), since it can't return()
-anything.
-
-Another #define will check to make sure a meta is indeed that type by
-verifying the type code and size. Theoretically, meta types can overlap
-with the same fourcc code, as long as they have different sizes. But I
-probably ought to have a global public registry so people writing things
-don't conflict. MSFT got that right, at least.
-
-So, a hairy problem is what to do when there are utility functions
-associated with one of these things. One option is to not bother with
-them. This is very likely a possible solution, since metadata is supposed
-to be flat memory of a given size. Not much to do to either free or copy
-it, is there?
diff --git a/docs/random/mutability b/docs/random/mutability
deleted file mode 100644
index 74a094449c..0000000000
--- a/docs/random/mutability
+++ /dev/null
@@ -1,30 +0,0 @@
-Mutability is the property of an object that defines whether or not you
-are allowed to modify it. In the context of GST, that means that if you
-want to mutilate a buffer, say to do an audio effect, you may have to do
-this on a copy of the buffer, if someone else has a reference on it.
-
-The simplest sequence of events in a decoder pipeline is as follows:
-
-1) create buffer
-2) allocate and fill data region, attach to buffer
-3) pass to next element
-4) decode the data into new buffer, free original buffer
-5) pass to next element
-6) buffer gets copied to output device (sound, video, whatever)
-
-Both of these buffers are created from malloc()'d memory, are referenced
-by one and only one element at a time, and are never modified in place.
-They have no special flags, and when ref==0, they're simply free()'d.
-
-An optimization in the case of the sound card or video double buffering,
-where the output buffer actually comes from the output device. In that
-case the element will be aware of such things.
-
-A more complex example is where the data is teed after being decoded, sent
-to an effects or visualization object.
-
-1) create buffer, fill from source
-2) hand to decoder
-3) create new buffer, decode into it, free old buffer
-4) hand to tee
-5) ref++, hand off to
diff --git a/docs/random/negotiation b/docs/random/negotiation
deleted file mode 100644
index 76dc1974ba..0000000000
--- a/docs/random/negotiation
+++ /dev/null
@@ -1,286 +0,0 @@
-
-Some notes on pad negotiation
-
-
-A "pad link" is a connection between two pads. It can be in one of
-two states, "negotiated" or "not negotiated". Pad links are created
-by gst_pad_link().
-
-A "pad link" is created when two pads are linked using gst_pad_link().
-When initially created, the link only specifies a src pad, a sink pad,
-and optionally a filter caps provided by the application.
-
-In order to pass data through a link, the peer pads must decide on
-what data format to use. This is called negotiation. Pads
-describe acceptable data formats by using a combination of pad
-template caps and (optionally) a pad->getcaps function.
-
-
-Negotiation can happen in one of two forms, directed or undirected.
-Directed negotiation happens when one element has decided (usually
-during negotiation on another pad) to ask for a specific format on
-a pad. This happens when a pad calls gst_pad_try_set_caps().
-Undirected negotiation happens when the core decides to negotiate
-a link, either due to a state change or a specific application
-request.
-
-Steps in undirected negotiation (core view):
-
- - core checks that both pad's parent elements are in the READY or
- higher state.
-
- - core calls gst_pad_get_caps() on each pad, intersects the two caps,
- and intersects again with the filter caps. If the intersection is
- empty, then the pads have no formats in common, and the link fails.
-
- - If the intersection caps is not fixed, there are multiple possible
- formats that the link could use. If this is the case, fixate
- functions are called until the caps are fixed. The fixate functions
- are called by priority -- src application fixate function, sink
- application fixate function, src and sink fixate functions, and
- the default core fixate function. The application fixate functions
- are implemented by the "fixate" signal on each pad. The core
- loops through the fixate functions until a fixed caps is decided
- on.
-
- - Each pad may have a pad_link function, which is called with the
- fixed caps. The pad_link function has the option of accepting,
- rejecting, or delaying the negotiation.
-
- - If both pads accept the caps, the link is then negotiated.
-
-Steps in directed negotiation (gst_pad_try_set_caps):
-
- - the originator is the pad that gst_pad_try_set_caps() is called
- on.
-
- - the elements owning both pads are assumed to be in a non-NULL state
-
- - the caps argument of try_set_caps() must be fixed.
-
- - gst_pad_get_caps() is called on the peer pad, and intersected with
- the originating pad's pad template caps and the filter caps. The
- caps argument is checked to be a subset of the intersection. (It's
- important that this intersection uses the pad _template_ caps.)
-
- - Fixate functions don't need to be called, since the caps are
- already fixed.
-
- - The peer's pad_link function is called.
-
- - If the peer's pad_link function accepts the caps, the link is then
- negotiated.
-
- - If the peer's pad_link function refuses the caps, and the link had
- already been negotiated, the peer's pad_link function is called
- with the caps of the old negotiation.
-
- - Note: the originator's pad_link function is _not_ called. The
- originator must take appropriate alternative steps.
-
-
-Notes about renegotiation:
-
- - same as negotiation. Note that get_caps() functions should always
- ignore the currently negotiated caps of a link.
-
- - if renegotiation fails, the previous negotiation is still in effect.
- If the renegotiation fails in the last pad_link step, the pad_link
- functions are called with the previously negotiated caps.
-
-
-Notes for sources and sinks:
-
- - sources and sinks that talk to hardware may not be able to fully
- describe their available formats, and thus need to rely on pad_link
- functions to test a particular format. FIXME: currently, the core
- completely fails negotiation if a pad_link function refuses a caps,
- instead of attempting with an alternate caps.
-
- Example: Assume osssink advertises rate=(int)[8000,48000], but
- the device cannot actually handle rate=44100 (unknown to osssink).
- Assume that the pad_link function is called with rate=44100 --
- ideally, the pad_link function should return GST_PAD_LINK_DELAYED,
- and future calls to getcaps should return {[8000,44099],[44101,
- 48000]}. I don't know how to make this easy and/or work well.
-
-
-Notes for decoders/demuxers:
-
- - Decoders will typically negotiate a sink pad, receive some data,
- determine the output format, and call try_set_caps() with the given
- format. If the output format is non-fixed, gst_pad_renegotiate()
- may be used instead, in order to have the fixate functions choose
- the optimal format. Note that this requires communication so that
- the pad's getcaps function returns the correct caps.
-
-Notes for converters:
-
- - Converters change one or more properties of the format of a data
- stream. A typical converter's getcaps function will call
- gst_pad_get_allowed_caps() for the opposite pad in the element,
- change one or more fields in the caps, and return the result.
-
- - getcaps function:
-
- - call gst_pad_get_allowed_caps() on the other pad in the element
-
- - for each possible format ("A") in the allowed caps, determine all
- the formats ("B") that your converter could convert the original
- format (A) to. The union of all these formats (all the B's) is
- the caps that should be returned. (This is how to do it
- _theoretically_, but an optimal implementation will probably be
- quite different.)
- For example, a simple way to do this for an element that can convert
- a given field of the caps is to remove the field(s) from the structure,
- then intersect with the pad template.
-
- - As an example, videoscale can convert any sized video to any other
- sized video. Its getcaps function iterates over each structure in
- the caps, and replaces the width and height with the range
- [1,MAXINT].
-
- - pad_link function:
-
- - the "otherpad" is the opposite pad in the element.
-
- - extract fields from the caps that are relevant for your converter
- handling the format. Store these in _local_ variables. (E.g,
- things like video size, sample rate, etc.)
-
- - If it's possible to pass buffers through without modifying them
- (passthrough), you should call gst_try_set_caps() with the caps
- that was specified as the parameter to the pad_link function. If
- this is successful, save the local variables to the element
- structure, perform whatever other setup is necessary for your
- element, and return GST_PAD_LINK_OK.
-
- - Otherwise, you're not using passthrough, and may need to
- change the caps on the otherpad to match the given format.
-
- - If the otherpad is not negotiated (!gst_pad_is_negotiated()),
- you shouldn't attempt to set a format on it. It will eventually
- be negotiated. Save the local variables to the element structure,
- perform whatever other setup is necessary, and return
- GST_PAD_LINK_OK.
-
- - At this point, the other pad is already negotiated, but won't
- accept the passthrough format, so you should combine the existing
- negotiated caps on the otherpad and the caps that was the pad link
- argument. This can either be done using existing information in the
- element that was saved during a previous pad_link call, or you can
- get the information from the negotiated caps
- (gst_pad_get_negotiated_caps()).
-
- As an example, consider the videoscale element. Assume that
- videoscale.src has already negotiated "video/x-raw-yuv,
- format=(fourcc)I420, width=320, height=240", and that the sink
- pad's link function is called with "video/x-raw-yuv,
- format=(fourcc)YUY2, width=640, height=480". Since it's the
- videoscale element, we can have different width and height
- fields on the pads, but the format must be the same. So we'll
- use the existing negotiated size (640x480), and the new format,
- and call gst_pad_try_set_caps() with "video/x-raw-yuv,
- format=(fourcc)I420, width=640, height=480".
-
- This may seem overkill, but most of the time, you'll end up
- calling try_set_caps() with the same caps that are currently
- negotiated -- try_set_caps() just returns GST_PAD_LINK_OK in
- this case.
-
- - If gst_pad_try_set_caps() returns GST_PAD_LINK_OK, save the
- local variables to the element structure. In any case, return
- the return value of gst_pad_try_set_caps().
-
-
-Notes for filters:
-
- - Filters can almost always use gst_pad_proxy_getcaps() as the
- getcaps function. This just returns gst_pad_get_allowed_caps()
- on the otherpad.
-
- - You may be able to use gst_pad_proxy_pad_link() as the pad link
- function, but only if you don't need to extract parameters from
- the caps.
-
-
-Notes for encoders/muxers:
-
- - Encoders and muxers should roughly work like converters. Many
- converters are symmetric; encoders and muxers obvious are not,
- thus it may make the code clearer to have separate src and sink
- getcaps and pad_link functions.
-
- - Encoders and muxers should handle multiple negotiations until
- the first buffer has been passed. After this point, it's unlikely
- that additional negotiations will happen in well-constructed
- pipelines, but it may be wise to "lock" the caps after the
- muxer has committed to a format. (FIXME: it's still unclear to
- me when the caps should get "unlocked". Obviously at EOS or
- PAUSED->READY transitions. Any others?)
-
- - Locking caps can be done by adding (near the top) of the getcaps
- function:
-
- if (my_element->lock_caps) {
- return gst_pad_get_negotiated_caps (pad);
- }
-
-
-Explicit caps:
-
- - There's a hack in the core to make the code for decoder elements
- a lot simpler. This hack can be used only for src pads of elements
- that get their srcpad capabilities directly from the data stream,
- i.e., decoders, demuxers, and typefind. This hack overrides the
- pad's getcaps() and pad_link() function, so that they work correctly
- in all decoder states.
-
- - To enable this hack on a pad, call gst_pad_use_explicit_caps().
-
- - To indicate that a decoder has found the format of the stream, call
- gst_pad_set_explicit_caps(pad,caps) with the caps of the stream.
- This caps must be fixed.
-
- - To indicate that a decoder has lost the format of the stream, i.e.,
- there's been a NEW_MEDIA event, call gst_pad_set_explicit_caps(pad,
- NULL).
-
- - If the explicit caps are set, the getcaps function will return that
- caps, and the pad_link function will return GST_PAD_LINK_OK. If
- the explicit caps are not set, the getcaps function returns the pad
- template caps, and the pad_link function returns GST_PAD_LINK_DELAYED.
-
-
-
-Other junk:
-
- - negotiation can happen at any time
-
- - negotiation can happen multiple times/often happens multiple times
-
- - initial negotiation can lead to strange caps
-
- - directed negotiation can happen in either direction (src to sink or
- sink to src)
-
- - Other considerations ignored, every pad should have a getcaps function.
-
- - If a pad's getcaps function returns the same caps in every
- circumstance, the getcaps function can be omitted.
-
- - If you use gst_pad_use_explicit_caps(), the getcaps function must
- be omitted.
-
- - fixate functions are a method for applications to exert influence
- on how a format is chosen from a caps. It's also used as a hack to
- allow elements to do the same. Element fixate functions are _not_
- intended to give good results for applications -- they're intended
- to give non-disgusting results in gst-launch. Don't attempt to
- make them do more than they're capable of.
-
- - Fixate functions should not be implemented on anything except source
- and sink elements.
-
-
-
diff --git a/docs/random/omega/EOS/chain-walkthrough b/docs/random/omega/EOS/chain-walkthrough
deleted file mode 100644
index b4c28bfb10..0000000000
--- a/docs/random/omega/EOS/chain-walkthrough
+++ /dev/null
@@ -1,19 +0,0 @@
-disksrc -> mp3parse -> mpg123 -> audiosink
-
-disksrc reads last 4K chunk from disk. Sets EOS on the buffer, pushes buffer out to peer.
-disksrc:src sets itself to EOS and chains to mp3parse.
-
- mp3parse takes the buffer and slices it up into several.
- Each buffer gets chained off to mpg123.
-
- mpg123 decoders each frame and sends it to audiosink.
-
- audiosink dumps the audio data to the sound card.
-
- mpg123 returns from the chain, and returns.
-
- mp3parse returns from each of the several chains,a nd returns.
-
-disksrc:src returns from the chain, signals EOS, and sets disksrc's EOS flag
-
-At this point the pipeline state code takes over and starts to set everyone to READY
diff --git a/docs/random/omega/IDEAS b/docs/random/omega/IDEAS
deleted file mode 100644
index ef79da4037..0000000000
--- a/docs/random/omega/IDEAS
+++ /dev/null
@@ -1,21 +0,0 @@
-Plugin Registry:
-Have a web-accessible database of plugins from everywhere, including their merit values. gstplugin.c
-and the autoplug code (should we abstract out autoplug into a special file? I think so) can make use of
-this database via another module (gstwebregistry.c?) to look up stuff. A copy of the registry (gzip'd
-XML) could even be cached. System and user options would determine whether this registry is checked
-and/or updated automatically. The registry could simply be merged with the local machine and user
-registries, with the state bit set to "don't even have it". Autodownload/install code should be
-provided, though designed such that it's not toolkit/OS specific.
-
-Merit:
-Plugins and even type definition should carry merit values, allowing the system to determine which
-plugin or type definition is better. This can be tied into the web registry setup, where merit values
-can be updated without requiring an update of the plugins themselves. They can also be guaranteed
-unique if there's a range reserved for registered objects. Unfortunately, that might get us into some
-political wars. We could leave that up to the users via some voting system.
-
-Scheduling state:
-All the scheduling info for a Bin should be contained in a single object of some form. This can be
-saved off and restored seamlessly at some point in the future. This would be ideal for some autoplug
-cases, where the rest of the pipeline simply ceases to exist temporarily, and scheduling entry points
-may need to be modified until the autoplug stage is finished.
diff --git a/docs/random/omega/TODO-0.1.0 b/docs/random/omega/TODO-0.1.0
deleted file mode 100644
index f07798a0b3..0000000000
--- a/docs/random/omega/TODO-0.1.0
+++ /dev/null
@@ -1,54 +0,0 @@
-mandatory:
------------
-
-(done) Fix compile warnings in gst/*
-
-Make sure all common media types work with autoplug and gstmediaplay
-
-Add interface to control the level of output, both compile- and run-time
-
-(done) Make sure the build is capable of being run without any debugging noise
-
-Make sure that the absence of any of the optional libraries will not be fatal
-
-Make sure all the old plugins don't build normally, via some configure option?
-
-Try to get more of the INFO calls worked out, removing gst_info
-
-Decide whether DEBUG should use the category system, and if so, implement it
-
-Icon for gstplay, .desktop file, etc.
-
-Build tarballs
-
-Build RPMs
-
-Figure out which docs we're going to make 'offical'
-
-UPDATE THE WEB SITE
-
-optional:
----------
-
-Fix gstreamer-launch to do named pads
-
-Need to fix EOS subsystem so -launch can play out
-
-gstreamer-launch should play out till EOS
-
-Finish LADSPA plugin to data-moving stage
-
-
-
-things to remember for the announcement:
----------------------------------------
-
-Build requirements list:
-libtool 1.3.5 or patched to "pass_all"
-optional:
-libcdparanoia
-libmp3lame
-libxaudio
-libXv
-libgdk-pixbuf
-libglade
diff --git a/docs/random/omega/TYPE_FOURCC b/docs/random/omega/TYPE_FOURCC
deleted file mode 100644
index dff7d7d574..0000000000
--- a/docs/random/omega/TYPE_FOURCC
+++ /dev/null
@@ -1,3 +0,0 @@
-#define GST_TYPE_FOURCC(f) \
-(((f)[0]) & ((f)[1] << 8) & ((f)[2] << 16) & ((f)[3] << 24))
-
diff --git a/docs/random/omega/build/TODO b/docs/random/omega/build/TODO
deleted file mode 100644
index fe0e5bea14..0000000000
--- a/docs/random/omega/build/TODO
+++ /dev/null
@@ -1,7 +0,0 @@
-Convert all the STREAMER_ version stuff to GSTREAMER_
-
-Convert all the --version-info lines to use GSTREAMER_LIBVERSION
-
-Fix all the masses of Makefile.am's that override CFLAGS
-
-Makefile.am's should use the library_la_CFLAGS and _LIBADD always
diff --git a/docs/random/omega/caps2 b/docs/random/omega/caps2
deleted file mode 100644
index 62647813e1..0000000000
--- a/docs/random/omega/caps2
+++ /dev/null
@@ -1,320 +0,0 @@
-The elementfactory for a given element will contain some information about the capabilities of element's
-pads or potential pads. An indication will be provided as to whether the pad always exists, always
-exists once data is present, or *might* exist once data is present (the latter case is for things like
-the MPEG system parsers, where an audio stream might or might not exist).
-
-
-First, an entirely normal example:
-----------------------------------
-
-(-----------) (----------) (-------------)
-! disksrc ! ! mpg123 ! ! audiosink !
-! src sink src sink !
-! ! ! ! ! !
-(-----------) (----------) (-------------)
-
-We start with only the disksrc. The typefind filter is attached to the disksrc, and via typefind magic
-the properties of the disksrc are found to be:
-
- disksrc->src->caps = {
- "audio/mp3",
- "layer", GST_CAPS_INT (3),
- "bitrate", GST_CAPS_INT (128),
- NULL
- };
-
-A look through the plugin registry shows that we have an element called mpg123 that has the following
-caps:
-
- static GstCapsFactory mpg123_sink_caps = {
- "audio/mp3",
- "layer", GST_CAPS_INT_RANGE (1, 3),
- "bitrate", GST_CAPS_INT_RANGE (8, 320),
- NULL
- };
-
-The caps of the disksrc fit within those parameters, so we instantiate an mpg123 and attach it to the
-disksrc. The connection succeeds negotiation and as a result the mpg123 specifies its output caps as:
-
- mpg123->src->caps = {
- "audio/raw",
- "format", GST_CAPS_BITFIELD (S16),
- "depth", GST_CAPS_INT (16),
- "rate", GST_CAPS_INT (44100),
- "channels", GST_CAPS_INT (2),
- NULL
- };
-
-Again from the plugin registry we find an element audiosink that has appropriate caps:
-
- static GstCapsFactory audiosink_src_caps = {
- "audio/raw",
- "format", GST_CAPS_BITFIELD (S16,....),
- "depth", GST_CAPS_INT (16),
- "rate", GST_CAPS_INT_RANGE (4000, 96000),
- "channels", GST_CAPS_INT_RANGE (1, 2),
- NULL
- };
-
-A copy of the audiosink is instantiated and attached, negotiation goes smoothly, and we're done. No
-dataflow has occurred, no failure found, etc. An ideal autoplug.
-
-
-Now, a slightly more convoluted example:
-----------------------------------------
-
-Start with the same graph:
-
-(-----------) (----------) (-------------)
-! disksrc ! ! mpg123 ! ! audiosink !
-! src sink src sink !
-! ! ! ! ! !
-(-----------) (----------) (-------------)
-
-Run typefind on the disksrc's output, get the same output caps:
-
- disksrc->src->caps = {
- "audio/mp3",
- "layer", GST_CAPS_INT (3),
- "bitrate", GST_CAPS_INT (128),
- NULL
- };
-
-Find and attach mpg123, get the following output caps this time:
-
- mpg123->src->caps = {
- "audio/raw",
- "format", GST_CAPS_BITFIELD (S16),
- "depth", GST_CAPS_INT (16),
- "rate", GST_CAPS_INT (44100),
- "channels", GST_CAPS_INT (1),
- NULL
- };
-
-Note that this time we have a mono output. A look into the audiosink caps shows that we have a match.
-So we instantiate a copy. Oops. We now find that the caps for the input pad on our audiosink have
-changed:
-
- mpg123->src->caps = {
- "audio/raw",
- "format", GST_CAPS_BITFIELD (S16,...),
- "depth", GST_CAPS_INT (16),
- "rate", GST_CAPS_INT (11025, 48000),
- "channels", GST_CAPS_INT (2),
- NULL
- };
-
-Whoops. It seems that the sound card we've got in this machine (FIXME how on earth to deal with
-multiple sound cards???) doesn't support mono output *at all*. This is a problem. We now find that we
-have no options as far as directly matching the mpg123 to the audiosink.
-
-A look through our (fictitious) plugin registry shows at least one element that at least has audio/raw
-on both input and output (since both mpg123 and audiosink have open pads with this mime type). A closerlook shows that its caps are:
-
- static GstCapsFactory mono2stereo_sink_caps = {
- "audio/raw",
- "channels", GST_CAPS_INT (1),
- NULL
- };
- static GstCapsFactory mono2stereo_src_caps = {
- "audio/raw",
- "channels", GST_CAPS_INT (2),
- NULL
- };
-
-Wow, that's a perfect match. Instantiate, attach to mpg123, no problems. Attach to audiosink, no
-problems. Done. When we start up the pipeline, we should get absolutely no callbacks from pads saying
-"help me, I've fallen and..., er, I don't like this buffer!".
