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January 2012

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ffmpeg-devel.log.20120120
by burek 21 Jan '12

21 Jan '12

[00:02] <ubitux> Compn: isn't it cigaes? [00:02] <ubitux> i don't remember seeing him on irc [01:22] <CIA-31> ffmpeg: 03Clément BSsch 07master * r9d35fa43ee 10ffmpeg/libavfilter/ (af_pan.c vf_fade.c vf_yadif.c vsrc_color.c vsrc_mandelbrot.c): lavfi: add missing periods in descriptions. [02:25] <CIA-31> ffmpeg: 03Diego Biurrun 07master * r45f2908d02 10ffmpeg/ (avconv.c avplay.c cmdutils.c cmdutils.h ffmpeg.c): cosmetics: Rename ffsink to avsink. [02:25] <CIA-31> ffmpeg: 03Diego Biurrun 07master * r395e6402fe 10ffmpeg/tests/ (42 files in 2 dirs): fate: Give some tests more sensible names. [02:25] <CIA-31> ffmpeg: 03Diego Biurrun 07master * re5903e2972 10ffmpeg/tests/fate/ (22 files): fate: add some shorthands to run groups of tests [02:25] <CIA-31> ffmpeg: 03Michael Niedermayer 07master * r48706f41e1 10ffmpeg/: (log message trimmed) [02:25] <CIA-31> ffmpeg: Merge remote-tracking branch 'qatar/master' [02:25] <CIA-31> ffmpeg: * qatar/master: [02:25] <CIA-31> ffmpeg: mpeg12: check for available bits to avoid an infinite loop [02:25] <CIA-31> ffmpeg: fate: add some shorthands to run groups of tests [02:25] <CIA-31> ffmpeg: fate: Give some tests more sensible names. [02:25] <CIA-31> ffmpeg: cosmetics: Rename ffsink to avsink. [02:25] <CIA-31> ffmpeg: 03JindYich Makovi ka 07master * ra85ce653fb 10ffmpeg/libavcodec/mpeg12.c: [02:25] <CIA-31> ffmpeg: mpeg12: check for available bits to avoid an infinite loop [02:25] <CIA-31> ffmpeg: Signed-off-by: Jindrich Makovicka <makovick(a)gmail.com> [02:25] <CIA-31> ffmpeg: Signed-off-by: Diego Biurrun <diego(a)biurrun.de> [02:29] <ubitux> michaelni: sorry for the merge conflict with the tests/fate/*.mak [02:29] <ubitux> michaelni: i didn't expect Diego to do exactly what i did, but putting the rename before& [02:32] <ubitux> i guess those reindent, cosmetics and rename are the most painful things to merge... :p [06:35] Action: Compn wonders why ronald emails one person about doing a joint venture with the two projects for soc instead of just making a post on -devel [06:36] Action: Compn wonders to himself :D [06:36] <ohsix> he used to be a cool guy [13:21] <Cervajz> Hello [13:21] <Cervajz> How can I tell "avformat_find_stream_info" to do not probe private streams in TS, please? [13:21] <Cervajz> This function starts to read whole TS file when it reaches my private stream [13:21] <Cervajz> Then it takes very long time [13:22] <Cervajz> The private streams are marked as "6" [13:38] <CIA-31> ffmpeg: 03Stefano Sabatini 07master * r72092188ef 10ffmpeg/ffmpeg.c: ffmpeg: remove unused and commented variable OutputStream.sync_ipts [13:38] <CIA-31> ffmpeg: 03Stefano Sabatini 07master * reef4b70450 10ffmpeg/ffprobe.c: ffprobe: reindent after previous commit [13:38] <CIA-31> ffmpeg: 03Stefano Sabatini 07master * r4601ad769d 10ffmpeg/ffprobe.c: [13:38] <CIA-31> ffmpeg: ffprobe: fix value_string() prefix printing for values with negative logarithm [13:38] <CIA-31> ffmpeg: The index for the binary_unit_prefixes array is expected to be positive, [13:38] <CIA-31> ffmpeg: so avoid to use negative indexes for accessing it. [13:38] <CIA-31> ffmpeg: 03Stefano Sabatini 07master * r58b10b4c7e 10ffmpeg/ffprobe.c: ffprobe: factorize common code in value_string() [13:38] <CIA-31> ffmpeg: 03Stefano Sabatini 07master * r23934e5f41 10ffmpeg/ffprobe.c: ffprobe: increase precision for the shown float values [15:24] <ubitux> saste: "Note tha a ratio" i think you missed a letter here [15:25] <saste> yes, the doxy contains even more errors [20:37] <CIA-31> ffmpeg: 03Michael Niedermayer 07master * r2d9e2c6464 10ffmpeg/libavutil/opt.c: [20:37] <CIA-31> ffmpeg: opt: Fix "libavutil/opt.c:811:20: warning: initialization discards const qualifier from pointer target type [enabled by default]" [20:37] <CIA-31> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [21:15] <mcbaine1> Hi there ::: I'm really excited about going to FOSDEM ... I put an advert in the national newspaper to see if anyone else is going and to get in touch etc... Anybody available for abit of travelling / FLOSS advice here atmo. ?? [21:28] <merbanan> mcbaine1: a few libav people are going [21:29] <mcbaine1> merbanan, All I hear is that it's the latest and greatest in EUROPE :) [21:30] Action: gnafu puts his hand up to shield his eyes from the sun as he peers off into the distance, looking at a smiling face some miles away. [22:32] <ubitux> michaelni: i have a lot of "Truncating packet of size XXX" XXX being 1 or a value between 400 and 1000+ when decoding MP3 since you added the av_log in ee181f84a31d5ba3efd28bd635f249cc5240f5a8 [22:32] <ubitux> is it a normal behaviour? [22:33] <ubitux> for instance, here i'm decoding a "large" amount of mp3, and got that kind of output: http://pastie.org/3221336 [22:34] <ubitux> i also wonder if the max_analyze_duration warning is really necessary [22:35] <ohsix> if the analysis fails it's good to know when it finally gives up ;] [22:35] <ubitux> these are normal mp3 files, generated with ffmpeg [23:03] <michaelni> ubitux, the mp3 demuxer reads in fixed size amounts and only afterwards the parser splits into mp3 frames [23:03] <michaelni> so the last read will get truncated [23:03] <michaelni> its no error [23:03] <michaelni> somehow we need to suppress the message ... [23:05] <cbsrobot> michaelni: in sws: yuv2planeX_10_c_template [23:05] <cbsrobot> there's: int shift = 11 + 16 - output_bits; [23:05] <cbsrobot> wheres do these numers come from ? [23:05] <cbsrobot> *numbers ? [23:06] <michaelni> ubitux, AVIOContext->maxsize= -1 somewhere in mp3dec.c might work [23:06] <cbsrobot> I'm looking whats needed to add yuv4XXp12* [23:07] <ubitux> michaelni: ok i'ill look into it, thank you [23:07] <michaelni> cbsrobot, just use trial and error with the numbers :) [23:07] <ubitux> but i won't have time until sunday [23:08] <michaelni> ubitux, well its fairly harmless spam so ill let you fix it when you have time ... [23:10] <ubitux> a lot of people are wondering what's wrong with the max_analyze_duration warning, and it's only a warning; here the spam is an error, so users might think something is wrong [23:11] <ohsix> oh, i didn't look at the log, i thought it was just the one :D [23:19] <mcbaine1> http://goo.gl/6knuF [23:23] <gnafu> http://longurl.org/expand?url=http%3A%2F%2Fgoo.gl%2F6knuF [23:28] <durandal_1707> wow hacking base64 got quite popular [23:29] <CIA-31> ffmpeg: 03Reimar Döffinger 07master * r686ba5f4ea 10ffmpeg/libavutil/base64.c: [23:29] <CIA-31> ffmpeg: Minor optimization of base64 decode. [23:29] <CIA-31> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de> [23:29] <CIA-31> ffmpeg: 03Reimar Döffinger 07master * r6af4c9814a 10ffmpeg/libavutil/base64.c: [23:29] <CIA-31> ffmpeg: Optimize output buffer size check in base64 decode. [23:29] <CIA-31> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de> [23:29] <CIA-31> ffmpeg: 03Reimar Döffinger 07master * rd60d718c49 10ffmpeg/libavutil/base64.c: [23:29] <CIA-31> ffmpeg: Slightly optimize base64 encode. [23:29] <CIA-31> ffmpeg: Move handling of last byte outside of innermost loop. [23:29] <CIA-31> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de> [23:29] <durandal_1707> is there variant for yasm yet? [00:00] --- Sat Jan 21 2012

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ffmpeg.log.20120120
by burek 21 Jan '12

21 Jan '12

[00:48] <jerro> Hi, does anyone here know if it's possible to stream webm with ffserver? How do I configure it to do that? [01:08] <shevy> I can use ffmpeg to join several .mp3 files right? [01:11] <treund> cat even [01:22] <Randy_T> I want to use mplayer or mencoder as a source. This works great, except for sources with variable framerate. I can only do video or audio at once so I encode an audio file first, then video, then copy both to an mp4, or I could include the encoded audio while while encoding the video. Mencoder does a decent job with harddup but I still have a video file length of 24:30, and an audio file length of 24:23. So what I really [01:22] <Randy_T> want is some frames to be dropped so video syncs to audio. Would vsync be a solution here and could someone give me an example in this case? The following is my best guess but it drops all the frames: "-vsync -i pipe: -i aud.mp4 -map 0:0,1:0" [01:25] <Randy_T> pipe: being the raw video output from mencoder. aud.mp4 being audio previously encoded with ffmpeg. [03:07] <jerro> Does anyone here know if it's possible to stream html 5 webm with ffserver? If it is, how do I configure it to do that? [07:52] <Arthur_D> for some reason, copying a file with ffmpeg and trimming it at the end with -t left me with no audible sound [07:52] <Arthur_D> I used ffmpeg -i file.mpg -codec copy file2.mpg [07:53] <Arthur_D> when looking at the codec info of the file in VLC, it has many audio channels and it gets more and more while I play the file [07:55] <Arthur_D> so something is wrong, but I don't know how to remedy this unfortunate situation. Care to enlighten me? :) [07:59] <Arthur_D> hm, importing it to PiTiVi works, sound and all... but when opening it in VLC and MPlayer, it doesn't give any sound [09:33] <Arthur_D> still looking for an answer to my audio stream problem [09:39] <sacarasc> Arthur_D: To quote the on join message: [09:39] <sacarasc> ¬ChanServ¬ [#ffmpeg] When asking a question be precise, detailed and patient. If you don't receive a response, try the users mailing list. PASTE TO http://ffmpeg.pastebin.com/ ! If you are asking about CLI usage, paste the complete command and output to the paste bin. [09:42] <Arthur_D> well, it's not like I'm hogging the channel. Noone else has asked for help in the meantime [09:42] <sacarasc> But you've not read it again. [09:42] <sacarasc> Where is the pastebin of the output of ffmpeg? [09:44] <Arthur_D> http://pastebin.com/RMYLff5E well it doesn't show anything wrong as far as I can see [09:45] <Arthur_D> though I guess I better leave that for others to decide :) [09:53] <rfx> Arthur_D: I don't see anything wrong but I'm just a user like you. The only thing I see is that your version about three months old, you might want to update that. What does ffmpeg -i on the new file show? [09:55] <Arthur_D> Duration: 00:10:27.99, start: 1.000000, bitrate: 2463 kb/s [09:55] <Arthur_D> Stream #0:0[0x1e0]: Video: mpeg2video (Main), yuv420p, 720x480 [SAR 8:9 DAR 4:3], 9000 kb/s, 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc [09:55] <Arthur_D> Stream #0:1[0x80]: Audio: ac3, 48000 Hz, stereo, s16, 320 kb/s [09:55] <Arthur_D> so everything seems to be in order [09:56] <Arthur_D> yet VLC and MPlayer doesn't output any sound [09:57] <rfx> Arthur_D: vlc has a log console somewhere, any output there maybe? [09:59] <Arthur_D> not really, but when comparing the original, working file's "module tree" with the copy, there is no audio output in it [10:00] <Arthur_D> funnily enough, it works all right, sound and everything when importing in PiTiVi video editor [10:00] <rfx> I'd try latest ffmpeg just in case .. [10:05] <Arthur_D> I'll try to upgrade it [10:09] <Cervajz> Hello :) [10:09] <Cervajz> Is it possible to switch off (avoid) continuity_check during openning TS file, please? [10:11] <Cervajz> I am trying to open TS file and the "avformat_find_stream_info" function takes a lot of time [10:35] <Arthur_D> rfx: updated ffmpeg, and the issue prevails [12:00] <Arthur_D> alrighty, kinda fixed my issue [12:01] <Arthur_D> used -target ntsc-dvd -vcodec copy to let it automatically choose audio stream output [12:01] <Arthur_D> and thus, it works [13:12] <Cervajz> Hello [13:17] <Cervajz> How can I tell "avformat_find_stream_info" to do not probe private streams in TS, please? [13:18] <Cervajz> This functions started to read whole TS file when it reaches my private stream [13:18] <Cervajz> Then it takes very long time [13:22] <Cervajz> The private streams are marked as "6" [14:07] <rfx> Arthur_D: thanks for sharing, interesting. [15:06] <Arthur_D> rfx: actually, when encoding with the aforementioned options, I got this in the terminal: Incompatible sample format 's16' for codec 'ac3', auto-selecting format 'flt' [15:07] <Arthur_D> thus suggesting that using -acodec copy tried to use the wrong sample format [15:09] <Arthur_D> anyway, it worked, and I'm happy ;) [16:36] <IchGucksLive> Hi all im from Germany i try to find a way to get a AVCHD file from a MOV to play the video from a camcorder on a panasonic screen [16:36] <IchGucksLive> there are 1000 google finds from MTS to ... but not one to the other side [16:37] <IchGucksLive> ffmpeg version git-2011-11-21-2c28e26 [16:39] <Mavrik> IchGucksLive, um... so you have a AVCHD file and you need to convert it to what exactly? [16:39] <IchGucksLive> no the other way i got a MOV from Kodak Zi8 [16:40] <Mavrik> and you need to make an AVCHD file? [16:40] <IchGucksLive> and i want that to play on a usb stick in a Panasonic flatscreen [16:40] <IchGucksLive> Mavrik: yes [16:41] <Mavrik> hmm... I'm not sure ffmpeg supports that... are your sure your Panasonic flatscreen doesn't support mp4 or something like that? [16:42] <IchGucksLive> it says .ps [16:42] <IchGucksLive> or .mts [16:43] <IchGucksLive> i going to have a look at the handbok [16:44] <IchGucksLive> AVCHD, MPEG2 und DivX as transport straem [16:44] <Mavrik> IchGucksLive, it doesn't seem like ffmpeg supports AVCHD encoding [16:44] <Mavrik> ah. [16:44] <Mavrik> MPEG-2 is ok :) [16:45] <IchGucksLive> the transport stream .ps is the container wright [16:45] <Mavrik> .ps is program stream, transport streams are in .ts usually [16:46] <Mavrik> IchGucksLive, ffmpeg.exe -i d:\Videos\MVI_0014.MOV.MP4 -codec:v mpeg2video -b:v 1500k -codec:a mp2 -b:a 128k test.ts [16:47] <Mavrik> that should create a valid stream for the TV [16:47] <IchGucksLive> im on ubuntu [16:47] <IchGucksLive> thanks [16:47] <Mavrik> you can also use "-codec:v libxvid" to encode to DivX for greater quality [16:47] <Mavrik> (if you have it compiled) [16:47] <Mavrik> change -b:v and -b:a parameters for bitrate for the quality you want [16:48] <IchGucksLive> thanks i try this [16:49] <rfx> Irritating nick :) [17:14] <IchGucksLive> Mavrik: thanks i got the problem that libxvidcore4 is on the git ffmpeg but no -codec:v libxvid in codecs [17:15] <IchGucksLive> http://ubuntuforums.org/showthread.php?t=1540254 i updated from medibuntu [17:15] <IchGucksLive> but still no codec [17:15] <IchGucksLive> do i need to recompile ffmpeg [17:21] <IchGucksLive> do i need to restart the system for this ß [17:21] <IchGucksLive> im going foer a restart [17:28] <rhizmoe> going through the build flags, what's the practical difference between faac and vo_aacenc? [18:37] <danverde> anyone using avformat / avcodec in with their own application? [18:40] <danverde> is there an libav user channel? [19:34] <teratorn> danverde: a lot of people use libavformat and libavcodec in their applications [19:35] <danverde> I am wonder where to ask about a memory leak I get when trying to close my format/codev contexts [19:36] <danverde> codec* [19:36] <danverde> I call avformat_close_input but I still am seeing memory leaks [19:36] <danverde> are there other "close" methods I need to be aware of? [19:37] <teratorn> what is being leaked? [19:37] <teratorn> and how do you know it's leaking? [19:38] <danverde> well, that's not real clear to me. I am developing with MSVC and precompiled libav libraries. [19:38] <teratorn> maybe you need to call avcodec_close() on your AVCodecContext [19:38] <danverde> ahh. I am not doing that yet [19:38] <danverde> I'll look at that [19:38] <teratorn> I suggest developing on a real OS that can run valgrind [19:39] <danverde> yeah. I may have to do that if codec close doesn't fix it for me [19:44] <danverde> @teratron: that was it. thx [21:15] <mcbaine1> Hi there ::: I'm really excited about going to FOSDEM ... I put an advert in the national newspaper to see if anyone else is going and to get in touch etc... Anybody available for abit of travelling / FLOSS advice here atmo. ?? [23:19] <mcbaine1> http://goo.gl/6knuF [23:42] <maujhsn> What has to be enabled to get this screencast command to work? http://pastebin.com/HF9KRCWE [23:44] <microchip_> maujhsn: --enable-libx264 [23:45] <maujhsn> microchip_ I thought so! [23:47] <maujhsn> microchip_ Is libx264 one of those "non-free" medibuntu packages from multi-verse? [23:47] <microchip_> i don't know, i don't use *buntu [23:48] <maujhsn> Can anybody else answer this question? [23:49] <Guest50148> um [23:50] <maujhsn> ummmmmmm! [23:50] <Mavrik> there's no reason you'd need libx264 to make a screenshot O.o [23:50] <Mavrik> you forgot to paste the actual command line you use [23:50] <maujhsn> OK! [23:50] <Mavrik> not to mention your ffmpeg version is beyond ancient [23:50] <cbreak> x264 is non-free? [23:50] <cbreak> maybe because it's GPL [23:51] <Mavrik> no, x264 is GPL [23:51] <cbreak> instead of LGPL like the rest of ffmpeg [23:51] <Mavrik> probably some Ubuntu dev has an "opinion" about patents [23:51] <Mavrik> but IIRC it shouldn't be classified as non-free [23:52] <maujhsn> So how is this command modified? "ffmpeg -f alsa -ac 2 -i pulse -f x11grab -r 30 -s 1024x768 -i :0.0 -qscale 1 -acodec pcm_s161e -vcodec libx264 -vpre lossless_ultrafast -threads 0 output.mkv". [23:52] <Mavrik> um [23:52] <Mavrik> what exactly are you trying to achieve? [23:53] <maujhsn> screencast [23:53] <Mavrik> ah. [23:53] <maujhsn> ahhhh! [23:53] <Mavrik> yuck, my mistake, I read "screenshot" -_- [23:53] <maujhsn> screencast [23:54] <Mavrik> maujhsn, anyway, your FFMpeg doesn't have libx264 compiled in and dates back to middle ages [23:54] <maujhsn> and [23:54] <Mavrik> update it to new version that has libx264 support compiled in [23:54] <cbreak> if you want to do live recording, maybe something simple like motion jpeg is ok [23:54] <cbreak> until you later compress it down to something more manageable [23:55] <Mavrik> yea, or that - use mjpeg, since you're not compressing the audio either :) [23:55] <maujhsn> So it is sudo apt-get ? [23:56] <cbreak> you could git clone it :) [23:56] <cbreak> (if you like compiling your software yourself) [23:58] <maujhsn> git clone it...Whats the command? [00:00] --- Sat Jan 21 2012

