MultimediaWiki - Recent changes [en]

Small FFmpeg Tasks

Ce — Wed, 28 Mar 2012 07:37:54 GMT

/* FLIC work */ Implemented by Paul B Mahol

Revision as of 07:37, 28 March 2012		Current revision
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	Codec copy of VC1 from m2ts currently doesn't work. Either extend the VC1 parser to output/fix timestamps, or fix the timestamps from m2ts demuxing.		Codec copy of VC1 from m2ts currently doesn't work. Either extend the VC1 parser to output/fix timestamps, or fix the timestamps from m2ts demuxing.

-	=== FLIC work ===

-	Revise the [[Flic Video]] decoder at libavcodec/flicvideo.c to support video transported in AVI files while making sure that data coming from the usual FLI files still works. 'AFLC' FourCC samples are linked from the [[Flic Video]] page.

	=== flip flag for upside-down codecs ===		=== flip flag for upside-down codecs ===

Libav Summer Of Code 2012

Jumpyshoes — Wed, 28 Mar 2012 05:30:29 GMT

Update asm project to be more student friendly

Revision as of 05:30, 28 March 2012		Current revision
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	1st tier project proposals are project ideas that are reasonably well defined '''AND''' have a mentor volunteered.		1st tier project proposals are project ideas that are reasonably well defined '''AND''' have a mentor volunteered.

-	=== Assembly Testing Framework ===	+	=== Assembly Unit Testing Framework ===
-		+	* Libav has a lot of assembly and not enough tests for it. Your job is to write a unit testing framework for assembly.
-	FFmpeg/Libav use an extensive amount of assembly to optimize video and audio decoding. For scientific applications and high-assurance systems (e.g. a spacecraft), accuracy and speed are vital. Assembly is difficult to code, and lends itself to mistakes. As FFmpeg/Libav has no framework for testing assembly functions, development of assembly is often hindered by each developer having to create his or her own tests. In addition, functions are not tested for speed. While FATE covers some functions, there are many that simply aren't tested. The samples in FATE do not cover all features in every decoder that has assembly functions (e.g. several 10-bit H.264 predict functions are not used in any FATE sample). FATE also does not test functions in isolation, which impedes assembly development. In addition, FATE might not cover all assembly versions of a function if multiple different levels of optimizations exist (e.g. MMX, MMX2, SSE, SSE2, SSSE3, SSE4, AVX, etc).	+
-		+
-	* Qualification task: describe how you will implement the system (probably based on x264's checkasm). It should test accuracy, speed, and be extensible. Also, contact me via IRC.	+
-	* Actual task: implement the system. At least all H.264 assembly functions should be covered and the framework should be easily extensible.	+
	* The framework should work across all supported architectures and operating systems.		* The framework should work across all supported architectures and operating systems.
	* The framework should measure exactly how fast an individual function is (e.g. using START/STOP_TIMER).		* The framework should measure exactly how fast an individual function is (e.g. using START/STOP_TIMER).
	* The framework should be able to test functions in isolation.		* The framework should be able to test functions in isolation.
	* x264's checkasm can be used as a reference.		* x264's checkasm can be used as a reference.
		+	* The qualification task will be to implement at least one unit test and have an idea of how to do the rest.

	''Mentor: Daniel Kang'' (Jumpyshoes on #libav-devel@chat.freenode.net; daniel.d.kang@gmail.com -- ping me on IRC and email me).		''Mentor: Daniel Kang'' (Jumpyshoes on #libav-devel@chat.freenode.net; daniel.d.kang@gmail.com -- ping me on IRC and email me).
-

	=== HEVC/H265 decoder ===		=== HEVC/H265 decoder ===

FFmpeg Summer of Code 2012

Reimar — Tue, 27 Mar 2012 19:02:52 GMT

Revision as of 19:02, 27 March 2012		Current revision
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	==Timeline==		==Timeline==
	March 12th-16th: Project application evaluation.		March 12th-16th: Project application evaluation.
		+
		+	NOTE: FFmpeg has not been accepted this year, so unless you want to do any of these, the small tasks or qualification tasks just for fun...

	==Contacting Developers==		==Contacting Developers==

Small FFmpeg Tasks

Pbm — Tue, 27 Mar 2012 18:06:48 GMT

/* FLIC work */

Revision as of 18:06, 27 March 2012		Current revision
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	=== FLIC work ===		=== FLIC work ===

-	Revise the [[Flic Video]] decoder at libavcodec/flicvideo.c to support video transported in AVI or MOV files while making sure that data coming from the usual FLI files still works. 'AFLC' and 'flic' FourCC samples are linked from the [[Flic Video]] page.	+	Revise the [[Flic Video]] decoder at libavcodec/flicvideo.c to support video transported in AVI files while making sure that data coming from the usual FLI files still works. 'AFLC' FourCC samples are linked from the [[Flic Video]] page.

	=== flip flag for upside-down codecs ===		=== flip flag for upside-down codecs ===
Line 135:		Line 135:

	This would add a free lossless intra-frame RGB codec for all supported platforms (most important OS X + Windows) which is often asked for video editing in video forums (e.g. slashcam.de)		This would add a free lossless intra-frame RGB codec for all supported platforms (most important OS X + Windows) which is often asked for video editing in video forums (e.g. slashcam.de)
-
-	=== Metal Gear Solid Video format demuxer ===
-	Write a demuxer to play video files harvested from the game Metal Gear Solid: The Twin Snakes. The format is described on the wiki page [[Metal Gear Solid VP3]] (which also contains links to samples). This page is based on observations and conjecture, so remember to engage the mailing list or IRC with questions.

	=== [[IFF#ANIM\|IFF ANIM]] decoder ===		=== [[IFF#ANIM\|IFF ANIM]] decoder ===

RTSP

Compn — Tue, 27 Mar 2012 17:36:04 GMT

add msrtsp spec and rtspdump program

Revision as of 17:36, 27 March 2012		Current revision
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	* Protocolhelper: rtsp://		* Protocolhelper: rtsp://
	* X-Purevoice Spec: http://www.faqs.org/rfcs/rfc2658.html		* X-Purevoice Spec: http://www.faqs.org/rfcs/rfc2658.html
		+	* MS RTSP Spec: http://msdn.microsoft.com/en-us/library/cc245238(PROT.10).aspx
		+	* MS rtspdump program: http://bisqwit.iki.fi/source/ms-rtsp-dump/

	'''Real Time Streaming Protocol''' (RTSP) is a client-server multimedia presentation control protocol, designed to address the needs for efficient delivery of streamed multimedia over IP networks. It leverages existing web infrastructure (for example, inheriting authentication and PICS from [[HTTP]]) and works well both for large audiences as well as single-viewer media-on-demand. RealNetworks, Netscape Communications and Columbia University jointly developed RTSP within the MMUSIC working group of the Internet Engineering Task Force (IETF). In April, 1998, it was published as a Proposed Standard by the IETF		'''Real Time Streaming Protocol''' (RTSP) is a client-server multimedia presentation control protocol, designed to address the needs for efficient delivery of streamed multimedia over IP networks. It leverages existing web infrastructure (for example, inheriting authentication and PICS from [[HTTP]]) and works well both for large audiences as well as single-viewer media-on-demand. RealNetworks, Netscape Communications and Columbia University jointly developed RTSP within the MMUSIC working group of the Internet Engineering Task Force (IETF). In April, 1998, it was published as a Proposed Standard by the IETF

Scream Tracker 3 Module

Nickysn — Mon, 26 Mar 2012 14:23:51 GMT

/* Ixy (%) */ documented tremor

Revision as of 14:23, 26 March 2012		Current revision
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	===Ixy (%)===		===Ixy (%)===
-	Tremor. ''TODO: this effect is weird.''	+	Tremor.

-	(Looking through what's released of the ST3 playback source code, tremor is not retriggered if there's a Dxy effect, or something like that, but this needs to be clarified.)	+	* "On" time is '''x + 1''' ticks, "off" time is '''y + 1''' ticks
		+	* This effect is updated on '''every tick'''.
		+	* Implemented with two decrementing counters per channel - the "on" counter and the "off" counter.
		+	* On each tick, if the "on" counter is greater than zero, it is decremented and if it reaches zero, the current volume is set to 0 and the "off" counter is set to the "off" time (y + 1). If the "on" counter was zero in the beginning of the update procedure, then the "off" counter is decremented and if it reached zero (or became less than zero), '''the current volume is set to the stored volume''' and the "on" counter is set to the "on" time (x + 1).
		+	* '''The "on" and "off" counters are never reset''', except in the tremor update procedure described above. Scream Tracker doesn't even reset them on playback start. Only on tracker startup are they reset.
		+	* If the current volume was 0 at the end of the effect and there is no tremor effect on the next row, '''the current volume stays 0'''. It isn't reset back to the stored volume or its previous value from before the tremor effect.
		+	* '''The stored volume isn't modified by this effect.'''