-
-
-A really messy case:
---------------------
-
-Start with a disksrc, typefind it, get the following:
-
- disksrc->src->caps = {
- "audio/mp3",
- "layer", GST_CAPS_INT (3),
- "bitrate", GST_CAPS_INT (128),
- NULL
- };
-
-Look through the plugin registry, find mpg123. Instantiate it, attach it. It spits out audio
-parameters as usual:
-
- mpg123->src->caps = {
- "audio/raw",
- "format", GST_CAPS_BITFIELD (S16),
- "depth", GST_CAPS_INT (16),
- "rate", GST_CAPS_INT (44100),
- "channels", GST_CAPS_INT (2),
- NULL
- };
-
-Now we instantiate an audiosink plugin. This time, we're sunk:
-
- mpg123->src->caps = {
- "audio/raw",
- "format", GST_CAPS_BITFIELD (S8,U8),
- "depth", GST_CAPS_INT (8),
- "rate", GST_CAPS_INT_RANGE (11025, 22050),
- "channels", GST_CAPS_INT (1),
- NULL
- };
-
-ACK! It's one of those Disney Sound Source things. We've got a problem here that isn't obviously
-solvable. However, there happens to be another mp3 decoder sitting around. It's got the same
-properties as mpg123, but a lower merit value. Let's instantiate one and attach it. We get the
-following output pad caps:
-
- mp3decoder->src->caps = {
- "audio/raw",
- "format", GST_CAPS_BITFIELD (S8,S16),
- "depth", GST_CAPS_INT_RANGE (8,16),
- "rate", GST_CAPS_INT_RANGE (8000, 44100),
- "channels", GST_CAPS_INT (1,2),
- NULL
- };
-
-Well, that matches the audiosink. We try attaching it, and during negotiation the mp3decoder finds
-sufficient common ground with the castrated audiosink and sets its output pad to match the best of the
-options: S8 at 22050 KHz.
-
-
-
-Next to impossible scenario: DVD
---------------------------------
-
-Start with a dvdsrc. It's output pad caps are:
-
- static GstCapsFactory dvdsrc_src_caps = {
- "video/mpeg",
- "mpegversion", GST_CAPS_INT (2),
- "systemstream", GST_CAPS_BOOLEAN (TRUE),
- NULL
- };
-
-The type would be classified as incomplete via some mechanism. This might cause the autoplug code to go
-and run the typefind function. It would flesh the type out to the following:
-
- dvdsrc->src->caps = {
- "video/mpeg",
- "mpegversion", GST_CAPS_INT (2),
- "systemstream", GST_CAPS_BOOLEAN (TRUE),
- "videostreams", GST_CAPS_INT (1),
- "audiostreams", GST_CAPS_INT (3),
- "bitrate", GST_CAPS_INT (40960),
- NULL,
- };
-
-Wow, that helped a lot. A check through the plugin registry shows that the mpeg2parse will match those
-properties:
-
- static GstCapsFactory mpeg2parse_sink_caps = {
- "video/mpeg",
- "mpegversion", GST_CAPS_INT (2),
- "systemstream", GST_CAPS_BOOLEAN (TRUE),
- NULL
- };
-
-(In retrospect, it may not be necessary to run typefind if there's match this good right away. Only run
-typefind when there's no exact match.)
-Since there are no output pads yet, we have to actually push data through the pipeline. The moment a
-buffer or two get to the mpeg2parse element, it promptly goes and creates an output pad, probably of the
-following caps:
-
- mpeg2parse_video_src_caps = {
- "video/mpeg",
- "mpegversion", GST_CAPS_RANGE (1,2),
- "systemstream", GST_CAPS_BOOLEAN (FALSE),
- NULL
- };
-
-This seems to be a task for typefind again. But since data is flowing, we have to be careful with the
-buffers. (This is the case in any typefind maneuver, but more so when one really can't rewind the
-source without consequences) The autoplug system attaches a special pseudo-element to mpeg2parse's new
-output pad, and attaches the typefind element to the end of that. The pseudo-element takes the buffer,
-stores it, and passes a copy off to the attached element, in this case typefind. This repeats until
-typefind has determined the type, at which point the typefind is removed, and the newly found element is
-attached instead.
-
-The pseudo-element is 'rewound' and the stored buffers flow out and into the newly attached element.
-When the cache of buffers is gone, a signal fires and the autoplug system removes the pseudo-element and
-reconnects the pipeline.
-
-In this case, the typefind function will find the following:
-
- mpeg2parse_video_src_caps = {
- "video/mpeg",
- "mpegversion", GST_CAPS_INT (2),
- "systemstream", GST_CAPS_BOOLEAN (FALSE),
- "bitrate", GST_CAPS_INT (36864),
- "width", GST_CAPS_INT (720),
- "height", GST_CAPS_INT (480),
- "framerate", GST_CAPS_FLOAT (29.97002997),
- "chromaformat", GST_CAPS_INT (1), [GST_CAPS_STRING ("4:2:0") ?]
- NULL
- };
-
-Back to the plugin registry, we find our only choice is mpeg2dec, which has input caps of:
-
- static GstCapsFactory mpeg2dec_sink_caps = {
- "video/mpeg",
- "mpegversion", GST_CAPS_RANGE (1,2),
- "systemstream", GST_CAPS_BOOLEAN (FALSE),
- NULL
- };
-
-Once again it just so happens that we really didn't need to do the typefind at all. But it can't hurt
-unless the typefind is slow and painful, which we can guess won't be the case since the choices are
-rather limited by the fact that there's already a MIME type attached, meaning we can drastically reduce
-the number of typefind functions we try (down to one, actually).
-
-However, since we *have* run the typefind, upon attachment of the input pad of mpeg2dec, the output pad
-looks like the following:
-
- mpeg2dec_src_caps = {
- "video/raw",
- "fourcc", GST_CAPS_LIST (
- GST_CAPS_FOURCC ("YV12"), [identical...]
- GST_CAPS_FOURCC ("IYUV"),
- GST_CAPS_FOURCC ("I420"),
- ),
- "width", GST_CAPS_INT (720),
- "height", GST_CAPS_INT (480),
- "framerate", GST_CAPS_FLOAT (29.97002997),
- NULL
- };
-
-Currently only videosink supports the output of video/raw. It claims a list of FOURCCs but nothing
-more:
-
- static GstCapsFactory videosink_sink_caps = {
- "video/raw",
- "fourcc", GST_CAP_LIST ( GST_CAPS_FOURCC ("YV12"),
- GST_CAPS_FOURCC ("IYUV"), GST_CAPS_FOURCC ("I420"),
- GST_CAPS_FOURCC ("YUY2"), GST_CAPS_FOURCC ("UYVY"),
- [ etc... ],
- ),
- NULL
- };
-
-When instantiated, we potentially have the same problem as with the audiosink: we don't necessarily know
-which hardware output to use. Somehow we have to solve the problem of setting some element arguments
-before we can get useful information out of them as to the properties. In this case anyway, if the
-videosink were to find only one output possibility, it would trim the list of FOURCCs it can deal with
-to what the hardware can handle, as well as add further properties:
-
- videosink_sink_caps = {
- "video/raw",
- "fourcc", GST_CAPS_LIST (GST_CAPS_FOURCC ("YV12"),
- GST_CAPS_FOURCC ("YUY2"),
- ),
- "width", GST_CAPS_INT_RANGE (4,1020),
- "height", GST_CAPS_INT_RANGE (4,1020),
- NULL
- };
-
-We can now connect the mpeg2dec output to the videosink, and we now have displaying video.
-
-. . . .
diff --git a/docs/random/omega/caps3 b/docs/random/omega/caps3
deleted file mode 100644
index 11dcf8c646..0000000000
--- a/docs/random/omega/caps3
+++ /dev/null
@@ -1,20 +0,0 @@
- /* fake mp3 where bitrates 8 through 32 can't be stereo */
-static GstCapsListFactory mpg123_sink_caps = {
- {
- "audio/mp3",
- "layer", GST_CAPS_INT_RANGE (1, 3),
- "bitrate", GST_CAPS_INT_RANGE (32, 320),
- "channels", GST_CAPS_INT_RANGE (1, 2),
- "framed", GST_CAPS_BOOLEAN (TRUE),
- NULL
- },
- {
- "audio/mp3",
- "layer", GST_CAPS_INT_RANGE (1, 3),
- "bitrate", GST_CAPS_INT_RANGE (8, 32),
- "channels", GST_CAPS_INT_RANGE (1),
- "framed", GST_CAPS_BOOLEAN (TRUE),
- NULL
- },
- NULL
-};
diff --git a/docs/random/omega/debug-commit b/docs/random/omega/debug-commit
deleted file mode 100644
index b1b4589708..0000000000
--- a/docs/random/omega/debug-commit
+++ /dev/null
@@ -1,30 +0,0 @@
-Changes made to the DEBUG system. New header file gstdebug.h holds the stuff to keep it out of gst.h's
-hair. DEBUG prints out the process id, cothread id, source filename and line number. Two new macros
-DEBUG_ENTER and DEBUG_LEAVE are used to show the entry and exit of a given function. This eventually
-might be used to construct call trace graphs, even taking cothreads into account. This would be quite
-useful in visualizing the scheduling mechanism.
-
-Minor changes to various debug messages.
-
-Also sitting in gstdebug.h is a prototypical DEBUG_ENTER that's capable of performing DEBUG_LEAVE
-automatically. It does this by utilizing a little-known GCC extension that allows one to call a
-function with the same parameters as the current function. The macro uses this to basically call
-itself. A boolean is used to ensure that when it calls itself it actually runs the body of the
-function. In the meantime it prints stuff out before and after the real function, as well as
-constructing a debugging string. This can be used eventually to provide call-wide data on the DEBUG
-lines, instead of having to replicate data on each call to DEBUG. More research is needed into how this
-would most cleanly be fit into some other chunk of code, like GStreamer (I think of this DEBUG trick as
-a separate project, sorta).
-
-Unfortunately, the aforementioned DEBUG trick interacts quite poorly with cothreads. Almost any time
-it's used in a function that has anything remotely to do with a cothread context (as in, it runs in
-one), a segfault results from the __builtin_apply call, which is the heart of the whole thing. If
-someone who really knows assembly could analyze the resulting code to see what's really going on, we
-might find a way to fix either the macro or the cothreads (I'm thinking that there's something we missed
-in constructing the cothreads themselves) so this works in all cases.
-
-In the meantime, please insert both DEBUG_ENTER and DEBUG_LEAVE in your functions. Be sure to put
-DEBUG_ENTER after your variable declarations and before any functional code, not to put the function
-name in any DEBUG strings (it's already there, trust me), and put a DEBUG_LEAVE if you care enough.
-
-Changes are going to happen in the way DEBUGs and other printouts occur, so stay tuned.
diff --git a/docs/random/omega/eos.old b/docs/random/omega/eos.old
deleted file mode 100644
index 8621d67fc3..0000000000
--- a/docs/random/omega/eos.old
+++ /dev/null
@@ -1,52 +0,0 @@
-OUTDATED
---------
-
-
-What I propose for EOS condition is th following: As is stands, EOS is a
-flag in a buffer. However, it's also a signal from an element, which is
-odd. What I propose is that EOS become a pad thing. Here's how it would
-work:
-
-When a source has a final buffer, it marks it as EOS. It passes this on
-through the pad, and the pads set a 'pending' EOS state on the way
-through. When another push or pull happens, only then does the pad signal
-EOS. This means that EOS doesn't happen until the buffer has passed all
-the way to the end, and a request for another buffer is starting to work
-its way back. It gets stopped almost immediately.
-
-This gets a bit messy in complex cases, haven't thought them all out.
-Assuming everything is reasonably symmetric, it should work out cleanly.
-We may have to add the ability to force buffers to pass even post EOS, but
-that sounds like a hack.
-
-
-
-Example: DVD
-============
-
-
- ------------ |-------|
- -| mpeg2dec |-----| |
- ------------------------ / ------------ | |
- | video0|------/ ------------- | | -------------
- | | -| subtitle0 |-----| merge |-------| videosink |
- | subtitle0|------/ ------------- | | -------------
----------- | | ------------- | |
-| dvdsrc |---| mpeg2parse subtitle1|--------| subtitle1 |-----| |
----------- | | ------------- |-------|
- | audio0|--------\ ----------- |----------|
- | | -| ac3dec0 |-----| |
- | audio1|-------\ ----------- | | -------------
- ------------------------ \ ----------- | switcher |----| audiosink |
- -| ac3dec1 |-----| | -------------
- ----------- |----------|
-
-When dvdsrc reads its last buffer, it sets EOS on the buffer and passes it. The pad (probably audio1,
-since video and audio are interleaved and subtitles don't usually end up at the end of a movie) would
-set EOS pending, and pass it to mpeg2dec which would pass the EOS to the merge. Merge would then ask
-for a buffer from subtitle1, since that's the one we're showing subtitles on right now. It would then
-pull from the mpeg2parse. This would cause a switch back the parser, who's next task is to go and set
-all the output pads to EOS. It may have to accomplish this by sending a zero-byte buffer.
-
-Well, this example isn't actually very good. I'll go ahead and implement it, and we can see what
-happens. The current setup is so lacking that anything is better....
diff --git a/docs/random/omega/filterfactory b/docs/random/omega/filterfactory
deleted file mode 100644
index 9936306a14..0000000000
--- a/docs/random/omega/filterfactory
+++ /dev/null
@@ -1,3 +0,0 @@
-A very useful feature would be a FilterFactory, i.e. some system whereby filter writers can avoid
-getting their hands messy with the details of building and operating a G[tk]Object. A couple of
-structures and a couple of functions should do it.
diff --git a/docs/random/omega/output_policies b/docs/random/omega/output_policies
deleted file mode 100644
index 81dfc803a9..0000000000
--- a/docs/random/omega/output_policies
+++ /dev/null
@@ -1,61 +0,0 @@
-GStreamer polices for various forms of library output, including error cases.
-
-
-g_return_...
-============
-The parameters of a function are checked for validity (non-NULL, correct type, etc.) with g_return[_val]_if_fail, i.e.:
-
-gst_element_connect (GstElement *src, gchar *srcpadname,
- GstElement *dest, gchar *destpadname) {
- g_return_if_fail (src != NULL);
- g_return_if_fail (GST_IS_ELEMENT(src));
- g_return_if_fail (srcpadname != NULL);
- g_return_if_fail (dest != NULL);
- g_return_if_fail (GST_IS_ELEMENT(dest));
- g_return_if_fail (destpadname != NULL);
-
-This will inform the user of the library of any basic problems with the arguments they pass, such as a NULL pointer.
-
-
-ERROR
-=====
-The ERROR macro will be used whenever there is some other type of flaw in the data passed, at a GStreamer-specific
-level:
-
- srcpad = gst_element_get_pad (src, srcpadname);
- if (srcpad == NULL) {
- ERROR(src,"source element has no pad \"%s\"",srcpadname);
- return;
- }
-
-An ERROR will generally cause ceasation of the application, and ideally launch gdb.
-
-
-INFO
-====
-The INFO macro will be used to output any interesting state from the library, such as plugin loading and various events
-of interest. They will be separated into categories that can be individually enabled and disabled.
-
-Categories (first draft of list, unordered):
-
-cothreads [cothreads.c]
-gst initialization [gst.c]
-autoplug results [autoplug.c]
-autoplug attempts [autoplug.c]
-bin parentage issues [gstbin.c]
-plan generation [gstbin.c]
-schedule generation [gstschedule.c]
-schedule implementation [gstpad.c, gstbin.c, etc.]
-buffer operations [gstbuffer.h]
-caps matching [gstcaps.c]
-clock stuff [gstclock.c]
-element pad operations [gstelement.c]
-elementfactory operations [gstelementfactory.c]
-pad creation/connection [gstpad.c]
-pipeline stuff [gstpipeline.c]
-plugin loading [gstplugin.c]
-plugin loading errors [gstplugin.c]
-properties operations [gstprops.c]
-thread creation/management [gstthread.c]
-type operations [gsttype.c]
-XML load/save [gstxml.c]
diff --git a/docs/random/omega/pad-negotiation b/docs/random/omega/pad-negotiation
deleted file mode 100644
index 6c124685f5..0000000000
--- a/docs/random/omega/pad-negotiation
+++ /dev/null
@@ -1,40 +0,0 @@
-OUTDATED, EARLY IDEA
---------------------
-
-When two pads are connected, a negotiation phase is going to have to
-happen. Ideally, the caps of the two pads will both be fully-specified,
-and match. That's the ideal case, but may rarely happen in practice.
-
-It'll work the following way:
-
-1) gst_pad_connect(pad1,pad2) is called by something
-2) pad1's negotiate() method is called, with pad2 as argument
-3) negotiate() repeatedly calls pad2's set_caps() method
-
-At some point, the two pads will agree on a set of caps, and proceed by
-returning TRUE from negotiate(), at which point gst_pad_connect()
-finishes. If it returns FALSE, gst_pad_connect() is forced to fail.
-
-Now, obviously the algorithm used to find matching caps can get
-complicated. But in some cases it'll be simple. Ideally, if there is no
-negotiate() function for pad1, there'll be a function that will go through
-the options and try to make pad1 and pad2 meet in the middle, with no
-specific knowledge of what the caps actually mean.
-
-Another detail is deciding which pads are pad1 and pad2. In the autoplug
-case, the code will sometimes know which of the pads have the more
-specific caps. In others, you may not. Either you can guess, and
-possibly lose to having the slower of the two pad's negotiate() methods do
-the work, or you might be able to actually guess at which is the most
-specific set of caps:
-
-For any given typeid in the union of both pads, look at all properties
- For each property, find the smallest range, assign this a 1.0
- For all other instances of this property, assign relative to 1.0
-For each pad1,pad2
- Take the assigned value of every property, and multiply together
-
-Whichever value is lower between pad1 and pad2 is most likely to be the
-most specific set of caps. The trick is implementing the above
-efficiently, but on the surface there appear to be more than enough
-potential optimizations.
diff --git a/docs/random/omega/padtemplates b/docs/random/omega/padtemplates
deleted file mode 100644
index 6b3ce4c19a..0000000000
--- a/docs/random/omega/padtemplates
+++ /dev/null
@@ -1,46 +0,0 @@
-OUTDATED
---------
-
-typedef gpointer GstCapsFactoryEntry;
-typedef GstCapsFactoryEntry GstCapsFactory[];
-typedef GstCapsFactory *GstCapsListFactory[];
-
-typedef gpointer GstPadFactoryEntry;
-typedef GstPadFactoryEntry GstPadFactory[];
-typedef GstPadFactory *GstPadListFactory[];
-
-#define GST_PADFACTORY_SRC GINT_TO_POINTER (GST_PAD_SRC)
-#define GST_PADFACTORY_ALWAYS GINT_TO_POINTER (GST_PAD_ALWAYS)
-
-typedef struct GstPadTemplate {
- gchar *name_template;
- gint direction;
- gint presence;
- GList *caps;
-};
-
-GstPadFactory mpg123_src_padfactory = {
- "src"
- GST_PADFACTORY_SRC,
- GST_PADFACTORY_ALWAYS,
-
- "audio/raw"
- "samplerate", GST_PROPS_INT (44100),
- . . .
- NULL
-};
-
-static GstPadTemplate *srcpadtemplate;
-
-mpg123_new(GstMpg123 *mpg123) {
- mpg123->srcpad = gst_pad_new_template("src", srcpadtemplate);
- . . .
-}
-
-plugin_init() {
- GstElementFactory *factory;
-
- factory = gst_elementfactory_new("mpg123",. . .);
- srcpadtemplate = gst_padfactory_new(mpg123_src_padfactory);
- get_elementfactory_add_padtemplate (srcpadtemplate);
-}
diff --git a/docs/random/omega/plan-generation b/docs/random/omega/plan-generation
deleted file mode 100644
index afc9e07a7a..0000000000
--- a/docs/random/omega/plan-generation
+++ /dev/null
@@ -1,87 +0,0 @@
-OUTDATED
---------
-
-Plan generation happens at transition from NULL to READY (and PLAYING to READY right now, need to fix
-that). By way of some logic in gst_bin_change_state(), gst_bin_create_plan() is only called for the
-outer Bin, usually a Pipeline. This keeps things from getting nasty later on.
-
-A major new concept in plan generation is that of the 'manager'. This is the element that is responsible
-for running a given element. In general, Pipelines and Threads are the only managing-capable elements
-(have the MANAGER flag set), since they are the only ones with real scheduling authority (because they
-have a process context to play with, basically).
-
-gst_bin_set_manager() is called to set the manager element of the bin and all it's children and their
-children. However, there's one important trick: it won't recurse into child Bins that have the MANAGER
-flag set. This avoids some highly redundant recursion.
-
-When create_plan() is called on the outside Pipeline, the first thing it does is call
-set_manager(self,self). As noted above, this recursion will not proceed into child Bins that have the
-MANAGER flag set.
-
-The next step is to recursively generate the plan (yes, head-recursive). This gives child Bins the
-opportunity to generate their plan first, causing a inside-to-outside sequence. This matches the way
-the scheduling is arranged now, where the plan for a Src/Connection outside a Bin is handled by that
-Bin, not it's parent. But we must be very careful not to stomp on that plan in the parent Bin.
-
-Because create_plan() is called on all Bins, but we can only set up scheduling state in MANAGER bins,
-create_plan() must perform create_plan() recursion, but not do anything else *unless* the MANAGER bit is
-set. It shouldn't even call set_manager() unless it's a MANAGER itself, because calling it otherwise
-would waste time doing the work again. Basically, from the standpoing of setting the manager,
-create_plan() recursion starts it when the current Bin is a MANAGER, and set_manager() stops when it
-finds the next one. create_plan()'s further recursion eventually starts the process back up again
-furtuer down the hierarchy, until everything is covered.