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ffmpeg-devel.log.20120119
by burek 20 Jan '12

20 Jan '12

[01:12] <Compn> http://cryptome.org/ [01:12] <Compn> cryptic [01:14] <kierank> <BIG><BIG><BIG><BIG><BIG><BIG>NO</BIG></BIG></BIG></BIG></BIG></BIG> [01:14] <kierank> HAHA [01:18] <funman> win 29 [01:24] <Compn> am i missing the joke ? [01:25] <Compn> kierank [01:25] <kierank> no, just a lot of BIG for some reason [01:35] <CIA-31> ffmpeg: 03Anton Khirnov 07master * rbbf0023bb6 10ffmpeg/doc/RELEASE_NOTES: RELEASE_NOTES: mention hiding private symbols in shared builds. [01:35] <CIA-31> ffmpeg: 03Anton Khirnov 07master * r26f71ef48c 10ffmpeg/doc/RELEASE_NOTES: RELEASE_NOTES: mention some notable API changes in 0.8 [01:35] <CIA-31> ffmpeg: 03Donald Ovcharov 07master * rac84f2edd4 10ffmpeg/libavcodec/error_resilience.c: [01:35] <CIA-31> ffmpeg: error_resilience: K&R formatting cosmetics [01:35] <CIA-31> ffmpeg: Signed-off-by: Diego Biurrun <diego(a)biurrun.de> [01:35] <CIA-31> ffmpeg: 03Alex Converse 07master * rd7420e0ac9 10ffmpeg/libavformat/mov.c: mov: Remove dead stores for spherical coordinates for channel position. [01:35] <CIA-31> ffmpeg: 03Anton Khirnov 07master * rf97cb45156 10ffmpeg/libavformat/avformat.h: [01:35] <CIA-31> ffmpeg: lavf: fix and extend av_interleaved_write_frame() doxy. [01:35] <CIA-31> ffmpeg: Specify that lavf is responsible for freeing the data. [01:35] <CIA-31> ffmpeg: 03Anton Khirnov 07master * raa2e4bb058 10ffmpeg/libavformat/utils.c: [01:35] <CIA-31> ffmpeg: lavf: free packets for muxers implementing interleave_packet(). [01:35] <CIA-31> ffmpeg: Fixes a memleak. [01:35] <CIA-31> ffmpeg: 03Gaurav Narula 07master * rb481bbc32a 10ffmpeg/tests/ (audiogen.c base64.c rotozoom.c tiny_psnr.c videogen.c): tests: K&R formatting cosmetics for test programs [01:35] <CIA-31> ffmpeg: 03Michael Niedermayer 07master * rc130428ab5 10ffmpeg/: (log message trimmed) [01:35] <CIA-31> ffmpeg: Merge remote-tracking branch 'qatar/master' [01:35] <CIA-31> ffmpeg: * qatar/master: [01:35] <CIA-31> ffmpeg: avprobe, cmdutils: K&R formatting cosmetics [01:39] <ubitux> saste: thanks for your reviews (and thanks for the L trick... :)) [01:41] <saste> ubitux: np :) [01:48] <ubitux> arh i want to continue the subtitles stuff but i have too much stuff in progress :( [01:48] <ubitux> (not only for ffmpeg unfortunately... :'( [03:38] <xxthink> does ffmpeg support http streaming output? [03:39] <durandal_1707> xxthink: ffmpeg -protocols [03:40] <xxthink> durandal_1707: great! [03:41] <xxthink> but I can't output it [03:41] <xxthink> ./ffmpeg -i mmsh://10.1.104.130:80/fxqiyi1m?MSWMExt=.asf -vcodec libx264 -preset fast -g 60 -vb 500000 -strict experimental -acodec aac -ab 64000 -ar 44100 -ac 2 -f flv http://10.1.30.82:8989 [03:41] <xxthink> [tcp @ 0x1c25fbe0] TCP connection to 10.1.30.82:8989 failed: Connection refused [03:41] <durandal_1707> there is ffserver [03:42] <durandal_1707> but it is in bad shape [03:42] <xxthink> durandal_1707: I should set up ffserver? [03:43] <durandal_1707> ffserver does streaming stuff [03:43] <xxthink> ok [04:13] <xxthink> durandal_1707: it says av_interleaved_write_frame(): Connection reset by peer [04:14] <durandal_1707> not enough information provided [04:16] <xxthink> I use ffmpeg and ffserver to transcode and stream a stream on the fly [04:16] <xxthink> the ffmpeg server is run [04:16] <xxthink> when I use [04:17] <xxthink> ./ffmpeg -i mmsh:/19.151.104.130:80/fxqiyi1m?MSWMExt=.asf -vcodec libx264 -preset fast -g 60 -vb 500000 -strict experimental -acodec aac -ab 64000 -ar 44100 -ac 2 -f flv http://127.0.0.1:8090/feed1.ffm [04:17] <xxthink> it says: av_interleaved_write_frame(): Connection reset by peer [04:18] <durandal_1707> cant you actualy transcode that mmsh it file? [04:18] <xxthink> http://pastebin.com/vFUJD8zn [04:18] <xxthink> this is the whole log [04:19] <xxthink> yes, I can transcode it to file [04:20] <durandal_1707> and can you stream such transcoded file? [04:20] <xxthink> sorry, I don't know how to stream this file [04:20] <xxthink> I can save this flv file. [04:21] <xxthink> do you need my ffserver config? [04:21] <durandal_1707> ffserver can be configured to stream from the file no? [04:22] <xxthink> let me first google how to stream it [04:24] <funman> win 29 [04:26] <xxthink> durandal_1707: [04:26] <xxthink> ffmpeg -i x.flv -s qcif -acodec copy -vcodec copy -f flv http://127.0.0.1:8090/feed1.ffm [04:26] <xxthink> I use this command to stream this flv [04:26] <xxthink> the same error occur [04:26] <xxthink> av_interleaved_write_frame(): Connection reset by peer [04:27] <durandal_1707> perhaps it is problem with flv, try another container, and what version are you using? [04:28] <xxthink> ffmpeg version N-35986-g0edf7eb [04:29] <xxthink> almost the latest version [04:30] <xxthink> mp4, mpegts have the same problem [04:31] <durandal_1707> is there some conf for ffserver which is guaranted to work? [04:32] <durandal_1707> maybe it is regression, in which case you would need to bisect for relevant commit [04:33] <xxthink> old version of ffmpeg is ok? [04:34] <durandal_1707> old version of ff* and libav* [04:37] <xxthink> 0.7.1 is ok [04:44] <xxthink> but only mpegts format [08:59] <CIA-31> ffmpeg: 03Michael Niedermayer 07master * r1ce8377c78 10ffmpeg/libavcodec/wmalosslessdec.c: [08:59] <CIA-31> ffmpeg: wmalossless: Fix incompatoble pointer type warnings [08:59] <CIA-31> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [09:16] <piece3> Hiya, my ffmpeg version is 0.8.7 (updated from fedora repo), was ticket #285 closed out by this version, as I'm getting a similar error [12:53] <Compn> At the Association for Computing Machinerys Symposium on Discrete Algorithms (SODA) this week, a group of MIT researchers will present a new algorithm that, in a large range of practically important cases, improves on the fast Fourier transform. Under some circumstances, the improvement can be dramatic a tenfold increase in speed. [12:53] <Compn> ten fold! :P [12:53] <Compn> http://web.mit.edu/newsoffice/2012/faster-fourier-transforms-0118.html [12:55] <av500> yes, in case you only have a few frequencies in the spectrum [12:56] <av500> if you know the freqs, you can do http://en.wikipedia.org/wiki/Goertzel_algorithm anyway [17:11] <ubitux> michaelni: nice the -fasan box [17:12] <ubitux> are you the one who added the mingw32 boxes btw? [17:12] <michaelni> yes [17:14] <ubitux> ok :) [17:43] <CIA-31> ffmpeg: 03Javier Cabezas 07master * r1406d9ddc6 10ffmpeg/libavcodec/dnxhddec.c: [17:43] <CIA-31> ffmpeg: DNxHD: frame multithreading [17:43] <CIA-31> ffmpeg: Signed-off-by: Javier Cabezas <jcabgz(a)gmail.com> [17:43] <CIA-31> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [17:43] <CIA-31> ffmpeg: 03ami_stuff 07master * ref9623f670 10ffmpeg/libavformat/riff.c: [17:43] <CIA-31> ffmpeg: Add "S263" FourCC [17:43] <CIA-31> ffmpeg: fixes ticket #923 [17:43] <CIA-31> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [19:41] <elupus> Hi, i have a question about avfilter_graph_parse [19:42] <elupus> the ouput open_input_ptr and open_output_ptr the docs say should be freed. But when should they be freed? Can they be freed directly? [19:42] <elupus> Hmm.. sorry this belongs in #ffmpeg [19:42] <elupus> will re-state it there [19:49] <CIA-31> ffmpeg: 03Nicolas George 07master * r4cf81d9f99 10ffmpeg/configure: libcelt: configure: check for recent version. [22:33] <Compn> they fuckin got megaupload! [22:33] <gnafu> D: [22:40] <Compn> if anyone doesnt think its not a big deal... even mplayer and ffmpeg users use megaupload to upload samples! [22:40] <Compn> and if you take one down, theres no stopping them from taking down the rest ! [22:42] <gnafu> I thought you weren't supposed to use file upload sites for samples. [22:42] Action: gnafu ducks. [22:42] <Tjoppen> I was about to say.. [22:42] <Tjoppen> is there some kind of news article on this? [22:43] <Compn> http://www.justice.gov/opa/pr/2012/January/12-crm-074.html [22:43] <gnafu> http://entertainment.slashdot.org/story/12/01/19/210201/megauploadcom-shut-… [22:43] <ubitux> usa, laughter creator since its birth [22:44] <Compn> five years in prison on the charge of conspiracy to commit copyright infringement [22:45] <Compn> ..... [22:46] <Tjoppen> wow [22:46] <Tjoppen> though I suspect you get stiffer sentences for say manslaughter. in sweden, not so much [22:47] <ubitux> btw, once again, centralization shows its weakness again [22:47] <Tjoppen> it's fascinating how worst practice laws rule the internet [22:48] <Compn> theres a few results in google for '5 years for manslaughter' in usa Tjoppen [22:48] <Compn> so drive drunk and kill a guy... 5 years. [22:48] <Compn> conspire to commit copyright infringement .. 5 years. [22:50] <Tjoppen> I'm alluding to the TPB guys being sentenced to far worse than what you get for manslaughter or rape (if I'm not mistaken) [22:51] <Compn> ah [22:51] <Compn> hows the tpb stuff going ? [22:51] <Tjoppen> I'm actually a bit surprised megaupload wasn't taken down sooner, considering they actually host files (unlike tpb) [22:51] <Compn> i havent heard anything about tpb [22:51] <Tjoppen> it's on appeal [22:51] <Compn> except that the appeal judges were all crooked [22:52] <Compn> and in copyright lobbies [22:52] <Tjoppen> court date is.. some time this year? [22:52] <Tjoppen> the entire justice system is in the lobby's hands [22:52] <Compn> what is sweden doing to fix that ? [22:53] <Tjoppen> you're implying the goverment and the riksdag aren't in cahoots with them [22:54] <Compn> i'm talking about its citizens :( [22:54] <Compn> pirate party didnt get many votes :( [22:54] <bcoudurier> michaelni, for very high bitrate [22:54] <Tjoppen> indeed. media preferred to present the neo nazis as the underdog party [22:54] <bcoudurier> I'm not convinced frame threading decoding is the right solution [22:54] <bcoudurier> ex: dnxhd and avcintra [22:54] <bcoudurier> slices work really better in my experience [23:02] <ohsix> mediafire is pretty good gr8 too, just no reason not to use megaupload while megaupload was around [23:04] <Compn> bcoudurier : since you are here , someone posted a patch to support something, ported from ffmbc [23:04] <Compn> but it was originally a patch from somewhere else [23:04] <Tjoppen> dvcprohd [23:04] <Compn> maybe you could look at it and see if it can be lgpld easily ? [23:04] <michaelni> quote from slashdot: "With SOPA, they can take your site down if you link to (or, presumably, mention) megaupload.com. Think about that one for a minute." [23:05] <michaelni> did we do that ? ;) [23:05] <Tjoppen> let's play six degrees of megaupload [23:05] <michaelni> bcoudurier, i agree slices are better if possible [23:05] <ohsix> the kim dotcom guy looked like an easy target, he's been indicted for stuff before; even though it's inadmissable they can probably hang a lot on him because of it [23:05] <durandal_1707> michaelni: grepping ffmpeg git complete history right now.... [23:07] <michaelni> htdocs/bugreports.html:Furthermore movie files uploaded to services like megaupload, rapidshare or [23:07] <durandal_1707> omg, delete that immediately do git rebase too [23:07] <Tjoppen> better use git-filter-branch to purge from history [23:07] <Compn> like i said, ffmpeg used megaupload [23:07] <ohsix> heh does git history run afoul of SOPA or PIPA? man that'd be awful [23:08] <Compn> this is important stuff :( [23:08] <ohsix> you must purge all of recorded online history! [23:09] <Tjoppen> filter-branch is fairly neat. used it at work to clean some huge samples from svn history when porting to git [23:31] <CIA-31> ffmpeg: 03Carl Eugen Hoyos 07master * rb4634a006c 10ffmpeg/libavcodec/ (targa.c version.h): Simplify targa decoding on big endian. [23:31] <CIA-31> ffmpeg: 03Carl Eugen Hoyos 07master * r34aadeaa2c 10ffmpeg/libavcodec/targa.c: [23:31] <CIA-31> ffmpeg: Simplify PIX_FMT_RGB555LE selection in targa decoder. [23:31] <CIA-31> ffmpeg: Suggested by Jean First. [23:31] <CIA-31> ffmpeg: 03Carl Eugen Hoyos 07master * rb614c14767 10ffmpeg/libavcodec/ (pnm.c pnmdec.c version.h): [23:31] <CIA-31> ffmpeg: Simplify 32bit pam decoding. [23:31] <CIA-31> ffmpeg: Reviewed-by: Paul B Mahol [23:31] <CIA-31> ffmpeg: 03Carl Eugen Hoyos 07master * r21a1006109 10ffmpeg/libavcodec/ (sgienc.c version.h): Support encoding 16bit pixel formats to sgi. [23:31] <CIA-31> ffmpeg: 03Carl Eugen Hoyos 07master * re209a37777 10ffmpeg/libavcodec/ (pngdec.c version.h): Simplify 32bit png decoding. [23:31] <CIA-31> ffmpeg: 03Michael Bradshaw 07master * rde07355026 10ffmpeg/libavcodec/libopenjpegenc.c: Changed indexing in libopenjpeg to shorten lines [23:31] <CIA-31> ffmpeg: 03Carl Eugen Hoyos 07master * rdf42dd7323 10ffmpeg/libavcodec/ (libopenjpegenc.c version.h): [23:31] <CIA-31> ffmpeg: Support gray8a encoding with libopenjpeg. [23:31] <CIA-31> ffmpeg: Reviewed-by: Michael Bradshaw [23:31] <CIA-31> ffmpeg: Reviewed-by: Paul B Mahol [23:45] <CIA-31> ffmpeg: 03Nicolas George 07master * ra9575ffc65 10ffmpeg/cmdutils.c: cmdutils: media type to letter conversion in a separate function. [23:45] <CIA-31> ffmpeg: 03Nicolas George 07master * rdcea7ca752 10ffmpeg/cmdutils.c: cmdutils: include type in filter list. [23:52] <Compn> is nicholas george on irc ? [23:52] Action: Compn always forgets nicks [00:00] --- Fri Jan 20 2012