	===Jxy (%)===		===Jxy (%)===

Libav Summer Of Code 2012

Kierank — Mon, 26 Mar 2012 10:58:02 GMT

Revision as of 10:58, 26 March 2012		Current revision
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	* It does not need to be SIMD-optimized, but preferably has relevant functions separated in DSP contexts so they're easy to optimize later on.		* It does not need to be SIMD-optimized, but preferably has relevant functions separated in DSP contexts so they're easy to optimize later on.
	* This task will depend on the current standardisation status of HEVC. Final publication is in 2013 but there may be parts which will not change much anyway.		* This task will depend on the current standardisation status of HEVC. Final publication is in 2013 but there may be parts which will not change much anyway.
-	* Draft spec: http://phenix.it-sudparis.eu/jct/doc_end_user/documents/8_San%20Jose/wg11/JCTVC-H1003-v21.zip	+	* Draft spec: http://www.itscj.ipsj.or.jp/sc29/open/29view/29n12612c.htm
	* HM (draft reference decoder) git mirror: http://hevc.kw.bbc.co.uk/git/w/jctvc-hm.git		* HM (draft reference decoder) git mirror: http://hevc.kw.bbc.co.uk/git/w/jctvc-hm.git

Libav Summer Of Code 2012

Ronald S. Bultje — Sun, 25 Mar 2012 16:28:35 GMT

/* HEVC/H265 decoder */

Revision as of 16:28, 25 March 2012		Current revision
Line 44:		Line 44:
	* It does not need to be SIMD-optimized, but preferably has relevant functions separated in DSP contexts so they're easy to optimize later on.		* It does not need to be SIMD-optimized, but preferably has relevant functions separated in DSP contexts so they're easy to optimize later on.
	* This task will depend on the current standardisation status of HEVC. Final publication is in 2013 but there may be parts which will not change much anyway.		* This task will depend on the current standardisation status of HEVC. Final publication is in 2013 but there may be parts which will not change much anyway.
		+	* Draft spec: http://phenix.it-sudparis.eu/jct/doc_end_user/documents/8_San%20Jose/wg11/JCTVC-H1003-v21.zip
		+	* HM (draft reference decoder) git mirror: http://hevc.kw.bbc.co.uk/git/w/jctvc-hm.git

	''Mentor: [[User:Ronald_S._Bultje\|Ronald S. Bultje]]''		''Mentor: [[User:Ronald_S._Bultje\|Ronald S. Bultje]]''
-

	=== DTS-HD decoder ===		=== DTS-HD decoder ===
Line 114:		Line 115:
	* [http://en.wikipedia.org/wiki/Digital_Negative Wikipedia Article]		* [http://en.wikipedia.org/wiki/Digital_Negative Wikipedia Article]
	* [http://wwwimages.adobe.com/www.adobe.com/content/dam/Adobe/en/products/photoshop/pdfs/dng_spec.pdf Specification]		* [http://wwwimages.adobe.com/www.adobe.com/content/dam/Adobe/en/products/photoshop/pdfs/dng_spec.pdf Specification]
-	* DNG samples can be created from other raw formats using the free [http://en.wikipedia.org/wiki/Digital_Negative DNG Converter] program	+	* DNG samples can be created from other raw formats using the free [http://www.adobe.com/products/photoshop/extend.displayTab2.html DNG Converter] program
	* A good place to find raw camera samples is http://www.imaging-resource.com		* A good place to find raw camera samples is http://www.imaging-resource.com

Line 190:		Line 191:
	''Mentor: Kieran Kunhya (kierank)''		''Mentor: Kieran Kunhya (kierank)''

-	=== 2nd Tier Project Proposals ===	+	== 2nd Tier Project Proposals ==

	=== DTS-LBR decoder ===		=== DTS-LBR decoder ===

Libav Summer Of Code 2012

Kierank — Wed, 21 Mar 2012 14:44:57 GMT

Revision as of 14:44, 21 March 2012		Current revision
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	''Mentor: Benjamin Larsson''		''Mentor: Benjamin Larsson''
-

	=== MPEG-4 ALS Roundup ===		=== MPEG-4 ALS Roundup ===
Line 190:		Line 189:

	''Mentor: Kieran Kunhya (kierank)''		''Mentor: Kieran Kunhya (kierank)''
		+
		+	=== 2nd Tier Project Proposals ===
		+
		+	=== DTS-LBR decoder ===
		+
		+	This is the DTS equivalent to E-AC3 but not technically related to DTS/DTS-HD. It is found in the following specification: http://www.etsi.org/deliver/etsi_ts/102100_102199/102114/01.03.01_60/ts_102114v010301p.pdf). The task is to find a way of making the official
		+	decoder decode just the LBR stream (and not mix it into the main audio) and use that to verify decoder compliance of the decoder you wrote.

	[[Category:FFmpeg]]		[[Category:FFmpeg]]

Small FFmpeg Tasks

Compn — Tue, 20 Mar 2012 18:59:43 GMT

add evrc task

Revision as of 18:59, 20 March 2012		Current revision
Line 234:		Line 234:
	Following marker codes are not supported by our mjpeg decoder:		Following marker codes are not supported by our mjpeg decoder:
	DAC, SOF9, SOF10, SOF11, SOF13, SOF14 and SOF15.		DAC, SOF9, SOF10, SOF11, SOF13, SOF14 and SOF15.
		+
		+	=== [[Enhanced Variable Rate Codec]] decoder ===
		+	samples and specs available.

	== Reverse Engineering Tasks ==		== Reverse Engineering Tasks ==

FFmpeg Summer of Code 2012

Compn — Tue, 20 Mar 2012 17:28:16 GMT

/* 1st Tier Project Proposals */ add gwenole's ideas, useful for future ideas.

Revision as of 17:28, 20 March 2012		Current revision
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	''Baptiste has also offered to mentor.''		''Baptiste has also offered to mentor.''
		+
		+	=== hwaccel: global architecture ===
		+
		+	Revisit my older v2 proposal to completely get rid of HW pixel formats. The advantages were: this simplifies things, clean ups code, allows to move more code to the codec library (FFmpeg), allow for fallbacks to SW decoding (at the stream level if HW cannot meet the requirements, not while decoding a particular stream). The key point was to separate identification of the pixel format from the underlying HW accelerator. This will also allow for reading HW surfaces back, when requested if the selected pixel format.
		+
		+	Add encoding and post processing infrastructure.
		+
		+	* VA-API (hwaccel)
		+
		+	:Codecs related:
		+	+ Add support {,M}JPEG decoding
		+	+ Add support for VC-1 interlaced acceleration
		+	+ Add support for H.264 interlaced
		+
		+	:Tools related:
		+	:Add VA support to ffplay (see my older patches)
		+	+ requires some VO infrastructure work
		+	+ requires hwaccel hooks
		+	+ requires enabling vaapi, dxva, bcm, tidsp, whatever
		+
		+	* Add VA support to ffmpeg
		+	+ requires VA/DRI API, for no X dependency
		+	+ dependencies: hwaccel v2 to construct pipelines (HW->HW, HW->SW, SW->HW, etc.), or at least allow for VA surface readback wherever necessary.
		+
		+	* VDPAU (hwaccel)
		+
		+	Update and push my older hwaccel-based VDPAU code. It should still work. :)
		+	:Priority: hwaccel v2 infrastructure work first, ffplay enabling second.
		+
		+	''Mentor: Gwenole Beauchesne''

	=== glplay ===		=== glplay ===

User:Ronald S. Bultje

Ronald S. Bultje — Sat, 17 Mar 2012 17:15:05 GMT

Revision as of 17:15, 17 March 2012		Current revision
Line 8:		Line 8:

	In daily life, I work for Google as part of the WebM team.		In daily life, I work for Google as part of the WebM team.
		+
		+	=Contact:=
		+	*Email: rsbultje (at) gmail (dot) com
		+	*IRC: BBB on #libav-devel (irc.freenode.net)

Libav Summer Of Code 2012

Kierank — Sat, 17 Mar 2012 15:12:11 GMT

/* Swscale Cleanup and Additions */

Revision as of 15:12, 17 March 2012		Current revision
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	''Mentor: [[User:lu_zero\|Luca Barbato]]''		''Mentor: [[User:lu_zero\|Luca Barbato]]''
		+
		+	=== Swscale Cleanup and Additions ===
		+
		+	Libswscale is an open source library for colourspace conversion. However, it requires significant cleanup to allow
		+	it to support modern features. Knowledge of x86 assembly is preferred.
		+
		+	* Support for more interleaved colourspaces. This is useful because interleaved chroma provides improved cache coherency and would form the basis for future optimisations in libav's decoders.
		+	* Support for the vast number of packed pixel formats (e.g r210, v210). Many are currently implemented as decoders but it is agreed that all uses of such formats are effectively colourspace conversions, they should be included in swscale. This task will involve collaborating with the student of "Adobe DNG Decoder" (if applicable) to rewrite the pixel format headers to support more complex pixel formats.
		+	* Other general cleanup
		+
		+	''Mentor: Kieran Kunhya (kierank)''

	[[Category:FFmpeg]]		[[Category:FFmpeg]]

Real Lossless Codec

Kostya — Sat, 17 Mar 2012 08:54:19 GMT

fixes

Revision as of 08:54, 17 March 2012		Current revision
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	val1 = t / 21;		val1 = t / 21;
	val2 = t - val1 * 21;		val2 = t - val1 * 21;
-	val1 = extend_code(val1, 10, bits);	+	val1 = extend_code(val1, 10, 0);
-	val2 = extend_code(val2, 10, bits);	+	val2 = extend_code(val2, 10, 0);
-	dst[i] = val1;	+	if (bits == 0) {
-	dst[i + 1] = val2;	+	dst[i] = val1;
		+	dst[i + 1] = val2;
		+	} else {
		+	dst[i] = (val1 << bits) \| get_bits(bits);
		+	dst[i + 1] = (val2 << bits) \| get_bits(bits);
		+	}
	}		}
	} else {		} else {
Line 148:		Line 153:
	val1 = t / 13;		val1 = t / 13;
	val2 = t - val1 * 13;		val2 = t - val1 * 13;
-	val1 = extend_code(val1, 12, 0);	+	val1 = extend_code(val1, 6, 0);
-	val2 = extend_code(val2, 12, 0);	+	val2 = extend_code(val2, 6, 0);
	dst[i] = val1;		dst[i] = val1;
	dst[i + 1] = val2;		dst[i + 1] = val2;

FFmpeg Summer of Code 2012

Michael — Thu, 15 Mar 2012 22:06:01 GMT

Add "Redesigning the protocol reads"

Revision as of 22:06, 15 March 2012		Current revision
Line 112:		Line 112:

	''Mentor: [[User:Pbm\|Paul B Mahol]]''		''Mentor: [[User:Pbm\|Paul B Mahol]]''
		+
		+	=== Redesigning the protocol reads ===
		+	Eliminate active polling while allowing interrupts, and allowing selection on multiple handles.
		+
		+	''Mentor: Nicolas George''

	== 2nd Tier (need mentor) Project Proposals ==		== 2nd Tier (need mentor) Project Proposals ==

Real Lossless Codec

Kostya — Wed, 14 Mar 2012 19:09:20 GMT

/* Block header */

Revision as of 19:09, 14 March 2012		Current revision
Line 43:		Line 43:
	Frame length code:		Frame length code:

-	64 - 11110	+	64 - 111110
-	128 - 11111	+	128 - 111111
-	256 - 1110	+	256 - 11110
-	512 - 110	+	512 - 1110
-	1024 - 10	+	1024 - 110
-	2048 - 0	+	2048 - 10
		+	4096 - 0

	Decoding mode specifies table sets, bits and channel recombining:		Decoding mode specifies table sets, bits and channel recombining:

CSF

Reimar — Wed, 14 Mar 2012 16:44:36 GMT

New page

Currently discussed and sample here: https://ffmpeg.org/trac/ffmpeg/ticket/1060
Copy-and-pasted description:
It is a format based on PNG and partial re-encoding, as you can see when you look at it with e.g. http://sourceforge.net/projects/extractor-gtk/
Possibly contains some other encoding types, too.
All container structures seem to be little-endian.
Each "frame" contains a 4 byte size at offset 8.
For the PNG encoded ones, offsets 16 - 24 seem to contain 4 16 bit values that indicate the offset and size of the image the PNG updates. After that seems to follow yet another size field containing the remaining bytes.
Start of the following packet is always aligned to 16 bytes.
There seems to be some kind of index at 0x190.
This and the data start seems to be referenced in the header, at bytes 24 and 28.