-
-For all MANAGER Bins, the last step is to actually create the scheduling plan. This is still one of the
-nastiest chunks of code in the whole project, and probably will do nothing but get worse from now on (it
-got better recently, but only because I took a chainsaw to the code and broke everything...). It will
-remain similar to what it is now, but with some definite differences.
-
-First task is now to find all the elements that we're responsible for. This is normally a recursive
-process, because the structure is an arbitrary tree. However, something like the following should work
-(bin is self):
-
- GSList *elements = NULL;
- GList *children;
- GSList *waiting_bins = NULL;
- GstBin *waiting_bin;
-
- waiting_bins = g_slist_prepend (waiting_bins,bin);
-
- while (waiting_bins) {
- // retrieve the top of the stack and pop it
- waiting_bin = GST_BIN (waiting_bins->data);
- waiting_bins = g_slist_remove (waiting_bins,waiting_bin);
-
- // walk the list of elements, and find bins
- children = waiting_bin->children;
- while (children) {
- // add it to the list of elements
- elements = g_slist_prepend (elements, children->data);
-
- // if it's a bin and it's not a managing bin,
- // shove it on the list of bins to recurse into
- if (GST_IS_BIN (children->data) &&
- !GST_FLAG_IS_SET (GST_ELEMENT (children->data)))
- waiting_bins = g_slist_prepend (waiting_bins,children->data);
-
- children = g_list_next (children);
- }
- }
-
-The code makes the assumption that the manager of every element is the same until such time as a
-different managing parent appears in the hierarchy. This is the result of the aforementioned nested
-recursion of create_plan() and set_manager(), but may not remain the case forever. The above loop
-should probably be slightly re-written to work solely on whether or not the Bin in question is the
-element's manager. This means that the child Bins are *always* recursed into, in case there's a rogue
-element inside of one of them that's supposed to be managed.
-
-At the same time all the elements managed by this bin are found (i.e. in the inner loop), we can
-determine some useful bits of information, such as testing for several cases that require the use of
-cothreads. The availability of manager information at this point may aid significantly in this
-decision.
-
-Finally, the scheduling plan is generated, based on all the elements to be managed by the Bin (the list
-of which may span several 'generations' of Bins and elements). Elements which have peers in child
-self-managed Bins are left alone on for the pad in that makes that connection. This should keep the
-parent Bins from stepping all over state set up by the child Bins, by establishing clear implicit
-ownership on the pad level, based on the managing Bins' relationship to the pad.
diff --git a/docs/random/omega/sched-case b/docs/random/omega/sched-case
deleted file mode 100644
index f5ebd2d90a..0000000000
--- a/docs/random/omega/sched-case
+++ /dev/null
@@ -1,20 +0,0 @@
-Case 1:
-
-
----------------------------------------------------
-| pipeline |
-| --------------- ---------------- |
-| | bin | | thread | |
-| | ----------- | ------------ | ------------ | |
-| | | fakesrc | | | queue | | | fakesink | | |
-| | | src>|-|--|<sink src>|--|-|<sink | | |
-| | ----------- | ------------ | ------------ | |
-| --------------- ---------------- |
----------------------------------------------------
-
-Pipeline manages: fakesrc, queue
-Thread manages: fakesink
-Both forced to use cothreads.
-
-First thing the thread does is try to pull from the queue. Because it's a chain function, it runs in
-_chain_wrapper, which calls gst_pad_pull().
diff --git a/docs/random/omega/sched-commit1 b/docs/random/omega/sched-commit1
deleted file mode 100644
index 4170674847..0000000000
--- a/docs/random/omega/sched-commit1
+++ /dev/null
@@ -1,103 +0,0 @@
-STATUS: pushregion/pullregion is gone
--------------------------------------
-
-Changed the way things are scheduled, especially sources. A Src used to
-have a push() function, and optionally a pushregion() to deal with async
-reads, etc. That whole thing has gone away, in favor of providing a
-pull() function for the output (Src) pad instead, ala chain functions.
-This makes constructing cothreaded schedules out of non-loop elements
-somewhat easier. Basically there was always a question as to which pad
-was being dealt with. In the pullregion case, cothread-specific data was
-used to try to pass the region struct to the right place, which is a slow
-hack. And in general, the push function severely limited the kind of
-tricks that could be played when there's more than one output pad, such as
-a multi-out file reader with async capabilities on each pad independently.
-
-This changes the way cothread scheduling occurs. Instead of the hack to
-deal with Src's by calling their push() function (or optionally the
-pushregion(), in certain cases), we now are working towards a general
-mechanism where pads are the only thing that are dealt with directly.
-
-An optimization was made in the process of doing this: the loopfunction
-actually run as the outer [stack] frame of the cothread is now set more
-intelligently in create_plan() based on what kind of element it is. We
-now have:
-
-loopfunc_wrapper: used for loop-based elements, it simply calls the
- loopfunc in a loop, paying attention to COTHREAD_STOPPING (see
- below). It currently does other, soon to be deprecated, stuff.
-
-pullsrc_wrapper: wraps a Src that's not loop-based (since your options
- are now loop- or pull-based)
-
-There will be a couple more to deal with other cases, such as Connections
-and chain-based elements. The general idea is that it's a lot more
-efficient to make the decisions once in create_plan than to keep doing
-this huge if/else chain in the wrapper. Just choose the right wrapper up
-front. It'll be most apparent performance-wise in the case of whichever
-element context is switched to first for each iteration, since the whole
-wrapper setup is done for every iteration.
-
-The tricky part is that there is now a bit of overloading of the function
-pointers in a pad. The current meanings (possibly to change a bit more
-soon) are:
-
-chainfunc: as always, chainfunc pointer is mirrored between peer pads
- (this may change, and the chain func may end up in pushfunc)
-pushfunc: SrcPad: gst_pad_pushfunc_proxy, cothread_switch to peer
- SinkPad: none (may take over chainfunc, see below) pullfunc:
-SrcPad: Src or Connection's function to construct buffers
- SinkPad: gst_pad_pullfunc_proxy, cothread_switch to peer
-
-There are a number of issues remaining with the scheduling, not the least
-of which is the fact that Connections are still dealt with the old way,
-with _push() functions and such. I'm trying to figure out a way to unify
-the system so it makes sense. Following the scheduling system is hard
-enough, trying to change it is murder.
-
-
-Another useful scheduling addition, mentioned above, is COTHREAD_STOPPING.
-It's an element flag that's used to signal whatever code is running in
-cothread context that it should be finishing up and exiting soon. An
-example of this is in plugins/cobin/spindentity.c. All the loops should
-now be composed of do/while loops, rather than while(1) loops:
-
- do {
- buf = gst_pad_pull(spindentity->sinkpad);
- gst_pad_push(spindentity->srcpad,buf);
- } while (!GST_ELEMENT_IS_COTHREAD_STOPPING(element));
-
-The reason for this is that COTHREAD_STOPPING may be set before the above
-loop ever gets started. It wouldn't do for the body of the loop to never
-once get called, that would simply stall the pipeline. Note that only the
-core library code is ever responsible for setting and unsetting this flag.
-All elements have to do is respond to it by cleanly exiting the loop and
-the function holding it.
-
-This is needed primarily to allow iterations to occur properly.
-Basically, there's a single entry point in the cothread scheduling loop,
-gst_bin_iterate_func() simply switches to this cothread. If the element
-in this context is allowed to loop infinitely, nothing would even switch
-back to the context from which the iterate() was originally called. This
-is a bit of a problem. The solution is for there to be an implicit switch
-back to the originating context. Now, even I'm not sure exactly how this
-works, but if the cothread that's switched to actually returns, execution
-returns back to the calling context, i.e. iterate_func().
-
-COTHREAD_STOPPING is therefore set just before switching into this
-(currently randomly chosen) context, on the assumption that it will return
-promptly after finishing its duties. The burden of clearing the flag
-falls to the various wrapper functions provided by the Bin code, thus
-element writers don't have to worry about doing that at all (and simply
-shouldn't).
-
-
-Related changes:
-All the sources in elements/ have been changed to reflect the new system.
-
-
-FIXMEs:
-1) gstpipeline.c calls gst_src_push at some point, dunno why, it's
-commented out now.
-2) any other sources, including vcdsrc, dvdsrc, and v4lsrc will break
-badly and need to be modified to work as pull-based sources.
diff --git a/docs/random/omega/sched/chains b/docs/random/omega/sched/chains
deleted file mode 100644
index 69af945feb..0000000000
--- a/docs/random/omega/sched/chains
+++ /dev/null
@@ -1,89 +0,0 @@
-A new concept in scheduling is that of chains of elements that need to be schedule separately, even in the same set of
-managed elements (which is the set of elements that the Bin in question [a pipeline or thread] is responsible for).
-An example would by anywhere you have a non-blocking queue in place for buffering. This kind of element might be
-useful in cases where the scheduling on a buffer level is tight enough that deadlocks might occur.
-
-The scheduler will find chains by traversing the pipeline through the list of managed elements. A chain boundary is
-anywhere you have a 'DECOUPLED' element. A DECOUPLED element is one where there is no direct correlation between the
-activities of the various pads. A source fits this description, although the normal single-pad source is the
-degenerate case. A queue more properly fits the bill, since pushing a buffer at the sink pad doesn't trigger anything
-on the src pad, and vice versa. A multi-src async source is probably the best example, since you want to leave the
-scheduling up to the elements connected to it.
-
-Anyway, first the simple case:
-
-fakesrc -> fakesink
-
-Both of them should probably have the DECOUPLED bit set, at least to be true to the nature of the actual fake elements.
-These two end up being a chain, and scheduling has to be set up for the chain. There are no cothreaded elements in the
-chain, which means it's relatively easy. The goal is to find a single entry into the chain, which can be called in a
-loop to get things done. Since the fakesrc is DECOUPLED, and we'd be messing with the source pad, it has lower
-priority than a DECOUPLED sink pad, so the fakesrc's sink pad is the ideal entry into the chain. This can be
-simplified into saying that the fakesink is the entry.
-
-In the end, the code to do this boils down to:
-
- buf = gst_pad_pull (fakesink->sinkpad);
- gst_pad_push (fakesink->sinkpad, buf);
-
-Because of the way things are no implemented for scheduling, turning it around and making the source the entry has no
-effect as far as the efficiency. That's because _get no longer directly calls gst_pad_push(), so we have to do it
-outside. No big deal, it boils down to the same thing I think, modulo a cache-line of stack (i.e. one or two fewer
-this way).
-
-If we put an identity in the middle:
-
-fakesrc -> identity -> fakesink
-
-then we have the same thing, except that there's now an element that isn't DECOUPLED, so it gets higher priority. That
-means the identity is now the entry, and when we push the buffer into its chain function, the fakesink gets called.
-
-Now, we can make this much more complex, with the following elementary echo meta-filter:
-
- |=====| -> delay1 -> |=====|
- | | | |
--> queue -> | tee | -> delay2 -> | mix | -> queue ->
- | | | |
- |=====| -> delay3 -> |=====|
-
-The tee takes a buffer in and spits three out, delay shifts the timestamps around and possibly reframes things to be
-friendly. mix takes the three buffers and simply sums them (they're all audio). The tee element takes one buffer in
-and promptly spits three out, one after another. Delay takes an element and immediately spits out a buffer (it
-zero-pads at the beginning [the duration of the delay] for the sake of argument). Mix in this case is chained, but
-assumes that buffer will arrive in order. On the last chain, it does a push of the newly mixed audio buffer.
-
-The queues are both DECOUPLED, so they have lower weight. That leaves a bunch of other elements sitting there ripe for
-entry. But if we were to take delay1, what would happen? Well we can't, since there's no _get function on the tee's
-src pads.
-
-This just re-enforces the idea that the left-most (closest to the source, for you right-to-left people) element should
-get to be the entry. But what if we have multiple left-most elements?:
-
--> queue -> eq1 -> |=====|
- | mix | -> queue
--> queue -> eq2 -> |=====|
-
-If eq1 is the only entry, we call pull on the queue, then chain over to mix. Mix then doesn't do anything with it,
-since it's waiting for another buffer before doing anything. That means we have to do the same with eq2, and have it
-chain to mix, at which point mix will do its magic and chain out to the right-hand side. Figure out to actually use
-both entries is hard, because the idea at this point is that there's only a single entry to a chain.
-
-Does this mean that we should make mix a DECOUPLED element? That would fix it to some extent, giving us three chains
-in the above case. Each eq chain would be driven by the eq element, pulling from the queue and pushing into the mixer.
-The mixer -> queue chain is problematic, because there is no possibly entry. The mixer side has no _get function
-(since the push always happens upon the receipt of a buffer from the second sink pad), which means that those two
-pads have no possible entrance.
-
-Cothreads make this case much easier, since the mix element would drive things, forcing the eq elements to pull and
-process buffers in order as needed. It may be that the best option in the case where there are any multi-sinkpad
-elements is to turn them into cothreads.
-
-
-Now, on to cothreaded cases. The simplest possible is to turn all the elements into cothreads. I may punt on this and
-do just that for the moment, but there's still the question of what to do at the ends of the chain, where the DECOUPLED
-elements are. The easiest is to simply always make then chained, so there's never any worry about who owns the
-cothread context for the element, simply because there never will be one.
-
-fakesrc -> queue -> @identity -> fakesink
-
-We just set it up so both ends of the queue are chained, and all is well.
diff --git a/docs/random/omega/sched/walkthrough-72 b/docs/random/omega/sched/walkthrough-72
deleted file mode 100644
index 6ea350485f..0000000000
--- a/docs/random/omega/sched/walkthrough-72
+++ /dev/null
@@ -1,39 +0,0 @@
-72)
-
- [-pipeline---------------------------------------------------------------------------------------------]
- ! [-bin-----------------------------] [-thread---------] !
- ! ! [--------] [---------] ! [------] [---------] [------] ! [--------] ! !
- ! ! !faksesrc! !identity1! ! !queue1! !identity2! !queue2! ! !fakesink! ! !
- ! ! ! src --- sink * src --- sink n src -- sink src -- sink src -- sink ! ! !
- ! ! [--------] [---------] ! [------] [---------] [------] ! [--------] ! !
- ! [---------------------------------] [----------------] !
- [------------------------------------------------------------------------------------------------------]
-
-
------
-Ideally, you'd end up with the following sub-pipelines
-
-pipeline:
-fakesrc -> identity1 -> queue1
-queue1 -> identity2 -> queue2
-
-thread:
-queue2 -> fakesink
-
-
-They'd be scheduled as following:
-
-fakesrc: passive chained, pulled by identity1
-identity1: loopfunc cothreaded, ENTRY
-queue1:sink: passive chained, pushed by identity1
-
-queue1:src: _get-based iteration, ENTRY
-identity2: chained by queue1
-queue2:src: passively chained, pushed by identity2
-
-queue2:sink: passively chained, pulled by fakesink
-fakesink: loopfunc cothreaded, ENTRY
-
-
------
-Most likely, we'd end up with the following
diff --git a/docs/random/omega/sched2 b/docs/random/omega/sched2
deleted file mode 100644
index 3bac20ac70..0000000000
--- a/docs/random/omega/sched2
+++ /dev/null
@@ -1,13 +0,0 @@
-Currently, when an element wants to push or pull, they call gst_pad_*. These functions then decide what
-to do based on the selection of pointers in the pad. There are at least 2 options for each, plus some
-header stuff to do various checking of flags. Unfortunately, because of the selection of pointers in
-the pad, there is at least one case where scheduling has to be tricked, by providing a pointer to a
-function with no body.
-
-What I propose is that these functions be replaced with macros that call a function pointer directly.
-The default functions (provided by GstPad) would be capable of chaining, that's about it. When a
-schedule is formed, these get replaced with more specific functions, provided by GstBin or a subclass.
-
-In the chain case, the pad_push_func might even be replaced brute force with the chain function, since
-they have the same prototype. In the cothreaded case, various functions would provide the ability to
-switch, deal with bufpens, etc.
diff --git a/docs/random/omega/scheduling b/docs/random/omega/scheduling
deleted file mode 100644
index 8da58f0758..0000000000
--- a/docs/random/omega/scheduling
+++ /dev/null
@@ -1,66 +0,0 @@
-STATUS: pull on srcpads is outdated, they use _get
---------------------------------------------------
-
-0) definitions:
-
-All pads without further specifiers are assumed to belong to the element
-in question. The pad's peer is always denoted with ->peer. If there's
-question, pads will be prefixed with self-> and other-> as necessary.
-
-All elements in this document have at most one source and one sink pad,
-called srcpad and sinkpad. Multi-pad cases are supposed to be simple
-extrapolations except in a couple strange cases, to be covered elsewhere.
-
-
-1) loop functions:
-
-A loop function will call gst_pad_pull(sinkpad), do something, and call
-gst_pad_push(srcpad).
-
-gst_pad_pull first checks to see if there's a buffer in the pen. If not,
-it calls that pad's pullfunc handler, passing it the peer pad. When that
-finishes, we check again and return the buffer. If no buffer, we squawk.
-The pullfunc handler simply causes a cothread switch to the peer pad's
-context;
-
-gst_pad_push places the buffer in the peer pad's pen, and calls the local
-pad's pushfunc. The pushfunc simply causes a switch to the peer pad's
-context.
-
-
-2) chain functions
-
-The loopfunc constructed around a chain function starts by finding all
-sink pads. For each sink pad, it calls gst_pad_pull. This causes a
-switch to the peer pad's context and a buffer to appear, which is
-returned. The pad is the handed off to the chain function, which
-presumably does some processing and calls gst_pad_push(), which causes a
-context switch to that pad's peer.