1 0

ffmpeg.log.20120119
by burek 20 Jan '12

20 Jan '12

[01:14] <blizzow> I'm sure this has been answered a million times before, but I don't see a quick google answer. Can ffmpeg read and convert the gotomeeting g2m4 codec to another format? [01:37] <posciak> hi, I'm trying to extract elementary stream from an mp4 container, I tried various combinations of ffmpeg -i foo.mp4 -vcodec copy -f h264 out, but it doesn't seem to be producing what I'm looking for... I can't find valid start code prefixes in output file... What am I doing wrong? [02:01] <zerwas> When i use -sameq to process a asf file, the audio quality drops extremely, why is that? [02:01] <zerwas> Is -sameq only valid for video and not audio? [02:07] <posciak> ok, figured it out, needed the h264_mp4toannexb filter [02:19] <iiu7> I'm converting a MKV file and somewhere while running I get this error and ffmpeg quits. [02:20] <iiu7> Application provided invalid, non monotonically increasing dts to muxer in stream 1: 401137 >= 401077 av_interleaved_write_frame(): Invalid argument [02:20] <LexSfX> i've seen that before too; never did figure it out :L [02:20] <LexSfX> (sorry for the unhelpful comment) [02:20] <iiu7> Is there a way to deal with it? [02:20] <iiu7> np [02:21] <LexSfX> i used a different tool instead, but that's obviously not ideal [02:21] <iiu7> I could try with mencoder, but I really prefer ffmpeg [02:22] <LexSfX> might try using mkvextract to extract the streams, then using ffmpeg from there [02:35] <iiu7> Gonna use mencoder I think, but thanks! [02:35] <iiu7> ;) [03:01] <linuxis> Excuse me, I've been trying to edit this bash script to pipe shoutcast streams through ffplay but I can't seem to do it right [03:02] <linuxis> No matter what, I get errors [03:05] <linuxis> ( http://yp.shoutcast.com/sbin/tunein-station.pls?id=93492 ) [03:05] <linuxis> Uhh sorry disregard that [03:14] <treund> does ffplay take stdin ffplay - [03:16] <xxthink> I use ffmpeg to stream a video on the fly [03:17] <xxthink> ./ffmpeg -i mmsh://10.1.104.130:80/fxqiyi1m?MSWMExt=.asf -vcodec libx264 -preset fast -g 60 -vb 500000 -strict experimental -acodec aac -ab 64000 -ar 44100 -ac 2 -f flv http://10.1.30.82:8989 [03:17] <xxthink> but if fails [03:17] <xxthink> [tcp @ 0x1c25fbe0] TCP connection to 10.1.30.82:8989 failed: Connection refused [03:17] <xxthink> How to give the output parameters? [03:19] <xxthink> ? [03:30] <xxthink> does ffmpeg support http streaming? [03:30] <xxthink> output [03:55] <EOF-sensei> http://se.xlu.be/thaicp.jpg [04:42] <llrcombs> Anyone know much about libavcodec on iOS? [08:38] <piece3> Hi [08:39] <spaam> hi there [08:39] <piece3> Excuse me ppl, is there a way to break 1 video into smaller videos by time? [08:40] <piece3> Say I have 1 minute video and I want to divide it into 6, 10s videos, the "-t 10" option only stops after the first video has been created [08:41] <piece3> e.g. ">ffmpeg -i in_video -t 10 out_video" [08:42] <spaam> then you needd to start at 10s and encode 10s.. then 20s encode 10s and so on. [08:43] <piece3> Ok, the issue I need to resolve is we have an rtsp feed that runs 24h 7 days a week. I need to break that feed up into 30minute segements [08:43] <spaam> -ss 10 .. -ss 20 [08:44] <spaam> ok. [08:44] <piece3> Not missing a frame [08:44] <spaam> then i dont know how you can do it. [08:46] <piece3> Hmm, ok, do you think it may be possible, or for sure it can't be done? [08:51] <piece3> I think I could program a script that every nth amount of time, it started another thread recording the rtsp and then stopped the current one [08:56] <piece3> another question if you don't mind. I keep getting the following error "timebase 1/90000 not supported by MPEG 4 standard, the maximum admitted value for the timebase denominator is 65535" [08:57] <piece3> Any clues on this, sometimes my tbr from my rtsp feeds are 90K other times they are 25 [08:57] <piece3> It seems to be random. [09:06] <piece3> Anybody got any clues on this one, or is this more a devel question? [09:24] <cbreak-work> piece3: 90000 fps? [09:27] <piece3> Yeah, something's not right [09:28] <rfx> I know this Q is not related to ffmpeg, but is there software to recommend (win or linux) which is able to edit containers/streams meta data? i.e. owner, copyright, name of title, etc.? [09:29] <cbreak-work> hex editor + wikipedia? :/ [09:29] <Mavrik_> rfx, depends on a container really [09:29] <Mavrik_> you use your container tools [09:30] <Mavrik_> (or ffmpeg :) ) [09:30] <piece3> cbreak: The media boxes sending out the rtsp feed are actually only sending 25fps, I don't know why sometimes the feeds register on ffmpeg as 25 tbr other times 90k tbr [09:49] <Milos> libaacplus: bad aac setting: br:128000, AACch:2, AACsr:24000 [09:49] <Milos> [libaacplus @ 0x9a1de0] libaacplus doesn't support this output format! [09:49] <Milos> Hmm? [09:49] <Milos> I'm trying to drop side channels, I want only stereo. [09:49] <Milos> The stream apparently has 6 channels. [09:50] <Mavrik_> Milos, libaacplus doesn't support encoding over 64k of bitrate [09:50] <Milos> Which causes "[libaacplus @ 0x8ec110] encoding 6 channel(s) is not allowed" [09:50] <Milos> So I used -ac 2 [09:50] <Milos> Ohhh [09:50] <Milos> Ok, I can change that by doing -ab 32k? [09:50] <Mavrik_> -ab 64k :) [09:50] <Milos> sure [09:50] <Milos> ok thanks [09:50] <Mavrik_> or, in new versions -b:a 64k is preferred [09:50] <Milos> you're super helpful, are you a dev? [09:50] <Milos> :) [09:50] <Mavrik_> I'm not an ffmpeg contributor no... but I do develop software that uses ffmpeg extensively ;) [09:51] <Milos> ah I see! :D [09:51] <Milos> you know a lot, and I thank you for continually giving really good answers, I've been here a few times [09:59] <rfx> Mavrik_: understood .. in my case, it's "only" mp4 and webm right now; just in case, any tools to recommend for those formats? [09:59] <rfx> I mean, wiping all meta data with ffmpeg and writing my own I do probably prefer, for automating ... hmm [10:00] <Mavrik_> rfx, I've heard mp4box is nice for mp4 files... wouldn't know what to use for WebM, sorry [10:03] <Milos> Is it possible to ignore errors like "incomplete frame" when encoding a live stream (which is what I am doing?) [10:04] <Milos> I don't want that to abort the process. [10:05] <Mavrik_> hmmm... it shouldn't fail when encoding [10:08] <Mavrik_> Milos, does it just give warnings or does it crash? [10:10] <Milos> Mavrik_, http://dpaste.com/690157/ [10:10] <Milos> Mavrik_, then it just stops [10:10] <Milos> Mavrik_, doesn't always happen at the same time, sometimes takes seconds, sometimes minutes [10:11] <Mavrik_> hmmm... [10:11] <Mavrik_> it seems your stream is so broken at times it cannot recover [10:11] <Milos> yeah can ffmpeg ignore that [10:11] <Milos> it's broken on purpose; the antenna is not perfect [10:11] <Mavrik_> I don't think so... it's already ignoring all errors it can recover from :\ [10:12] <Milos> I see [10:12] <Milos> well fair enough [10:12] <Milos> although, VLC can do it [10:12] <Milos> if I restream from VLC it works just fine [10:12] <Mavrik_> yeah, ffmpeg is kinda wierd when it comes to receiving TS streams [10:12] <Milos> I see. [10:12] <Mavrik_> which version are you using? [10:12] <Milos> 0.9.1 [10:13] <Mavrik_> is that a DVB-T input? [10:13] <Milos> correct [10:13] <Milos> I basically want to make it crappy [10:13] <Mavrik_> hrm... can't help you there sadly. [10:14] <Milos> it comes in at 1080p, all flash and HD [10:14] <Milos> I just wanna make it crappier so I can stream it off myself [10:14] <Milos> etc [10:14] <Milos> and no worries [10:14] <Milos> I will figure something out! [10:14] <Mavrik_> I've been using ffmpeg to grab MPEG2-TS streams for IPTV and it was kinda capable of recovering from buffer underruns... but badly [10:14] <Milos> 1080i I mean. [10:15] <Milos> Mavrik_, is it supposed to say the input is mpegts though? [10:15] <Milos> Mavrik_, because DVB-T is x264 [10:16] <Mavrik_> yeah, DVB-T is MPEG-TS stream [10:16] <Milos> oh ok [10:16] <Mavrik_> with either H.264 or MPEG-2 video [10:16] <Milos> I'm not so great with codec terminology yet [10:16] <Milos> ah right [10:16] <Mavrik_> depending on country you live in :) [10:16] <Milos> yupyup [10:16] <Milos> well it says x264 [10:16] <Mavrik_> the problem ffmpeg is having with those stream is that it doesn't have a large enough decoder buffer [10:16] <Mavrik_> looking at source it only allocates 64k [10:16] <Milos> aha [10:17] <Milos> and when it runs out? [10:17] <Milos> boom? [10:17] <Mavrik_> you get errors [10:17] <Mavrik_> blocky video [10:17] <Mavrik_> or failure :) [10:17] <Milos> I love blocky video [10:17] <Milos> :D [10:17] <Mavrik_> I've avoided that issue by writing my own IP receiving and buffering routine, but I was developing a live transcoder app [10:17] <Mavrik_> that's probably the reason why VLC can cope [10:17] <Milos> ooo [10:17] <Milos> cool [10:18] <Mavrik_> You'll probably have to use VLC to reencode (it can do that)... even though VLC is a crappy piece of software for that .) [10:18] <Milos> the reason is, VLC is on my laptop here, which gets overutilised especially when restreaming so I'm trying to