CinemaDNG

Elte — Wed, 14 Mar 2012 10:35:14 GMT

initial version

New page

=Cinema Digital Negative=

* Company: [[Adobe]]

==Links==
* overview: http://labs.adobe.com/downloads/cinemadng.html#docs
* spezification: http://download.macromedia.com/pub/labs/cinemadng/cinemadng_spec_090211.pdf
* file format: http://download.macromedia.com/pub/labs/cinemadng/cinemadng_summary.pdf

[[Category:Video Codecs]]
[[Category:Container Formats]]

FFmpeg Summer of Code 2012

Pbm — Wed, 14 Mar 2012 01:59:13 GMT

/* Libavdevice API */

Revision as of 01:59, 14 March 2012		Current revision
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	''Mentor: Benjamin Larsson''		''Mentor: Benjamin Larsson''
		+
		+	=== Libavdevice API ===
		+	Currently libavdevice use libavformat API. This is far from being useful for real-time video rendering. Your task is to write new API which resolves this and similar issues and port all already present input and output devices to the new API.
		+
		+	New API should have:
		+	* Support for output video devices which do not provide own scaler.
		+	* Minimal latency.
		+
		+	Qualification task: write one new video outdev, for example any of, but not limited to: aalib, caca, x11, xv, xover, vesa or gl2 using current libavdevice API.
		+
		+	''Mentor: [[User:Pbm\|Paul B Mahol]]''

	== 2nd Tier (need mentor) Project Proposals ==		== 2nd Tier (need mentor) Project Proposals ==

FFmpeg Summer of Code 2012

Compn — Tue, 13 Mar 2012 13:49:10 GMT

/* AAC decoder improvements */ add sample url

Revision as of 13:49, 13 March 2012		Current revision
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	=== AAC decoder improvements ===		=== AAC decoder improvements ===
	Our AAC decoder does not support low-delay. Part of this task will be to also finish last year's BSAC task. A possible qualification task is to fix a crash in the current BSAC code with one of the samples from the BSAC testing suite.		Our AAC decoder does not support low-delay. Part of this task will be to also finish last year's BSAC task. A possible qualification task is to fix a crash in the current BSAC code with one of the samples from the BSAC testing suite.
		+	:Sample: https://ffmpeg.org/trac/ffmpeg/ticket/113

	=== AAC encoder improvements ===		=== AAC encoder improvements ===

Talk:FFmpeg Summer of Code 2012

Compn — Tue, 13 Mar 2012 13:42:33 GMT

i dont think a student can complete a wavelet decoder (H265).

New page

==HEVC/H265 decoder==
previous wavelet decoder SOC projects have failed multiple times. i think this is too difficult for students. also i believe its better to spend time on codecs that are actually used. --[[User:Compn|Compn]] 23:01, 4 March 2012 (EST)

FFmpeg Summer of Code 2012

Michael — Mon, 12 Mar 2012 12:38:47 GMT

ben volunteers to mentor DTS-HD in either project, so its 1st-tier

Revision as of 12:38, 12 March 2012		Current revision
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	''Mentor: [[User:reimar\|Reimar Döffinger]]''		''Mentor: [[User:reimar\|Reimar Döffinger]]''
		+
		+	=== DTS-HD decoder ===
		+
		+	* ETSI released specifications (http://www.etsi.org/deliver/etsi_ts/102100_102199/102114/01.03.01_60/ts_102114v010301p.pdf). Your job is to complete the existing decoder with the following features.
		+
		+	(1) Add support for mixed Core + DTS-HD stream structure
		+	(DtsCoreFrame+DtsHdFrame+DtsCoreFrame+DtsHdFrame+...), used by Blu-Ray main
		+	and commentary tracks.
		+	(2) Add support for XXCh extension (6.1 and 7.1 channels).
		+	(3) Add support for X96 extension (96khz).
		+	(4) Add support for XLL extension (lossless).
		+	(5) Add support for a pure DTS-HD stream structure
		+	(DtsHdFrame+DtsHdFrame+DtsHdFrame+...), used by Blu-Ray PiP tracks.
		+	(6) Add support for XBR extension (extra bitrate).
		+
		+	''Mentor: Benjamin Larsson''

	== 2nd Tier (need mentor) Project Proposals ==		== 2nd Tier (need mentor) Project Proposals ==

FFmpeg Summer of Code 2012

Suxen drol — Mon, 12 Mar 2012 09:43:36 GMT

various typo & caps fixes

FFmpeg Summer of Code 2012

Michael — Mon, 12 Mar 2012 07:37:34 GMT

MVC is now 1st tier

Revision as of 07:37, 12 March 2012		Current revision
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	* We need support for alternate conversion functions (e.g. sample format conversion with or without dithering)		* We need support for alternate conversion functions (e.g. sample format conversion with or without dithering)
	* fix bugs in current design (none known but there sure are some)		* fix bugs in current design (none known but there sure are some)
		+
		+	''Mentor: [[User:Michael\|Michael Niedermayer]]''
		+
		+	=== Implement a H.265 / High efficiency video coding (HEVC) decoder ===
		+
		+	* Write a basic decoder supporting I P and if time permits B slices.
		+	* It does not need to be ASM/SIMD optimized but its high level structure must permit such optimizations to be easily added later.
		+	* As a qualification task you need to implement parsing headers and maybe a bit beyond that to demonstrate that you are qualified and understand the HEVC specification. This project requires a solid understanding of video coding and C, its not something for the average SOC student.
		+	* A draft spec is available at: http://phenix.it-sudparis.eu/jct/doc_end_user/documents/8_San%20Jose/wg11/JCTVC-H1003-v21.zip
		+
		+	''Mentor: [[User:Michael\|Michael Niedermayer]]''
		+
		+	=== H.264 MVC ===
		+
		+	* Add MVC support to our H.264 decoder. MVC is used in 3D Blu-Rays.
		+	* As qualification you have to do some work that demonstrates your understanding of MVC and that is a subpart of the whole MVC implementation.

	''Mentor: [[User:Michael\|Michael Niedermayer]]''		''Mentor: [[User:Michael\|Michael Niedermayer]]''
Line 71:		Line 87:

	== 2nd Tier (need mentor) Project Proposals ==		== 2nd Tier (need mentor) Project Proposals ==
-	some following proposals might be done by libav, still waiting for dust to settle.	+	Some of the following proposals are also proposed by other organizations, we will try to coordinate this with them so as to avoid duplicate work.
		+	We are also happy to hear your personal project ideas ...

-	=== AAC ===	+	=== AAC decoder improvments ===
	Our AAC decoder does not support low-delay. Part of this task will be to also finish last year's BSAC task. A possible qualification task is to fix a crash in the current BSAC code with one of the samples from the BSAC testing suite.		Our AAC decoder does not support low-delay. Part of this task will be to also finish last year's BSAC task. A possible qualification task is to fix a crash in the current BSAC code with one of the samples from the BSAC testing suite.
		+
		+	=== AAC encoder improvments ===
		+	Our AAC encoder does not produce competetive quality per bitrate.
		+	Improve the encoder to be better than some other commonly used encoder like libfaac.
		+	This requires solid understanding of things like psychoacoustics and rate distortion.
		+	A qualification task for this could be to improve the encoder by at least 5% bitrate at the same quality
		+	meassured by some objective measure.
		+
		+	=== FF Fuzzer ===
		+	Write a system like FATE that fuzzes and tests these fuzzed multimedia files under address sanitizer and valgrind with ffmpeg and ffplay.
		+	When a crash, or other anomaly is found, it would use git bisect to identify which exact commit introduced the bug.
		+	And either display this via some web frontend (similar to fate.ffmpeg.org) or just automatically send an email to some
		+	dedicated mailing list. The system has to be robust (there will be infinite loops, OOM conditions and randomly occuring
		+	crashes). Its also important that the system is easy to maintain and can filter out duplicates of the same issue.

	=== VC1 interlaced ===		=== VC1 interlaced ===
Line 114:		Line 145:
	* implement rls-lms in the encoder		* implement rls-lms in the encoder
	* implement float support		* implement float support
		+
		+	=== Fix and improve FFserver ===
		+	FFserver has been part of FFmpeg since a long time but due to lack of a motivated maintainer its a bit buggy.
		+	For this project you would have to debug and fix many bugs. It requires good skills at reading and understanding other peoples code.
		+	As a qualification task you will have to write functioning regression tests for FFserver which implicates some bugfixing to make ffserver produce the same output on all supported platforms.