-
-
-
-3) source
-
-The loopfunc must repeatedly call the srcpad's pull
-
-
-
-
-
---------------------------------- ---------------------------------
- srcpad | | sinkpad
- | |
- GstPad *peer; | <-p-> | GstPad *peer;
- pointer to peer pad | | pointer to peer pad
- | |
- funcptr *pushfunc; | --s-> | funcptr *pushfunc;
- causes switch to peer ctx | | element uses buffer in pen,
- | |
- funcptr *pullfunc; | <-s-- | funcptr *pullfunc;
- element puts buffer in pen, | | causes switch to peer ctx
- calls gst_pad_push() | |
- | |
- funcptr *pullregion func | <-s-- | funcptr *pullregionfunc;
- element puts region in pen | | causes switch to peer ctx
- calls gst_pad_push() | |
---------------------------------- ---------------------------------
diff --git a/docs/random/omega/testing/Makefile b/docs/random/omega/testing/Makefile
deleted file mode 100644
index 691d5cd7f1..0000000000
--- a/docs/random/omega/testing/Makefile
+++ /dev/null
@@ -1,3 +0,0 @@
-gstobject: gstobject.c
- libtool gcc -o gstobject gstobject.c -I../../../.. ../../../../gst/libgst.la \
-`gtk-config --cflags --libs` `xml-config --cflags --libs`
diff --git a/docs/random/omega/testing/framework b/docs/random/omega/testing/framework
deleted file mode 100644
index 4ef4519fcb..0000000000
--- a/docs/random/omega/testing/framework
+++ /dev/null
@@ -1,144 +0,0 @@
-Construction
-Validation
-Testing
-
-Construction will generate some state
-Validation will ensure that everything is kosher
-Tests use combinations of the above to check things
-
-##### Example:
-parent = gst_object_get_parent(object);
-
-setup:
- create: new object
- action: object = gst_object_new();
- validation: object != NULL, object->parent == NULL, etc
- [cleanup: gst_object_destroy(object);]
- create: new parent
- action: parent = gst_object_new();
- validation: parent != NULL, parent->parent == NULL, etc
- [cleanup: gst_object_destroy(parent);]
- create: set object's parent
- precondition: object->parent == NULL
- action: gst_object_set_parent(object,parent);
- validation: object->parent = parent
-preconditions:
- nothing
-action:
- curparent = gst_element_get_parent(object);
-validation:
- curparent == object->parent
- curparent == parent
-cleanup:
- nothing
-
-
-##### Resulting code:
-
-///// setup
-// new object
-object = gst_object_new();
-ASSERT(object != NULL);
-ASSERT(object->parent == NULL);
-// new object
-parent = gst_object_new();
-ASSERT(parent != NULL);
-ASSERT(parent->parent == NULL);
-// set object parent
-ASSERT(object->parent == NULL);
-gst_object_set_parent(object,parent);
-ASSERT(object->parent != NULL);
-
-///// preconditions
-
-///// action
-curparent = gst_element_get_parent(object);
-
-///// validation
-ASSERT(object->parent == parent);
-
-///// cleanup
-gst_object_destroy(parent);
-gst_object_destroy(object);
-
-
-
-##### XML descriptions
-
-<construct name="new object">
- <variable>
- GstObject *object;
- </variabls>
- <action>
- object = gst_object_new();
- </action>
- <validation>
- <assert>
- object != NULL
- </assert>
- <assert>
- GST_IS_OBJECT(object)
- <assert>
- object->parent == NULL
- </assert>
- </validation>
- <cleanup>
- <validation>
- <assert>
- object != NULL
- </assert>
- <assert>
- GST_IS_OBJECT(object)
- </assert>
- </validation>
- <action>
- gst_object_destroy(object);
- </action>
- </cleanup>
-</construct>
-
-<construct name="set object parent">
- <variable>
- GstObject *object;
- </variable>
- <variable>
- GstObject *object;
- </variable>
- <precondition>
- <assert>
- object->parent == NULL
- </assert>
- </precondition>
- <action>
- gst_object_set_parent(object,parent);
- </action>
- <validation>
- <assert>
- object->parent == parent
- </assert>
- </validation>
-</construct>
-
-<test name="set object parent">
- <variable>
- GstObject *object;
- <variable>
- </variable>
- GstObject *parent;
- <variable>
- </variable>
- GstObject *curparent;
- </variable>
- <setup>
- object = gst_object_new();
- parent = gst_object_new();
- gst_object_set_parent(object,parent);
- </setup>
- <action>
- curparent = gst_element_get_parent(object);
- </action>
- <validation>
- curparent == object->parent
- curparent == parent
- </validation>
-</test>
diff --git a/docs/random/omega/testing/gstobject.c b/docs/random/omega/testing/gstobject.c
deleted file mode 100644
index d188f49c14..0000000000
--- a/docs/random/omega/testing/gstobject.c
+++ /dev/null
@@ -1,314 +0,0 @@
-#include <gst/gst.h>
-
-static gchar *_subject, *_category;
-static gint _testnum = 0;
-static gboolean _passed;
-static gint _total_tests = 0, _passed_tests = 0;
-static gint _random_size;
-
-void
-tabpad (gchar * str, gint width)
-{
- int i;
-
- for (i = 0; i < width - strlen (str); i++)
- fprintf (stderr, " ");
-}
-
-#define TEST_SUBJECT(subject) fprintf(stderr,"Subject: %s\n",subject),_subject = subject
-#define TEST_CATEGORY(category) fprintf(stderr,"\n\nCategory: %s\n",category)
-
-#define TEST(test) fprintf(stderr,"Test %d: %s...\n",_testnum,test),_passed = TRUE
-#define ASSERT(expr) G_STMT_START{ \
- fprintf(stderr,"\t%s:",#expr);tabpad(#expr,50); \
- if (!(expr)) { \
- fprintf(stderr,"FAILED\n"); \
- _passed = FALSE; \
- } else { \
- fprintf(stderr,"passed\n"); \
- } \
-}G_STMT_END;
-#define ENDTEST() G_STMT_START{ \
- _testnum++; \
- if (_passed) { \
- fprintf(stderr,"\tpassed.\n"); \
- _passed_tests++; \
- } else { \
- fprintf(stderr,"\tFAILED.\n"); \
- } \
- _total_tests++; \
-}G_STMT_END;
-
-void
-SETUP_RANDOM_SIZE (void *random, gint size)
-{
- int i;
-
- if (random)
- g_free (random);
- _random_size = size;
- random = g_malloc (_random_size);
- for (i = 0; i < _random_size; i++)
- ((unsigned char *) random)[i] = i;
-}
-
-#define SETUP_RANDOM(random,type) SETUP_RANDOM_SIZE(random,sizeof(type))
-
-gboolean
-RANDOM_OK (void *random)
-{
- int i;
-
- for (i = 0; i < _random_size; i++) {
- if (((unsigned char *) random)[i] != i) {
- SETUP_RANDOM_SIZE (random, _random_size);
- return FALSE;
- }
- }
- return TRUE;
-}
-
-int
-main (int argc, char *argv[])
-{
- GstObject *object;
- GstObject *parent;
- GstObject *newparent;
- GtkObject *gtkobject;
- GstObject *curparent;
-
- gst_init (&argc, &argv);
-
- TEST_SUBJECT ("GstObject");
-
-
- TEST_CATEGORY ("Creation");
-
- TEST ("create object");
- /* setup */
- /* action */
- object = gst_object_new ();
- /* assertions */
- ASSERT (object != NULL);
- ASSERT (GST_IS_OBJECT (object));
- /* cleanup */
- g_free (object);
- ENDTEST ();
-
-
- /* new category */
- TEST_CATEGORY ("Refcounting");
- /* category setup */
- object = gst_object_new ();
-
- TEST ("new object");
- /* setup */
- /* action */
- /* assertions */
- ASSERT (object->refcount == 1);
- ASSERT (GTK_OBJECT_FLOATING (object) == TRUE);
- /* cleanup */
- ENDTEST ();
-
- TEST ("increment refcount");
- /* setup */
- /* action */
- gst_object_ref (object);
- /* assertions */
- ASSERT (object->refcount == 2);
- ASSERT (GTK_OBJECT_FLOATING (object) == TRUE);
- /* cleanup */
- ENDTEST ();
-
- TEST ("sink object");
- /* setup */
- /* action */
- gst_object_sink (object);
- /* assertions */
- ASSERT (object->refcount == 1);
- ASSERT (GTK_OBJECT_FLOATING (object) == FALSE);
- /* cleanup */
- ENDTEST ();
-
- TEST ("increment refcount after sink");
- /* setup */
- /* action */
- gst_object_ref (object);
- /* assertions */
- ASSERT (object->refcount == 2);
- ASSERT (GTK_OBJECT_FLOATING (object) == FALSE);
- /* cleanup */
- ENDTEST ();
-
- TEST ("decrement refcount after sink");
- /* setup */
- /* action */
- gst_object_unref (object);
- /* assertions */
- ASSERT (object->refcount == 1);
- ASSERT (GTK_OBJECT_FLOATING (object) == FALSE);
- /* cleanup */
- ENDTEST ();
-
- /* category cleanup */
- g_free (object);
-
-
-
- /* new category */
- TEST_CATEGORY ("Parentage");
- /* category setup */
- object = gst_object_new ();
- parent = gst_object_new ();
- newparent = gst_object_new ();
- gtkobject = gtk_type_new (gtk_object_get_type ());
- /* category assertions */
- ASSERT (object != NULL);
- ASSERT (object->refcount == 1);
- ASSERT (object->parent == NULL);
- ASSERT (parent != NULL);
- ASSERT (newparent != NULL);
- ASSERT (gtkobject != NULL);
- ASSERT (!GST_IS_OBJECT (gtkobject));
-
- TEST ("gst_object_set_parent: null object");
- /* setup */
- /* action */
- gst_object_set_parent (NULL, NULL);
- /* assertions */
- ASSERT (object->parent == NULL);
- /* cleanup */
- ENDTEST ();
-
- TEST ("gst_object_set_parent: invalid object");
- /* setup */
- /* action */
- gst_object_set_parent ((GstObject *) gtkobject, NULL);
- /* assertions */
- ASSERT (object->parent == NULL);
- /* cleanup */
- ENDTEST ();
-
- TEST ("gst_object_set_parent: null parent");
- /* setup */
- /* action */
- gst_object_set_parent (object, NULL);
- /* assertions */
- ASSERT (object->parent == NULL);
- /* cleanup */
- ENDTEST ();
-
- TEST ("gst_object_set_parent: invalid parent");
- /* setup */
- /* action */
- gst_object_set_parent (object, (GstObject *) gtkobject);
- /* assertions */
- ASSERT (object->parent == NULL);
- /* cleanup */
- ENDTEST ();
-
- TEST ("gst_object_set_parent: valid object, parent is object");
- /* setup */
- /* action */
- gst_object_set_parent (object, object);
- /* assertions */
- ASSERT (object->parent == NULL);
- /* cleanup */
- ENDTEST ();
-
- TEST ("gst_object_set_parent: valid object and parent");
- /* setup */
- /* action */
- gst_object_set_parent (object, parent);
- /* assertions */
- ASSERT (object->parent == parent);
- /* cleanup */
- ENDTEST ();
-
- TEST ("gst_object_set_parent: parent already set");
- /* setup */
- /* action */
- gst_object_set_parent (object, newparent);
- /* assertions */
- ASSERT (object->parent != newparent);
- ASSERT (object->parent == parent);
- /* cleanup */
- g_free (object);
- ENDTEST ();
-
-
- TEST ("gst_object_get_parent: null object");
- /* setup */
- /* action */
- curparent = gst_object_get_parent (NULL);
- /* assertions */
- ASSERT (curparent == NULL);
- /* cleanup */
- ENDTEST ();
-
- TEST ("gst_object_get_parent: invalid object");
- /* setup */
- /* action */
- curparent = gst_object_get_parent ((GstObject *) gtkobject);
- /* assertions */
- ASSERT (curparent == NULL);
- /* cleanup */
- ENDTEST ();
-
- TEST ("gst_object_get_parent: no parent");
- /* setup */
- object = gst_object_new ();
- /* action */
- curparent = gst_object_get_parent (object);
- /* assertions */
- ASSERT (curparent == NULL);
- /* cleanup */
- ENDTEST ();
-
- TEST ("gst_object_get_parent: valid parent");
- /* setup */
- gst_object_set_parent (object, parent);
- /* action */
- curparent = gst_object_get_parent (object);
- /* assertions */
- ASSERT (curparent == parent);
- /* cleanup */
- g_free (object);
- ENDTEST ();
-
-
- TEST ("gst_object_unparent: null object");
- /* setup */
- /* action */
- gst_object_unparent (NULL);
- /* assertions */
- /* NONE - FIXME! */
- /* cleanup */
- ENDTEST ();
-
- TEST ("gst_object_unparent: invalid object");
- /* setup */
- /* action */
- gst_object_unparent ((GstObject *) gtkobject);
- /* assertions */
- /* NONE - FIXME! */
- /* cleanup */
- ENDTEST ();
-
- TEST ("gst_object_unparent: no parent");
- /* setup */
- object = gst_object_new ();
-
-
- /* category cleanup */
- g_free (object);
- g_free (parent);
- g_free (newparent);
- g_free (gtkobject);
-
-
-
- fprintf (stderr, "\n\nTotal tests:\t%d\n", _total_tests);
- fprintf (stderr, "Total passed:\t%d\n", _passed_tests);
- fprintf (stderr, "Total FAILED:\t%d\n", _total_tests - _passed_tests);
-}
diff --git a/docs/random/omega/testing/gstobject.txt b/docs/random/omega/testing/gstobject.txt
deleted file mode 100644
index 35f0396ed1..0000000000
--- a/docs/random/omega/testing/gstobject.txt
+++ /dev/null
@@ -1,95 +0,0 @@
-Subject: GstObject
-
-Areas to test
-=============
-
-Creation
-Refcounting
-Destruction
-Flags
-Locking
-Parentage
-Path string
-
-
-Tests
-=====
-
-Creation
---------
-Create an object
- Does it return !NULL
- GST_IS_OBJECT() ?
-
-Refcounting
------------
-Create new object
- object->refcount == 1, GTK_OBJECT_FLOATING(object) == TRUE
-Increment refcount
- object->refcount == 2
-Sink object
- object->refcount == 1, GTK_OBJECT_FLOATING(object) == FALSE
-Increment refcount
- object->refcount == 2
-Decrement refcount
- object->refcount == 1
-
-Destruction
------------
-???
-
-Flags (start with new object)
------
-Create new object
- Verify that all flags are unset
-Set a flag
- Verify it's set
-Unset a flag
- Verify it's not set
-
-Locking (start with new object)
--------
-Lock an object
- Try to lock, get false
-
-Parentage (start with new object, check refcount == 1)
----------
-gst_object_set_parent: (start with new parent object)
- Pass NULL...
- Pass !NULL, but not Object...
- Pass NULL parent...
- Pass !NULL parent, but not Object...
- Pass valid Object, and parent == object
- object->refcount == 1
- GTK_OBJECT_FLOATING(object) == TRUE
- object->parent == NULL
- Pass valid Object
- object->refcount == 1
- GTK_OBJECT_FLOATING(object) == FALSE
- object->parent == parent
- The "parent_set" signal should fire with the object and parent as args
- Pass Object with parent already set
- object->parent should not equal new parent
- object->refcount == 1
-gst_object_get_parent:
- Pass NULL...
- Pass !NULL, not Object...
- Pass valid object with no parent
- Get NULL
- Pass valid object with parent
- Get parent pointer
-gst_object_unparent:
- Pass NULL, with no parent
- no effect
- Pass !NULL, not Object, NULL parent
- pointer not mangled
- Pass valid object, with no parent
- object->parent == NULL
- object->refcount = 1
- Pass NULL, with valid parent
- no effect
- Pass !NULL, not Object, with valid object as parent
- pointer not mangled
- Pass valid object, with valid parent
- object->parent == NULL
- object->refcount == 0
diff --git a/docs/random/omega/type-properties b/docs/random/omega/type-properties
deleted file mode 100644
index b216470ca0..0000000000
--- a/docs/random/omega/type-properties
+++ /dev/null
@@ -1,77 +0,0 @@
-OUTDATED
---------
-
-
-A type properties system might look like following:
-
-
-1) Type definition includes the properties and their ranges:
-
-audio/mp3
- layer: 1 - 3
- bitrate: 8 - 320
-
-audio/raw
- format: bitfield (using asound.h definitions)
- depth: 8 - 32
- rate: 4000 - 96000
- channels: 1 - n
- interleave: boolean
-
-video/raw
- format: 32-bit FOURCC
- bpp: 1 - 32
- width: 1 - n
- height: 1 - n
- framerate: 32-bit float
-
-etc.
-
-
-2) An element can specify what subtypes it can deal with by creating a list of property tables:
-
-mpg123: audio/mp3
- layer: 1 - 3
- bitrate: 8 - 320
-
-osssink:
- format: S8, S16, etc.
- depth: 8 - 16
- rate: 8000 - 48000
- channels: 1 - 2
- interleave: true
-
-And you could list several of these, so for instance if the card only supports 8-bit at up to 22KHz in
-mono, you can remove S8 from the above list and add a second entry:
-
-osssink:
- format: S8
- depth: 8
- rate: 8000 - 22050
- channels: 1
- interleave: false (irrelevant)
-
-The obvious problem with these examples is that the rate isn't really 8000 - 48000, it's 8000, 11025,
-16000, 22050, 44100, and 48000. However, we may be able to leave these to pad connect time.
-
-
-
-
-
-struct _type_definition {
- char *mime_type;
-
- ....
-
- GData *properties;
-}
-
-gst_type_add_property_int(_type *type,gchar *propname,int min,int max) {
- struct _type_prop_int prop_int;
- GQuark quark = g_quark_from_string(propname);
-
- prop_int->id = quark;
- prop_int->min = min;
- prop_int->max = max;
- g_datalist_id_set_data(type->properties,quark,&prop_int);
-}
diff --git a/docs/random/phonon-gst b/docs/random/phonon-gst
deleted file mode 100644
index 9f9b93479f..0000000000
--- a/docs/random/phonon-gst
+++ /dev/null
@@ -1,269 +0,0 @@
-Phonon backend
---------------
-
- The phonon design is based around forming graphs using 3 basic components:
-
- - a source component that generates raw audio/video/subtitle data, aka
- MediaObject.
- - an effect component that applies effects to raw audio/video known as
- AudioPath/VideoPath respectively. Subtitles are routed to a VideoPath
- - output components that render audio or video called AudioOutput and
- VideoOutput.
-
- there is also a special input object that allows for feeding raw data in the
- pipeline and specialized sinks to retrieve audio samples and video frames from
- the pipeline.
-
- A typical graph or a source that produces an audio and a video stream that
- need to be played. The VideoPath and AudioPath typically contain no filters
- in this case:
-
- +----+ +---
- | FX | | ...
- +----+ +---
- V V
- +-----------+ +-------------+
- ----->| VideoPath |----->| VideoOutput |
- | +-----------+ +-------------+
- +-------------+
- | MediaObject |
- +-------------+
- | +-----------+ +-------------+
- +---->| AudioPath |----->| AudioOutput |
- +-----------+ +-------------+
- ^ ^
- +----+ +---
- | FX | | ...
- +-+--+ +---
-
- - This is very similar to a regular gstreamer playback pipeline.
-
- A typical graph of playing and crosfading two sources:
-
- +--------+
- | volume |
- +--------+
- V
- +-------------+ +-----------+
- | MediaObject |--->| AudioPath |\
- +-------------+ +-----------+ \ +-------------+
- ---->| AudioOutput |
- +-------------+ +-----------+ / +-------------+
- | MediaObject |--->| AudioPath |/
- +-------------+ +-----------+
- ^
- +--------+
- | volume |
- +--------+
-
- - As soon as two audio paths are connected to one sink, the input signals are
- mixed before sending them to the sink. The mixing is typically done in the
- audio sink by an element such as adder.
-
- Other types of graphs are possible too:
-
- +-----------+
- /| AudioPath |\
- +-------------+ / +-----------+ \ +-------------+
- | MediaObject |-- ---->| AudioOutput |
- +-------------+ \ +-----------+ / +-------------+
- \| AudioPath |/
- +-----------+
-
- - This graph sends the same out data to 2 effect filter graphs and then mixes
- it to an audio output. The splitting of the graph typically happens with a
- tee element after the media object.
-
-
-Questions
----------
-
- 1) do the following chains run
-
- - synchronized with a shared clock?
-
- +-------------+ +-----------+ +-------------+
- | MediaObject |--->| AudioPath |--->| AudioOutput |
- +-------------+ +-----------+ +-------------+
-
- +-------------+ +-----------+ +-------------+
- | MediaObject |--->| VideoPath |--->| VideoOutput |
- +-------------+ +-----------+ +-------------+
-
- - no API to set both MediaObjects atomically to play so it is assumed that
- the playback starts and follows the rate of the global clock as soon as
- the MediaObject is set to play. This makes unconnected chains run as if
- they were in different GstPipelines.
-
- 2) Threading:
-
- - Can signals be emitted from any thread?
- - what operations are permitted from a signal handler?
-
- 3) Error reporting
-
- - How does error reporting work?
- * an audio/video device/port is busy.
- * a fatal decoding error occurred.
- * a media type is not supported
-
-
-General
--------
-
- - Setting up KDE and Phonon build environment
- - Testing, identifying test applications, building test cases
- - Asking questions to Phonon maintainers/designers
-
-Essential classes
------------------
-
-These classes are essential to implement a backend and should be implemented
-first.
-
-Phonon::BackendCapabilities
-
- Mostly exposes features in the registry like available decoders and effects.
-
-Phonon::Factory
-
- Entry point for the GStreamer backend. Provides methods to create instances of
- object from our backed.
-
-
-Simple playback
----------------
-
-The following classes need to be implemented in order to have simple playback
-capabilities from the backend.
-
-Phonon::AudioOutput
-
- - Wrapper around audiosinks. Also needs provision for rate and format
- conversions.
- - Mixing capabilities in the case when 2 audio paths are routed to it.
-
- Notes:
-
- * is the volume related to the device or to the connection to the device.
-
-Phonon::VideoWidget
-
- - Wrapper around videosinks. Also needs provision for colorspace and size
- conversions. Extends QWidget and probably needs to hook into the XOverlay
- stuff to draw in the QT widget. Supports fullscreen mode with a switch.
-
- - Needs mixing capabilities in the case when 2 video paths are routed to it.
-
-Phonon::AbstractMediaProducer
-
- - contains stream selection
- - play/pause/stop
- - seeking
- - periodically performs tick callbacks.
-
-Phonon::MediaObject:
-
- - The object that decodes the media into raw audio/video/subtitle.
- This object will use the GStreamer decodebin element to perform the
- typefinding and decoding.
-
-Phonon::AudioPath/Phonon::VideoPath
-
- - Simple container for audio/video effect plugins.
- - Handles adding/removing of effects, making sure that the streaming is not
- interrupted and the formats are all compatible.
-
-
-Effect support
---------------
-
-Phonon::Visualization
-
- Connects an AudioPath to a VideoWidget and allows for selection of a
- visualisation plugin.
-
-Phonon::AudioEffect/Phonon::VideoEffect
-
- Base classes
-
-Phonon::VolumeFaderEffect
-
- Allows fade-in and fade-out with a configurable curve and time. Needs
- GstController.
-
-Phonon::BrightnessControl
-
- Controls the brightness of video.
-
-
-Playlist support
-----------------
-
-Phonon::MediaQueue:
-
- ?? don't know yet where this fits in.
-
-
-Capture
--------
-
-Phonon::AvCapture
-
- Synchronized audio and video capture.
-
-Phonon::AudioWriter
-
- Compress audio.
-
-
-Advanced features
------------------
-
-Phonon::ByteStream
-
- Feed raw data into the pipeline. Used for streaming network access.
-
- Implementation:
-
- Possibly a specialized source element connected to a decodebin.
-
- Notes:
-
- * Phonon::ByteStream::writeData
- - can it block?
-
- * Phonon::ByteStream::setStreamSeekable
- - If called before starting the ByteStream, decodebin might operate in pull
- based mode when supported. Else the source is activated in push mode.
- - If called after starting ByteStream, the Phonon::ByteStream::seekStream
- signal can be called for push-based seekable streams.
-
- * Can the signals be emitted from a streaming thread?
-
-
-Phonon::AudioDataOutput/Phonon::VideoDataOutput/
-
- Receive raw audio/video data from the pipeline. Used to allow applications to
- deal with the raw data themselves.
-
- Implementation:
-
- Possibly a specialized sink element.
-
- Notes :
-
- * Phonon::AudioDataOutput::dataReady
- - can this be emitted from the streaming threads?
-
- * Phonon::AudioDataOutput::endOfMedia
- - can this be emitted from the streaming threads?
- - We need to grab this EOS message synchronously from the bus.
- - should be emitted _before_ sending the last dataReady. This means we need
- to cache at least one dataReady.
-
- * Phonon::AudioDataOutput::setDataSize
- - can this be a _suggested_ data size or does every callback need to be of
- this size?
-
-
diff --git a/docs/random/plugins b/docs/random/plugins
deleted file mode 100644
index ad98aa2987..0000000000
--- a/docs/random/plugins
+++ /dev/null
@@ -1,93 +0,0 @@
-SOMEWHAT OUTDATED
------------------
-
-
-We describe how the plugin system works.
-
-Plugins
--------
-
-plugins are basically shared libraries with a plugin_init
-function.
-
-they provide the GStreamer core library with the following
-information:
-
- - element factories
- - type factories
- - metadata factories?
-
-ElementFactory
---------------
-
-element factories provide the core library with elements (duh)
-
-an element factory has the following information:
-
- - a unique name for the element factory
- - strings describing the element (name, description, copyright,...)
- - a description of the media types it accepts (as capabilities)
- - a description of the media types it outputs (as capabilities)
-
-
-TypeFactory
------------
-
-has the following properties:
-
- - a mime type
- - file extensions that may provide a hint for this type
- - a function to detect this type
- - a string to detect this type (file(1) like)
-
-
-XML registry
-------------
-
-The complete plugin tree will be exported into an XML description so
-that the definitions of the factories can be obtained without having
-to load and plugin_init the plugins.