offload it to a bored quad-core xeon [10:18] <Milos> which is command-line only [10:18] <Milos> but I'll see [10:23] <Mavrik_> you can do command-line only with VLC [10:23] <Mavrik_> it'll leak memory a little (if they didn't fix it by now) [10:23] <Mavrik_> Milos, you can also try appending ?pkt_size=13160 to your http URL to see if it helps [10:24] <Milos> Thanks! I'll see how long it can run for before it stuffs up. [10:25] <Mavrik_> (I'm not sure if that parameter works on HTTP input, it does on UDP :) ) [10:25] <Milos> Another question though. Unfortunately the audio if off, probably because of the frame delays/skips. [10:25] <Mavrik_> yeah [10:25] <Milos> Do you know of a way to get the audio fixed or is that totally impossible [10:25] <Milos> when I say off by the way I mean at the wrong time, not completely turned off [10:25] <Mavrik_> yeah, that's because the video is broken [10:26] <Mavrik_> and timestamps start drifting [10:26] <Mavrik_> if you fix the video problem, the sync problem will go away [10:26] <Mavrik_> you can also try something like -async 50 to tell ffmpeg to try to resync [10:26] <Mavrik_> but that won't help if your video keeps throwing errors [10:26] <Milos> what if a random environmental issue causes the video to mess up for 2 seconds, will everything after that be permanently 2 seconds late? [10:28] <Milos> Mavrik_, that did it!!! [10:28] <Mavrik> does it work? :) [10:29] <Milos> Mavrik, it fixes the sync issue yup! [10:29] <Milos> even with continus frame problems; once the frames recover the audio is correct [10:29] <Milos> is the 50 the frame rate? [10:29] <Milos> or what [10:29] <Milos> since it's 25 interlaced [10:29] <Mavrik> that's the max. number of audio samples ffmpeg can fix per "packet" [10:29] <Milos> ic [10:30] <Mavrik> meaning, if you have a packet of 44100 audio samples [10:30] <Mavrik> it can add/remove/break 50 samples to try to resync [10:30] <Milos> wow [10:30] <Milos> well it does the job [10:30] <Mavrik> if you give a large number it'll recover from drift very fast... but you'll hear the audio "slow" or "speed up"... like the slowmo scenes with exaggerated audio ;) [10:31] <Milos> haaaaahahahah [10:34] <Milos> now is this as easy to restream as typing -f mpegts udp://[interface-ip]:port ? [10:34] <Mavrik> yup [10:34] <Mavrik> if you're doing H.264 [10:35] <Milos> yup [10:35] <Milos> well I hope I am [10:35] <Milos> does ffmpeg support any other protocols? [10:35] <Mavrik> then you'll also need -bsfs mp4_h264toannexb [10:35] <Mavrik> Milos, RTSP and RTP afaik... RTMP as well maybe [10:35] <Milos> ok cool [10:35] <Mavrik> but I've found MPEG-TS to work the most rliably [10:36] <Milos> thanks I will look into all of that [10:36] <Milos> your help is so much appreciated, you have no idea [10:36] <Mavrik> I really need to write a blog post on that topic :P [10:36] <Milos> :D [10:38] <Milos> Mavrik, man that's so cool, the sound is all robotic when it's fixing the sync [10:39] <Mavrik> ^^ [10:39] <Mavrik> btw, how are you getting input over HTTP? [10:39] <Mavrik> do you have a HTTP server on your capture card? [10:39] <Milos> Mavrik, I am using mumudvb [10:39] <Milos> I didn't want to use multicast [10:39] <Milos> so I used http unicast [10:39] <Milos> works really well except it's 1080i and I can't stream that out of my house [10:39] <Milos> I only have ADSL2+ [10:40] <Milos> so I have to re-encode into a lower quality/bitrate and then I can. [10:40] <Mavrik> if you can, try setting up RTSP [10:40] <Mavrik> that could fix your problems [10:40] <Milos> sweet [10:40] <Mavrik> or at least UDP [10:40] <Milos> wouldn't you think that UDP is even worse [10:40] <Milos> because it's a connectionless protocol [10:41] <Mavrik> not really, pretty all IPTV is broadcasted over UDP [10:41] <Mavrik> you don't have all the connection and negotiation overhead [10:41] <Mavrik> and if you miss a packet you don't really care to retransmit it [10:41] <Mavrik> so UDP is an overall better choice [10:41] <Milos> btw, been running for at least 15 mins now without crashing, so either the signal's just better or your additional parameter solved it [10:41] <Milos> and cool, I will look into making it UDP [10:41] <Mavrik> hm [10:42] <Mavrik> it seems RTSP is in alpha on mumudvb [10:42] <Mavrik> Milos, I suggested RTSP because it has transmission control [10:42] <Milos> aha [10:42] <Mavrik> meaning, receiver can tell server "hey, stop sending" [10:42] <Mavrik> which you don't have on HTTP and UDP [10:42] <Mavrik> that could fix your problems [10:42] <Milos> ohoo [10:42] <Milos> s/h// [10:42] <Milos> I see [10:42] <Milos> thanks for that info [10:53] <Milos> final bit of advice... [10:53] <Milos> ffmpeg -i http://192.168.2.1:4242/bynumber/2?pkt_size=13160 -s 711x400 -codec:a libaacplus -ac 2 -b:v 700k -async 1000 -ar 44100 -b:a 32k -f mpegts udp://192.168.2.2:4243 [10:54] <Milos> I've set iptables input policy to ACCEPT so it's not a firewall issue but [10:54] <Milos> when I try connect using VLC it does nothing [10:54] <Mavrik> to the UDP address? [10:54] <Milos> the problem with UDP is you can't get help, because it's connection-less [10:54] <Milos> yeah [10:54] <Mavrik> hmmm [10:54] <Mavrik> usually you need a streaming server for that [10:54] <Mavrik> lemme check [10:54] <Milos> :O [10:55] <Mavrik> hrm [10:55] <Mavrik> -_- [10:55] <Milos> ya [10:55] <Mavrik> no idea, it should work :\ [10:55] <Milos> D: ! [10:55] <Milos> I'll keep trying. [10:56] <Mavrik> I have to run now, g'luck :) [10:56] <Milos> Thanks a lot! [10:56] <Milos> Bye :D [14:46] <Cervajz> Hello guys :) [14:46] <spaam> Hey girl! [14:46] <spaam> wzup? [14:48] <Cervajz> Could anyone tell me what could be wrong if "avformat_find_stream_info" needs a lot of time during opeenning MPEGTS file and my log is full of messages like this: [14:48] <Cervajz> 6060) [mpegts @ 0084F3A0]Continuity check failed for pid 100 expected 15 got 6 [14:48] <Cervajz> (6060) [mpegts @ 0084F3A0]PES packet size mismatch [14:48] <Cervajz> Thank you [14:48] <Cervajz> openning* [14:48] <Mavrik> seems like your input is broken [14:48] <Mavrik> did you pass correct packet_size to demuxer? [14:49] <Cervajz> It could be wrongly "assembled" TS stream, because we do it "manually" in our device. [14:50] <Cervajz> I mean - Our company develops video recorder for aircraft and the output of this recorder is TS stream [14:50] <Mavrik> hmm [14:50] <Mavrik> do you have any way of inspecting that? [14:50] <Mavrik> Cervajz, either your packet sizes are wrong or your continuinity counters are wrong [14:50] <Mavrik> I'd guess the first :) [14:52] <Cervajz> Mavrik: Thank you. "packet sizes" - what should I inspect? I am guy who develop's player for this device - "client side". So should I inspect my side or side where the TS is recorded? [14:53] <Cervajz> who develop* [14:53] <Cervajz> Where the TS is "assembled", a.k.a. in the HW device [14:59] <Mavrik> Cervajz, I suggest you save a stream to a .ts file [14:59] <Mavrik> and find a .TS inspector to check the file [15:00] <Mavrik> to see if the file is OK [15:00] <Mavrik> that will also tell you what the packet size is (it should be 1316) [15:00] <Mavrik> which you can pass to the ffmpeg when reading the stream via ?packet_size=<number> if you're using UDP/HTTP [15:01] <Cervajz> Mavrik: I have got TS in .ts file. This device records it to sdcard [15:16] <Cervajz> Mavrik: Thank you, I will try to focus on the packet size [15:16] <Cervajz> Mavrik: Interesting is, that VLC doesn't have problem and open this file quickly [15:18] <Mavrik> yeah, VLC can survive more garbage than FFMPEG's demuxe [15:18] <rfx> Hmm ... -ab is not documented on http://ffmpeg.org/ffmpeg.html ; is this report-worthy? [15:18] <Mavrik> btw, which ffmpeg version are you using? [15:18] <Mavrik> rfx, -ab is deprecated [15:18] <Mavrik> rfx, new notation is -b:a [15:18] <Mavrik> pretty much all such parameters have been changed to <paramerer>:<v|a> notation so they don't get mixed up between video and audio [15:18] <rfx> Mavrik: achso .. because it's still listed in ffmpeg -h (using snapshot from a few days) but not marked as deprecated there [15:19] <rfx> Mavrik: thanks a lot [15:19] <Mavrik> so now you have -codec:v, -codec:a, -profile:v, -profile:a, -b:v, -b:a etc :) [15:19] <Mavrik> rfx, yeah, it's a slow change, most of the "old" parameters still work [15:19] <Mavrik> except in some cases ;) [15:19] <rfx> :D [15:20] <Cervajz> Mavrik: 0.9.1 [15:20] <Mavrik> (like if you want to use -profile with libx264 and you also have libfaac :P ) [15:21] <rfx> Mavrik: is it worth reporting to mark -ab deprecated in the ffmpeg? or shall i just forget about it [15:21] <Mavrik> hm [15:22] <Mavrik> rfx, I'm not sure what the "official" stance on this is ;) [15:22] <Mavrik> maybe I shoudl start reading -devel lsit :\ [19:42] <elupus> Hi, i have a question about avfilter_graph_parse, the ouput open_input_ptr and open_output_ptr the docs say should be freed. But when should they be freed? Can they be freed directly without it breaking the open graph? [19:43] <elupus> ffplay never seem to free them [21:09] <linuxis> Is anyone here? [21:16] <spaam> no [00:00] --- Fri Jan 20 2012