	=== Reverse engineer [[TAK]] codec and write decoder for it ===		=== Reverse engineer [[TAK]] codec and write decoder for it ===

Libav Summer Of Code 2012

Jumpyshoes — Mon, 12 Mar 2012 05:24:31 GMT

Adding checkasm project

Revision as of 05:24, 12 March 2012		Current revision
Line 24:		Line 24:
	== 1st Tier Project Proposals ==		== 1st Tier Project Proposals ==
	1st tier project proposals are project ideas that are reasonably well defined '''AND''' have a mentor volunteered.		1st tier project proposals are project ideas that are reasonably well defined '''AND''' have a mentor volunteered.
		+
		+	=== Assembly Testing Framework ===
		+
		+	FFmpeg/Libav use an extensive amount of assembly to optimize video and audio decoding. For scientific applications and high-assurance systems (e.g. a spacecraft), accuracy and speed are vital. Assembly is difficult to code, and lends itself to mistakes. As FFmpeg/Libav has no framework for testing assembly functions, development of assembly is often hindered by each developer having to create his or her own tests. In addition, functions are not tested for speed. While FATE covers some functions, there are many that simply aren't tested. The samples in FATE do not cover all features in every decoder that has assembly functions (e.g. several 10-bit H.264 predict functions are not used in any FATE sample). FATE also does not test functions in isolation, which impedes assembly development. In addition, FATE might not cover all assembly versions of a function if multiple different levels of optimizations exist (e.g. MMX, MMX2, SSE, SSE2, SSSE3, SSE4, AVX, etc).
		+
		+	* Qualification task: describe how you will implement the system (probably based on x264's checkasm). It should test accuracy, speed, and be extensible. Also, contact me via IRC.
		+	* Actual task: implement the system. At least all H.264 assembly functions should be covered and the framework should be easily extensible.
		+	* The framework should work across all supported architectures and operating systems.
		+	* The framework should measure exactly how fast an individual function is (e.g. using START/STOP_TIMER).
		+	* The framework should be able to test functions in isolation.
		+	* x264's checkasm can be used as a reference.
		+
		+	''Mentor: Daniel Kang'' (Jumpyshoes on #libav-devel@chat.freenode.net; daniel.d.kang@gmail.com -- ping me on IRC and email me).
		+

	=== HEVC/H265 decoder ===		=== HEVC/H265 decoder ===

FFmpeg Summer of Code 2012

Suxen drol — Sun, 11 Mar 2012 22:14:03 GMT

capitalise headings

Revision as of 22:14, 11 March 2012		Current revision
Line 10:		Line 10:
	== Qualification Tasks ==		== Qualification Tasks ==
	To be eligible for a Summer of Code project, we ask you to do a small programming task to prove you know the basics. FFmpeg is a large, complicated collection of code and its not easy for beginners. There are some ideas for tasks on the [[Small FFmpeg Tasks]] page.		To be eligible for a Summer of Code project, we ask you to do a small programming task to prove you know the basics. FFmpeg is a large, complicated collection of code and its not easy for beginners. There are some ideas for tasks on the [[Small FFmpeg Tasks]] page.
-

	== 1st Tier Project Proposals ==		== 1st Tier Project Proposals ==
	These are proposals with a mentor attached.		These are proposals with a mentor attached.

-	Baptiste has also offered to mentor.	+	''Baptiste has also offered to mentor.''

	=== glplay ===		=== glplay ===
	Add OpenGL output to ffplay, this should allow for better performance (and less bugs at least for some hardware / driver combinations). This could be a new application (glplay), but it is probably simpler to extend ffplay to use OpenGL. You can use code from MPlayer's OpenGL vo module which you may relicense under the LGPL.		Add OpenGL output to ffplay, this should allow for better performance (and less bugs at least for some hardware / driver combinations). This could be a new application (glplay), but it is probably simpler to extend ffplay to use OpenGL. You can use code from MPlayer's OpenGL vo module which you may relicense under the LGPL.

-	Mentor: Reimar Döffinger	+	''Mentor: Reimar Döffinger''

	=== Improve the audio resampling/rematrixing/converting code ===		=== Improve the audio resampling/rematrixing/converting code ===
Line 31:		Line 30:

	''Mentor: [[User:Michael\|Michael Niedermayer]]''		''Mentor: [[User:Michael\|Michael Niedermayer]]''
-

	=== Libavfilter extension ===		=== Libavfilter extension ===
Line 49:		Line 47:

	''Mentor: Stefano Sabatini - saste on IRC (possibly with a backup mentor)''.		''Mentor: Stefano Sabatini - saste on IRC (possibly with a backup mentor)''.
-

	=== Bayer colorspace formats ===		=== Bayer colorspace formats ===
Line 65:		Line 62:
	''Mentor: [[User:Suxen_drol\|Peter Ross]]''		''Mentor: [[User:Suxen_drol\|Peter Ross]]''

-	=== extend image formats support ===	+	=== Extend image formats support ===

	Improve FFmpeg support for image formats, adding missing formats (e.g. XPM) and extending support for the current ones (e.g. animated GIF, GIF compression, fix PNG todos, add support to animated PNG) etc.		Improve FFmpeg support for image formats, adding missing formats (e.g. XPM) and extending support for the current ones (e.g. animated GIF, GIF compression, fix PNG todos, add support to animated PNG) etc.
Line 98:		Line 95:
	(DtsHdFrame+DtsHdFrame+DtsHdFrame+...), used by Blu-Ray PiP tracks.		(DtsHdFrame+DtsHdFrame+DtsHdFrame+...), used by Blu-Ray PiP tracks.
	(6) Add support for XBR extension (extra bitrate).		(6) Add support for XBR extension (extra bitrate).
-

	=== MPEG-4 ALS Roundup ===		=== MPEG-4 ALS Roundup ===
Line 119:		Line 115:
	* implement float support		* implement float support

-	=== reverse engineer [[TAK]] codec and write decoder for it ===	+	=== Reverse engineer [[TAK]] codec and write decoder for it ===
	TAK format is partially already documented.		TAK format is partially already documented.
	You need to revisit that documentation and update and/or fix any missing/wrong inforrmation.		You need to revisit that documentation and update and/or fix any missing/wrong inforrmation.
	Reverse engineer codec and write working bitexact decoder.		Reverse engineer codec and write working bitexact decoder.

-	=== support for more subtitle formats ===	+	=== Support for more subtitle formats ===
	we have libass support now, either a parser (from mplayer) to convert subs into ass or something else.		we have libass support now, either a parser (from mplayer) to convert subs into ass or something else.
	* http://mailman.videolan.org/pipermail/vlc-devel/2011-September/081803.html		* http://mailman.videolan.org/pipermail/vlc-devel/2011-September/081803.html
	* http://ale5000.altervista.org/subtitles.htm		* http://ale5000.altervista.org/subtitles.htm

-	=== mkv ordered chapters / playlist support ===	+	=== MKV ordered chapters / playlist support ===
	get playlist stuff into ffmpeg. playlist is blocking a few things like quicktime edit list and .asx / .pls files.		get playlist stuff into ffmpeg. playlist is blocking a few things like quicktime edit list and .asx / .pls files.

-	=== adobe fragmented http in/out ===	+	=== Adobe fragmented http in/out ===
	adobe has a new streaming format.		adobe has a new streaming format.
	* info/spec: http://www.adobe.com/products/httpdynamicstreaming/		* info/spec: http://www.adobe.com/products/httpdynamicstreaming/
Line 139:		Line 135:
	make filters work with higher bitrate codecs/colorspaces		make filters work with higher bitrate codecs/colorspaces

-	=== fix copying video between formats ===	+	=== Fix copying video between formats ===
	lots of h264 streams in flv, mp4, mkv, mpegts. people wish to remux these into various formats so it works on their hardware (ps3, ipod etc).		lots of h264 streams in flv, mp4, mkv, mpegts. people wish to remux these into various formats so it works on their hardware (ps3, ipod etc).
	https://ffmpeg.org/trac/ffmpeg/ticket/796		https://ffmpeg.org/trac/ffmpeg/ticket/796
Line 146:		Line 142:
	https://ffmpeg.org/trac/ffmpeg/ticket/976		https://ffmpeg.org/trac/ffmpeg/ticket/976

-	=== extend paletted format support ===	+	=== Extend paletted format support ===

	Cleanup framework for handling better with paletted format, write a posterize filter, add support to libswscale palette output (possibly making use of libavcodec/elbg), add support for reading and saving a palette to a file and apply them to the input video (e.g. by creating ad-hoc filters).		Cleanup framework for handling better with paletted format, write a posterize filter, add support to libswscale palette output (possibly making use of libavcodec/elbg), add support for reading and saving a palette to a file and apply them to the input video (e.g. by creating ad-hoc filters).

-	=== port formats/colorspace support from dcraw or make dcraw wrapper ===	+	=== Port formats/colorspace support from dcraw or make dcraw wrapper ===
	[[Dcraw]] supports many raw camera formats that ffmpeg may not. port or make a wrapper for this project.		[[Dcraw]] supports many raw camera formats that ffmpeg may not. port or make a wrapper for this project.
		+
		+	[[Category:FFmpeg]]

FFmpeg Summer of Code 2012

Ce — Sun, 11 Mar 2012 21:11:37 GMT

/* 2nd Tier (need mentor) Project Proposals */

Revision as of 21:11, 11 March 2012		Current revision
Line 17:		Line 17:
	Baptiste has also offered to mentor.		Baptiste has also offered to mentor.

		+	=== glplay ===
		+	Add OpenGL output to ffplay, this should allow for better performance (and less bugs at least for some hardware / driver combinations). This could be a new application (glplay), but it is probably simpler to extend ffplay to use OpenGL. You can use code from MPlayer's OpenGL vo module which you may relicense under the LGPL.
		+
		+	Mentor: Reimar Döffinger

	=== Improve the audio resampling/rematrixing/converting code ===		=== Improve the audio resampling/rematrixing/converting code ===
Line 71:		Line 75:
	== 2nd Tier (need mentor) Project Proposals ==		== 2nd Tier (need mentor) Project Proposals ==
	some following proposals might be done by libav, still waiting for dust to settle.		some following proposals might be done by libav, still waiting for dust to settle.
		+
		+	=== AAC ===
		+	Our AAC decoder does not support low-delay. Part of this task will be to also finish last year's BSAC task. A possible qualification task is to fix a crash in the current BSAC code with one of the samples from the BSAC testing suite.
		+
		+	=== VC1 interlaced ===
		+	FFmpeg has code for interlaced VC1, but nearly all samples still do not decode correctly. The task is to finish last year's project. You should be able to find a possible qualification task by testing interlaced samples.