-
-Loading of plugins
-------------------
-
-at some point, the plugin will need to be read into memory before
-any elements it provides can be used. the XML description of the
-loaded plugin will need to be updated.
-
-We will have the following methods for (implicitly) loading plugins:
-
- gst_plugin_load_all()
-
- remove the complete XML tree and read all the plugins in the
- paths. The plugin_init method will be called and the XML tree
- will be rebuild in memory.
-
- gst_plugin_load (name)
-
- The plugin will be located in the tree and if it already loaded, the
- function returns. If it is not loaded, the XML entry is removed and
- the plugin is loaded. The plugin_init is called so that the XML tree
- is reconstructed.
-
- gst_elementfactory_create (factory, name);
-
- The plugin providing the element will be located, if the plugin is
- already loaded, an element with the given name is created.
- if the plugin is not loaded, gst_plugin_load is called.
- the loaded factory is located again and the element is created.
-
- The typefind function is called
-
- When the plugin description is read from the XML file, the typefind
- function is hooked up to a dummy typefind function. The dummy typefind
- function will locate the plugin it belongs to and call gst_plugin_load.
- after the loading of the plugin, the real typefind function is called.
-
-
-
-
-
-
-
-
diff --git a/docs/random/plugins.dia b/docs/random/plugins.dia
deleted file mode 100644
index e7c42060b2..0000000000
--- a/docs/random/plugins.dia
+++ /dev/null
@@ -1,941 +0,0 @@
-<?xml version="1.0"?>
-<diagram xmlns:dia="http://www.lysator.liu.se/~alla/dia/">
- <diagramdata>
- <attribute name="background">
- <color val="#ffffff"/>
- </attribute>
- <attribute name="paper">
- <composite type="paper">
- <attribute name="name">
- <string>#A4#</string>
- </attribute>
- <attribute name="tmargin">
- <real val="2.82"/>
- </attribute>
- <attribute name="bmargin">
- <real val="2.82"/>
- </attribute>
- <attribute name="lmargin">
- <real val="2.82"/>
- </attribute>
- <attribute name="rmargin">
- <real val="2.82"/>
- </attribute>
- <attribute name="is_portrait">
- <boolean val="true"/>
- </attribute>
- <attribute name="scaling">
- <real val="1"/>
- </attribute>
- <attribute name="fitto">
- <boolean val="false"/>
- </attribute>
- </composite>
- </attribute>
- <attribute name="grid">
- <composite type="grid">
- <attribute name="width_x">
- <real val="1"/>
- </attribute>
- <attribute name="width_y">
- <real val="1"/>
- </attribute>
- <attribute name="visible_x">
- <int val="1"/>
- </attribute>
- <attribute name="visible_y">
- <int val="1"/>
- </attribute>
- </composite>
- </attribute>
- <attribute name="guides">
- <composite type="guides">
- <attribute name="hguides"/>
- <attribute name="vguides"/>
- </composite>
- </attribute>
- </diagramdata>
- <layer name="Background" visible="true">
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- <real val="11.4656"/>
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- <attribute name="name">
- <string>#GstPlugin#</string>
- </attribute>
- <attribute name="stereotype">
- <string/>
- </attribute>
- <attribute name="abstract">
- <boolean val="false"/>
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- <attribute name="suppress_attributes">
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- <attribute name="visible_operations">
- <boolean val="false"/>
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- <attribute name="name">
- <string>#name#</string>
- </attribute>
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- <string>#gchar *#</string>
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- <attribute name="value">
- <string/>
- </attribute>
- <attribute name="visibility">
- <enum val="0"/>
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- <attribute name="class_scope">
- <boolean val="false"/>
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- <composite type="umlattribute">
- <attribute name="name">
- <string>#longname#</string>
- </attribute>
- <attribute name="type">
- <string>#char *#</string>
- </attribute>
- <attribute name="value">
- <string/>
- </attribute>
- <attribute name="visibility">
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- <attribute name="class_scope">
- <boolean val="false"/>
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- <composite type="umlattribute">
- <attribute name="name">
- <string>#filename#</string>
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- <attribute name="type">
- <string>#gchar *#</string>
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- <string/>
- </attribute>
- <attribute name="visibility">
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- <attribute name="class_scope">
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- <composite type="umlattribute">
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- <string/>
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- <attribute name="visibility">
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- <composite type="umlattribute">
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- <string>#elements#</string>
- </attribute>
- <attribute name="type">
- <string>#GList *#</string>
- </attribute>
- <attribute name="value">
- <string/>
- </attribute>
- <attribute name="visibility">
- <enum val="0"/>
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- <attribute name="class_scope">
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- <composite type="umlattribute">
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- <string>#types#</string>
- </attribute>
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- <string>#GList *#</string>
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- <attribute name="value">
- <string/>
- </attribute>
- <attribute name="visibility">
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- <attribute name="class_scope">
- <boolean val="false"/>
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- <composite type="umlattribute">
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- <string>#autopluggers#</string>
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- <string>#GList *#</string>
- </attribute>
- <attribute name="value">
- <string/>
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- <attribute name="template">
- <boolean val="false"/>
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- <attribute name="templates"/>
- </object>
- <object type="UML - Class" version="0" id="O1">
- <attribute name="obj_pos">
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- </attribute>
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- <attribute name="name">
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- </attribute>
- <attribute name="stereotype">
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- <attribute name="abstract">
- <boolean val="false"/>
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- <attribute name="suppress_attributes">
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- <attribute name="visibility">
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- <boolean val="false"/>
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- </composite>
- <composite type="umlattribute">
- <attribute name="name">
- <string>#type#</string>
- </attribute>
- <attribute name="type">
- <string>#GtkType#</string>
- </attribute>
- <attribute name="value">
- <string/>
- </attribute>
- <attribute name="visibility">
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- <string>#details#</string>
- </attribute>
- <attribute name="type">
- <string>#GstElementDetails *#</string>
- </attribute>
- <attribute name="value">
- <string/>
- </attribute>
- <attribute name="visibility">
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- <attribute name="abstract">
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- <attribute name="class_scope">
- <boolean val="false"/>
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- <composite type="umlattribute">
- <attribute name="name">
- <string>#padtemplates#</string>
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- <attribute name="templates"/>
- </object>
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- <point val="12.3156,9.25"/>
- </attribute>
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- <rectangle val="10.8656,7.8;18.45,11.15"/>
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- <attribute name="direction">
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- <string/>
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- <string>#*#</string>
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- <attribute name="aggregate">
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- </attribute>
- </composite>
- </attribute>
- <connections>
- <connection handle="0" to="O0" connection="17"/>
- <connection handle="1" to="O1" connection="3"/>
- </connections>
- </object>
- <object type="UML - Class" version="0" id="O3">
- <attribute name="obj_pos">
- <point val="21,2"/>
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- <rectangle val="20.95,1.95;34.9396,6.85"/>
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- <attribute name="elem_height">
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- <attribute name="name">
- <string>#GstPadTemplate#</string>
- </attribute>
- <attribute name="stereotype">
- <string/>
- </attribute>
- <attribute name="abstract">
- <boolean val="false"/>
- </attribute>
- <attribute name="suppress_attributes">
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- <attribute name="visible_operations">
- <boolean val="false"/>
- </attribute>
- <attribute name="attributes">
- <composite type="umlattribute">
- <attribute name="name">
- <string>#name_template#</string>
- </attribute>
- <attribute name="type">
- <string>#gchar *#</string>
- </attribute>
- <attribute name="value">
- <string/>
- </attribute>
- <attribute name="visibility">
- <enum val="0"/>
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- <attribute name="class_scope">
- <boolean val="false"/>
- </attribute>
- </composite>
- <composite type="umlattribute">
- <attribute name="name">
- <string>#direction#</string>
- </attribute>
- <attribute name="type">
- <string>#GstPadDirection#</string>
- </attribute>
- <attribute name="value">
- <string/>
- </attribute>
- <attribute name="visibility">
- <enum val="0"/>
- </attribute>
- <attribute name="abstract">
- <boolean val="false"/>
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- <attribute name="class_scope">
- <boolean val="false"/>
- </attribute>
- </composite>
- <composite type="umlattribute">
- <attribute name="name">
- <string>#presence#</string>
- </attribute>
- <attribute name="type">
- <string>#GstPadPresence#</string>
- </attribute>
- <attribute name="value">
- <string/>
- </attribute>
- <attribute name="visibility">
- <enum val="0"/>
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- <composite type="umlattribute">
- <attribute name="name">
- <string>#caps#</string>
- </attribute>
- <attribute name="type">
- <string>#GstCaps *#</string>
- </attribute>
- <attribute name="value">
- <string/>
- </attribute>
- <attribute name="visibility">
- <enum val="0"/>
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- <attribute name="abstract">
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- </attribute>
- <attribute name="operations"/>
- <attribute name="template">
- <boolean val="false"/>
- </attribute>
- <attribute name="templates"/>
- </object>
- <object type="UML - Association" version="0" id="O4">
- <attribute name="obj_pos">
- <point val="31.8592,13.3"/>
- </attribute>
- <attribute name="obj_bb">
- <rectangle val="30.4092,1.25;38.45,14.75"/>
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- <string>#1#</string>
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- </attribute>
- </composite>
- <composite>
- <attribute name="role">
- <string/>
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- <attribute name="multiplicity">
- <string>#*#</string>
- </attribute>
- <attribute name="arrow">
- <boolean val="false"/>
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- <attribute name="aggregate">
- <enum val="0"/>
- </attribute>
- </composite>
- </attribute>
- <connections>
- <connection handle="0" to="O1" connection="15"/>
- <connection handle="1" to="O3" connection="4"/>
- </connections>
- </object>
- <object type="UML - Class" version="0" id="O5">
- <attribute name="obj_pos">
- <point val="17.1,14.85"/>
- </attribute>
- <attribute name="obj_bb">
- <rectangle val="17.05,14.8;32.494,18.9"/>
- </attribute>
- <attribute name="elem_corner">
- <point val="17.1,14.85"/>
- </attribute>
- <attribute name="elem_width">
- <real val="15.344"/>
- </attribute>
- <attribute name="elem_height">
- <real val="4"/>
- </attribute>
- <attribute name="name">
- <string>#GstTypeFactory#</string>
- </attribute>
- <attribute name="stereotype">
- <string/>
- </attribute>
- <attribute name="abstract">
- <boolean val="false"/>
- </attribute>
- <attribute name="suppress_attributes">
- <boolean val="false"/>
- </attribute>
- <attribute name="suppress_operations">
- <boolean val="false"/>
- </attribute>
- <attribute name="visible_attributes">
- <boolean val="true"/>
- </attribute>
- <attribute name="visible_operations">
- <boolean val="false"/>
- </attribute>
- <attribute name="attributes">
- <composite type="umlattribute">
- <attribute name="name">
- <string>#mime#</string>
- </attribute>
- <attribute name="type">
- <string>#gchar *#</string>
- </attribute>
- <attribute name="value">
- <string/>
- </attribute>
- <attribute name="visibility">
- <enum val="0"/>
- </attribute>
- <attribute name="abstract">
- <boolean val="false"/>
- </attribute>
- <attribute name="class_scope">
- <boolean val="false"/>
- </attribute>
- </composite>
- <composite type="umlattribute">
- <attribute name="name">
- <string>#exts#</string>
- </attribute>
- <attribute name="type">
- <string>#gchar *#</string>
- </attribute>
- <attribute name="value">
- <string/>
- </attribute>
- <attribute name="visibility">
- <enum val="0"/>
- </attribute>
- <attribute name="abstract">
- <boolean val="false"/>
- </attribute>
- <attribute name="class_scope">
- <boolean val="false"/>
- </attribute>
- </composite>
- <composite type="umlattribute">
- <attribute name="name">
- <string>#typefindfunc#</string>
- </attribute>
- <attribute name="type">
- <string>#GstTypeFindFunc#</string>
- </attribute>
- <attribute name="value">
- <string/>
- </attribute>
- <attribute name="visibility">
- <enum val="0"/>
- </attribute>
- <attribute name="abstract">
- <boolean val="false"/>
- </attribute>
- <attribute name="class_scope">
- <boolean val="false"/>
- </attribute>
- </composite>
- </attribute>
- <attribute name="operations"/>
- <attribute name="template">
- <boolean val="false"/>
- </attribute>
- <attribute name="templates"/>
- </object>
- <object type="UML - Association" version="0" id="O6">
- <attribute name="obj_pos">
- <point val="12.3156,10.05"/>
- </attribute>
- <attribute name="obj_bb">
- <rectangle val="10.8656,8.6;18.55,17"/>
- </attribute>
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- <point val="15.7,10.05"/>
- <point val="15.7,15.55"/>
- <point val="17.1,15.55"/>
- </attribute>
- <attribute name="orth_orient">
- <enum val="0"/>
- <enum val="1"/>
- <enum val="0"/>
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- <attribute name="name">
- <string/>
- </attribute>
- <attribute name="direction">
- <enum val="0"/>
- </attribute>
- <attribute name="ends">
- <composite>
- <attribute name="role">
- <string/>
- </attribute>
- <attribute name="multiplicity">
- <string>#1#</string>
- </attribute>
- <attribute name="arrow">
- <boolean val="false"/>
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- <attribute name="aggregate">
- <enum val="2"/>
- </attribute>
- </composite>
- <composite>
- <attribute name="role">
- <string/>
- </attribute>
- <attribute name="multiplicity">
- <string>#*#</string>
- </attribute>
- <attribute name="arrow">
- <boolean val="false"/>
- </attribute>
- <attribute name="aggregate">
- <enum val="0"/>
- </attribute>
- </composite>
- </attribute>
- <connections>
- <connection handle="0" to="O0" connection="19"/>
- <connection handle="1" to="O5" connection="3"/>
- </connections>
- </object>
- <object type="UML - Class" version="0" id="O7">
- <attribute name="obj_pos">
- <point val="17.065,19.975"/>
- </attribute>
- <attribute name="obj_bb">
- <rectangle val="17.015,19.925;27.5201,24.025"/>
- </attribute>
- <attribute name="elem_corner">
- <point val="17.065,19.975"/>
- </attribute>
- <attribute name="elem_width">
- <real val="10.4051"/>
- </attribute>
- <attribute name="elem_height">
- <real val="4"/>
- </attribute>
- <attribute name="name">
- <string>#GstAutoplugFactory#</string>
- </attribute>
- <attribute name="stereotype">
- <string/>
- </attribute>
- <attribute name="abstract">
- <boolean val="false"/>
- </attribute>
- <attribute name="suppress_attributes">
- <boolean val="false"/>
- </attribute>
- <attribute name="suppress_operations">
- <boolean val="false"/>
- </attribute>
- <attribute name="visible_attributes">
- <boolean val="true"/>
- </attribute>
- <attribute name="visible_operations">
- <boolean val="false"/>
- </attribute>
- <attribute name="attributes">
- <composite type="umlattribute">
- <attribute name="name">
- <string>#name#</string>
- </attribute>
- <attribute name="type">
- <string>#gchar *#</string>
- </attribute>
- <attribute name="value">
- <string/>
- </attribute>
- <attribute name="visibility">
- <enum val="0"/>
- </attribute>
- <attribute name="abstract">
- <boolean val="false"/>
- </attribute>
- <attribute name="class_scope">
- <boolean val="false"/>
- </attribute>
- </composite>
- <composite type="umlattribute">
- <attribute name="name">
- <string>#longdesc#</string>
- </attribute>
- <attribute name="type">
- <string>#gchar *#</string>
- </attribute>
- <attribute name="value">
- <string/>
- </attribute>
- <attribute name="visibility">
- <enum val="0"/>
- </attribute>
- <attribute name="abstract">
- <boolean val="false"/>
- </attribute>
- <attribute name="class_scope">
- <boolean val="false"/>
- </attribute>
- </composite>
- <composite type="umlattribute">
- <attribute name="name">
- <string>#type#</string>
- </attribute>
- <attribute name="type">
- <string>#GtkType#</string>
- </attribute>
- <attribute name="value">
- <string/>
- </attribute>
- <attribute name="visibility">
- <enum val="0"/>
- </attribute>
- <attribute name="abstract">
- <boolean val="false"/>
- </attribute>
- <attribute name="class_scope">
- <boolean val="false"/>
- </attribute>
- </composite>
- </attribute>
- <attribute name="operations"/>
- <attribute name="template">
- <boolean val="false"/>
- </attribute>
- <attribute name="templates"/>
- </object>
- <object type="UML - Association" version="0" id="O8">
- <attribute name="obj_pos">
- <point val="12.3156,10.85"/>
- </attribute>
- <attribute name="obj_bb">
- <rectangle val="10.8656,9.4;18.515,22.125"/>
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- <point val="17.065,20.675"/>
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- <enum val="0"/>
- <enum val="1"/>
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- <attribute name="name">
- <string/>
- </attribute>
- <attribute name="direction">
- <enum val="0"/>
- </attribute>
- <attribute name="ends">
- <composite>
- <attribute name="role">
- <string/>
- </attribute>
- <attribute name="multiplicity">
- <string>#1#</string>
- </attribute>
- <attribute name="arrow">
- <boolean val="false"/>
- </attribute>
- <attribute name="aggregate">
- <enum val="2"/>
- </attribute>
- </composite>
- <composite>
- <attribute name="role">
- <string/>
- </attribute>
- <attribute name="multiplicity">
- <string>#*#</string>
- </attribute>
- <attribute name="arrow">
- <boolean val="false"/>
- </attribute>
- <attribute name="aggregate">
- <enum val="0"/>
- </attribute>
- </composite>
- </attribute>
- <connections>
- <connection handle="0" to="O0" connection="21"/>
- <connection handle="1" to="O7" connection="3"/>
- </connections>
- </object>
- </layer>
-</diagram>
diff --git a/docs/random/porting-to-0.11.txt b/docs/random/porting-to-0.11.txt
deleted file mode 100644
index 66f850aea4..0000000000
--- a/docs/random/porting-to-0.11.txt
+++ /dev/null
@@ -1,5 +0,0 @@
-This guide has been renamed to porting-to-1.0.txt.
-
-The latest version can be found at:
-
-http://cgit.freedesktop.org/gstreamer/gstreamer/tree/docs/random/porting-to-1.0.txt
diff --git a/docs/random/queue b/docs/random/queue
deleted file mode 100644
index a76f235742..0000000000
--- a/docs/random/queue
+++ /dev/null
@@ -1,38 +0,0 @@
- thread 1 thread2
-
- // the queue is empty
- while (!queue->level_buffers) {
- STATUS("queue: %s U released lock\n");
- GST_OBJECT_UNLOCK (queue);
-
- // thread1 is scheduled and puts a lot of buffers
- // in the queue
-
- // thread1 has put the last buffer on the queue
- // here. A signal is going to be emitted
-
- tosignal = (queue->level_buffers >= 0);
- queue->level_buffers++;
-
- /* we can unlock now */
- GST_OBJECT_UNLOCK (queue);
-
- if (tosignal) {
- g_mutex_lock (queue->emptylock);
- g_cond_signal (queue->emptycond);
- g_mutex_unlock (queue->emptylock);
- }
-
- g_mutex_lock (queue->emptylock);
- // wait forever
- g_cond_wait (queue->emptycond, queue->emptylock);
- g_mutex_unlock (queue->emptylock);
- GST_OBJECT_LOCK (queue);
- }
-
-
- // thread 1 will also wait forever because the
- // queue is filled....
-
-
-
diff --git a/docs/random/richardb/syncmail b/docs/random/richardb/syncmail
deleted file mode 100644
index d45adf3352..0000000000
--- a/docs/random/richardb/syncmail
+++ /dev/null
@@ -1,3 +0,0 @@
-The syncmail script in CVSROOT is a modified version of the script
-taken from the sourceforge snippets page, at:
-http://sourceforge.net/snippet/detail.php?type=snippet&id=100414
diff --git a/docs/random/rtp b/docs/random/rtp
deleted file mode 100644
index b0dbfdce60..0000000000
--- a/docs/random/rtp
+++ /dev/null
@@ -1,44 +0,0 @@
-RTP client subsystem proposal.
-Ramon Garcia Fernandez
-
-A brief description of the RTP protocol.
-
-The RTP protocol uses two connections: one for passing data and another for control. The control connection is used for starting, finishing, and passing statistics of lost packets. On the data connection, packets are transmitted containing the media. These packets have a 7 bit field that says what codec is the data transmitted with. This codec is variable. However data cannot change from audio to video in packets of the same connection. All the packets belong to the same logical stream, that is, for instanace, to the same speech, or to the same song. Different codecs in the same stream is used, for instance, to insert comfort noise.
-
-Each codec is packeted in a specific way in RTP packets. This is necessary to minimize the damage produced by lost packets. Therefore, packets should be arranged so that each packet can be decoded independently. If a packet is lost, that shouldn't preclude the decoding of following packets.
-
-
-Suggested implementation:
-
-The implementation that I suggest contains an rtpdec element. This element has one sink pad for the data, one src pad for the decoded data, and a pair of src and sink pads for control messages. The decoded data that comes from rtpdec has no RTP dependency. It is in the format expected by the codec that it decodes it. Therefore, rtpdec must do codec specific processing of data to take into account that different codecs are stored differently in RTP.
-
-A possible pipeline is:
-
-udpcon name="data" udpcon name="control" data.src!rtpdec.datasrc control.src!rtpdec.controlin control.sink!rtpdec.controlout rtpdec name="rtpdec" ! mp3decode ! osssink
-
-udpcon is an element (not yet written) that provides a src and a sink pad to read and write to a socket. There are two udpcon, one for reading the data socket and another for reading and writing to the control socket.
-
-To handle variable types, the following would work (the UDP part left out for clarity is identical to the one above):
-
-[udp] rtpdec ! spider ! osssink
-
-rtpdec has a property that tells it the mapping of the payload bits to mime type. This is necessary, because that mapping is not fixed. It has to be obtained at runtime through other mechanisms, such as a SDP file obtained by http or rtsp.