1 0

ffmpeg-devel.log.20120118
by burek 19 Jan '12

19 Jan '12

[00:00] <durandal_1707> there is single corrupted frame near at end with second sample(did not tried first one) [00:01] <durandal_1707> cbsrobot: so make sure to test both samples [00:05] <cbsrobot> durandal_1707: first sample has artifacts at the bottom in vlc, but not in ffplay [00:05] <cbsrobot> second sample is as you say - but no artifacts [00:08] <durandal_1707> second sample in mplayer have strange last line, so maybe that is regression... [00:09] <durandal_1707> it appears to have no color at all [00:10] <CIA-31> ffmpeg: 03Stefano Sabatini 07master * r54661219c1 10ffmpeg/ffprobe.c: [00:10] <CIA-31> ffmpeg: ffprobe: fix printing of unit values which cannot be contained in an int [00:10] <CIA-31> ffmpeg: Use long long int to contain such values instead of an int, which is [00:10] <CIA-31> ffmpeg: required to contain at least 64 bits, so it is guaranteed to contain also [00:10] <CIA-31> ffmpeg: int64_t values, which are used by some fields. [00:10] <CIA-31> ffmpeg: In particular, should fix trac ticket #921. [00:17] <cbsrobot> it looks like there are some color artifacts at the top row, left side [00:17] <cbsrobot> michaelni: do you see that too ? [00:25] <michaelni> cbsrobot, which video with which player and where on the timeline ? [00:25] <cbsrobot> both from https://trac.videolan.org/vlc/ticket/5872 [00:26] <cbsrobot> a bit everywhere - just the top row left side [00:26] <cbsrobot> ffplay [00:26] <michaelni> there are 2 videos in that ticket [00:26] <michaelni> cmi & outlaw [00:26] <cbsrobot> both [00:27] <michaelni> is this also vissible if you convert to jpg/png with ffmpeg ? [00:27] <michaelni> because if yes, please upload a picture of these artifacts [00:29] <cbsrobot> no I cant see them [00:35] <michaelni> i cant reproduce this :( [00:45] <CIA-31> ffmpeg: 03Stefano Sabatini 07master * r140a9afcf2 10ffmpeg/ffprobe.c: [00:45] <CIA-31> ffmpeg: ffprobe: add more safe casts in value_string() [00:45] <CIA-31> ffmpeg: Second attempt at fixing ticket #921. [00:45] <CIA-31> ffmpeg: 03Stefano Sabatini 07master * r2c8d37c2fb 10ffmpeg/ffprobe.c: [00:45] <CIA-31> ffmpeg: ffprobe: use safer type for index variable in value_string() [00:45] <CIA-31> ffmpeg: There is no guarantee that the casted double which is assigned to the [00:45] <CIA-31> ffmpeg: variable will be contained in an int (also if it is almost sure for most [00:45] <CIA-31> ffmpeg: non-alien architectures). [02:37] <CIA-31> ffmpeg: 03Anton Khirnov 07master * r9993284793 10ffmpeg/avconv.c: [02:37] <CIA-31> ffmpeg: avconv: fix -frames for video encoders with delay. [02:37] <CIA-31> ffmpeg: Frames must be counted when they are passed to the encoder, not when [02:37] <CIA-31> ffmpeg: they come out. [02:37] <CIA-31> ffmpeg: Fixes Bug 202. [02:37] <CIA-31> ffmpeg: 03Justin Ruggles 07master * r01ed1c390d 10ffmpeg/libavcodec/tta.c: [02:37] <CIA-31> ffmpeg: tta: cast output data pointer to the correct type [02:37] <CIA-31> ffmpeg: fixes "warning: assignment from incompatible pointer type" [02:37] <CIA-31> ffmpeg: 03Michael Niedermayer 07master * re191f1f414 10ffmpeg/: [02:37] <CIA-31> ffmpeg: Merge remote-tracking branch 'qatar/master' [02:37] <CIA-31> ffmpeg: * qatar/master: [02:37] <CIA-31> ffmpeg: tta: cast output data pointer to the correct type [02:37] <CIA-31> ffmpeg: avconv: fix -frames for video encoders with delay. [02:37] <CIA-31> ffmpeg: Merged-by: Michael Niedermayer <michaelni(a)gmx.at> [04:46] <piece3> Hi all, does ffmpeg currently offer muxing more than 2 video streams simultaneously? or is this feature not yet implemented? [04:49] <piece3> Excuse my last question, wrong channel, plus I have just confirmed it does :) have a good day [06:32] <Compn> wikipedia is still up [06:32] <Compn> what a liar jimbo is [06:32] <Compn> unless i'm stuck in some cache [06:36] <t1g3r> blockd from here [07:07] <michaelni> Compn, the wiki blackout only works with javascript enabled AFAIK [07:08] <michaelni> also its just the english wikipedia [08:28] <av500> ffmpeg.org is also blacked out? [08:50] <ubitux> making wikipedia available when you have no js is a real joke& [08:51] <av500> ubitux: its not about making it unavailable [08:51] <av500> they even state in the help section how to disable the blackout [08:51] <av500> like using m.wikipedia [08:51] <ubitux> yes i read it [08:52] <av500> and most non-geek users have js enabled and these are the ones they try to reach [08:52] <ubitux> if it doesn't annoy people i guess the effect will be way less effective, but just my PoV [08:53] <av500> it will annoy enough people [08:53] <ubitux> it would have been nice to blacklist USA instead [08:53] <ubitux> to force them to use another country as proxy [08:53] <ubitux> could have been fun :) [08:56] <av500> ubitux: https://twitter.com/#!/herpderpedia [08:57] <ubitux> haha [09:00] <ubitux> "I think Wikipedia planned this shit." [09:00] <ubitux> there are good ones :') [09:00] <av500> yep [13:42] <Compn> michaelni : ah, i usually browse without javascript [13:42] <Compn> since wikipedia pages have become so bloated [14:33] Action: Compn writes another run on sentence [14:33] <Compn> grammar getting worse :) [16:33] <durandal_1707> hmm, ffplay gives bmp little darker and less sharp than feh/ImageMagick [16:34] <durandal_1707> happens with any image format, so sws is really busted [16:39] <CIA-31> ffmpeg: 03Dmitry Volyntsev 07master * rb29ec002bb 10ffmpeg/libavformat/ (rtsp.c rtsp.h): [16:39] <CIA-31> ffmpeg: fix rtp 916 issue [16:39] <CIA-31> ffmpeg: Reviewed-by: Luca Abeni <lucabe72(a)email.it> [16:39] <CIA-31> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [16:39] <CIA-31> ffmpeg: 03Dmitry Volyntsev 07master * r4259d1eb99 10ffmpeg/libavformat/rtsp.c: [16:39] <CIA-31> ffmpeg: fix rtp 916 issue, patch 2 [16:39] <CIA-31> ffmpeg: Reviewed-by: Luca Abeni <lucabe72(a)email.it> [16:39] <CIA-31> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [16:39] <CIA-31> ffmpeg: 03Michael Niedermayer 07master * re49780f749 10ffmpeg/libavcodec/utils.c: (log message trimmed) [16:39] <CIA-31> ffmpeg: lavc: call ff_init_buffer_info() from default reget buffer. [16:39] <CIA-31> ffmpeg: This is needed in case the get_buffer() callback doesnt set [16:39] <CIA-31> ffmpeg: width/height. [16:39] <CIA-31> ffmpeg: Ideally all decoders would make calls through some wraper [16:39] <CIA-31> ffmpeg: to the callbacks and that wraper would call ff_init_buffer_info() [16:39] <CIA-31> ffmpeg: But until thats done, the default reget buffer must call this [16:39] <CIA-31> ffmpeg: 03Michael Niedermayer 07master * r2fb65d3e55 10ffmpeg/libavcodec/utils.c: [16:40] <CIA-31> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [16:42] <michaelni> durandal_1707, ffplay uses SDL fixed in YUV420 mode so RGB gets its chroma downconverted which surely doesnt help sharpness [16:43] <michaelni> the darker could have various reasons from bugs in sws over bugs in SDL to gamma correction differences [16:43] <ubitux> (hey fate is almost all green!) [16:44] <ubitux> (time to add a valgrind tracking undef read) [16:45] Action: michaelni is working on adding asan [16:45] <ubitux> oh? great :) [16:45] <ubitux> are you expecting it to raise things valgrind doesn't handle? [16:46] <michaelni> a quick preliminary tests had something in eac3 failing IIRC [16:46] <ubitux> oh? ok [16:46] <michaelni> asan is able to detect stack issues better than valgrind and valgrind is better at leaks and undef reads AFAIK [16:47] <michaelni> i also tried safecode yesterday [16:47] <ubitux> this make me thing we should have one build with -fstack-protector-all or sth like that [16:47] <ubitux> but i think it's the default on openbsd [16:47] <michaelni> but it deadlocked somehow in mpeg2 -therads > 1 [17:24] <durandal_1707> anybody have working Xorg 1 or 4 depth display please upload xwd -root dump [17:24] <durandal_1707> or find xwd with gray colorspace [18:35] <durandal_1707> xy image format is ugly to support in ffmpeg [21:22] <CIA-31> ffmpeg: 03Reimar Döffinger 07master * rf9ced97543 10ffmpeg/libavcodec/ (fraps.c mpc.h mpc7.c): [21:22] <CIA-31> ffmpeg: Use av_fast_padded_malloc in fraps and mpc decoders. [21:22] <CIA-31> ffmpeg: Fixes FATE failures due to uninitialized reads under valgrind [21:22] <CIA-31> ffmpeg: for these two codecs. [21:22] <CIA-31> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de> [21:23] <j-b> m [21:23] <j-b> oops [21:41] <gnafu> Moops? [21:42] <funman> moo process show : a CowOS command [21:44] Action: gnafu prefers BullOS. [21:46] <CIA-31> ffmpeg: 03Michael Niedermayer 07master * r126b04196e 10ffmpeg/libpostproc/postprocess.h: [21:46] <CIA-31> ffmpeg: pp: bump major in git master to match what debian contains. [21:46] <CIA-31> ffmpeg: The ABI differences are negligible, but its easier for all if [21:46] <CIA-31> ffmpeg: all distros have libpostproc HEAD under the same soname and [21:46] <CIA-31> ffmpeg: debian bumped soname without consulting upstream, so as silly as [21:46] <CIA-31> ffmpeg: it is following this is probably the least pain for all. [21:46] <CIA-31> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [22:04] <CIA-31> ffmpeg: 03Reimar Döffinger 07master * re8b060f7e7 10ffmpeg/libavcodec/ (mimic.c truemotion2.c): [22:04] <CIA-31> ffmpeg: Use av_fast_padded_malloc in mimic and truemotion2 decoders. [22:04] <CIA-31> ffmpeg: Fixes use of uninitialized data errors under valgrind. [22:04] <CIA-31> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de> [22:04] <CIA-31> ffmpeg: 03Reimar Döffinger 07master * r6fd4b8a6ef 10ffmpeg/libavcodec/fraps.c: [22:04] <CIA-31> ffmpeg: Replace a av_fast_malloc with av_fast_padded_malloc. [22:04] <CIA-31> ffmpeg: This one was missed in the previous fraps fix, the [22:04] <CIA-31> ffmpeg: allocation is exactly the same in both cases. [22:04] <CIA-31> ffmpeg: Fixes fraps-v5 under valgrind. [22:04] <CIA-31> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de> [00:00] --- Thu Jan 19 2012