	=== H.264 MVC ===		=== H.264 MVC ===

Real Lossless Codec

Kostya — Sun, 11 Mar 2012 17:26:41 GMT

Revision as of 17:26, 11 March 2012		Current revision
Line 1:		Line 1:
	* FOURCCs: ralf		* FOURCCs: ralf
	* Company: [[Real\|RealNetworks]]		* Company: [[Real\|RealNetworks]]
-	* Samples: http://samples.mplayerhq.hu/A-codecs/lossless/ (luckynight.rmvb)	+	* Samples: http://samples.libav.org/A-codecs/lossless/ (luckynight.rmvb)

	This is RealNetworks lossless audio codec.		This is RealNetworks lossless audio codec.
Line 69:		Line 69:
	=== Block data ===		=== Block data ===

-	code from tree 0 (0..642) - something to do with filter length	+	==== Variable-length codes ====
-	also if code == 642 just read raw data and return it	+

		+	RALF uses hybrid coding scheme for signed integers: coded value in range <code>1..N-1</code> maps to <code>-N/2+1..N/2-1</code> and values <code>0</code> and <code>N</code> are used as escape values.
		+	In case an escape value is occurred, an additional Elias Gamma code (aka exp-Golomb) is read and added to or subtracted from the result.
		+	Also some low bits can be read explicitly.
		+
		+	extend_code(code, range, add_bits)
		+	{
		+	if (code == 0) {
		+	len = get_unary(); // 000...001
		+	code = -range + 1 - get_bits(len + 1);
		+	} else if (code == range*2) {
		+	len = get_unary(); // 000...001
		+	code = range - 1 + get_bits(len + 1);
		+	} else {
		+	code -= range;
		+	}
		+	if (add_bits > 0)
		+	code = (code << add_bits) \| get_bits(add_bits);
		+	return code;
		+	}
		+
		+	==== Channel data ====
		+
		+	Channel data is organised that way:
		+
		+	filter_parameter = get_vlc(filter_parameter_table);
		+	if (filter_parameter == 642) {
		+	bias_val = 0;
		+	all samples are coded raw
		+	}
		+	bias_val = get_vlc(bias_table);
		+	bias_val = extend_code(bias_val, 127, 4); // input is 0..254, make it signed, read escape values if needed and low 4 bits
		+	if (filter_parameter == 0) {
		+	all samples are zero
		+	return
		+	}
		+	if (filter_parameter > 1) {
		+	filter_bits = (filter_parameter - 2) >> 6; // range 0..9
		+	filter_len = filter_parameter - 1 - (filter_bits << 6); // range 1..64
		+	coeff = 0;
		+	coeff_mode = 0;
		+	for (i = 0; i < filter_len; i++) {
		+	t = get_vlc(filt_coef_table[filter_bits][coeff_mode + 5]);
		+	t = extend_code(t, 21, filter_bits);
		+	if (coeff_mode == 0)
		+	coeff -= 12 << filter_bits;
		+	coeff = t - coeff;
		+	filter[i] = coeff;
		+	coeff_mode = coeff >> filter_bits;
		+	if (coeff_mode < 0)
		+	coeff_mode = max(-5, -1 - log2(-coeff_mode));
		+	else if (coeff_mode > 0)
		+	coeff_mode = min(5, 1 + log2(coeff_mode));
		+	}
		+	}
		+	coding_mode = get_vlc(coding_mode_tree);
		+	if (coding_mode >= 15) { // big codes
		+	bits = (coding_mode / 5 - 3) / 2;
		+	if (bits > 9) {
		+	bits = 9;
		+	if ((coding_mode % 5) == 2)
		+	bits++;
		+	}
		+	for (i = 0; i < num_samples; i += 2) {
		+	t = get_vlc(long_codes_trees[coding_mode - 15]);
		+	val1 = t / 21;
		+	val2 = t - val1 * 21;
		+	val1 = extend_code(val1, 10, bits);
		+	val2 = extend_code(val2, 10, bits);
		+	dst[i] = val1;
		+	dst[i + 1] = val2;
		+	}
		+	} else {
		+	for (i = 0; i < num_samples; i += 2) {
		+	t = get_vlc(short_codes_trees[coding_mode]);
		+	val1 = t / 13;
		+	val2 = t - val1 * 13;
		+	val1 = extend_code(val1, 12, 0);
		+	val2 = extend_code(val2, 12, 0);
		+	dst[i] = val1;
		+	dst[i + 1] = val2;
		+	}
		+	}
		+
		+	=== Restoring channel data ===
		+
		+	* apply LPC filter
		+	* add bias value to all samples
		+	* re-correlate stereo channels if needed
		+
		+	==== LPC filter ====
		+
		+	filter_bits = ((filter_parameter - 2) >> 6) + 3;
		+	range_bits = bits_in_channel; // i.e. 16 or 17
		+	bias = 1 << (filter_bits - 1);
		+	pos_clip = (1 << range_bits) - 1;
		+	neg_clip = ~pos_clip;
		+
		+	for (i = 0; i < num_samples; i++) {
		+	acc = 0;
		+	for (j = 0; j < min(i, filter_length); j++)
		+	acc += filter[j] * src[i - j - 1];
		+	if (acc < 0)
		+	val = (acc + bias - 1) >> filter_bits;
		+	else
		+	val = (acc + bias) >> filter_bits;
		+	val = clip(val, neg_clip, pos_clip);
		+	src[i] += val;
		+	}

-	[[Category:Undiscovered Audio Codecs]]
	[[Category:Lossless Audio Codecs]]		[[Category:Lossless Audio Codecs]]
	[[Category:Audio Codecs]]		[[Category:Audio Codecs]]

FFmpeg Summer of Code 2012

Compn — Sun, 11 Mar 2012 15:39:18 GMT

remove old message, derp

Revision as of 15:39, 11 March 2012		Current revision
Line 1:		Line 1:
-	:Please detail the deadlines for applying and being a student/mentor.
-
	FFmpeg Summer of Code Proposals and Qualification Tasks.		FFmpeg Summer of Code Proposals and Qualification Tasks.

FFmpeg Summer of Code 2012

Pbm — Sun, 11 Mar 2012 14:58:14 GMT

/* reverse engineer [[TAK]] format */ better explain task

Revision as of 14:58, 11 March 2012		Current revision
Line 111:		Line 111:
	* implement float support		* implement float support

-	=== reverse engineer [[TAK]] format ===	+	=== reverse engineer [[TAK]] codec and write decoder for it ===
-	there is some interest in this format. maybe a qual task?	+	TAK format is partially already documented.
		+	You need to revisit that documentation and update and/or fix any missing/wrong inforrmation.
		+	Reverse engineer codec and write working bitexact decoder.

	=== support for more subtitle formats ===		=== support for more subtitle formats ===

Small FFmpeg Tasks

Compn — Sun, 11 Mar 2012 13:33:38 GMT

move emblaze task to rev eng section

Revision as of 13:33, 11 March 2012		Current revision
Line 198:		Line 198:
	[[SNES-SPC700 Sound Format]]		[[SNES-SPC700 Sound Format]]
	VGM, VGZ		VGM, VGZ
-
-	=== port [[Ut Video]] encoder ===
-	External and internal decoder are available, encoder (native or via libutvideo) is missing.

	=== libswscale PAL8 output ===		=== libswscale PAL8 output ===
Line 233:		Line 230:

	Add support to the AAC decoder for object type ER AAC LC or AAC LC 960.		Add support to the AAC decoder for object type ER AAC LC or AAC LC 960.
-
-	=== emblaze demuxer/decoder from java code ===
-	samples and java decoder: http://samples.mplayerhq.hu/internets/emblaze/

	=== arithmetic decoding (and encoding) for mjpeg ===		=== arithmetic decoding (and encoding) for mjpeg ===
Line 248:		Line 242:
	=== adobe http f4f segmented fragmentation dynamic streaming format ===		=== adobe http f4f segmented fragmentation dynamic streaming format ===
	sample streams on http://www.fox.com . command line instructions for creating such files: http://help.adobe.com/en_US/HTTPStreaming/1.0/Using/WS9463dbe8dbe45c4c-c126f3b1260533756d-7ffc.html . spec is available under adobe NDA. not to be confused with freely available F4V specification.		sample streams on http://www.fox.com . command line instructions for creating such files: http://help.adobe.com/en_US/HTTPStreaming/1.0/Using/WS9463dbe8dbe45c4c-c126f3b1260533756d-7ffc.html . spec is available under adobe NDA. not to be confused with freely available F4V specification.
		+
		+	=== pick a random binary codec from mplayer ===
		+	[[MPlayer]] has over 100 binary codecs which have no opensource decoder. pick one, find a sample and try to reverse engineer it. note that some work has been done on some codecs, and its a good idea to ask on the mailing list before starting.
		+
		+	=== emblaze demuxer/decoder from java code ===
		+	samples and java decoder: http://samples.mplayerhq.hu/internets/emblaze/

FFmpeg Summer of Code 2012

Compn — Sun, 11 Mar 2012 13:11:53 GMT

add dcraw wrapper or camera raw format support idea

Revision as of 13:11, 11 March 2012		Current revision
Line 57:		Line 57:
	* Complete PhotoCINE demuxer to support Bayer format; (or another format of your choosing)		* Complete PhotoCINE demuxer to support Bayer format; (or another format of your choosing)
	* SIMD optimisations of the libswscale transformations		* SIMD optimisations of the libswscale transformations
		+	* decoders/specs may be available in the [[Dcraw]] project

	Qualification task: TBD		Qualification task: TBD
Line 139:		Line 140:
	Cleanup framework for handling better with paletted format, write a posterize filter, add support to libswscale palette output (possibly making use of libavcodec/elbg), add support for reading and saving a palette to a file and apply them to the input video (e.g. by creating ad-hoc filters).		Cleanup framework for handling better with paletted format, write a posterize filter, add support to libswscale palette output (possibly making use of libavcodec/elbg), add support for reading and saving a palette to a file and apply them to the input video (e.g. by creating ad-hoc filters).

-	=== g2m4 decoder ===	+	=== port formats/colorspace support from dcraw or make dcraw wrapper ===
-	lots of videolan users are requesting this codec be supported. there is a binary decoder in mplayer. its possible maxim is already working on this.	+	[[Dcraw]] supports many raw camera formats that ffmpeg may not. port or make a wrapper for this project.

FFmpeg Summer of Code 2012

Reimar — Sun, 11 Mar 2012 08:52:26 GMT

/* extend image formats support */

Revision as of 08:52, 11 March 2012		Current revision
Line 62:		Line 62:
	''Mentor: [[User:Suxen_drol\|Peter Ross]]''		''Mentor: [[User:Suxen_drol\|Peter Ross]]''

		+	=== extend image formats support ===
		+
		+	Improve FFmpeg support for image formats, adding missing formats (e.g. XPM) and extending support for the current ones (e.g. animated GIF, GIF compression, fix PNG todos, add support to animated PNG) etc.
		+
		+	Qualification task: TBD (possibly finally fixing and integrating Måns' zlib decoder that has been unmerged since ages? Or just starting with some small part of the task itself, or implementing format autodetection for imagepipe demuxer)
		+
		+	''Mentor: [[User:reimar\|Reimar Döffinger]]''

	== 2nd Tier (need mentor) Project Proposals ==		== 2nd Tier (need mentor) Project Proposals ==
Line 127:		Line 134:
	https://ffmpeg.org/trac/ffmpeg/ticket/954		https://ffmpeg.org/trac/ffmpeg/ticket/954
	https://ffmpeg.org/trac/ffmpeg/ticket/976		https://ffmpeg.org/trac/ffmpeg/ticket/976
-
-	=== extend image formats support ===
-
-	Improve FFmpeg support for image formats, adding missing formats (e.g. XPM) and extending support for the current ones (e.g. animated GIF, GIF compression, fix PNG todos, add support to animated PNG) etc.