-
-A possible implementation of rtpdec is a bin that contains two elements: a media independent _rtpparse and a media dependent rtp-dec-mp3. Thus the pipeline would be
-
-udpsrc ! _rtpparse ! rtp-dec-mp3 ! mp3decode ! osssink
- \________________________/
- rtpdec bin
-
-Another possible implementation is, that rtpdec opens a plugin that contains the code to convert the codec specific packets into the input that the normal decoder for that codec expects.
-
-Ronald said that this mechanism would be complicated because it would require to duplicate the functionality of the plugin loader. It shouldn't be like that because it should use the normal plugin loader. However, this is an issue that I do not fully understand.
-
-Ronald suggested to use inheritance. Thus the user should insert a codec specific rtp element in the pipeline. Such as:
-
-udpsrc ! rtp-mp3 ! mp3decode ! osssink
-
-Reuse of RTP logic would be achieved through inheritance.
-
-This looks more logical, because inheritance reflects the fact that rtp-mp3 "is an" specialization of rtp. However, there are several issues. As stated above, it is possible in RTP to switch the encoding of the media at any time. If this happens, some state must be kept, such as statistics of packets received and sent.
-
diff --git a/docs/random/signal b/docs/random/signal
deleted file mode 100644
index 679a36c122..0000000000
--- a/docs/random/signal
+++ /dev/null
@@ -1,12 +0,0 @@
-Random notes on signals:
-
-* Use a BOXED (and not a POINTER) marshaller when your signal will have a
- GstBuffer argument:
-
-g_signal_new ("handoff", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
- G_STRUCT_OFFSET (GstIdentityClass, handoff), NULL, NULL,
- gst_marshal_VOID__BOXED, G_TYPE_NONE, 1,
- GST_TYPE_BUFFER | G_SIGNAL_TYPE_STATIC_SCOPE);
-
-* For GstBuffers arguments, consider using G_SIGNAL_TYPE_STATIC_SCOPE as it
- can prevent an extra copy of the buffer
diff --git a/docs/random/sources b/docs/random/sources
deleted file mode 100644
index 3e5c466225..0000000000
--- a/docs/random/sources
+++ /dev/null
@@ -1,324 +0,0 @@
-There are a number of different ways of coding a GstSrc. I'll try to
-outline them and how the function here:
-
-1a) Simple push-function based with single output
-
-
- *------* *------
- ! ! !
- ! src !--->--! plugin
- ! ! !
- *------* *------
-
-This is the style that all the existing sources use. There is a single
-output pad, and a _push function that's global to the whole element. The
-_push function simply constructs buffers and pushes them out the pad.
-
-Chained (current implementation):
-
-
- bin src pad1 pad2 plugin
- ! (= pad1->peer)
- gst_bin_iterate
- !
- ! (find entry)
- !
- ! gst_src_push
- !---------------->!
- ! (create buffer)
- !
- ! gst_pad_push
- !---------------->!
- !
- ! pad1->chainfunc (pad1->peer)
- !------------------------------->!
- ! !
- : :
- (more chaining)
- : :
- !<-------------------------------!
- !<----------------!
- !<-----------------!
- !
- iteration ends
- !
- ---
- -
-
-Typically this will be the/an entry into the Bin. The Bin's iterate
-function simply calls the Src's _push function. When the _push function
-pushes a buffer out it's pad, the chain function of the peer pad is
-called, presumably causing a push out the other side of that element, and
-eventually data gets to the other end. The stack unrolls, and the
-iteration ends for that Src.
-
-
-
-Cothreaded:
-
-Again, the source would generally be an entry into the Bin. A loopfunc
-will be constructed around it, which will simply loop calling the Src's
-_push function as in the non-cothreaded case. When the _push function
-pushes a buffer, it finds a pushfunc attached to the pad, drops the buffer
-in the pen, and calls the pushfunc provided by the Bin. This causes a
-switch to the next element, then the next, to the end, at which point a
-buffer pull will travel back down the chain. The _push function gets
-context and finishes, at which point the loopfunc wrapper simply calls it
-again in the next iteration.
-
-(current implementation):
-
-
- bin cothread1 src pad1 pad2 cothread2 plugin
- ! (src) (= pad2->peer) (plugin)
- gst_bin_iterate
- !
- ! (first entry)
- !
- ! cothread_switch
- !---------------->!
- ! gst_src_push
- !---------------->!
- ! (create buffer)
- !
- ! gst_pad_push (pad1)
- !
- !--------------------!
- ! (fill bufpen)
- !
- ! cothread switch
- ----------------------->!
- ! gst_pad_pull (pad2)
- !
- !<----------!
- !
- ! (get buffer from bufpen)
- !
- !---------->!
- ! pad2->chainfunc
- !------------->!
- !
- :
-
- :
- !<-------------!
- ! gst_pad_pull (pad2)
- !<----------!
- !
- ! (bufpen empty)
- !<----------!
- !<-------------------------------------! cothread switch
- !<----------------!
- !
- iteration ends
- !
- :
-
- :
- !
- next iteration
- !
- ! cothread_switch
- !---------------->!
- ! gst_src_push
- !---------------->!
- ! (create buffer)
- !
- ! gst_pad_push (pad1)
- !
- !--------------------!
- ! (fill bufpen)
- !
- ! cothread switch
- !---------->!
- ! (get buffer from bufpen)
- !
- !---------->!
- ! pad2->chainfunc
- !------------->!
- !
- :
-
- :
- !<-------------!
- ! gst_pad_pull (pad2)
- !<----------!
- !
- ! (bufpen empty)
- !<----------!
- !<-------------------------------------! cothread switch
- !<----------------!
- !
- iteration ends
- !
- :
-
-
------------------------------------------------------------------------------------------------
-1b) Simple push-function based with multiple output
-
-Chained:
-
-Similar to the single output variant, except several chains are spawned
-off, one per push, hanging off whichever pad the buffer is pushed off of.
-The stack will grow and unwind as many times as buffers are pushed out.
-
-(current implementation)
-
- bin src pad1 pad2 plugin
- ! (= pad1->peer)
- gst_bin_iterate
- !
- ! (find entry)
- !
- ! gst_src_push
- !---------------->!
- ! (create buffer)
- !
- ! gst_pad_push
- !---------------->!
- !
- ! pad1->chainfunc (pad1->peer)
- !------------------------------->!
- ! !
- : :
- (more chaining)
- : :
- !<-------------------------------!
- !<----------------!
- ! (create buffer)
- !
- ! gst_pad_push
- !---------------->!
- !
- ! pad1->chainfunc (pad1->peer)
- !------------------------------->!
- ! !
- : :
- (more chaining)
- : :
- !<-------------------------------!
- !<----------------!
- :
- (more pushes)
- :
- !<-----------------!
- !
- iteration ends
- !
- ---
- -
-
-
-Cothreaded:
-
-Also similar to the single output variant. When the pull winds its way
-back from the first push, execution returns to the Src's _push function,
-which simply goes off and pushes out another buffer, causing another
-series of context switches. Eventually the loopfunc wrapper starts over,
-round and round we go.
-
-
-
------------------------------------------------------------------------------------------------
-2) Pull-function based with single output
-
-Similar to a regular filter with a chain function associated with each
-pad, this kind of source doesn't provide a src-wide push function, but
-does provide pullfuncs for its pad. A pullfunc puts a buffer in the pen
-and exits.
-
-Chained:
-
- bin src pad1 pad2 plugin
- ! (= pad1->peer)
- gst_bin_iterate
- !
- ! (find entry)
- !
- ! (find src pad
- ! of entry element)
- !
- ! gst_pad_pull
- !------------------------------------------------>!
- ? !
- !<------------------------------!
- !
- ! (create buffer)
- !
- ! gst_pad_push
- !------------------------------>!
- ! (bufpen filled)
- (return buffer) !
- !<------------------------------------------------!
- !
- ! gst_pad_chain
- !------------------------------------------------------------------->!
- !
- :
- (more chaining)
- :
- !<-------------------------------------------------------------------!
- !
- iteration ends
- !
- ---
- -
-
-As usual, is likely to be an entry into a Bin. The Bin iterate code must
-explicitly pull a buffer and pass it on to the peer.
-
-Cothreaded:
-
-
- bin cothread1 src pad1 pad2 cothread2 plugin
- ! (src) (= pad2->peer) (plugin)
- gst_bin_iterate
- !
- ! (first entry)
- !
- ! cothread_switch
- !---------------->! gst_pad_pull
- !------------------------------------------------>!
- ? !
- !<------------------------------!
- !
- ! (create buffer)
- !
- ! gst_pad_push
- !------------------------------>!
- ! (bufpen filled)
- (return buffer) !
- !<------------------------------------------------!
- !
- ! pad_chain
- !--------------------------------------! cothread switch
- |---------------------->!
- ! gst_pad_pull (pad2)
- !
- !<----------!
- !
- ! (get buffer from bufpen)
- !
- !---------->!
- ! pad2->chainfunc
- !------------->!
- !
- :
-
- :
- !<-------------!
- ! gst_pad_pull (pad2)
- !<----------!
- !
- ! (bufpen empty)
- !<-----------!
- ! cothread switch
- !-------------------!
- !<----------------!
- !<----------------!
- !
- iteration ends
- !
- :
-
diff --git a/docs/random/status-0.11-14-jun-2011.txt b/docs/random/status-0.11-14-jun-2011.txt
deleted file mode 100644
index 48ffaba270..0000000000
--- a/docs/random/status-0.11-14-jun-2011.txt
+++ /dev/null
@@ -1,192 +0,0 @@
-0.11 status 1 jun 2011
-----------------------
-
-Hello again GStreamer hackers,
-
-
-Here's another status update from the 0.11 branch. This one is packed
-with new stuff because it really took too long between the last
-update, but well..
-
-
-Misc Core Changes
------------------
-
-The first set of big core change was done by über hacker Sebastian Dröge.
-He removed our special GST_BOILERPLATE macro. Because of some API strangeness
-we needed to do actions in the base_init method, which the GST_BOILERPLATE
-macro set up for us. The base_init concept is unfortunately not supported
-by almost all language bindings so we wanted to clean this up and use the
-regular G_DEFINE_TYPE macros. By inheriting metadata and pad templates
-from the parent, the GST_BOILERPLATE macro could be removed.
-
-
-Removal of gst_pad_alloc_buffer
--------------------------------
-
-The next big change was the removal of gst_pad_alloc_buffer along with
-all the pad_alloc functions. The pad_alloc function were used to both
-implement allocation of buffers and notify upstream elements of new media
-formats. The functionality is replace by 3 new things: GstBufferPool (as
-noted in the previous status update), a new ALLOCATION query and a new
-RECONFIGURE event.
-
-The ALLOCATION query is used to query the downstream elements about the
-buffer properties they would like. These properties include the size,
-prefix and alignment of the memory. The downstream element can also propose
-a GstBufferPool. It is then the upstream element that decides how it
-will allocate buffers. This allows us, for example, to make the video
-decoders specify their required memory alignment to the videosink.
-
-Since new format suggestions are no longer piggybacked upstream with the
-allocation, something new was needed to notify upstream elements of a
-format change. It started with bringing in the new RENEGOTIATE event
-that Thiago Santos had been experimenting with in the 0.10 branch. The
-new event basically lets upstream elements know that new formats are
-possible somewhere downstream and it would make them renegotiate a new
-format before pushing a new buffer.
-
-New format changes can also happen when new elements are linked, so
-gst_pad_link() will also send this RENEGOTIATE event upstream. As it turns
-out, adding new elements could also signal the availability of a new
-bufferpool or different ALLOCATION properties. The RENEGOTIATE event was
-thus renamed to the more generic RECONFIGURE event.
-
-The most impressive result of these changes is that most of the
-complicated code in GstBaseTransform could simply be removed.
-
-
-Sticky Events
--------------
-
-The idea for the sticky events comes from the observation that a lot of the
-serialized downstream events (SEGMENT, EOS, TAGS, ..) are really context
-for the buffers that follow. The design called for making these events
-sticky on the pads, meaning that they become a property of the pads
-they travel over.
-
-The result is that the current timing information (SEGMENT event) or the
-current stream medatadata (TAG event) that is handled by the pad are
-directly accessible by looking at the last event that traveled on the pad.
-
-By making the events stick on the pads, we can propagate them downstream
-automatically when new pads are linked. This should solve one of the
-biggest 0.10 problems when dealing with dynamic pipelines, the loss of
-events (mostly NEWSEGMENT events).
-
-It was then only natural to also make the CAPS event a sticky downstream
-serialized event. CAPS are indeed also context for the buffers that follow.
-The caps property was also removed from GstBuffer, making caps negotiation
-separate from the buffer dataflow again.
-
-
-GstSegment changes
-------------------
-
-For the sticky events to work with SEGMENT events, we needed to change
-the SEGMENT event so that it became selfcontained. The complicated segment
-accumulation logic of 0.10 was simply removed and replaced with pad offsets.
-
-It is now possible to tweak the timing of the data coming from a pad by
-using the pad offset property.
-
-
-GstCaps optimizations
----------------------
-
-It doesn't look like we'll be able to implement GstCaps iterators to 0.11 so
-Sebastian was looking for other ways to improve caps performance. One of the
-low hanging fruits was to pass the complete GstCaps structure to the
-GstBaseTransform transform_caps method. This allows for better and smarter
-implementations of the function at the cost of marginally more complex code.
-
-Sebastian also added a filter caps parameter to the gst_pad_get_caps()
-function. This allows the caller to pass preferences and possible caps to
-the function. The result is that the pad can make much better negotiation
-decisions.
-
-Sebastian also optimized gst_caps_is_subset()in the 0.10 branch and ported
-the results to 0.11 as well.
-
-with the still planned caps simplifications later, these changes should
-substantially improve caps negotiation speed.
-
-
-Pad probes
-----------
-
-Quite a few iterations were needed to reimplement the pad blocking and pad
-probes. The new infrastructure allows you to add callbacks for the various
-stages in the data transfer on a pad.
-
-It's now possible to be notified when no data is going over a pad with the
-IDLE probe. This should also fix one of the big problems with implementing
-dynamic pipelines and the old pad blocking.
-
-Not all features that we want to add to the new probes are implemented yet
-but they can be added later without much trouble.
-
-
-Other gems
-----------
-
-A new SCHEDULING query was added to get more information about the upstream
-elements scheduling properties.
-
-Tim-Philipp Müller finally removed our dependency on libxml2. Back in the
-day, we used XML for serialization of objects and the registry. Object
-serialization had long been deprecated and the XML registry was replaced
-with a much faster binary registry.
-
-Tim also removed the unversioned gst-launch, gst-inspect and gst-typefind
-scripts. They were always confusing and annoying for packagers and users,
-it is just easier
-
-Edward Hervey spent some time keeping the unit tests working and making
-sure the API docs are up to data.
-
-GstIterator was made more binding friendly By Johan Dahlin and Sebastian.
-
-
-And what about the plugins..
-----------------------------
-
-As usual, all -base plugins are kept in sync with the core API updates and
-the new features. They are always a good place to see how the new API can
-be used and how to port things.
-
-
-What's next
------------
-
-Almost all of the features laid out in the GStreamer conference last year
-are now implemented. We also have quite a few new cool features that
-slipped in. Things are shaping up indeed.
-
-The last big missing piece is the redesign of the caps fields for raw audio
-and video. We plan to finish that in the next weeks.
-
-There are also a few smaller things we would like to do: use GstQuery
-for the various caps functions, do GstFlowReturn for the query/event
-functions, ...
-
-Meanwhile we start the stabilization phase and we will do a first prerelease
-of 0.11 this week to bring things to a wider audience. Now is the time to
-catch up and start porting your plugins to 0.11.
-
-There is still a lot of work to be done to port plugins to 0.11 before we
-can release 1.0. I ask everyone again (and especially maintainers) to help
-us porting plugins, it's really a lot of work!
-
-
-Have a nice hacking week,
-
-Wim
-
-
-
-
-
-
-
-
diff --git a/docs/random/styleguide b/docs/random/styleguide
deleted file mode 100644
index 6fadda1f19..0000000000
--- a/docs/random/styleguide
+++ /dev/null
@@ -1,18 +0,0 @@
-Some notes about various stylistic issues
------------------------------------------
-
-(This file should end up as a nice web page eventually)
-
-CODING STYLE
-------------
-- No //-style comments allowed. Never. Reason: Forte doesn't support it.
-- Our indentation style is codified in gstreamer/tools/gst-indent
-- Don't mix tabs with spaces when writing new code.
-- use GST_*_OBJECT as much as possible so it's clear which object you're
- logging for
-
-SUBMITTING CODE
----------------
-- new code submitted to core, plugins-base, or plugins-good needs to pass a
- review from another committer.
-- submitting new base classes also means you look at targets for assimilation.
diff --git a/docs/random/testing/syntax b/docs/random/testing/syntax
deleted file mode 100644
index f66c12631e..0000000000
--- a/docs/random/testing/syntax
+++ /dev/null
@@ -1,37 +0,0 @@
-The test program
-----------------
-
-We need a test program that:
-
- - can read a file with the test specs
- - construct a pipeline using the pipeline definition in the spec file
- (using gst_parse_launch ())
- - add signals to elements in the pipeline
- - run the pipeline for a fixed (in time/number of iterations) period
- - record the fireing of the signals
- - compare the signals to the expected results.
-
-
-spec file contents:
--------------------
-
-tcN: name-of-the-tescase
-tcP: pipeline of the testcase in gst_parse_launch () syntax.
-tcS: id1, element, signalname (attach to signal in an element)
-tcS: id2, element, signalname
-...
-tcI: the number of iterations on the top bin
-tcT: a timeout value in mSecs
-tcR: id1,1,id2,1,.. (the pattern of signals triggered)
-or
-tcR: id1==id2,... (denote an equal number of signals)
-/n
-
-
-signal trigger patterns
------------------------
-
-
-
-
-
diff --git a/docs/random/thaytan/opengl b/docs/random/thaytan/opengl
deleted file mode 100644
index a796629ca1..0000000000
--- a/docs/random/thaytan/opengl
+++ /dev/null
@@ -1,20 +0,0 @@
-# OpenGL usage
-How to set up openGL using plugins in such a way that they can
-render to an on-screen window, or offscreen equally?
-
-eg:
-superfoo3d ! glwindow
-or
-superfoo3d ! gloffscreen ! xvideowindow
-
-This would imply that there is some mime type which connects a glwindow/gloffscreen and a GL using component sensibly. I don't see that there is any actual
-data to send, however - the only purpose of the glwindow/gloffscreen is to
-set up an openGL context and then 'output' it somehow one the superfoo3d has
-finished drawing each frame.
-
-In the case of glwindow, 'output' it means translate into an on-screen window,
-but for gloffscreen it means produce a video packet and go.
-
-These components really need some other 2 way communication, rather than the
-pads metaphor, I think?
-
diff --git a/docs/random/thaytan/video-overlays b/docs/random/thaytan/video-overlays
deleted file mode 100644
index af0697b1cd..0000000000
--- a/docs/random/thaytan/video-overlays
+++ /dev/null
@@ -1,28 +0,0 @@
-# Compositing and video overlays.
-We want a way to composite a video frame and an overlay frame. The video frame is as we expect, the overlay is a normal video frame + an alpha mask. In the following monologue, please consider that I mean RGB/I420/YV12/YUY2 wherever I say RGB.
-
-So we have a plugin with 2 sinks and one src:
- +----------------+
- |video |
- | video|
- |overlay |
- +----------------+
-
-What should the format for the overlay data look like?
-Ideas:
- * RGBA - 4 byte per pixel video frame
- * RGB+A - 3 Bytes per pixel video frame, then 1 byte per pixel overlay frame.
- A new mime type, or a variation on video/raw? I'm not sure
- * Overlay is actually 2 sinks, one takes a RGB/YUV data the other the alpha channel.
-
-I'm not sure which approach is better, but I think it is probably neatest to
-use RGB+A, and then have a separate plugin which has 2 sinks and converts an
-RGB/YUV stream plus an alpha stream into an RGB+A stream. The benefit of RGB+A over RGBA in this scenario, is that it is easier (correct me if I'm wrong) to optimise 2 block copies which appends an alpha frame to a RGB frame than it is to
-do the many copies required to interleave them into an RGBA stream.
-
-So, I see this producing a few new plugins:
-videooverlay - takes an RGB and an RGB+A frame from 2 sinks, does the overlay (according to some properties) and outputs a result frame in RGB or RGB+A (as negotiated) format on 1 src.
-rgb2rgba - takes 1 RGB frame and one A frame from 2 sinks and outputs RGB+A on 1 src. If the A sink is not connected, we just add a fixed alpha channel based on an object-property.
-rgb2rgba - takes an RGB+A frame and discards the RGB component.
-textoverlay - This plugin, instead of taking a video frame and overlaying text, can just output an RGB+A stream with appropriate timestamping. This prevents duplicating code to composite with an alpha mask and allows us to optimise it in one spot only.
-
diff --git a/docs/random/thomasvs/0.10 b/docs/random/thomasvs/0.10
deleted file mode 100644
index ee3e466a04..0000000000
--- a/docs/random/thomasvs/0.10
+++ /dev/null
@@ -1,121 +0,0 @@
-general
--------
-- write script to check for:
- invalid // comments
- invalid %lld
- invalid LL for 64bit ints
- wrong whitespacing
- trailing space
-
-gstreamer
----------
-- reorganize tests and examples into
- - testsuite
- - check: unit tests
- - examples: example code
- - interactive: interactive tests
-- move gst/base to libs/gst/base ?
- (but elements link against them)
-- move elements out of gst/ dir ?