1 0

ffmpeg.log.20120118
by burek 19 Jan '12

19 Jan '12

[00:07] <YellowOnion> is there anyway to get the hq denoiser to not look like 'airbrushed crayola(tm)' [02:38] <piece3> Hi all [02:39] <piece3> Does anybody have experience creating an mkv with 2 or 3 channels(views/angles) with ffmpeg? [02:43] <newl> 3D? [02:45] <piece3> Yeah, well kind of, we're wanting to put 3 channels together, this worked with mkvtoolnix in post processing, when playing back all 3 channels play in sync, but haven't been able to replicate this with ffmpeg [02:47] Action: newl has his 3D glasses on waiting to watch [02:50] <freeminds> can I force ffmpeg to use one particular time pattern? Sometimes its like: time=3.29 other times it is like this: 00:00:03.29 [02:54] <piece3> This is what I'm currently trying: [02:54] <piece3> "ffmpeg -i rtsp://192.168.1.60:554/udpstream_ch1_stream1 -i rtsp://192.168.1.61:554/udpstream_ch1_stream1 -an -vcodec copy -metadata stereo_mode=left_right -t 10 test_020.mkv" [03:07] <piece3> Does anybody know a good place to find tutorials on ffmpeg and matroska? [03:09] <newl> you got a shell, an editor and can do loops ? who needs a tutorial [03:09] <newl> start with man ffmpeg [03:10] <newl> post here when it is buzzing with challangeable knowledge [03:10] <piece3> Went through it, the matroska section specs that stero_mode should yield 2 channels, however the output doesn't [03:11] <piece3> Ok, cheers [03:11] <newl> -ac 2 ?? [03:11] <newl> i thought i always get stereo .. [03:12] <piece3> Stereo video [03:12] <piece3> or multi channel video [03:12] <newl> oh yea sorry [03:15] <piece3> np, what time is it most active in here [03:16] <newl> use to be when the bars in berlin let out but .... [03:17] <piece3> Great, I'm on the outher side of the world, so guess it will be midinight for me [06:54] <theperfectpunk> Hey [06:54] <theperfectpunk> Guys [06:55] <theperfectpunk> Anybody? [06:56] <theperfectpunk> Blister [07:02] <arbin> how do I set the profile to baseline using x264opts? [07:52] <dies_irae> hei [07:52] <dies_irae> ei [07:53] <dies_irae> ppl awokened at this time of the date? [07:54] <sacarasc> arbin: profile=baseline I think. [07:54] <dies_irae> ok [07:54] <arbin> that didn't work [07:55] <dies_irae> how to fix 240p youtube vids?? I mean up-scale them to better rez. [07:55] <arbin> i ended up doing -profile baseline and it worked somehow [07:55] <dies_irae> 240p quality is absolutely attrocious [07:55] <arbin> dies_irae, yes, do -s 9000x9000 [07:55] <arbin> but don't expect better quality [07:55] <sacarasc> dies_irae: You won't gain quality by upscaling, in fact, you'll probably lose some. [07:56] <dies_irae> no algorithm exist to fix them? like the up-scaling algos on emulators for 8-bit and 16-bit. [08:23] Last message repeated 1 time(s). [08:23] <dies_irae> was this message delivered?? ^^^ [08:34] <dies_irae> best command for tegra mp4 for maximum quality and minimum size at the same time?? [08:35] <dies_irae> and for ipad2? [09:02] <piece3> Can anybody assist on muxing Subtitles? [10:33] <rfx> Hello [10:49] <rfx> Pardon me, what is the proper way to adjust the quality when using x264 for an mp4? I though using "qscale" but it gives me same result for values 1 or 31 ... am I completely wrong? Using current ffmpeg git head. [10:50] <JEEB> crf is closest to 'constant quality' [10:50] <JEEB> (for libx264) [10:51] <JEEB> start from around 24 or so, and proceed downwards if it looks bad, and upwards if it looks good [10:51] <JEEB> the idea is to pick the highest number that still looks good for you [10:51] <rfx> JEEB: ok, thank you. So qscale is actually not applicable when using x264 .. now I looked about crf and I see it in the help output, mentionig x264 and vp8 .. interesting. [10:52] <JEEB> ffmpeg -i input -vcodec libx264 -preset derp -tune herp -crf wee -acodec whatever output.mp4 [10:53] <JEEB> replace derp with the slowest preset you can take from libx264, herp with possible tuning that matches your source, and crf with the highest value that still looks good for you [10:53] <JEEB> http://mewiki.project357.com/wiki/X264_Settings#preset <- list of libx264 presets [10:54] <JEEB> (everything but the borderline presets should be more or less OK to use, placebo and ultrafast are generally for your benchmarks) [10:54] <rfx> Ok .. thanks for the help. I was on this page but I wasn't able to connect the pieces together, how much from x264 is applicable when using ffmpeg command line .. I see now. So the "-tune" is actually not from ffmpeg but libx264 ? [10:55] <JEEB> most commonly used settings such as crf, preset, tune, profile can be used in ffmpeg easily with their own setting in ffmpeg [10:56] <JEEB> basically, a ffmpeg setting that maps to a libx264 setting [10:56] <rfx> Ah ... [10:56] <JEEB> then there's the setting for setting specific libx264 flags, too [10:57] <rfx> Ah .. ok, I think I understand now. On http://ffmpeg.org/ffmpeg.html#Options-4 I see "x264opts" .. [10:57] <JEEB> yeah [10:57] <JEEB> x264opts is separate [10:58] <JEEB> and you usually don't need it [10:58] <JEEB> I only needed it to set certain settings for a certain hardware device that had extra limitations on top of the usual limitations of profiles and levels [10:58] <rfx> The think is, I googled a lot and saw a lot of examples and by reading ffmpeg documentation I would have definitly assumed qscale would influence the quality for x264; weird. [10:58] <rfx> s/think/thing [10:59] <JEEB> -qscale seems to be used in other internal ffmpeg codecs, but I have no idea of it and how it maps with libx264 [10:59] <JEEB> now that ffmpeg has a -crf options, I don't see any reason to use -qscale [10:59] <JEEB> .-. [10:59] <JEEB> *option [10:59] <rfx> thanks, noted :) [11:00] <JEEB> if you are encoding for PC playback, you can just let loose with settings, no need to set level/profile. if you are encoding for hardware, then you might want to set profile and level (profile = limits features, level = limits the amount of memory the decoder has to have available) accordingly [11:01] <JEEB> then there's special stuff such as blu-ray compatibility which has its own switch :) [11:02] <JEEB> (this is only needed if you are encoding a raw H.264 stream for muxing into a commercial blu-ray) [11:03] <rfx> actually .. I'm encoding for the internetz; like iphone/ipad/android .. the usual non-exiting stuff [11:03] <JEEB> well, in that case you will need to set level and profile matching to the lowest feature set you want to support [11:04] <JEEB> if you want to support android phones that came in during early 2009 f.ex. [11:04] <rfx> I've already gathered some examples and also got things properly playing back on even iPhone3G (by adjusting the profile/level); but I'm far from properly understand all these examples where people through rediculious command lines at your without explaining a single thing :p [11:04] <JEEB> you'd need to set -profile baseline and -level 3 [11:04] <JEEB> yeah, up to 3G it's profile baseline and level 3 [11:05] <JEEB> 3GS has a much better ASIC and can handle high profile (default, doesn't have to be set) and level 4/4.1 [11:05] <JEEB> :) [11:05] <rfx> I dug out my ./to_mp4_iphone script and I have this in there: "-vprofile baseline -level 13". hmm [11:05] <JEEB> level 1.3 .-. [11:05] <JEEB> sure is limited [11:05] <JEEB> but if you really need that, sure [11:06] <JEEB> I know for a fact that my old phones supported level 3 [11:06] <rfx> jesus. no, I've actually two (!) -level arguments, one to 30 and one 13 .. definitly not intentional. [11:06] <JEEB> hah [11:06] <rfx> a topic copy/paste/concatenate thing .. argh [11:06] <rfx> s/topic/typical/ [11:06] <JEEB> also, if you're using faac or any other aac encoder you'll probably want to use -profile:v and thus specify that you're setting the setting only for the video encoder [11:07] <JEEB> because aac encoders tend to eat the -profile setting as well, and then end up erroring out [11:07] <JEEB> because there is no 'baseline' or 'main' profile in aac [11:07] <JEEB> lol [11:08] <rfx> Thanks, I was about to ask about the "-vprofile", because .. it's all so complex it seems, when I look at http://ffmpeg.org/ffmpeg.html or http://mewiki.project357.com/wiki/X264_Settings there is no "vprofile" at all anywhere. Is this old, for compatibility? [11:08] <JEEB> vprofile is the old name for profile [11:08] <JEEB> and still kept around [11:08] <JEEB> like vpre for preset [11:08] <JEEB> and yes, it's ffmpeg specific so it's not in libx264 naturally [11:09] <JEEB> anyways, the older devices you'll try to support the less compression you get, but at least with H.264 there's usually no black magic involved :) [11:10] <JEEB> mpeg-4 part 2 on the other thand... there's a reason why I will never go back into encoding stuff with it even for random plastic boxes :D [11:10] <rfx> I wish there would be a page provider "default" examples for all the common devices out there and explaining what to tweak in case one wants to tweak something. [11:12] <rfx> About the "vprofile" example: it's not documented on one of the both URLs currently; but "profile" is documented on both pages. How do I know which one I'm using? I guess using -profile:v is just ffmpeg specific, but -profile seems ambiguous to me. How can I better see the distinction? [11:16] <JEEB> eh [11:16] <JEEB> I don't get the "How do I know which one I'm using" [11:16] <JEEB> since currently vprofile is just a legacy name for profile .-. [11:16] <JEEB> and -profile is just sets the "profile" setting for whatever for a given encoder/decoder [11:16] <JEEB> libx264 being one of them [11:17] <JEEB> and what exactly the "profile" setting does is encoder/decoder specific (format-specific) [11:18] <JEEB> so in a way -profile is ffmpeg-specific as it maps to different things in different codecs, but if you are using libx264 it maps to its --profile setting [11:18] <JEEB> I don't see what exactly is hard in that .-. [11:18] <JEEB> -profile:v just specifies that you only want to set the setting to the video encoder/decoder (if you set it before the -i it's for the decoder, otherwise it's for the encoder) [11:20] <JEEB> also, "for all the common devices" in most cases is just a matter of setting the lowest point you support [11:20] <JEEB> and then encoding for it [11:20] <JEEB> since the amount of devices with VeryWeirdASICs is rather small [11:22] <JEEB> you can pretty much assume that a relatively low-res baseline/level 3 encode will work on most devices you will pretty much ever meet that support H.264 [11:22] <rfx> Thanks, that helps understanding. Well, I assume you've worked quite a lot with ffmpeg and know it, I wasn't aware of those though I tried to carefully read its documentation. See, when I get slammed those examples found on the internet, I try to lookup every parameter to understand it. 'crf' example: when I look at the web page, I see it's documented only for libvpx and says " VPX_CQ, VP8E_SET_CQ_LEVEL [11:23] <rfx> ", but when I look the command line I see another documention (which is more helpful, btw). Maybe my approach is wrong, I don't know. [11:24] <rfx> OTOH, no one said the video encoding topic is a simple one :) [11:25] <JEEB> I've more worked with x264 than ffmpeg :D [11:25] <JEEB> I used to think that I'll never use libx264 with ffmpeg [11:26] <rfx> Yeah, lol, that's also an interesting thing: having done some ffmpeg example the past week, I just realized that libx264 has it's own binary and capable of doing the encoding itself .. didn't knew that. [11:26] <JEEB> now that ffmpeg actually has sane switches that make using libx264 easier, it's usable [11:27] <JEEB> (-profile/-preset/-crf/-tune) [11:28] <JEEB> the reason why people wanted to f.ex. implement video filtering and audio encoding into x264cli (the 'x264' tool that you use from command line) was exactly because using libx264 via ffmpeg used to be a mess [11:28] <rfx> oh [11:44] <rfx> JEEB: well, many many thanks for your time! need to move on; nice channel so far :) [11:45] <JEEB> no problem [12:25] <niro_> Hi, can anyone tell me how to set the compression ratio when saving to a jpeg from a usb cam... [12:40] <solexious> Hello, is there a way to pipe the information from cropdetect back into ffmpeg to crop the video to that value? [13:01] <LedM> Hello: i have a simple 7MB binary of ffmpeg that runs on absolutel every linux distribution, and want to know about a problem to sinchronize with the ffplay (Staic) [13:01] <LedM> (Static) [13:06] <ikonia> you shouldn't have a single binary for every linux distro [13:06] <ikonia> you should have a binariy built for (and probably by) your distribtion [13:08] <ikonia> ooh, he's gone [13:33] <rfx> zeranoe: are you here? [14:23] <rfx> JEEB: I just made a test using profile baseline and no specific level given and plays fine on iphone 3g [14:24] <JEEB> well yeah, libx264 nowadays calculates the level from what the preset sets + profile's limitations + resolution + frame rate [14:24] <JEEB> you should see it in libx264's output that's shown in the terminal [14:25] <JEEB> with small resolution/frame rate etc. you probably don't need to worry about it too much [14:25] <JEEB> but it's still nice to set the maximum level you want, as that will make sure that libx264 will not go over it [14:25] <JEEB> in any case (unless you specifically set options by hand) [14:30] <Xedecimal> I've got a series of avi files with 2 audio streams in them, I was using ffmpeg to -map 0:0 -map 0:2 (skipping 0:1) to another file, when using both -acodec copy and -vcodec copy, it'll still change the audio and screw it all up (as in it's just random noise), these are xvid avis [14:30] <Xedecimal> what I mean by change the audio is that it's coming from 128k, but in vlc opening the result video file and looking at codec information it suddenly says it's a 98kbit audio [14:31] <rfx> JEEB: I also noted: such an mp4 can play on android but only works directly streamed over HTTP if I use "MP4Box -hint" to add some hinting to it. Given that, I was able to create an .mp4 I can play from ios/ipad over android even in ie9 ... [14:31] <Xedecimal> if I run -vn and export the audio out to an mp3 file, it comes out flawlessly [14:33] <JEEB> Xedecimal, could be a lulzy avi file. VBR audio in avi to begin with is a hack [14:33] <JEEB> (as in, you have to mux it in as a video track as audio IIRC is CBR only in AVI) [14:33] <Xedecimal> should I just re-encode the audio ? [14:33] <JEEB> nah, if you can first extract the audio and that's fine, just do that first [14:34] <JEEB> and then remux to matroska or something capable of taking in VBR audio (mp4/mkv/flv come to mind) [14:34] <Xedecimal> ahhhhh! pull the mp3, then do the video without audio and theeeeen put them back together, duh [14:34] <JEEB> well, you can set multiple inputs after all :) [14:34] <JEEB> as far as I know [14:35] <JEEB> so you can mux the video and audio together in one step [14:35] <JEEB> although it could also of course be a bug / not a supported "feature" in ffmpeg, so you might check out the ffmpeg bug tracker [14:35] <JEEB> and submit it there to be checked, with a sample file [14:36] <JEEB> oh, and make sure it still happens with ffmpeg's HEAD [14:36] <JEEB> as in, current revision [14:36] <JEEB> rfx, you should be able to play it in all of those via HTTP if you add the index atom into the beginning of the file... I don't really get what you're trying to say tbqh [14:36] <Xedecimal> is N-36890-g67f5650 the latest ? [14:36] <Xedecimal> (windows binary that is too) [14:37] <JEEB> e191f1f [14:37] <JEEB> is the current commit [14:37] <JEEB> http://git.videolan.org/?p=ffmpeg.git;a=commit;h=e191f1f4141aec6b40d12af483… [14:37] <JEEB> http://git.videolan.org/?p=ffmpeg.git;a=shortlog [14:56] <dies_irae> best command for ipad2 mp4 (h264) for maximum