	=== extend paletted format support ===		=== extend paletted format support ===

FFmpeg Summer of Code 2012

Suxen drol — Sun, 11 Mar 2012 02:47:48 GMT

Revision as of 02:47, 11 March 2012		Current revision
Line 47:		Line 47:

	''Mentor: Stefano Sabatini - saste on IRC (possibly with a backup mentor)''.		''Mentor: Stefano Sabatini - saste on IRC (possibly with a backup mentor)''.
		+
		+
		+	=== Bayer colorspace formats ===
		+
		+	Several image and video format store pixels using Bayer-pattern colorspaces. Supporting these format would broaded FFmepg's applicability to RAW still and video photography processing. Tasks:
		+	* Implement bayer transformations in libswscale (plain C)
		+	* Add bayer formats to the libavutil pixfmt enumeration routines
		+	* Extend TIFF decoder to support DNG-Bayer format
		+	* Complete PhotoCINE demuxer to support Bayer format; (or another format of your choosing)
		+	* SIMD optimisations of the libswscale transformations
		+
		+	Qualification task: TBD
		+
		+	''Mentor: [[User:Suxen_drol\|Peter Ross]]''
		+

	== 2nd Tier (need mentor) Project Proposals ==		== 2nd Tier (need mentor) Project Proposals ==

FFmpeg Summer of Code 2012

Compn — Sun, 11 Mar 2012 01:44:10 GMT

add deadline date

Revision as of 01:44, 11 March 2012		Current revision
Line 2:		Line 2:

	FFmpeg Summer of Code Proposals and Qualification Tasks.		FFmpeg Summer of Code Proposals and Qualification Tasks.
		+
		+	==Timeline==
		+	March 12th-16th: Project application evaluation.

	=Contacting Devels=		=Contacting Devels=

Real Lossless Codec

Kostya — Sat, 10 Mar 2012 16:17:12 GMT

/* Frame format */

Revision as of 16:17, 10 March 2012		Current revision
Line 13:		Line 13:

	bytes 0- 3 - codec FOURCC, always "LSD:"		bytes 0- 3 - codec FOURCC, always "LSD:"
-	bytes 4- 5 - version? Always 0x103	+	bytes 4- 5 - version
-	bytes 8- 9 - number of channels?	+	bytes 8- 9 - number of channels
-	bytes 10-11 - bits per sample? Always 16	+	bytes 10-11 - bits per sample
	bytes 12-15 - sampling rate		bytes 12-15 - sampling rate
	bytes 16-19 - block size in samples?		bytes 16-19 - block size in samples?
-	bytes 20-23 - always zero?	+	bytes 20-23 - ignored?
		+
		+	Known decoder support only 1/2 channel 16-bit audio with sampling rate up to 96 kHz.
		+
		+	== Frame format ==
		+
		+	Data is read MSB.
		+
		+	block header
		+	channel data 1
		+	if (stereo)
		+	channel data 2
		+
		+	Each channel is then restored by applying filter and in stereo case channels are recombined.
		+
		+	=== Block header ===
		+
		+	get frame length code
		+	if (stereo)
		+	decoding_mode = get_bits(2) + 1;
		+	else
		+	decoding_mode = 0;
		+
		+	Frame length code:
		+
		+	64 - 11110
		+	128 - 11111
		+	256 - 1110
		+	512 - 110
		+	1024 - 10
		+	2048 - 0
		+
		+	Decoding mode specifies table sets, bits and channel recombining:
		+
		+	{\| border="1" cellpadding="5" style="border-collapse: collapse; border-style: dashed; border-color: #2f6fab;"
		+	\|- bgcolor="#f0f0f0" \|
		+	! Mode !! tableset1 !! bits1 !! tableset2 !! bits2 !! Channel decoupling
		+	\|-
		+	\| align="center" \| 0 \|\| align="center" \| 0 \|\| align="center" \| 16 \|\| align="center" \| n/a \|\| align="center" \| n/a \|\| align="center" \| mono
		+	\|-
		+	\| align="center" \| 1 \|\| align="center" \| 0 \|\| align="center" \| 16 \|\| align="center" \| 0 \|\| align="center" \| 16 \|\| align="center" \| L, R
		+	\|-
		+	\| align="center" \| 2 \|\| align="center" \| 0 \|\| align="center" \| 16 \|\| align="center" \| 2 \|\| align="center" \| 17 \|\| align="center" \| L+R, R
		+	\|-
		+	\| align="center" \| 3 \|\| align="center" \| 1 \|\| align="center" \| 16 \|\| align="center" \| 2 \|\| align="center" \| 17 \|\| align="center" \| L, R-L
		+	\|-
		+	\| align="center" \| 4 \|\| align="center" \| 2 \|\| align="center" \| 17 \|\| align="center" \| 2 \|\| align="center" \| 17 \|\| align="center" \| (L+R)/2, (L-R)/2
		+	\|}
		+
		+	=== Block data ===
		+
		+	code from tree 0 (0..642) - something to do with filter length
		+	also if code == 642 just read raw data and return it
		+

	[[Category:Undiscovered Audio Codecs]]		[[Category:Undiscovered Audio Codecs]]
	[[Category:Lossless Audio Codecs]]		[[Category:Lossless Audio Codecs]]
	[[Category:Audio Codecs]]		[[Category:Audio Codecs]]

Libav Summer Of Code 2012

Jruggle — Fri, 09 Mar 2012 21:09:14 GMT

add dng decoder project idea

Revision as of 21:09, 9 March 2012		Current revision
Line 83:		Line 83:
	* Handle CELT, SILK, and Hybrid modes (including transitions)		* Handle CELT, SILK, and Hybrid modes (including transitions)
	* (optional) Handle more than 2 channels		* (optional) Handle more than 2 channels
		+
		+	''Mentor: [[User:Jruggle\|Justin Ruggles]]''
		+
		+
		+	=== Adobe DNG Decoder (Basic Support) ===
		+
		+	Adobe Digital Negative (DNG) is an attempt at a universal file format for raw camera images. Most camera manufacturers have their own proprietary raw image format. Adobe provides tools for converting these to DNG with minimal or no loss of information for more reliable long-term support in a format with an open specification.
		+
		+	The project goal would be to add features required for basic support of DNG files. Some of these include:
		+	* test/improve TIFF and LJPEG 16-bit decoding support
		+	* implement both variants of JPEG-in-TIFF in the TIFF decoder
		+	* add basic handling for Bayer CFA pixel format(s), including demosaicing
		+	* conversion from camera colorspace to RGB
		+	* export of DNG/TIFF/Exif metadata
		+
		+	Resources:
		+	* [http://en.wikipedia.org/wiki/Digital_Negative Wikipedia Article]
		+	* [http://wwwimages.adobe.com/www.adobe.com/content/dam/Adobe/en/products/photoshop/pdfs/dng_spec.pdf Specification]
		+	* DNG samples can be created from other raw formats using the free [http://en.wikipedia.org/wiki/Digital_Negative DNG Converter] program
		+	* A good place to find raw camera samples is http://www.imaging-resource.com

	''Mentor: [[User:Jruggle\|Justin Ruggles]]''		''Mentor: [[User:Jruggle\|Justin Ruggles]]''

Special:Log/delete

DonDiego — Fri, 09 Mar 2012 19:27:22 GMT

deleted "[[Talk:FFmpeg / Libav Summer Of Code]]": content was: '== reverting intro paragraph ==could you explain why you reverted this change Diego?http://wiki.multimedia.cx/index.php?title=FFmpeg_/_Libav_Summer...'

Libav Summer Of Code 2012

Ronald S. Bultje — Fri, 09 Mar 2012 19:26:37 GMT

remove WIP edit notice

Revision as of 19:26, 9 March 2012		Current revision
Line 2:		Line 2:

	Google's Summer of Code program is simple: you (the student) work on a project, full-time, during the whole summer, and you get assistance (advice, mentoring) from a seasoned Libav developer who knows his way around the project and has considerable standing in the community. By doing so, you'll learn to operate in an opensource project, you'll get relevant coding experience, and you'll have a chance at earning money while doing fun stuff during the summer. So, you need a project, a mentor, do a qualification task (see below) so we can quickly assess how good a candidate we feel you'll be for the program, and then you can apply.		Google's Summer of Code program is simple: you (the student) work on a project, full-time, during the whole summer, and you get assistance (advice, mentoring) from a seasoned Libav developer who knows his way around the project and has considerable standing in the community. By doing so, you'll learn to operate in an opensource project, you'll get relevant coding experience, and you'll have a chance at earning money while doing fun stuff during the summer. So, you need a project, a mentor, do a qualification task (see below) so we can quickly assess how good a candidate we feel you'll be for the program, and then you can apply.
-
-	:Please detail the deadlines for applying and being a student/mentor.

	=== Selecting a project ===		=== Selecting a project ===
Line 23:		Line 21:
	=== Applying ===		=== Applying ===
	While you're working on your qualification task, apply at http://www.google-melange.com/. The degree (and detail) with which you've finished your qualification task will determine how likely your project is to be selected. In the past few years, students that completely finished their qualification task always got selected as Summer of Code students, but that may vary depending on the number of spots we get assigned by Google, and the number of students that apply.		While you're working on your qualification task, apply at http://www.google-melange.com/. The degree (and detail) with which you've finished your qualification task will determine how likely your project is to be selected. In the past few years, students that completely finished their qualification task always got selected as Summer of Code students, but that may vary depending on the number of spots we get assigned by Google, and the number of students that apply.
-

	== 1st Tier Project Proposals ==		== 1st Tier Project Proposals ==

FFmpeg Summer of Code 2012

Compn — Fri, 09 Mar 2012 16:05:59 GMT

Revision as of 16:05, 9 March 2012		Current revision
Line 117:		Line 117:

	Cleanup framework for handling better with paletted format, write a posterize filter, add support to libswscale palette output (possibly making use of libavcodec/elbg), add support for reading and saving a palette to a file and apply them to the input video (e.g. by creating ad-hoc filters).		Cleanup framework for handling better with paletted format, write a posterize filter, add support to libswscale palette output (possibly making use of libavcodec/elbg), add support for reading and saving a palette to a file and apply them to the input video (e.g. by creating ad-hoc filters).
		+
		+	=== g2m4 decoder ===
		+	lots of videolan users are requesting this codec be supported. there is a binary decoder in mplayer. its possible maxim is already working on this.