-- check/gst/pipelines: currently disabled, random failures
-
-gst-plugins-base
-----------------
-- gst-libs/gst/audio:
- - DONE: audiofilter folded in
-- gst:
- - adder: needs docs, an example, and a test
- - audioconvert: ok
- - audiorate: needs docs and tests
- - audioresample: David needs to fix this
- - audioscale: needs to go
- - audiotestsrc: ok
- - ffmpegcolorspace: needs a test
- - playback: example
- - sine: removed, DONE
- - subparse:
- - works, but how do we link it so we can display it ?
- - example ?
- - test with different files/sources ? can be inlined
- - tags:
- - contained a very small code file that wasn't built, and a copy of a header
- that was in the tag lib; removed; DONE
- - tcp:
- - works
- - need tests
- - need docs
- - need possible porting to GNet (later)
- - typefind:
- - works
- - need tests - this definitely could use it
- - is there any way they can be documented ?
- - should the plugin docs show a list of them ?
- - videorate:
- - needs tests, docs
- - videoscale:
- - needs tests
- - negotiation
- - par conversion
- - different scale algorithms
- - needs docs
- - negotiation with five out of six free variables (src/sink w/h/par)
- - videotestsrc:
- - could use tests for all possible caps
- - volume: OK
-- ext:
- - alsa:
- - needs docs; esp. params and common practices
- - needs interactive tests; depends on having such a setup available
- - cdparanoia:
- - needs docs, and interactive test
- - remains in -base until cdio is proven to be better on all counts
- - gnomevfs:
- - needs docs (how to use proxy, link to gnomevfs docs, explanation
- about need for homedir to create .gnome2 dir in, ...)
- - needs test; test could use local files and http urls
- - libvisual
- - needs docs (easy)
- - needs test
- - ogg, vorbis, theora
- - needs docs
- - needs test
-- sys
- - v4l
- - needs interactive test
- - needs lots of docs
- - ximage
- - interactive test should go somewhere
- - docs ok
- - xvimage
- - interactive test should go somewhere
- - docs ok
-
-gst-plugins-good
-----------------
-
-- gst:
- - alpha, alphacolor: document with example
- - auparse: crashes on e.g.
- gst-launch -v filesrc location=/usr/share/emacs/site-lisp/emacspeak/sounds/default-8k/ask-short-question.au ! auparse ! osssink
- -> will move to bad
- - autodetect: OK
- - videofilter:
- - is the lib still needed, given basetransform ?
- yes
- - currently installs a lib; should not install, or move to a dir, with pc
- file, possibly in -base
- DONE: moved to -base
-- ext:
- - aasink: properties need looking at
- - width, height: why are they there ? caps don't match
- - frames-displayed: in base class ?
- - frame-time: what's this ?
- - cairo:
- - cairotimeoverlay works
- - cairotextoverlay ? pipeline ?
- - flac:
- - flacenc:
-gst-launch -v audiotestsrc wave=white-noise ! flacenc ! filesink location=white-noise.flac does not produce a correct file
- - flacdec works, but
-gst-launch gnomevfssrc location=http://gstreamer.freedesktop.org/media/small/dark.441-16-s.flac ! flacdec ! autoaudiosink
- does not
diff --git a/docs/random/thomasvs/0.4.0 b/docs/random/thomasvs/0.4.0
deleted file mode 100644
index ddbc777cd9..0000000000
--- a/docs/random/thomasvs/0.4.0
+++ /dev/null
@@ -1,108 +0,0 @@
-Set of 0.4.0 proposals
-
-* module versioning
- - we need an agreement on how we are going to version the separate modules.
- This needs to meet some requirements :
- a) we shouldn't be forced to release all of the modules together
- every time (if the plugins have been updated, and still work against
- a previous core, then only release plugins, for example)
- We should start treating the different modules as separate projects
- (albeit still closely tied of course)
- b) it should be clear for other people what packages they need to download
- c) major API breakage needs to be clear; we don't expect really old players
- to keep working with really new cores.
-
- - my idea (which others seemed to agree with) would be to let the micro
- numbers run independently. Only when the core API has changed to the
- point that other modules are not compatible with them anymore
- should we upgrade the minor version.
- We can assume/assert that modules with the same minor number will be able
- to work with each other.
-
- - Example schedule :
- * we release 0.4.0 versions of all of the modules
- * plugins get updated a few times : 0.4.1 and 0.4.2
- * core gets a new scheduler, doesn't affect other modules : 0.4.1
- * gst-player gets rapid releases due to arik's recovery: 0.4.1-0.4.5
- * core gets a major new update re: events : 0.5.0
- * some days later, other modules have been updated to the new core
- and the new core starts being useful to other people as well
-
-* release practice
- - we should have a simple way to cut releases; something that makes
- the necessary adaptions to the source tree
- This could also be a simple check list of things that need to pass
- - cvs tarballs/packages should be easily distinguishable from pre-release
- tarballs/packages and actual released ones.
-
- my idea here would be :
- a) releases are named as normal
- e.g. gstreamer-0.4.0.tar.gz
- gstreamer-0.4.0-1.i386.rpm
-
- b) as soon as the release is made, we are back in "cvs" mode
- i'd use a ".1" for a fourth version number for all cvs versions
- so as soon as we release 0.4.0, I'd add a fourth ".1" version number.
- this one would be used during the whole of the cvs period, no
- reason to up this manually in between
- The packages (rpms in any case, don't know about debs) would still
- keep the date as the revision number like they have now, in order
- to make it easy to work with snapshots.
-
- c) when we are ready to release this module, I would go back to maintainer
- mode, but keep the fourth version number and increase that for each cut.
- So we'd stop developing the module, switch to maintainer mode, up the
- version number to
- gstreamer-0.4.0.2.tar.gz
- and test that.
-
- d) when we are happy with the precuts, we drop the fourth number and make
- a release
-
- Summary :
- * all "official" released versions have sane versioning with three numbers
- * all "cvs" versions are clearly recognizable by the fourth .1
- * all "precuts" are also recognizable
- * no tarballs will accidentally spill pretending to be real releases ;)
- * upgrading rpm's all through this process is easy
-
-* build code stuff duplicated between various modules
- - how do we integrate these ? this pertains to stuff in autogen.sh,
- duplicate stuff in configure.ac (which should probably be moved out to
- custom .m4 files and yanked in), and maybe testing stuff
-
-* what possible ways are there to build gstreamer ?
- I would like to take stock of the combinations of deps possible to build
- gstreamer. While most people want only glib2, I think there is merit
- in still making glib1 work and willing to put in the effort. I just need
- the possibilities mapped out once and for all ;) IMO the effort going into
- making gstreamer build without libxml is the same as making it work with
- older glib too. I mean, as soon as you allow variation, it isn't that hard
- to allow for more than one variation. The bigger step is from zero to one.
-
- so, what are they ?
- - glib1 & gtk1
- - glib1 & gtk1 & libxml
- - glib2 & libxml2
-
-* media suite
- - media files on the site should be renamed to simple uniform names
- - and split up based on size
- - described by features/content
-
-* when to branch in CVS ?
-
-* automatic build testing
- - tinderbox ? useful ?
- - small build scripts
-
-* automatic functionality testing
- - an md5sink would be useful to do this
- - something automated is needed; it should check if you have the plugins
- that are needed to test other plugins
-
-* media player
-
-* packaging
- - dependency libs should be easily available
-
diff --git a/docs/random/thomasvs/TODO b/docs/random/thomasvs/TODO
deleted file mode 100644
index fce3d74f5c..0000000000
--- a/docs/random/thomasvs/TODO
+++ /dev/null
@@ -1,3 +0,0 @@
-GSTREAMER
----------
-- gst-i18n-lib.h is included funnily from base classes and elements
diff --git a/docs/random/thomasvs/docreview b/docs/random/thomasvs/docreview
deleted file mode 100644
index 0fced3cc0e..0000000000
--- a/docs/random/thomasvs/docreview
+++ /dev/null
@@ -1,37 +0,0 @@
-Documentation review
-
-* gstbuffer
-- What are the flags in GstBuffer ? used anywhere ? defined how ?
-
-
-* General
-- how can we define common terms and make them cros-ref'd ?
- e.g. timestamps in buffer, do we say everywhere that they're in nanosec ?
-
-
-* Style
-- when in doubt, try to conform to GTK+ reference docs
- (in the gtk-doc tarball, doc/style-guide.txt)
-- GtkMisc and GtkFontSelectionDialog are example templates.
-
-- in the arg clarification, use as much cross-reffing as possible. Do it
- only where it is useful in the explanation text.
-
-- examples
-
- - use active form instead of imperative describing functions; we describe
- what the function does.
- good : creates a new buffer.
- bad : create new buffer
- - use singular for enum names; this makes it more natural to reference to
- it in the API docs
- good : GstBufferFlag
- bad : GstBufferFlags
- - in arg clarification, use a period and start with a small letter.
- Call the object you work on "a" instead of "the". Call the other objects
- "the".
- If the object in question is the return value, this means you call the
- return value "a". If the object in question is the first argument
- of the call, you call this argument "a" and the rest "the".
- good : @buf: a pointer to query.
- bad : @buf: The pointer to query
diff --git a/docs/random/thomasvs/features b/docs/random/thomasvs/features
deleted file mode 100644
index 2d52339751..0000000000
--- a/docs/random/thomasvs/features
+++ /dev/null
@@ -1,15 +0,0 @@
-Here's a list of features in GStreamer which plug-ins can make use of.
-For each feature we will try to find a few plug-ins that show a good way
-of implementing them.
-
-seeking
-caps negotiation
-timestamps
-clock interaction
-signals
-object argument handling
-chain-based
-loop-based
-request pads
-sometimes pads
-
diff --git a/docs/random/thomasvs/guadec-4 b/docs/random/thomasvs/guadec-4
deleted file mode 100644
index f7965dd1a2..0000000000
--- a/docs/random/thomasvs/guadec-4
+++ /dev/null
@@ -1,16 +0,0 @@
-Presentation ideas for GUADEC 4 (thomasvs, April 8 2002)
-
-* use gst-editor to create pipelines that make a karaoke machine
- in different steps and using different features
-
-1) pipeline 1: play the free software song by Richard Stallman
-2) pipeline 2: do the same but add a visualization plugin
-3) create a small video using actual RMS footage
-4) pipeline 3: play this video and the song together
-5) Stallman is a bit hard to understand. We want text.
- pipeline 4: use the subtitle reader to overlay text
- maybe also do a bouncing ball overlay !
-6) Stallman can't sing. Let's pitch-shift him.
- this will need MIDI or dynparams to control a pitch shifter
-7) It sounds better, but still not quite there. Replace him with a festival
- voice doing the pitch shifting.
diff --git a/docs/random/thomasvs/pthread b/docs/random/thomasvs/pthread
deleted file mode 100644
index 8047c7ca16..0000000000
--- a/docs/random/thomasvs/pthread
+++ /dev/null
@@ -1,75 +0,0 @@
-Some notes on use of pthreads in GStreamer
-------------------------------------------
-
-First off, pthreads are HARD. Remember that.
-
-1) How I learned to debug glibc and pthreads and add debug code to it.
-
-You have to trick your GStreamer test app in running against a modified
-glibc.
-
-I used Red Hat 7.3, downloaded the .src.rpm, installed it, applied the
-patches included, and ran
- ./configure --prefix=/home/thomas/cvs --with-add-ons
- make
- make install
-
-After quite some time this left me with recompiled libc and libpthread
-libraries in /home/thomas/cvs/lib, as well as a new ld-linux.so.2
-
-Now you need to use this new ld-linux.so ld loader to run your app,
-preferably from inside of gdb so you can tell what's going on when it
-crashes.
-
-You can use ld-linux.so.2 to call your binaries:
- ld-linux.so.2 .libs/thread1
-to run the thread1 program with the new glibc.
-
-If this is a GStreamer app, chances are it might not find some libraries
-it needs that you could safely use from /usr/lib (like, zlib and popt).
-
-Also, you want it to run in gdb, so this is my full line:
-
-LD_LIBRARY_PATH=/usr/lib /home/thomas/cvs/lib/ld-linux.so.2 \
- /usr/bin/gdb .libs/thread1
-
-At this point you can start adding debug code to the pthreads implementation
-in your glibc source tree. Just change, re-run make install, and restart
-the test app in gdb.
-
-Helpful --gst-mask is 0x00200100 to get thread info and scheduling info
-(with mem alloc from cothreads)
-
-2) What GStreamer does with pthreads.
-
-Apps create a thread with gst_thread_new. This just allocates the GstThread
-structure without actually doing much with it.
-
-When a thread goes from NULL to READY, the gst_thread_change_state function
-creates the actual pthread.
- - we lock the thread->lock mutex
- - we create attributes for the pthread
- - by default the pthread is JOINABLE
- - we ask the thread's scheduler for a preferred stack size and location
- (FIXME: if the scheduler doesn't return one, what do we do ?)
- - we create the pthread with the given attributes
- - the pthread id is stored in thread->thread_id
- - the created pthread starts executing gst_thread_main_loop (thread)
- - the change_state function does a g_cond_wait
- - this means it unlocks the mutex, waits until thread->cond is set
- (which happens in gst_thread_main_loop),
- then relocks the mutex and resumes execution
-
-From the point of view of the created pthread, here's what happens.
-gst_thread_main_loop (thread) gets called
- - the thread's mutex gets locked
- - the thread's scheduler's policy gets examined
- - the scheduler gets set up (invokes the scheduler object's setup method)
- FIXME: what are the prereqs of this _setup method ?
- - basic and fast scheduler both call do_cothread_context_init
- - basic: this calls cothread_context_init
- - cothread_context_init
- - fast: this calls cothread_create (NULL, 0, NULL, NULL))
-
-(FINISHME)
-(FOLDMEBACKTOREALDOCS)
diff --git a/docs/random/thomasvs/pwg b/docs/random/thomasvs/pwg
deleted file mode 100644
index c51b7671ea..0000000000
--- a/docs/random/thomasvs/pwg
+++ /dev/null
@@ -1,39 +0,0 @@
-Stuff for the PWG
------------------
-
-* arguments
-
- - how to add arguments
- - create an identifier in the enum, starting with ARG_
- example: ARG_RATE
- - add the property by adding a
- g_object_class_install_property line
- FIXME: what is name/nick/blurb in the paramspec ?
- - if the argument is readable, a block of code for it needs to be added
- to the _get_property function.
-
- - default value
- - default value should be set in _init function
- - default value can be specified in paramspec (but I don't think this
- is used anywhere)
-
-
- - things to check/possible problems
- - do you have a _get_property function ?
- - do you have a _set_property function ?
- - do both have a default handler that handles invalid property ID's ?
- - are the _get/_set_property handlers assigned to the class's struct ?
- - do you have a g_object_class_install_property line for the argument ?
- - are there restrictions on when your parameters can change ?
- e.g. sample rate is not something that should be changed while PLAYING,
- so it can only be changed in the NULL state
- - did you use ARG_ ... consistently everywhere ?
-
-
- - my own problems:
- - how to set defaults and make the paramspec be right for them ?
-
-
-* audio
- - explanation of difference of width and depth
- - guidelines on how to implement this properly
diff --git a/docs/random/thomasvs/registry b/docs/random/thomasvs/registry
deleted file mode 100644
index 5287133767..0000000000
--- a/docs/random/thomasvs/registry
+++ /dev/null
@@ -1,36 +0,0 @@
-Reviewing the registry (thomasvs, April 8 2002)
-
-* added a --gst-registry flag to the core which allows any gst app
- to specify a different registry for loading/saving
-
- some stuff to do this went into gstreamer/gst/gstregistry.h
-
-* What location is used for writing ? (gst-register)
-
- - if specified (using --gst-registry) then use the specified location
- - if not specified :
- - it can be written in the global location, do it there
- (which should be sysconfdir/gstreamer) and reg.xml
- - if not writable, then try ~/.gstreamer/reg.xml
-
-* What location is used for reading ? (gst-whatever)
-
- - if specified (using --gst-registry) then use the specified location
- - if not specified :
- - right now :
- if local exists, only read local
- if not, read global
-
- - TODO: try reading GST_CONFIG_DIR/reg.xml first
- then try reading ~/.gstreamer/reg.xml
- AND replace every namespace collision with the new one
-
-* actual variables stuff (gstregistry.c)
- - use gst_registry_write_get to get a GstRegistryWrite struct back
- listing the right location of dir, file and tmp file
- - use gst_registry_read_get to get a GstRegistryRead struct back
- listing the path of global and local file to read
-
-* gst-register signals it's going to write to the registry (causing it to
- be unlinked before the read in post_init ()) by setting a global variable,
- _gst_init_write_registry
diff --git a/docs/random/types b/docs/random/types
deleted file mode 100644
index eac3012e3f..0000000000
--- a/docs/random/types
+++ /dev/null
@@ -1,33 +0,0 @@
-GstTypes exist to try to make sure data eveyrone is talking about the
-right kind of data. They aid quite a bit in autoplugging, in fact make it
-possible. Each well-formed type includes a function (typefind) that will
-take one or more buffers and determine whether or not it is indeed a
-stream of that type, and possible a metadata to go with it. It may
-provide related metadata structure IDs (and must if it provides metadata
-from the typefind function).
-
-Because multiple elements and plugins are very likely to be using the same
-types, the process of creating/finding types is designed to be done with a
-single function call. All operations on GstTypes occur via their guint16
-ID numbers, with the GstType structure "private" to the GST library. A
-plugin wishing to use a give type would contain a static struct of type
-GstTypeFactory, which lists the MIME type, possible extensions (which may
-overlap the mime-types file), a typefind function, and any other cruft I
-decide to add.
-
-A plugin init function would take this typefactory and hand it to the
-gst_type_new() (FIXME: badly named) function, which would first search for
-that same MIME type in the current list. If it found one, it would
-compare the two to see if the new one is "better". Better is defined as
-having more extensions (to be merged) or a typefind function verses none.
-
-The point of returning an existing MIME type is a result of the goal of
-unifying types enough to guarantee that, for instance, all MP3 decoders
-will work interchangeably. If MP3 decoder A says "MIME type 'audio/mpeg'
-with extensions 'mpeg3'" and decoder B says "MIME type 'audio/mpeg' with
-extensions 'mp3'", we don't want to have two types defined, possibly with
-two typefind functions. If we did, it's not obvious which of the two would
-be tried first (luck) and if both would really identify streams as mp3
-correctly in all cases. And whichever wins, we're stuck using the
-associated decoder to play that stream. We lose the choice between any
-valid mp3 decoder, and thus the whole point of the type system.
diff --git a/docs/random/types2 b/docs/random/types2
deleted file mode 100644
index d2515e59f1..0000000000
--- a/docs/random/types2
+++ /dev/null
@@ -1,282 +0,0 @@
-OUTDATED
---------
-
-
-1. Introduction
----------------
-
-The type system is used to attach meaning to the bytes in a GstBuffer.
-A plugin can decide to add metadata to the GstBuffer, this metadata
-will carry an associated typeid.
-
-Types are also used by the plugins to expose the type of their pads to
-the type system.
-
-Types are essential for autoplugging.
-
-We will explain the inner workings of the type system in this document, as
-well as the API used in the plugins.
-
-
-2. Type properties
-------------------
-
-a) identification of the type
-
- Types are identified by one or more MIME types.
-
-b) detection of types
-
- The type of any given GstBuffer can be detected using
-
- - a typefind function: a function that will inspect the bytes
- in the buffer and return a gboolean indicating the
- buffer is of the given type.
-
- - a template for the bytes/bits in the data that must be
- satisfied in order for the GstBuffer to be of the given
- type.
-
- - other properties that act more like a hint like:
- the extension of the source filename.
- the URI of the source.
-
-
-3. Type registration
---------------------
-
-a) the core libraries will create to types:
-
- video/raw image/raw
- audio/raw
-
-b) all other types will be provided by the plugins
-
-before a type can become available to other plugins, the plugin
-has to register the type.
-
-The system will keep a directed graph of the types and the plugins
-that operate on them.
-
-example:
-
- video/mpeg
- !
- mpeg1parse
- / \
- video/mpeg1 audio/mp3 -----\
- ! ! \ !
- mpeg_play mpeg2dec mpg123 xing ...
- \ / \ /
- video/raw audio/raw-----\
- ! ! ! !
- videosink SDLsink audiosink alsasink ...
-
-
-The system can find the needed plugins to convert video/mpeg to
-audio/raw using this graph.
-
-
-4. type equivalence
--------------------
-
-some types can have the same meaning for example:
-
- audio/mp3 and audio/mpeg
-
-or
-
- video/raw and image/raw
-
-
-a plugin that exposes its output type as audio/mpeg and another plugins
-with input type audio/mp3 can be connected. The system has to know about
-the equivalence of both types, even it they have a different mime type.
-
-
-5. type hierarchy
------------------
-
-some plugins can output a specific subset of an already existing type.
-
-example:
-
- mp3parse inputs audio/mp3 and packs the stream into mp3 audio frames
- with mime type: audio/mp3-frame
-
- mpg123 only accepts audio/mp3-frame but not audio/mp3.
-
- another mp3 decoder (libmpg123) can accept audio/mp3 (and thus also
- audio/mp3-frame)
-
- it must be possible to connect both libmpg123 and mpg123 to the mp3parse
- element.
- it must not be possible to connect mpg123 to an element that outputs only
- audio/mp3 but not audio/mp3-frame.
-
-
-We say that audio/mp3-frame is a more specific subset of type audio/mp3.
-
-we can create a type hierarchy:
-
- audio/mp3
- / \
- audio/mp3-frame audio/mp3-layer12
- / \
- audio/mp3-layer1 audio/mp3-layer2
-
-
-6. multi-type pads
-------------------
-
-certain plugins might accept multiple non equivalent types in one of their
-input pads. Consequently a plugin might output non equivalent types in
-its output pad.
-
-It must therefore be possible to specify multiple types for a pad.
-
-example:
-
- mpegdemux may be able to demux both MPEG1 and MPEG2 system streams.
- we show the type hierarchy of the video/mpeg as follows:
-
- video/mpeg
- / \
- video/mpeg1 video/mpeg2 ---------------\
- / \ / \ !