quality and minimum size at the same time?? [14:57] <rfx> JEEB: I was tryint to say: the .mp4 created with ffmpeg/x264 in this case did not play in one of two cases on Android (2.2 in my case): first case) download mp4, play, works ; second case) do not "download" but rather stream directly -> does not work without using the MP4Box application. [14:58] <cbreak-work> maximal quality and minimal size are conflicting requirements [14:58] <dies_irae> my Android tablet plays mp4 files without any modification though. [14:58] <cbreak-work> you will NEVER get both [14:58] <JEEB> dies_irae, -vcodec libx264 -preset placebo -crf <highest crf that still looks good for you> [14:58] <rfx> dies_irae: likely that tables also have newer android versions ... [14:58] <cbreak-work> dies_irae: that's not entirely it [14:59] <cbreak-work> dies_irae: considering mp4 is a container format which can contain weird stuff [14:59] <JEEB> rfx, to load straight up you don't need hinting, you need the index atom to be in the beginning of the file for playback to begin right away :P [14:59] <dies_irae> JEEB: lemme try that one [14:59] <cbreak-work> and considering that x264 can create h.264 with unrealistically high numbers of consecutive reference and b frames [14:59] <JEEB> qt-faststart [14:59] <JEEB> dies_irae, I think the ipad 2 has a rather good ASIC, but if it doesn't work, try adding -level 41 [15:00] <JEEB> that should limit the amount of memory needed for decoding [15:00] <rfx> JEEB: so x264 or mpg does not put the index atom at the beginning? Or, rephrased: how would I add it? [15:01] <rfx> sorry, that should read "x264 or ffmpeg" [15:01] <JEEB> rfx, you must know the whole contents of the file to make an index [15:01] <JEEB> and it's mp4 we're talking about [15:01] <JEEB> so basically ffmpeg writes the data and then the index to the end [15:01] <rfx> ah.... [15:01] <JEEB> use qt-faststart (included with ffmpeg's sources) for automation of putting the index into the beginning [15:01] <rfx> O_O [15:03] <ubitux> about qt-faststart, i wonder if it couldn't be a muxing option in the mov [15:04] <rfx> hmm, it's not included in the windows binaries and it seems it's not automatically built when compiling ffmpeg? I used this configure line, but qt-faststart isn't compiled to a binary: --enable-static --enable-libvpx --enable-libvorbis --enable-libx264 --enable-gpl --enable-libvo-aacenc --enable-version3 --enable-libtheora [15:04] <Tjoppen> ubitux: such an option already exists [15:04] <JEEB> oh? [15:04] <JEEB> already implemented? [15:04] <ubitux> Tjoppen: oh? [15:04] <Tjoppen> there's an option for reserving space. lemme see.. [15:05] <Tjoppen> -moov_size [15:05] Action: JEEB scratches head [15:05] <Tjoppen> -moov_size 100000 or so [15:05] <JEEB> doesn't sound like anything stable [15:06] <JEEB> I'd rather have ffmpeg go through the file once again [15:06] <Tjoppen> it's hard to figure out in advance, though not impossible [15:06] <Tjoppen> at work I've experimented with calculating the size based on which streams are CBR and VBR etc., but it's highly shitty [15:07] <JEEB> rfx, no it doesn't get built by default with normal ffmpeg compilation [15:07] <Tjoppen> FFmbc does something similar. the biggest problem is figuring out the number of packets in each stream in advance [15:07] <JEEB> rfx, there's also a python script for it https://github.com/danielgtaylor/qtfaststart [15:07] <dies_irae> trying to get a balance [15:07] <rfx> JEEB: ok, thx; it self contained and 'gcc -o qt-selfstart qt-selfstart.c' did it [15:07] <dies_irae> I am getting this [NULL @ 0x1d63e40] [Eval @ 0x7fff71fa7c70] Undefined constant or missing '(' in 'tevas.mp4' [15:08] <dies_irae> 'tevas.mp4' is the output [15:08] <dies_irae> :/ [15:08] <JEEB> dies_irae, command line and all that, pastebin, please [15:09] <dies_irae> -crf needs an option?? [15:09] <JEEB> so you didn't set any -crf value? [15:09] <dies_irae> JEEB: exactly what you gave me [15:09] <dies_irae> no [15:09] <JEEB> anyways [15:09] <JEEB> just freaking pastebin it all [15:09] <JEEB> please [15:10] <JEEB> easier than trying to esper what exactly you tried to do [15:10] <JEEB> unfortunately I am not exactly good at ESP [15:10] <rfx> JEEB: awesome, you were absolutely right (of course); after qt-fasttrack it also works on my Android ! [15:10] <dies_irae> ffmpeg -i jean.mp4 -vcodec libx264 -vpre placebo -crf tevas.mp4 [15:11] <JEEB> also I only posted the video-related stuff [15:11] <JEEB> and yes [15:11] <JEEB> you didn't set any crf value .-. [15:11] <dies_irae> I got a tegra2 tablet. [15:11] <JEEB> uhh [15:11] <JEEB> you just said ipad 2 [15:11] <JEEB> tegra 2 is well known to have a crappy ASIC [15:11] <JEEB> (read: doesn't support more than baseline) [15:12] <dies_irae> yes, the command is for ipad2 [15:12] <JEEB> ok [15:12] <dies_irae> JEEB: which -crf value would be good?? [15:12] <JEEB> highest that looks good for you [15:12] <JEEB> start from 24 [15:12] <JEEB> go down if it looks bad [15:12] <JEEB> go up if it looks good [15:13] <JEEB> also you lack the audio settings y'know [15:13] <JEEB> just letting you know [15:14] <dies_irae> I think the same as original is just fine. youtube don't be having best quality anyways. [15:14] <dies_irae> I just need to compress the image a lil bit. [15:14] <JEEB> uhh, youtube is already all crapped onto it .-. [15:14] <JEEB> and original audio is -acodec copy [15:15] <dies_irae> JEEB: excellent, working now. [15:16] <dies_irae> mm... anyone here with an AMD x8 bulldozer?? [15:16] <dies_irae> phenom x4 not to fast doing ffmpeg stuff. [15:16] <dies_irae> does ffmpeg have GPU accel? [15:17] <JEEB> sir [15:17] <JEEB> you asked for maximum compression [15:17] <JEEB> I gave you -preset placebo [15:17] <JEEB> if you need speed, you will most probably want to lower that [15:17] <JEEB> :P [15:17] <JEEB> http://mewiki.project357.com/wiki/X264_Settings#preset [15:17] <JEEB> ^ list of libx264 presets [15:18] <JEEB> in other words, find the slowest preset that you can take [15:18] <dies_irae> JEEB: this is a different question, I want maximum quality at minimum time. [15:18] Action: dies_irae guesses everyone's wants that. [15:19] <JEEB> well, quality is more or less controlled by crf [15:19] <JEEB> different algorithms count the crf values a bit differently, but it more or less should keep similar [15:19] <JEEB> so what preset controls is mostly just compression [15:21] <JEEB> (and the amount of time taken by the algorithms) [15:21] <JEEB> slower presets generally compress more and use more features [15:21] <JEEB> faster presets generally end up compressing less and using less features [15:22] <dies_irae> so x8 bulldozer be cool? [15:22] <JEEB> XOP is nice, but I'd still put my money onto intel [15:22] <JEEB> intel is generally faster for libx264 [15:38] <dies_irae> JEEB: what about GPU accel?? [15:40] <JEEB> nope [15:42] <dies_irae> guess youtube's optimizing code didn't do a good job, still shaved 3MB. [15:43] <dies_irae> I could take his job, with JEEB on irc helping though. [15:43] <JEEB> youtube is set up for a quick job [15:44] <JEEB> that's all they care about [15:44] <JEEB> set bitrate, 2pass, quick settings [15:44] <JEEB> youtube was never about quality [16:37] <dies_irae> on the hood, rumors has it that youtube uses ffmpeg. [16:37] <JEEB> it uses libavcodec, that's for sure [16:39] <Tjoppen> see mike's youtube experiments somewhere on http://multimedia.cx/eggs/ [16:39] <Tjoppen> they use their own fork of ffmpeg based on what game formats etc. they support [16:41] <dies_irae> DAAYUUM!!! --> "pushed development forward at a blinding pace." [16:42] Action: dies_irae tries to picture coding at 'blinding pace' [16:48] <rfx> Last week I found some example how to make mp4 "compatible for the web"; back the I had no idea what it means, it looked scary like this: "-codec:v libx264 -crf 22 -flags +loop -cmp +chroma -partitions +parti4x4+partp8x8+partb8x8 -me_method hex -me_range 16 -subq 6 -g 250 -sc_threshold 40 -i_qfactor 0.71 -b_strategy 1" ; I still don't know what most of it means and why it's there. I just know, I use "-f mp4 [16:48] <rfx> -c:v libx264 -profile:v baseline" and still get something which plays everywhere. I wonder *why* such examples are out there. Because,afaics, no one cared to explain why these options are there. [16:49] <dies_irae> rfx: wtf that means?? 'mp4 compatible with the web' [16:51] <rfx> dies_irae: it was meant as a metapher; so many example call them compatible but fail to explain what they actually do until the point you start thinking maybe they even don't know what they were doing and just copied it from somewhere else ... [16:53] <JEEB> rfx, oh boy -- command lines like that were the reason why I didn't want to use ffmpeg for libx264 encoding up until lately [16:53] <JEEB> it's probably a bit over 1.5y ago that it got better [16:53] <JEEB> before that it was lolblackmagic [16:54] <JEEB> and most of those people don't even know what they set .-. [16:54] <rfx> That's what I started to think .. and it seems many examples are just old/obsolete/etc. and it's hard to find insightful reliable information [16:54] <JEEB> well, at least now you can more or less leave it to libx264 because ffmpeg 'lets you push the right buttons' [16:55] <rfx> ;) [16:57] <JEEB> preset: speed/compression -- sets defaults that will get limited by profile/level, profile: limits features, level: limits the amount of memory needed to decode the stream [16:57] <JEEB> that's the basics that you need to know today :) [16:58] <JEEB> I think the most quirky thing I have is the PSP, and H.264-wise http://x264.fushizen.eu/random/psp_encoding.txt is all it needs :) [17:13] <Bombo> hmm how do i do "ffmpeg -i foo.mp4 -vcodec copy -i foo.mp3 -acodec copy muxed.mp4" in new ffmpeg? i get an error now: "Unknown decoder 'copy'" [17:14] <JEEB> ffmpeg -i foo.mp4 -i foo.mp3 -vcodec copy -acodec copy muxed.mp4 [17:14] <JEEB> I would guess something like this [17:14] <JEEB> because if you set a vcodec before -i [17:14] <JEEB> it sets it for the decoder [17:16] <Bombo> ok that worked, thx JEEB [17:16] <JEEB> yeah, -vcodec copy is an 'encoder' [17:16] <JEEB> so you have to put it after the -i [17:17] <Bombo> ok, so someone changed that, as it worked like this -i -vcodec -i -acodec with the old version (last year) [17:18] <Bombo> i get a red "Truncating packet of size 1024 to 1" how bad is that? resulting muxed.mp4 looks ok ... [17:19] <JEEB> if it sounds and looks OK [17:19] <JEEB> I'd guess it's OK [17:19] <Bombo> yeah but why is it red ;) [17:19] <Bombo> red is bad [17:19] <JEEB> "could be problematic in some cases" [17:20] <Bombo> i'll check if its synched [17:22] <Bombo> hmm its ok, so *ignore* [19:56] <comm2k-n> according to mediainfo these are the details: http://pastebin.com/ZRhvCmar [19:57] <comm2k-n> how can I set ID and gop (M/N)? (ignore the NTSC things that is definately wrong) [19:58] <Mavrik> comm2k-n, you set GOP with -g [19:58] <Mavrik> why would you want to set ID? [19:59] <comm2k-n> I'm not sure if I want to - basically these are supposed to become 'HDV-compliant' so that my editing software will take them without reencoding.. I'm trying to approximate as closely as possible (God knows what trips it off) [20:00] <Mavrik> output of which tool is this? [20:01] <comm2k-n> adobe premiere 2 [20:01] <Mavrik> comm2k-n, btw, 0xE0 is an ID of a MPEG-1 or MPEG-2 video stream, so I strongly recommend against trying to change that ;) [20:04] <Mavrik> it seems you also need to use -vtag HDV5 to mark the video as HDV [20:22] <comm2k-n> arghh.. I think it is lost battle.. file1, created by premiere is ok, send this file through ffmpeg (using a/v copy) and the file is not accepted anymore. even though media info output is identical [20:23] <Mavrik> comm2k-n, did you use the vtag parameter? [20:23] <comm2k-n> yes didn't make much difference [20:25] <Mavrik> and why exactly doesn't it accept your video? [20:26] <comm2k-n> that is good question, basically it does accept them but wants to reencode them - this is the problem... [21:42] <rhizmoe> hi all [21:42] <rhizmoe> is ffserver really as janky as the docs would make it appear? [21:43] <Mavrik> yeah [21:43] <rhizmoe> sorry to drop a turd my first minute in-channel [21:43] <rhizmoe> just researching the build options [21:45] <Mavrik> basically, ffserver is very very obsolete :) [21:46] <j-b-r> >janky [21:46] <j-b-r> That might be the first time I've ever read this word [21:51] <rhizmoe> Mavrik: that's the confirmation i was looking for :) [21:51] <rhizmoe> j-b-r: it's a good one :) [21:51] <rhizmoe> actually, i should confirm that it doesn't have an offensive etymology [21:52] <j-b-r> If used in the right way, it sounds like the kind of word where you could trick people into ASSUMING that it was offensive [21:52] <rhizmoe> niggardly [21:52] <j-b-r> Yes, quite like that [21:53] <rhizmoe> :) ok, i'm out [21:54] <EOF-sensei> ? [22:43] <creaux_> hi, is there a option for ffmpeg to keep reading a input file that is open and growing? [23:05] <linuxis> Anyone here? [23:10] <linuxis> I downloaded a bashscript which allows streaming shoutcast. Now by default it has radiotray as player and mplayer as alternate but I want to stream with FFMPEG [23:10] <linuxis> But I'm not sure how to edit the script [23:11] <linuxis> to use ffmpeg to stream shoutcast [23:14] <linuxis> Why is this room so dead? [23:14] <linuxis> ?? [23:16] <linuxis> Where is everyone? [23:18] Action: linuxis1 hates disconnecting [23:24] <linuxis1> How do I stream shoutcast from the commandline using FFMPEG? [23:24] <linuxis1> Or even Icecast [23:26] <Mavrik> do you want to stream AS shout/icecast or TO shout/icecast server? [23:28] <linuxis1> Ummm no I want to play the radio stations from terminal [23:28] <linuxis1> Wait I didn't understand, [23:29] <linuxis1> I want to stream the Shout/Icecast from command line [23:29] <linuxis1> I don't want to broadcast [23:30] <Mavrik> stream to who? [23:30] <linuxis1> Well, to a player? Or can FFMPEG play the streams? [23:31] <linuxis1> I'll put things into context. [23:31] <Mavrik> so you want to use ffmpeg to receive shoutcast stream and play it? [23:31] <linuxis1> yes [23:31] <linuxis1> I have a script that can search shoutcast radio directories [23:32] <linuxis1> But I want to forward the searches to FFPEG to play the stations [23:34] <Mavrik> I see [23:34] <Mavrik> you do "ffplay -i <http url of station>" [23:34] <Mavrik> to play the station [23:34] <Mavrik> or ffmpeg -i <http url of stream> ... to convert it [23:34] <linuxis1> convert it to what? [23:34] <Mavrik> to whatever you want :P [23:35] <Mavrik> ffmpeg is just a media encoder [23:35] <linuxis1> So I can convert an mp3 ac3 stream to Ogg in realtime? [23:35] <Mavrik> yeah [23:35] <Mavrik> ffmpeg -i <your station> -codec:a vorbis -b:a 128k station_recording.ogg [23:35] <linuxis1> Ok, so I should add ffplay to the script [23:36] <Mavrik> and it'll start encoding the station to ogg and dumping it to file [23:36] <linuxis1> Though how can I forward the script's choices to ffplay? [23:36] <Mavrik> "forward script choices?" [23:38] <linuxis1> Ok, I didn't write the script [23:38] <linuxis1> By default it uses Radiotray and Mplayer but I don't want to use them [23:38] <linuxis1> So via the script if I search a station, how can I pipe the station to ffmpeg? [23:40] <Mavrik> um [23:40] <Mavrik> if you pass "pipe:" to -i it'll read from pipe [23:40] <linuxis1> It'll be so much easier if you could just see the script, it's freely available from crunchbang's forums [23:41] <linuxis1> Then you could just give me better advice [23:41] <linuxis1> http://crunchbanglinux.org/forums/post/150615/ [23:46] <linuxis1> I'm wondering if ffplay() would work [23:46] <linuxis1> I'm still learning so I'm not really sure [00:00] --- Thu Jan 19 2012

