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b51bb8f341
Original commit message from CVS: add hacking notes
131 lines
6.6 KiB
Text
131 lines
6.6 KiB
Text
THE GOAL
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--------
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What we are trying to achieve:
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satisfy:
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patching of CVS checkout using our patch files placed in our CVS
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passing of
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make
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make distcheck
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non-srcdir build (ie, mkdir build; cd build; ../configure; make)
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THE SETUP
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---------
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There is a "mirror" root CVS module that contains "ffmpeg".
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This directory contains a vendor-branch checkout of upstream FFmpeg CVS
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of a given day.
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On head, the following things have been commited on top of this:
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* patches/, which is a directory with a set of patches, and a series file
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listing the order, as generated by quilt
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* .pc/, which is a tree of files that quilt uses to keep control of its state.
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It contains a list of applied patches, and one directory per patch,
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containing a tree of hardlinked files that were added to the patchset, and
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a .pc file listing all files part of the patchset.
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* the result of having all these patches commited (ie, quilt push -a) to the
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ffmpeg tree.
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Both the actually patched CVS ffmpeg code as well as the .pc dir need to be
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commited to CVS so the state of quilt wrt. the source is in sync.
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THE WAY
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-------
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- If you want to hack on our copy of the FFmpeg code, there are some basic
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rules you need to respect:
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- you need to use quilt. If you don't use quilt, you can't hack on it.
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- we separate patches based on the functionality they patch, and whether
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or not we want to send stuff upstream. Make sure you work in the right
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patch. use "quilt applied" to check which patches are applied.
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- before starting to hack, run cvs diff. There should be NO diffs, and
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NO files listed with question mark. If there are, somebody before you
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probably made a mistake. To manage the state correctly, it is vital that
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none of the files are unknown to CVS.
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- if you want to add a file to a patchset, you need to:
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- be in the right patchset
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- quilt add (file)
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- cvs add .pc/(patchsetname)/(file)
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- cvs commit .pc/(patchsetname) (to update the state of quilt in cvs)
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- edit the file
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- quilt refresh
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- quilt push -a (This one is IMPORTANT, otherwise you'll have a huge diff)
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- cvs commit
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- if you want to add a patchset, you need to:
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- go over the procedure with thomas to check it's correct
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- decide where in the stack to put it. ask for help if you don't know.
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- go there in the patch stack (use quilt pop/push)
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- quilt new (patchsetname).patch (don't forget .patch !)
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- quilt add (files)
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- cvs add .pc/(patchsetname) the whole tree
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- cvs commit .pc/(patchsetname)
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- quilt refresh
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- quilt push -a
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- cvs commit
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- cvs diff (to check if any of the files are unknown to CVS; if they are,
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you need to add them to CVS)
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THE PLUGIN
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----------
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Some notes on how ffmpeg wrapping inside GStreamer currently works:
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* gstffmpeg{dec,enc,demux,mux}.c are wrappers for specific element types from
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their ffmpeg counterpart. If you want to wrap a new type of element in
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ffmpeg (e.g. the URLProtocol things), then you'd need to write a new
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wrapper file.
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* gstffmpegcolorspace.c is a wrapper for one specific function in ffmpeg:
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colorspace conversion. This works different from the previously mentioned
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ones, and we'll come to that in the next item. If you want to wrap one
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specific function, then that, too, belongs in a new wrapper file.
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* the important difference between all those is that the colorspace element
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contains one element, so there is a 1<->1 mapping. This makes for a fairly
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basic element implementation. gstffmpegcolorspace.c, therefore, doesn't
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differ much from other colorspace elements. The ffmpeg element types,
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however, define a whole *list* of elements (in GStreamer, each decoder etc.
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needs to be its own element). We use a set of tricks for that to keep
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coding simple: codec mapping and dynamic type creation.
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* ffmpeg uses CODEC_ID_* enumerations for their codecs. GStreamer uses caps,
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which consists of a mimetype and a defined set of properties. In ffmpeg,
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these properties live in a AVCodecContext struct, which contains anything
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that could configure any codec (which makes it rather messy, but ohwell).
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To convert from one to the other, we use codec mapping, which is done in
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gstffmpegcodecmap.[ch]. This is the most important file in the whole
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ffmpeg wrapping process! It contains functions to go from a codec type
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(video or audio - used as the output format for decoding or the input
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format for encoding), a codec id (to identify each format) or a format id
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(a string identifying a file format - usually the file format extension)
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to a GstCaps, and the other way around.
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* to define multiple elements in one source file (which all behave similarly),
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we dynamically create types for each plugin and let all of them operate on
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the same struct (GstFFMpegDec, GstFFMpegEnc, ...). The functions in
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gstffmpeg{dec,enc,demux,mux}.c called gst_ffmpeg*_register() do this.
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The magic is as follows: for each codec or format, ffmpeg has a single
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AVCodec or AV{Input,Output}Format, which are packed together in a list of
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supported codecs/formats. We simply walk through the list, for each of
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those, we check whether gstffmpegcodecmap.c knows about this single one.
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If it does, we get the GstCaps for each pad template that belongs to it,
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and register a type for all of those together. We also leave this inside
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a caching struct, that will later be used by the base_init() function to
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fill in information about this specific codec in the class struct of this
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element (pad templates and codec/format information). Since the actual
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codec information is the only thing that really makes each codec/format
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different (they all behave the same through the ffmpeg API), we don't
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really need to do anything else that is codec-specific, so all other
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functions are rather simple.
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* one particular thing that needs mention is how gstffmpeg{mux,demux}.c and
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gstffmpegprotocol.c interoperate. ffmpeg uses URLProtocols for data input
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and output. Now, of course, we want to use the *GStreamer* way of doing
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input and output (filesrc, ...) rather than the ffmpeg way. Therefore, we
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wrap up a GstPad as a URLProtocol and register this with ffmpeg. This is
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what gstffmpegprotocol.c does. The URL is called gstreamer://%p, where %p
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is the address of a GstPad. gstffmpeg{mux,demux}.c then open a file called
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gstreamer://%p, with %p being their source/sink pad, respectively. This
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way, we use GStreamer for data input/output through the ffmpeg API. It's
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rather ugly, but it has worked quite well so far.
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* there's lots of things that still need doing. See the TODO file for more
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information.
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