- mpeg1-system* mpeg1-video mpeg2-ts mpeg2-ps* mpeg2-video
-
-
- the mpegdemux element might specify the type of the input pad as
- one of video/mpeg1-system and video/mpeg2-ts
-
-
-
-7. definition of types
-----------------------
-
-A plugin will provide the following information to the type system:
-
- - a mime type string describing the hierarchy and where the types
- they provide are located in that hierarchy.
-
- - typefind functions for some of the types.
-
- - extensions for some of the types
-
-We will propose a syntax to define the type hierarchy
-
-a) equivalent types :
-
- separated with a | sign
-
- ( audio/mp3 | audio/mpeg )
-
-b) type hierarchy :
-
- in braces:
-
- ( audio/mp3 ( audio/mp3-frame))
-
-c) multi-type pads
-
- ( mpegdemux ( video/mpeg1-system + video/mpeg2-ps) )
-
- the pad will have type mpegdemux which is the parent for
- type video/mpeg1-system or video/mpeg2-ps
-
- mpegdemux
- / \
- video/mpeg1-system video/mpeg2-ps
-
-
-
-once the type hierarchy has been registered, the typeid of each
-element can be obtained with:
-
- guint16 gst_type_find_by_mime (gchar *mime)
-
-extra typefind functions and/or extensions can be added to the
-typeid afterwards.
-
-
-8. type matching
-----------------
-
-The more specific typefind functions, the functions associated with
-types in the leaf nodes, are tried first.
-
-when a specific type has been found ex. video/mpeg1-system elements
-that can handle this type or one of its parents are selected:
-
- mpegdemux: mpeg1parse: supermpegdecoder:
-
- video/mpeg video/mpeg video/mpeg
- ! !
- mpegdemux video/mpeg1-system
- !
- video/mpeg1-system
-
-In the above case, also the super mpeg element is selected because it stated
-that it can handle all sorts of video/mpeg data.
-
-
-example 2:
-
- suppose the typefind functions finds an mp3 stream, following elements
- are selected:
-
- libmpg123 mp3parse:
-
- audio/mp3 audio/mp3
-
- mpg123 is not selected because its pad type is too specific (audio/mp3-frame):
-
- mpg123
-
- audio/mp3
- !
- audio/mp3-frame
-
- if the typefind would find a mp3-frame type, all three objects would be selected.
-
-
-8. consideration
-----------------
-
-It is clear that clear indications have to be given to the type hierarchy,
-especially for the top nodes.
-
-The more specific an element is in its mime type specification, the more flexible
-and extendible the plugin system can be.
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
diff --git a/docs/random/types3 b/docs/random/types3
deleted file mode 100644
index 5d84677d22..0000000000
--- a/docs/random/types3
+++ /dev/null
@@ -1,177 +0,0 @@
-SOMEWHAT OUTDATED, design still holds though
---------------------------------------------
-
-
-1. Introduction
----------------
-
-The type system is used to attach meaning to the bytes in a GstBuffer.
-A plugin can decide to add metadata to the GstBuffer, this metadata
-will carry an associated typeid.
-
-Types are also used by the plugins to expose the type of their pads to
-the type system.
-
-Types are essential for autoplugging.
-
-We will explain the inner workings of the type system in this document, as
-well as the API used in the plugins.
-
-2. major types
---------------
-
-major types are identified with mime types and are used to denote a
-family of types.
-
-More specific information about the type is given using properties. This
-will allow us to be very specific without creating a lot of mime types.
-
-3. API
-------
-
-Both a simple array based specification and a real API will be
-provided to build the capabilities.
-
-In the array based approach, we basically build an array of pointers.
-Some macros will be available to specify ranges, boolean values, lists
-and id's. (not sure if this can be done)
-
-#define GST_TYPE_INT_RANGE(a, b) GST_TYPE_RANGE,(a),(b)
-#define GST_TYPE_BOOLEAN(a) GST_TYPE_BOOLEAN,(a)
-#define GST_TYPE_LIST(a...) GST_TYPE_LIST,(##a),NULL
-
-for example:
-
- static GstTypeCapsFactory mpg123_sink_caps[] = {
- "audio/mp3",
- "layer", GST_TYPE_INT_RANGE (1, 3),
- "bitrate", GST_TYPE_INT_RANGE (8, 320),
- "framed", GST_TYPE_BOOLEAN (true),
- NULL
- };
-
-will expand to the array:
-
- static GstTypeCapsFactory mpg123_sink_caps[] = {
- "audio/mp3",
- "layer", GST_TYPE_RANGE,1,3,
- "bitrate", GST_TYPE_RANGE,8,320,
- "famed", GST_TYPE_BOOLEAN,true,
- NULL,
- };
-
-when we register the caps factory, the strings will be converted
-into GQuarks and be stored into a GData Keyed Data List.
-
-This will result in a GstTypeCaps structure:
-
-struct _GstTypeCaps {
- guint16 id; // if of the major type
-
- GData *properties;
-}
-
-4. example using arrays
------------------------
-
-mpg123: an mpeg audio decoder.
-
- // a factory for the major type we use
- static GstTypeFactory mp3factory = {
- "audio/mp3", // major type
- ".mp3 .mp2 .mp1 .mpga", // extensions
- NULL, // typefind function
- };
-
- // capabilities of the sink pad
- static GstTypeCapsFactory mpg123_sink_caps[] = {
- "audio/mp3",
- "layer", GST_TYPE_INT_RANGE (1, 3),
- "bitrate", GST_TYPE_INT_RANGE (8, 320),
- "framed", GST_TYPE_BOOLEAN (true),
- NULL
- };
-
- // capabilities of the source pad
- static GstTypeCapsFactory mpg123_src_caps[] = {
- "audio/raw",
- "rate", GST_TYPE_INT_RANGE (8000, 48000),
- "channels", GST_TYPE_INT_RANGE (1, 2),
- NULL
- };
-
- static GstTypeCaps *sinkcaps = NULL, *rawcaps = NULL;
-
- static gst_mpg123_init (GstMpg123 *mpg123)
- {
- mpg123->sinpad = gst_pad_new ("sink", GST_PAD_SINK);
- gst_element_add_pad (GST_ELEMENT (mpg123), mpg123->sinkpad);
- gst_pad_set_caps (mpg123->sinkpad, sinkcaps);
-
- ...
- }
-
- GstPlugin *plugin_init (GModule *module)
- {
- ...
- plugin = gst_plugin_new ("mpg123");
-
- gst_plugin_add_type_factory (plugin, mp3factory);
-
- ...
- sinkcaps = gst_type_register_caps (mpg123_sink_caps, NULL);
- rawcaps = gst_type_register_caps (mpg123_src_caps, NULL);
- ...
- }
-
-mpeg2dec: an mpeg video decoder that can do mpeg1 and mpeg2.
-
- static GstTypeFactory mpegfactory = {
- "video/mpeg", // major type
- ".mpg .mpeg", // extensions
- NULL, // typefind function
- };
-
- static GstTypeCapsFactory mpeg2dec_sink_caps[] = {
- "video/mpeg",
- "mpegtype", GST_TYPE_LIST (
- GST_TYPE_INT(1),
- GST_TYPE_INT(2),
- ),
- NULL
- };
-
- static GstTypeCapsFactory mpeg2dec_src_caps[] = {
- "video/raw",
- "fourcc", GST_TYPE_LIST (
- GST_TYPE_INT32 (0x32315659),
- GST_TYPE_INT32 (0x32...),
- ),
- "width", GST_TYPE_INT_RANGE (16, 4096),
- "height", GST_TYPE_INT_RANGE (16, 4096),
- NULL
- };
-
- static GstTypeCaps *sinkcaps = NULL, *rawcaps = NULL;
-
- GstPlugin *plugin_init (GModule *module)
- {
- ...
- plugin = gst_plugin_new ("mpeg2dec");
-
- ...
- sinkcaps = gst_type_register_caps (mpeg2dec_sink_caps, NULL);
- rawcaps = gst_type_register_caps (mpeg2dec_src_caps, NULL);
- ...
- }
-
-
-5. capability compatibility
---------------------------
-
-Two pads are compatible if:
-
-- The major types are equal
-- range of the sink pad contains the range of the src pad
-
-
diff --git a/docs/random/use-cases-0.11.txt b/docs/random/use-cases-0.11.txt
deleted file mode 100644
index 6ba1d4dda2..0000000000
--- a/docs/random/use-cases-0.11.txt
+++ /dev/null
@@ -1,21 +0,0 @@
-Use cases for 0.11
-------------------
-
-- improve performance
-
- * incremental caps
- * cleanup caps fields
- * Make miniobject a simple boxed type
- * reverse negotiation with an event
-
-- buffer metadata
-
- * Handle strides for video
- * express crop, scaling, region of interest, ...
- * Better support for foreign objects such as Cairo, OpenGL, OMX, ...
-
-- make dynamic pipelines easier
-
- * sticky event propagation
- * newsegment simplifications
- * per-pad time offsets
diff --git a/docs/random/usecases b/docs/random/usecases
deleted file mode 100644
index 5fa895afe0..0000000000
--- a/docs/random/usecases
+++ /dev/null
@@ -1,44 +0,0 @@
-Use cases for audio applications
---------------------------------
-
-ANALYSIS
-========
-
-* find start and end -x dB boundary
-
-* find overall volume for complete track (peak/RMS/average)
-
-* do level-triggered playback
-
-CUE FILES
-=========
-
-Given a long audio file and a .cue file:
-- an element should be able to read an audio input stream and a cue input
- stream and send new-media events based on the .cue file information
-- a audiocdsink should be able to record this stream straight to disk
-
-Use cases for core functionality
---------------------------------
-
-THREADS
-=======
-
-* { sinesrc } ! { queue ! osssink }
- - play thread
- - pause input thread
- - output thread should keep playing as long as queue not empty
- - play input thread again
- - no gaps should be heard
-
-Use cases for video applications
---------------------------------
-
-THUMBNAILING
-============
-- load file
-- figure out if it's playable
-- ask length of file
-- seek to middle
-- snapshot one frame
-- save to disk
diff --git a/docs/random/vis-transform b/docs/random/vis-transform
deleted file mode 100644
index aa80a535ba..0000000000
--- a/docs/random/vis-transform
+++ /dev/null
@@ -1,78 +0,0 @@
-Duh, an 'easy' way to replicate Giess's behavior:
-
-For each frame, you have to mutate it by a transform matrix. This is
-easy, thought not cheap. First you precalculate the transform matrix how
-you want it, based on whatever rotations or whatever you want.
-
-The data stored in each spot on the matrix tells you how to transform a
-single pixel. The simple case is dx,dy, where both are relatively small.
-The probably ought to be a byte in any case, so you can scale the
-transform matrix on slow machines. A more complex case is some trick
-whereby a single pixel ends up splattered in several places. Idea below.
-
-The matrix consists of some number of 8bit arrays of the same size as the
-image. They'd probably be line-interleaved or better to help with cache
-effects (which are VERY serious here). Each channel represents some
-aspect of the transform. The first two would likely be dx and dy, the
-third might be a multiplier if that wasn't done statically.
-
-The idea: any number of transform sets could be applied, given available
-processing power. Just set the static scalar or the multiplier matrices
-so you don't completely swamp the output pixels.
-
-Note that this is fastest in 8-bit, but theoretically could be applied to
-32 bit. 15 and 16 are hard, since you can't easily apply the multipliers
-unless they're 1/2^n, and even then it's significantly heavier (you'd have
-to mask the top n bits of each color out).
-
-This SCREAMS for MMX, in case you haven't figured it out yet.
-Unfortunately, MMX is only directly useful for the scalar matrix, unless
-you do a trick where all the pixels in that fit in 64 bits (8 8bit, 4
-16bit, or 2 32bit) are always moved in a group. This is very possible,
-and might be a significant perf increase by being able to use MMX all the
-way through. Otherwise you have to place each pixel by extracting the MMX
-stuff back into normal registers, and that just plain sucks.
-
-A pseudo-C implementation:
-
------ BEGIN -----
-gint x,y; /* image x and y size */
-guchar old_image[x][y]; /* original image */
-guchar new_image[x][y]; /* new image */
-gchar x_xform[x][y]; /* dx matrix */
-gchar y_xform[x][y]; /* dy matrix */
-guchar s_xform[x][y]; /* intensity scalar matrix */
-guchar scalar; /* global scalar */
-
-gint i,j; /* indixes */
-gulong p; /* pixel value in question */
-guchar u,v,w; /* modifier variables */
-
-/* clear the new image, we don't want anything getting in the way */
-/* NOT NECESSARILY A GOOD THING, THOUGH */
-memset(new_image,0,x*y);
-
-/* loop through all the lines in the image */
-for (j=0;j<y;j++) {
- /* loop through all the pixels in the line */
- for (i=0;i<x;i++) {
- p = old_image[i][j];
- u = x_xform[i][j];
- v = y_xform[i][j];
- w = s_xform[i][j];
- new_image[i+u][j+v] = (guchar)((p<<14) / (w * scalar));
- }
-}
------ END -----
-
-Note that the above really, *REALLY* sucks performance-wise. Throw it a
-80x60 image and it'll swamp my poor laptop. Also note that I simply set
-the pixel value, not merge it. That means you'd better be sure your
-transform matrix doesn't have overlapping destinations.
-
-Other notes about the above code: x_xform and y_xform are signed chars,
-which means pixels can move in all directions. The intensity matrix is
-unsigned, with a range from 0 to 255, so is the global scalar. Note the
-shift of 14bits (2 * 7bits), then divide by each. That means identity for
-both scalars is at 128. The FP range of each is thus 0.0 to 2.0. Very
-handy.
diff --git a/docs/random/wingo/porting-plugins-to-0.9 b/docs/random/wingo/porting-plugins-to-0.9
deleted file mode 100644
index dc56f73323..0000000000
--- a/docs/random/wingo/porting-plugins-to-0.9
+++ /dev/null
@@ -1,32 +0,0 @@
--*- outline -*-
-
-
-Some collected notes porting plugins to GStreamer 0.9.
-
-
-* General notes
-
-Consider using the base classes. There are base classes for sources,
-sinks, and "transformers", which are plugins that have the same caps on
-their source and sink sides.
-
-These base classes offer interfaces that are much more appropriate to
-the function of specific plugins than the general GstElement interface.
-
-
-* Functions that changed
-
-** gst_pad_use_explicit_caps
-** gst_pad_set_explicit_caps
-
-Not needed; just set the caps on the first buffer you push out.
-
-** gst_element_set_loop_function
-
-Use gst_pad_set_loop_function. Note that loop-based elements participate
-more in the scheduling process in 0.9, so you'll need to add hooks to
-the activate function to start up a task.
-
-refcounting:
-_get_ accessors
-GValue, GstObject, and refcounts (arrrrgh) \ No newline at end of file
diff --git a/docs/random/wingo/threadsafe-properties b/docs/random/wingo/threadsafe-properties
deleted file mode 100644
index 7e9306f9ff..0000000000
--- a/docs/random/wingo/threadsafe-properties
+++ /dev/null
@@ -1,76 +0,0 @@
--*- Mode: text -*-
-
-Sometimes it's difficult to use GStreamer in real applications where the GUI and
-the GStreamer pipeline are in different threads. You have to somehow make sure
-that the object is not modifying its properties while you get and set them. This
-document is a brief outline of a potential fix to this issue.
-
-* When is it safe to get or set properties?
-
-Well, it is only safe to do so when other threads are not accessing an object.
-So, from the main thread, it is only safe to query or modify properties on an
-object when it is not in the RUNNING state, because when it is RUNNING its
-properties are potentially changing in the thread of execution.
-
-The only place that it is safe to get or set properties while in the RUNNING
-state is from within the thread of execution. Thus, something within the
-iteration loop of the thread must check to see if there are pending property
-changes or queries. There are two places this could be done, from within
-gstthread itself and from within the scheduler. Doing it from gstthread sounds
-like a good idea because it keeps more code out of the scheduler, but is also
-bad in a way because it requires a gstthread-specific api to proxy prop_set and
-prop_get. Setting it in the scheduler sounds good because of its finer
-granularity, but might be bad because it would clutter the scheduler a bit more.
-
-I propose to go with the scheduler-based solution based on system response time
-and out of potentials for integration with gst_clock_wait.
-
-* Implementation
-
-We do want to preserve some measure of generality, however. Considering that
-more threadsafety issues could pop up in the future, it would be nice to have
-one function to call from the scheduler, in the interests of code
-simplification. This function will be present in some elements and not in
-others, and will be modified at run time, so we will make it a function pointer
-within the GstElement struct.
-
-struct _GstElement {
- ...
- void (*pre_run_func) (GstElement *);
- void (*post_run_func) (GstElement *);
-}
-
-Only the managing bin of an element is allowed to set that function, because
-presumably that bin would know something about how to schedule the element.
-Then, in the scheduler, before we call chain functions and before we switch into
-loop functions:
-
-if (element->pre_run_func)
- element->pre_run_func (element);
-
-Then, to get or set properties, we use the new functions gst_element_get or
-gst_element_set. _set would add the property name and a gvalue onto a queue
-(probably a GAsyncQueue). Then the pre_run_func would go ahead and set the
-properties. _get is a little more tricky; _set doesn't hardly block at all,
-although it's not instantaneous. With _get though, you really don't know what
-the properties are until you query them. The best thing would be to connect to
-the ::notify signal, which executes within the thread of interest. However, say
-you really want to use _get. Hmm. I think that it would have to block. On what?
-Well, probably on an element's mutex. So it seems we might need a
-post_run_func too, to unlock the mutex. We can use the GstObject lock for this.
-
-But we need a little more. How do we know whether or not just calling
-g_object_get/set is ok? I'm thinking this whole prop set/get proxy thing should
-not be abstracted away, that it should be contained in gstelement.c. There are
-more kinds of bins that need threadsafety than just gstthread (I'm thinking
-about jack here). So, we can add on a GAsyncQueue *prop_set_queue; to the
-GstElement struct, and only initialize it in certain managing bins. Then, we can
-set a flag on gstelement, GST_ELEMENT_USE_THREADSAFE_PROPERTIES. If this flag is
-set (by the managing bin), do all this complicated mess; otherwise use the
-gobject native functionality.
-
-So, this is the plan. We'll see how the implementation goes. This should make MT
-gst programming much easier.
-
-wingo.
-25 May 2002.
diff --git a/docs/random/wingo/without-factories b/docs/random/wingo/without-factories
deleted file mode 100644
index 07e38857e3..0000000000
--- a/docs/random/wingo/without-factories
+++ /dev/null
@@ -1,62 +0,0 @@
--*- outline -*-
-
-* Creating Elements Without Factories
-
-** The purpose of factories
-
-On a typical GStreamer system, there are approximately 6.022*10^23
-plugins. GStreamer knows about all of them because of the registry. The
-goal is to avoid initializing each one of them, when maybe for your
-application you only need one or two.
-
-The problem becomes, how do you create an instance of the plugin? The
-normal way to instantiate a class is via g_object_new (TYPE, ARGS...).
-In the case that the plugin isn't loaded, you don't know its type, and
-can't even get it from the type name.
-
-Element factories exist to solve this problem by associating names (like
-"sinesrc" or "identity") with certain types that are provided by the
-plugin. Then when the user asks for "sinesrc", the appropriate plugin is
-loaded, its types are initialized, and then gst_element_factory_create
-creates the object for you.
-
-** Why not factories?
-
-To review, factories (1) allow plugins to remain unloaded if not
-necessary, and (2) make it easy to create elements.
-
-If you are writing an application that has custom elements (as is the
-case with most serious applications), you will probably have the plugin
-loaded up already, and you will have access to the type of the element.
-To muck about creating a plugin for the app, registering the element
-with the plugin, and then creating it with the element factory API
-actually takes more work than the normal way.
-
-** g_object_new
-
-So you want to avoid factories. To create objects with a simple
-g_object_new call is our strategy. However, to preserve the same
-semantics as gst_element_factory_create, we need to know what else is
-needed to initialize a GStreamer element.
-
-The other things that gst_element_factory_create does are as follows:
-
-*** Sets the ->elementfactory member on the element class
-
-Note that anything trying to get the factory won't work (e.g.
-gst_element_get_factory). Thankfully this is less of a problem after the
-0.7 plugin system changes.
-
-*** Initializes the name of the element
-
-To do this ourselves, we either call gst_object_set_name, or when we
-set the "name" property when creating the object.
-
-** Summary
-
-To create a GStreamer element when you know the type, you can just use
-
-g_object_new (get_type_of_my_element (),
- "name", the_name_you_want_possibly_null,
- ... any other properties ...
- NULL);
diff --git a/docs/random/zaheerm/dvb-interface.txt b/docs/random/zaheerm/dvb-interface.txt
deleted file mode 100644
index 1cc479bffb..0000000000
--- a/docs/random/zaheerm/dvb-interface.txt
+++ /dev/null
@@ -1,26 +0,0 @@
-Interface:
-- type (string like caps) channels/x-mpegts-pat
-- GstStructure to describe the what the list is, name, where from etc. (how many entries you can select, if multiple tried, first one picked)
-- method to list entries (channels)
-- method to get current entries
-- method to set list of entries
-- each entry contains an id (32 bit int say), GstStructure (with additional stuff)
-
-signals to say when entry list changes
- to say when current entry changes
-
-So for a transport stream, this interface would be exported and it could contain a list of interfaces...eg it would export PAT, and PAT list would contain interfaces in the GstStructure for the PMT.
-
-dvbsrc -> list of frequencies
-flutsdemux -> list of programs (PAT) and list of PMT streams per program
-
-get list of frequencies, select frequency
-get list of programs, select programs
-get list of streams, select streams
-
-only give program details in PAT list, when we have seen the PMT for that program.
-
-possibly another interface to select multiple list interfaces from an element. (interface selector interface)
-
---
-other option is to tag