FFmpeg Summer of Code 2012

Stefanosa — Thu, 08 Mar 2012 11:17:03 GMT

Add two more random proposals.

Revision as of 11:17, 8 March 2012		Current revision
Line 109:		Line 109:
	https://ffmpeg.org/trac/ffmpeg/ticket/954		https://ffmpeg.org/trac/ffmpeg/ticket/954
	https://ffmpeg.org/trac/ffmpeg/ticket/976		https://ffmpeg.org/trac/ffmpeg/ticket/976
		+
		+	=== extend image formats support ===
		+
		+	Improve FFmpeg support for image formats, adding missing formats (e.g. XPM) and extending support for the current ones (e.g. animated GIF, GIF compression, fix PNG todos, add support to animated PNG) etc.
		+
		+	=== extend paletted format support ===
		+
		+	Cleanup framework for handling better with paletted format, write a posterize filter, add support to libswscale palette output (possibly making use of libavcodec/elbg), add support for reading and saving a palette to a file and apply them to the input video (e.g. by creating ad-hoc filters).

Libav/ReleaseHowto

DonDiego — Thu, 08 Mar 2012 08:44:04 GMT

Replace backtics by superior $() syntax.

Revision as of 08:44, 8 March 2012		Current revision
Line 17:		Line 17:
	echo '$VERSION' > VERSION<br/>		echo '$VERSION' > VERSION<br/>
	git add VERSION<br/>		git add VERSION<br/>
-	git commit -m"Update Version identification for `cat VERSION`"	+	git commit -m"Update Version identification for $(cat VERSION)"
	</code>		</code>

Line 50:		Line 50:

	* Install tarballs to libav.org:		* Install tarballs to libav.org:
-	** <code>scp /tmp/libav-`cat VERSION`.* libav.org:releases</code>	+	** <code>scp /tmp/libav-$(cat VERSION).* libav.org:releases</code>

	=== Publish the release ===		=== Publish the release ===

FFmpeg Metadata

Multimedia Mike — Tue, 06 Mar 2012 20:42:41 GMT

/* Matroska */ MKV muxer accepts free-form key/value pairs

Revision as of 20:42, 6 March 2012		Current revision
Line 132:		Line 132:
	* “comment”		* “comment”
	* "rating"		* "rating"
		+
		+	Beyond these keys, the ASF muxer accepts free-form key/value metadata keys to be encoded into the header. Further, libavformat encodes its version using the key "WM/EncodingSettings".

	== AVI ==		== AVI ==
-	FFmpeg’s AVI muxer honors the following metadata keys and maps them to these FourCCs in the file header:	+	FFmpeg’s AVI muxer honors the following metadata keys, writing them into FourCC chunks in the file header:

-	* “Title” -> “INAM”	+	* "IARL"
-	* “Artist” or “Author” -> “IART”	+	* "IART", "artist"
-	* “Copyright” -> “ICOP”	+	* "ICMS"
-	* “Comment” -> “ICMT”	+	* "ICMT", "comment"
-	* “Album” -> “IPRD”	+	* "ICOP", "copyright"
-	* “Genre” -> “IGNR”	+	* "ICRD", "date"
-	* “Track” -> “IPRT”	+	* "ICRP"
		+	* "IDIM"
		+	* "IDPI"
		+	* "IENG"
		+	* "IGNR", "genre"
		+	* "IKEY"
		+	* "ILGT"
		+	* "ILNG", "language"
		+	* "IMED"
		+	* "INAM", "title"
		+	* "IPLT"
		+	* "IPRD", "album"
		+	* "IPRT", "track"
		+	* "ISBJ"
		+	* "ISFT", "encoder" - note that this is automatically filled in by libavformat
		+	* "ISHP"
		+	* "ISRC"
		+	* "ISRF"
		+	* "ITCH", "encoded_by"

	== FLV ==		== FLV ==
Line 171:		Line 191:
	* “description”		* “description”
	* “language”		* “language”
		+
		+	Beyond these keys, the Matroska muxer also accepts free-form key/value metadata pairs.

	== MP3 ==		== MP3 ==

Category talk:Audio Codecs

Compn — Tue, 06 Mar 2012 17:10:18 GMT

Revision as of 17:10, 6 March 2012		Current revision
Line 1:		Line 1:
	== list of acm codecs ==		== list of acm codecs ==
-	:rodo. put these in a nice table with format ids and mplayer/ffmpeg status	+	:todo. put these in a nice table with format ids and mplayer/ffmpeg status

	msacm.13acm=13codeca.acm		msacm.13acm=13codeca.acm
Line 34:		Line 34:
	msacm.dvacm=Dvacm.acm		msacm.dvacm=Dvacm.acm
	msacm.dvmpega=dvacma.dll		msacm.dvmpega=dvacma.dll
-	msacm.dvmpega=dvacmau.dll0	+	msacm.dvmpega=dvacmau.dll
-	msacm.enc=ITIG726. acm	+	msacm.enc=ITIG726.acm - Ingenient Technologies, Inc. G.726 CODEC
	msacm.fraunhoferacm=l3codecp.acm		msacm.fraunhoferacm=l3codecp.acm
	msacm.g723=g723.acm		msacm.g723=g723.acm

FFmpeg Summer of Code 2012

Compn — Mon, 05 Mar 2012 04:12:12 GMT

add popular bug/wish

Revision as of 04:12, 5 March 2012		Current revision
Line 102:		Line 102:
	=== libavfilter 9/10bit support ===		=== libavfilter 9/10bit support ===
	make filters work with higher bitrate codecs/colorspaces		make filters work with higher bitrate codecs/colorspaces
		+
		+	=== fix copying video between formats ===
		+	lots of h264 streams in flv, mp4, mkv, mpegts. people wish to remux these into various formats so it works on their hardware (ps3, ipod etc).
		+	https://ffmpeg.org/trac/ffmpeg/ticket/796
		+	https://ffmpeg.org/trac/ffmpeg/ticket/822
		+	https://ffmpeg.org/trac/ffmpeg/ticket/954
		+	https://ffmpeg.org/trac/ffmpeg/ticket/976

Talk:Libav Summer Of Code 2012

Compn — Mon, 05 Mar 2012 04:01:46 GMT

my opinions on proposals

Revision as of 04:01, 5 March 2012		Current revision
Line 26:		Line 26:
	</pre>		</pre>
	--[[User:Compn\|Compn]] 10:53, 20 February 2012 (EST)		--[[User:Compn\|Compn]] 10:53, 20 February 2012 (EST)
		+
		+
		+	==HEVC/H265 decoder==
		+	previous wavelet decoder SOC projects have failed multiple times. i think this is too difficult for students. also i believe its better to spend time on codecs that are actually used. --[[User:Compn\|Compn]] 23:01, 4 March 2012 (EST)
		+
		+	==Opus Decoder==
		+	is this used anywhere? i think rare use/samples is not a good candidate. think dirac.--[[User:Compn\|Compn]] 23:01, 4 March 2012 (EST)
		+
		+	==On2 VP7 decoder==
		+	skype has no way to record calls so the vp7 video is never found on users harddrive. also microsoft bought skype and i'm sure will torpedo it fast. --[[User:Compn\|Compn]] 23:01, 4 March 2012 (EST)
		+
		+	==RTMP[E\|S\|T\|TE] protocol implementation==
		+	i do not understand why you want to duplicate librtmp. adobe already has "dynamic http streaming". i'm guessing they are switching to it and dropping this rtmp crap since flv has no encryption? --[[User:Compn\|Compn]] 23:01, 4 March 2012 (EST)
		+
		+	==spin off build system into a separate project==
		+	useful thing, but is it good task for SOC? --[[User:Compn\|Compn]] 23:01, 4 March 2012 (EST)
		+
		+	== generic build system ==
		+	it would be a good idea to find a project that wants to switch to this build system. this would give a clearer goal, 'make generic build system and port project XXX to use it'. i think it would be a bad idea to make a big generic build system only to find that its missing some feature that everyone needs. there are autotool alternatives, maybe good for finding obstacles and required features: http://stackoverflow.com/questions/600274/alternatives-to-autoconf-and-autotools --[[User:Compn\|Compn]] 22:23, 4 March 2012 (EST)

Advanced Audio Coding

AConverse — Sun, 04 Mar 2012 23:19:44 GMT

/* Decoder Features */ elem_id 0 is now garbled instead of crashing in WinAMP

Revision as of 23:19, 4 March 2012		Current revision
Line 145:		Line 145:
	! WinAMP		! WinAMP
	\| Yes		\| Yes
-	\| Crash	+	\| Garbled
	\| No		\| No
	\| Yes		\| Yes

Windows Media Audio 9 Lossless

Multimedia Mike — Sun, 04 Mar 2012 22:02:47 GMT

lives in the open source projects now; no longer undiscovered

Revision as of 22:02, 4 March 2012		Current revision
Line 7:		Line 7:
	[[Category:Audio Codecs]]		[[Category:Audio Codecs]]
	[[Category:Lossless Audio Codecs]]		[[Category:Lossless Audio Codecs]]
-	[[Category:Undiscovered Audio Codecs]]	+	[[Category:Multichannel Audio Codecs]]
-	[[Category: Multichannel Audio Codecs]]	+

User:Compn

Compn — Sat, 03 Mar 2012 11:14:41 GMT

Revision as of 11:14, 3 March 2012		Current revision
Line 1:		Line 1:
-	The [[MPlayer]] IRC and bug report monkey.	+	[[FFmpeg]] and [[MPlayer]] developer.
		+	*email: compn@ffmpeg.org
		+

	-------------------		-------------------

FATE failures

Ronald S. Bultje — Thu, 01 Mar 2012 19:07:24 GMT

/* H.264 */

Revision as of 19:07, 1 March 2012		Current revision
Line 4:		Line 4:

	=== H.264 ===		=== H.264 ===
-
-	I've briefly analyzed MR3_TANDBERG_B and it appears that the first 41 frames are bitexact. After 42 frames there is a reset of POCs, and we drop all frames after that with a POC smaller than the POC of before the POC reset, causing a total of 41 frames to be dropped. A patch to fix this is pending.
-	MAIN/MR2_TANDBERG_E.264 ... JF ( yuv420p @ 176x144 / 25 fps) not equal
-	MAIN/MR3_TANDBERG_B.264 ... JF ( yuv420p @ 176x144 / 25 fps) not equal

	tandberg 4/5 have large codec delays because of misordered frames, and the delay jumps randomly within the file. We don't preroll well enough to know the correct delay in advance and so at each point where we return a frame too far ahead in the future, we drop the preceeding frames once we get around to decoding them because we've already passed that point in units returned to the application. The correct solution for these would be to increase delay dynamically whenever we encounter a gap until either we're at max-delay, or until we've resolved the gap. almost certainly, max-delay should be settable in user applications so we can disable it for live/low-latency purposes. Q: should we return old frames also? The user may want to reorder himself (although this doesn't make much sense). Fate currently uses -strict 1 as a workaround for these samples, which simply forces codec-delay to 16. This is not the correct solution.		tandberg 4/5 have large codec delays because of misordered frames, and the delay jumps randomly within the file. We don't preroll well enough to know the correct delay in advance and so at each point where we return a frame too far ahead in the future, we drop the preceeding frames once we get around to decoding them because we've already passed that point in units returned to the application. The correct solution for these would be to increase delay dynamically whenever we encounter a gap until either we're at max-delay, or until we've resolved the gap. almost certainly, max-delay should be settable in user applications so we can disable it for live/low-latency purposes. Q: should we return old frames also? The user may want to reorder himself (although this doesn't make much sense). Fate currently uses -strict 1 as a workaround for these samples, which simply forces codec-delay to 16. This is not the correct solution.

Revision as of 09:43, 12 March 2012		Current revision
Line 4:		Line 4:
	March 12th-16th: Project application evaluation.		March 12th-16th: Project application evaluation.

-	=Contacting Devels=	+	==Contacting Developers==
	Find us on irc, server: irc.freenode.net channel: #ffmpeg-devel		Find us on irc, server: irc.freenode.net channel: #ffmpeg-devel
	or contact us by subscribing to the mailing list https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel/		or contact us by subscribing to the mailing list https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel/

	== Qualification Tasks ==		== Qualification Tasks ==
-	To be eligible for a Summer of Code project, we ask you to do a small programming task to prove you know the basics. FFmpeg is a large, complicated collection of code and its not easy for beginners. There are some ideas for tasks on the [[Small FFmpeg Tasks]] page.	+	To be eligible for a Summer of Code project, we ask you to do a small programming task to prove you know the basics. FFmpeg is a large, complicated collection of code and it's not easy for beginners. There are some ideas for tasks on the [[Small FFmpeg Tasks]] page.

	== 1st Tier Project Proposals ==		== 1st Tier Project Proposals ==
Line 25:		Line 25:
	* right now, we're using libswresample to resample/rematrix audio (samplerate / channels) and to resample the audio format (int, float, 16-bit, 32-bit).		* right now, we're using libswresample to resample/rematrix audio (samplerate / channels) and to resample the audio format (int, float, 16-bit, 32-bit).
	* both interleaved and planar audio sample formats are already supported		* both interleaved and planar audio sample formats are already supported
-	* We need SIMD optimization of popular conversions (float-int16, int16-float), (stereo-mono-5.1) and anything else thats frequently used.	+	* We need SIMD optimization of popular conversions (float-int16, int16-float), (stereo-mono-5.1) and anything else that's frequently used.
	* We need support for alternate conversion functions (e.g. sample format conversion with or without dithering)		* We need support for alternate conversion functions (e.g. sample format conversion with or without dithering)
	* fix bugs in current design (none known but there sure are some)		* fix bugs in current design (none known but there sure are some)
Line 66:		Line 66:
	=== Bayer colorspace formats ===		=== Bayer colorspace formats ===

-	Several image and video format store pixels using Bayer-pattern colorspaces. Supporting these format would broaded FFmepg's applicability to RAW still and video photography processing. Tasks:	+	Several image and video format store pixels using Bayer-pattern colorspaces. Supporting these format would broaden FFmepg's applicability to RAW still and video photography processing. Tasks:
	* Implement bayer transformations in libswscale (plain C)		* Implement bayer transformations in libswscale (plain C)
	* Add bayer formats to the libavutil pixfmt enumeration routines		* Add bayer formats to the libavutil pixfmt enumeration routines
	* Extend TIFF decoder to support DNG-Bayer format		* Extend TIFF decoder to support DNG-Bayer format
	* Complete PhotoCINE demuxer to support Bayer format; (or another format of your choosing)		* Complete PhotoCINE demuxer to support Bayer format; (or another format of your choosing)
-	* SIMD optimisations of the libswscale transformations	+	* SIMD optimizations of the libswscale transformations
	* decoders/specs may be available in the [[Dcraw]] project		* decoders/specs may be available in the [[Dcraw]] project

Line 90:		Line 90:
	We are also happy to hear your personal project ideas ...		We are also happy to hear your personal project ideas ...

-	=== AAC decoder improvments ===	+	=== AAC decoder improvements ===
	Our AAC decoder does not support low-delay. Part of this task will be to also finish last year's BSAC task. A possible qualification task is to fix a crash in the current BSAC code with one of the samples from the BSAC testing suite.		Our AAC decoder does not support low-delay. Part of this task will be to also finish last year's BSAC task. A possible qualification task is to fix a crash in the current BSAC code with one of the samples from the BSAC testing suite.

-	=== AAC encoder improvments ===	+	=== AAC encoder improvements ===
-	Our AAC encoder does not produce competetive quality per bitrate.	+	Our AAC encoder does not produce competitive quality per bitrate.
	Improve the encoder to be better than some other commonly used encoder like libfaac.		Improve the encoder to be better than some other commonly used encoder like libfaac.
	This requires solid understanding of things like psychoacoustics and rate distortion.		This requires solid understanding of things like psychoacoustics and rate distortion.
	A qualification task for this could be to improve the encoder by at least 5% bitrate at the same quality		A qualification task for this could be to improve the encoder by at least 5% bitrate at the same quality
-	meassured by some objective measure.	+	measured by some objective measure.

	=== FF Fuzzer ===		=== FF Fuzzer ===
Line 104:		Line 104:
	When a crash, or other anomaly is found, it would use git bisect to identify which exact commit introduced the bug.		When a crash, or other anomaly is found, it would use git bisect to identify which exact commit introduced the bug.
	And either display this via some web frontend (similar to fate.ffmpeg.org) or just automatically send an email to some		And either display this via some web frontend (similar to fate.ffmpeg.org) or just automatically send an email to some
-	dedicated mailing list. The system has to be robust (there will be infinite loops, OOM conditions and randomly occuring	+	dedicated mailing list. The system has to be robust (there will be infinite loops, OOM conditions and randomly occurring
	crashes). Its also important that the system is easy to maintain and can filter out duplicates of the same issue.		crashes). Its also important that the system is easy to maintain and can filter out duplicates of the same issue.

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	=== DTS-HD decoder ===		=== DTS-HD decoder ===

-	* ETSI released specifcations (http://www.etsi.org/deliver/etsi_ts/102100_102199/102114/01.03.01_60/ts_102114v010301p.pdf). Your job is to complete the existing decoder with the following features.	+	* ETSI released specifications (http://www.etsi.org/deliver/etsi_ts/102100_102199/102114/01.03.01_60/ts_102114v010301p.pdf). Your job is to complete the existing decoder with the following features.

	(1) Add support for mixed Core + DTS-HD stream structure		(1) Add support for mixed Core + DTS-HD stream structure
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	=== Reverse engineer [[TAK]] codec and write decoder for it ===		=== Reverse engineer [[TAK]] codec and write decoder for it ===
	TAK format is partially already documented.		TAK format is partially already documented.
-	You need to revisit that documentation and update and/or fix any missing/wrong inforrmation.	+	You need to revisit that documentation and update and/or fix any missing/wrong information.
	Reverse engineer codec and write working bitexact decoder.		Reverse engineer codec and write working bitexact decoder.

	=== Support for more subtitle formats ===		=== Support for more subtitle formats ===
-	we have libass support now, either a parser (from mplayer) to convert subs into ass or something else.	+	We have libass support now, either a parser (from mplayer) to convert subs into ass or something else.
	* http://mailman.videolan.org/pipermail/vlc-devel/2011-September/081803.html		* http://mailman.videolan.org/pipermail/vlc-devel/2011-September/081803.html
	* http://ale5000.altervista.org/subtitles.htm		* http://ale5000.altervista.org/subtitles.htm

	=== MKV ordered chapters / playlist support ===		=== MKV ordered chapters / playlist support ===
-	get playlist stuff into ffmpeg. playlist is blocking a few things like quicktime edit list and .asx / .pls files.	+	Get playlist stuff into ffmpeg. Playlist is blocking a few things like quicktime edit list and .asx / .pls files.

	=== Adobe fragmented http in/out ===		=== Adobe fragmented http in/out ===
-	adobe has a new streaming format.	+	Adobe has a new streaming format.
	* info/spec: http://www.adobe.com/products/httpdynamicstreaming/		* info/spec: http://www.adobe.com/products/httpdynamicstreaming/

	=== libavfilter 9/10bit support ===		=== libavfilter 9/10bit support ===
-	make filters work with higher bitrate codecs/colorspaces	+	Make filters work with higher bitrate codecs/colorspaces

	=== Fix copying video between formats ===		=== Fix copying video between formats ===
-	lots of h264 streams in flv, mp4, mkv, mpegts. people wish to remux these into various formats so it works on their hardware (ps3, ipod etc).	+	Lots of h264 streams in flv, mp4, mkv, mpegts. People wish to remux these into various formats so it works on their hardware (ps3, ipod etc).
	https://ffmpeg.org/trac/ffmpeg/ticket/796		https://ffmpeg.org/trac/ffmpeg/ticket/796
	https://ffmpeg.org/trac/ffmpeg/ticket/822		https://ffmpeg.org/trac/ffmpeg/ticket/822
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	=== Port formats/colorspace support from dcraw or make dcraw wrapper ===		=== Port formats/colorspace support from dcraw or make dcraw wrapper ===
-	[[Dcraw]] supports many raw camera formats that ffmpeg may not. port or make a wrapper for this project.	+	[[Dcraw]] supports many raw camera formats that ffmpeg may not. Port or make a wrapper for this project.

	[[Category:FFmpeg]]		[[Category:FFmpeg]]