1 0

ffmpeg-devel.log.20120117
by burek 18 Jan '12

18 Jan '12

[00:06] <CIA-31> ffmpeg: 03Michael Niedermayer 07release/0.9 * r63945e2226 10ffmpeg/libavcodec/indeo4.c: [00:06] <CIA-31> ffmpeg: indeo4: fix LE reader define for backport [00:06] <CIA-31> ffmpeg: Fixes Ticket920 [00:06] <CIA-31> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [00:17] <CIA-31> ffmpeg: 03Clément BSsch 07master * r72ad228ada 10ffmpeg/tests/Makefile: fate: add GEN tag for mapchan sw tests. [00:42] <CIA-31> ffmpeg: 03Carl Eugen Hoyos 07master * rdc2f652ae4 10ffmpeg/libavcodec/imgconvert.c: [00:42] <CIA-31> ffmpeg: Fix pix_fmt_info for yuva420. [00:42] <CIA-31> ffmpeg: This allows correct output pix_fmt auto-selection [00:42] <CIA-31> ffmpeg: if the input pix_fmt is yuva420. [00:42] <CIA-31> ffmpeg: Fixes ticket #290. [00:42] <CIA-31> ffmpeg: 03Carl Eugen Hoyos 07master * r328e79329a 10ffmpeg/libavcodec/ (pnm.c pnmdec.c): Support gray8a pam decoding. [00:42] <CIA-31> ffmpeg: 03Carl Eugen Hoyos 07master * r00430075bd 10ffmpeg/libavcodec/ (pnm.c pnmdec.c): Support 64bit pam decoding. [00:55] <durandal_1707> unfortunately it looks like i would need to write new adpcm decoder, because THP adpcm variant (obviously match NGC_DSP in vgmstream) need something which brstm does not have [00:57] <durandal_1707> michaelni: i have dumb -af patch for ffmpeg but i couldn't prove it is useful at all [00:57] <ubitux> there is an existing -af patch in some lost branch [00:57] <ubitux> you should ask stefano; i don't remember why it's not upstream [00:58] <ubitux> (except there was some issues with it) [01:02] <ubitux> it seems the commit isn't available anymore in saste's tree [01:02] <ubitux> you may want to ask him about it [01:10] <michaelni> durandal_1707, i agree with ubitux, stefano is the one to ask about -af [01:41] <CIA-31> ffmpeg: 03Michael Niedermayer 07master * r905c4dc2b0 10ffmpeg/ (libavcodec/dpxenc.c tests/ref/lavf/dpx): [01:41] <CIA-31> ffmpeg: dpxenc: Dont store lavc indent when bitexact is set. [01:41] <CIA-31> ffmpeg: This avoids fate breakage from bumping version. [01:41] <CIA-31> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [02:34] <slipperz> I've just started taking a look at the source. If I wanted to get familiar with the libavformat codebase where should I start? [02:47] <durandal_1707> does FFmpeg have splay tree implementation? [02:51] <CIA-31> ffmpeg: 03Christophe GISQUET 07master * r3eeb755763 10ffmpeg/libavcodec/rv34.c: [02:51] <CIA-31> ffmpeg: rv34: Inter/intra MB code split [02:51] <CIA-31> ffmpeg: Split inter/intra macroblock handling code. This will allow further [02:51] <CIA-31> ffmpeg: optimizations such as performing inverse transform and block reconstruction [02:51] <CIA-31> ffmpeg: in a single pass as well as specialize code. [02:51] <CIA-31> ffmpeg: Signed-off-by: Janne Grunau <janne-libav(a)jannau.net> [02:51] <CIA-31> ffmpeg: 03Justin Ruggles 07master * rb2c75b6e63 10ffmpeg/libavcodec/ (avcodec.h internal.h pcm.c utils.c version.h): [02:51] <CIA-31> ffmpeg: avcodec: Add avcodec_encode_audio2() as replacement for avcodec_encode_audio() [02:51] <CIA-31> ffmpeg: This allows audio encoders to optionally take an AVFrame as input and write [02:51] <CIA-31> ffmpeg: encoded output to an AVPacket. [02:51] <CIA-31> ffmpeg: This also adds AVCodec.encode2() which will also be usable by video and [02:51] <CIA-31> ffmpeg: subtitle encoders once support is implemented in the public functions. [02:51] <CIA-31> ffmpeg: 03Justin Ruggles 07master * r5ee5fa021f 10ffmpeg/libavcodec/ (avcodec.h utils.c): [02:51] <CIA-31> ffmpeg: avcodec: add a public function, avcodec_fill_audio_frame(). [02:51] <CIA-31> ffmpeg: This is a convenience function for the user to fill audio AVFrame information. [02:52] <CIA-31> ffmpeg: This is a preparation for adding SMJPEG muxer. [02:52] <CIA-31> ffmpeg: Signed-off-by: Kostya Shishkov <kostya.shishkov(a)gmail.com> [02:52] <CIA-31> ffmpeg: 03Paul B Mahol 07master * rffa0923eee 10ffmpeg/ (6 files in 3 dirs): [02:52] <CIA-31> ffmpeg: add SMJPEG muxer [02:52] <CIA-31> ffmpeg: Signed-off-by: Kostya Shishkov <kostya.shishkov(a)gmail.com> [02:52] <CIA-31> ffmpeg: 03Michael Niedermayer 07master * r67f5650a78 10ffmpeg/: (log message trimmed) [02:52] <CIA-31> ffmpeg: Merge remote-tracking branch 'qatar/master' [02:52] <michaelni> slipperz, start with the avformat.h header [02:53] <CIA-31> (12 lines omitted) [02:53] <slipperz> ty [02:54] <michaelni> durandal_1707, no but ive thought about adding support for that to tree.c/h a long time ago [02:55] <michaelni> though you make me a bit curious, what do you want to do with splay trees ? [02:55] <durandal_1707> xpm [02:55] <durandal_1707> you have better idea? [02:57] <michaelni> iam not sure i understand what you want to do with a splay tree in xpm (x pixmap ?) [02:58] <durandal_1707> it is for picking color, imagemagick do that [03:00] <durandal_1707> for small icons there would be no difference but for images with huge number of pixels&colors it would become really slow [03:03] <michaelni> i suspect a hashtable might be faster than a splay tree [03:05] <michaelni> but of course i might be wrong and this depends on the data [03:15] <slipperz> In avformat.h, I see some documentation but then there are some titles like, ""Reading from an opened file" and "Seeking" that are empty. Are these documented somewhere else? Or will I just have to figure it out ? [03:26] <michaelni> slipperz, our docs are not as good as they should be sadly, but improvments are very welcome! [03:27] <michaelni> so when you figure something out that wasnt documented, please document it and submit a patch [03:27] <slipperz> michaelni, will do :) [03:31] <michaelni> thx [03:36] <durandal_1707> what about floating point pixels? [03:40] <michaelni> what are floating point pixels ? i get no match for "float" in xpm.pdf [03:41] <michaelni> but generically, if R/G/B color values are floats nothing should change [03:41] <michaelni> if the "index" is a float, well a float is just a fancy 32bit value [03:41] <michaelni> one can just read it as integer, calculate a hash and use a hashtable [03:43] <slipperz> michaelni, quick question, what does pts stand for? [03:43] <michaelni> presentation time stamp [03:44] <durandal_1707> michaelni: reading about RGBE image format, imagemagick cant show file [03:45] <slipperz> ty [03:48] <durandal_1707> heh this reminds me of OpenEXR patch [04:23] <Daemon404> orite [04:23] <Daemon404> openexr could use fun stuff [04:23] <Daemon404> like 16-bit floats for values [04:23] <Daemon404> "fun" [05:23] <Compn> hmm wheres pross [05:23] <Compn> did sigmadesigns ever release their 'xvid' source ? [05:54] <slipperz> anyone around? [07:21] <ubitux> youhou valgrind fate in the green \o/ [08:42] <CIA-31> ffmpeg: 03Reimar Döffinger 07master * r7e62315c91 10ffmpeg/libavcodec/x86/rv34dsp.asm: [08:42] <CIA-31> ffmpeg: Use correct register size. [08:42] <CIA-31> ffmpeg: Fixes compilation with NASM. [08:42] <CIA-31> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de> [09:59] <mateo`> j-b: hi, i am currently working on extracting the starting timecode from MXF files and Tim Nicholson proposed me to upload some sample. Is there an offcial ftp server to upload those samples and is it possible to have access ? thx [10:30] <Ti-CoeuR> salut [11:12] <funman> mateo`: ftp://streams.videolan.org/incoming at least [11:13] <funman> though the server is crashy [12:02] <CIA-31> ffmpeg: 03Michael Bradshaw 07master * ra02694c926 10ffmpeg/libavcodec/libopenjpegenc.c: libopenjpegenc: Simplify gray encoding. [12:02] <CIA-31> ffmpeg: 03Michael Bradshaw 07master * r0abe25aa7b 10ffmpeg/libavcodec/libopenjpegenc.c: libopenjpegenc: Add error messages. [12:02] <CIA-31> ffmpeg: 03Michael Bradshaw 07master * ree33eb40fc 10ffmpeg/libavcodec/libopenjpegenc.c: libopenjpegenc: Rename four conversion functions. [12:02] <CIA-31> ffmpeg: 03Michael Bradshaw 07master * r56af084cb5 10ffmpeg/libavcodec/libopenjpegenc.c: libopenjpegenc: Remove two redundant function parameters. [12:03] <mateo`> funman: thx [12:11] <CIA-31> ffmpeg: 03Stefano Sabatini 07master * r9f7144b44c 10ffmpeg/libavcodec/mpegvideo_enc.c: [12:11] <CIA-31> ffmpeg: lavc/mpegvideo_enc: improve log message in case of too large GOP [12:11] <CIA-31> ffmpeg: Change log level from ERROR to WARNING and provide more specific [12:11] <CIA-31> ffmpeg: information in the message itself. [12:11] <CIA-31> ffmpeg: 03Stefano Sabatini 07master * ra798c20a76 10ffmpeg/ (3 files in 2 dirs): lavfi/testsrc: add "decimals" option to the testsrc filter [12:11] <CIA-31> ffmpeg: 03Stefano Sabatini 07master * r999495734b 10ffmpeg/ (4 files in 3 dirs): [12:11] <CIA-31> ffmpeg: lavu/eval: add if() and ifnot() eval functions [12:11] <CIA-31> ffmpeg: They allow to implement the if/then/else logic, which cannot be [12:11] <CIA-31> ffmpeg: implemented otherwise. [12:11] <CIA-31> ffmpeg: For example the expression: [12:11] <CIA-31> ffmpeg: A*B + not(A)*C [12:11] <CIA-31> ffmpeg: always evaluates to NaN if B is NaN, even in the case where A is 0. [12:11] <CIA-31> ffmpeg: 03Stefano Sabatini 07master * r24860a913c 10ffmpeg/ (libavutil/Makefile tools/ffeval.c): [12:11] <CIA-31> ffmpeg: tools: add ffeval tool [12:11] <CIA-31> ffmpeg: The tool is useful for testing the internal arithmetic evaluation engine [12:11] <CIA-31> ffmpeg: (indeed I plan to use it in FATE), and provides a handy calculator when [12:12] <CIA-31> ffmpeg: 03Stefano Sabatini 07master * re4e291061c 10ffmpeg/doc/eval.texi: doc/eval.texi: add precisation about the assumption for "trueness" in conditional expressions [12:57] <ubitux> saste: durandal_1707 was talking about a -af patch for ffmpeg; i looked at your gitorious branches, but it seems not available anymore [12:57] <ubitux> did you drop it? [12:58] <saste> ubitux: given the number of reworking done in the audio engine I preferred to delay its proper integration [12:58] <ubitux> ok [12:58] <saste> so I stopped to update it in my audio-filters branches [12:59] <saste> it should still be available if you look in the oldest branches, but it obviously doesn't apply anymore [12:59] <ubitux> it is available in the oldest branch, but the commit seems not readable [13:00] <saste> which one? [13:01] <ubitux> https://gitorious.org/~saste/ffmpeg/sastes-ffmpeg/commits/audio-filters-201… last one here [13:02] <ubitux> it's not really important, i was just curious about it [13:03] <saste> uhm... weird [13:05] <saste> i still have it in my local branches, i'll try to revive it in a new audio-filters branch [13:05] <saste> give me a few days, or i can send you the patches [13:06] <ubitux> i just wanted to point it to durandal_1707 [13:06] <ubitux> after "< durandal_1707> michaelni: i have dumb -af patch for ffmpeg but i couldn't prove it is useful at all" [13:07] <saste> -af patch for ffmpeg was also changing the fate output, since it uses by default the lavfi path when lavfi is enabled [13:07] <saste> so it needs thorough testing [13:45] <ubitux> bcoudurier: since you are not satisfied with the timecode api, if you want to comment on the new one, you should do it now, it will be harder to change it later [13:45] <ubitux> ffmpeg-devel / "Rework and move timecode API to lavutil" patchset [15:23] <durandal_1707> is there function in ffmpeg to get number of unique colors in picture? [15:24] <av500> no [15:24] <durandal_1707> something like fast bitcount.... [15:24] <av500> fact_colorcount? [15:25] <av500> and with unique you mean unique 24bit RGB Values? [15:25] <av500> or whatever colorspace [15:27] <saste> durandal_1707: write a filter for that! [15:28] <durandal_1707> i googled little [15:28] Action: durandal_1707 going to see what ImageMagick does [15:29] <ubitux> saste: speaking of this, i wonder if vf showinfo could have stuff like http://silicontrip.net/~mark/lavtools/#yuvvalues [15:29] <ubitux> it will likely make it more useful over ffprobe -show_frames for instance [15:31] <ubitux> the other tools might be interesting as well [15:34] <durandal_1707> for counting number of unique colors in 24bit image we just need 2^24 bits [15:43] <saste> we may design common functions for that [15:44] <saste> but i'm not sure i want to make ffprobe depend on lavfi [15:44] <saste> (well maybe conditionally...) [15:46] <ubitux> saste: i meant having vf showinfo being more useful that ffprobe -show_frames by providing more insight/analyzed informations [15:46] <ubitux> different purpose [15:47] <saste> ubitux: or maybe extend ffprobe as well, although i'm not sure it is a good idea provided what ffprobe is from the philosophical pov (just a thin wrapper around what lavc/lavf shows about packets/streams/frames) [15:47] <ubitux> -show_frames: used to grab global/generic frames informations in easily exploitable formats, and vf showinfo more information [15:48] <saste> while extending showinfo is fine of course [15:48] <ubitux> indeed, i don't think ffprobe -show_frames should be extented to print more information [15:48] <ubitux> but afaik, showinfo is somehow a limited ffprobe -show_frames [15:48] <ubitux> so the link was some proposition on how to improve it ;) [15:49] <saste> ubitux: patches are welcome ;-) [15:50] <ubitux> heh i know ;) [15:50] <saste> i'm fighting with timestamps at the moment [17:22] <ubitux> saste: you should decide weither to put the number or the string in first :p [17:22] <ubitux> @item passthrough, 0 [...] @item 1, cfr [17:22] <saste> yes, i noticed one second after sending the patch [17:23] <saste> that's why i send patches, for getting a post-send-button enlightenment [17:23] <ubitux> :D [19:58] <durandal_1707> drop darwin [20:06] <Daemon404> ? [20:07] <durandal_1707> drop darwin support [20:10] <CIA-31> ffmpeg: 03Michael Niedermayer 07master * ra129622390 10ffmpeg/libavcodec/utils.c: [20:10] <CIA-31> ffmpeg: lavc: Check CODEC_CAP_VARIABLE_FRAME_SIZE && !frame [20:10] <CIA-31> ffmpeg: This combination is quite odd and almost certainly a bug if [20:10] <CIA-31> ffmpeg: it happens. [20:10] <CIA-31> ffmpeg: Reviewed-by: Justin Ruggles <justin.ruggles(a)gmail.com> [20:10] <CIA-31> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [20:10] <CIA-31> ffmpeg: 03Michael Niedermayer 07master * raf5004b557 10ffmpeg/libavcodec/utils.c: [20:10] <CIA-31> ffmpeg: lavc: use av_freep() for freeing extended_data [20:10] <CIA-31> ffmpeg: This is safer [20:10] <CIA-31> ffmpeg: Reviewed-by: Justin Ruggles <justin.ruggles(a)gmail.com> [20:10] <CIA-31> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [20:13] <berdario> Hello, I have a problem with ffmpeg... I downloaded the sources and I'm trying to dig up the issue by debugging it... I'm trying to supply an .sdp to ffplay, and the file is not recognized... after some calls to av_probe_input_format2 and 3 it fails by returning AVERROR_INVALIDDATA [20:14] <berdario> can anyone help me? [20:15] <berdario> I stopped here since av_probe_input_format3 is completely devoid of comments [20:16] <michaelni> berdario, can we see the sdp file ? [20:16] <berdario> michaelni, http://paste.pocoo.org/show/536262/ [20:18] <CIA-31> ffmpeg: 03Reimar Döffinger 07master * r3b55429d56 10ffmpeg/libavcodec/ (asv1.c avcodec.h eatqi.c mjpegdec.c utils.c version.h): (log message trimmed) [20:18] <CIA-31> ffmpeg: Add and use av_fast_padded_malloc. [20:18] <CIA-31> ffmpeg: The same as av_fast_malloc but uses av_mallocz and keeps extra [20:18] <CIA-31> ffmpeg: always-0 padding. [20:18] <CIA-31> ffmpeg: This does not mean the memory will be 0-initialized after each call, [20:18] <CIA-31> ffmpeg: but actually only after each growth of the buffer. [20:18] <CIA-31> ffmpeg: However this makes sure that [20:18] <CIA-31> ffmpeg: 03Reimar Döffinger 07master * r71a2c9b265 10ffmpeg/ffmpeg.c: (log message trimmed) [20:18] <CIA-31> ffmpeg: ffmpeg: check tcgetattr result. [20:18] <CIA-31> ffmpeg: This fixes parallel FATE (make fate -j4) failing under valgrind with: [20:18] <CIA-31> ffmpeg: Syscall param ioctl(TCSET{S,SW,SF}) points to uninitialised byte(s) [20:18] <CIA-31> ffmpeg: at 0x5D98B23: tcsetattr (tcsetattr.c:88) [20:18] <CIA-31> ffmpeg: by 0x43D66C: term_init (ffmpeg.c:734) [20:18] <CIA-31> ffmpeg: by 0x43CD8D: main (ffmpeg.c:5071) [20:21] <michaelni> berdario, look at libavformat/rtsp.c:sdp_probe() but note iam not the SDP expert ... [20:23] <berdario> uhhhhh, thanks... do you know who I should ask further questions to? [20:24] <michaelni> luca abeni [20:24] <michaelni> but he isnt on IRC AFAIK [20:25] <berdario> michaelni, "c" is an optional field... I'll try to modify my .sdp to get it working, but maybe this should be considered a bug and get fixed in rtsp.c? (or is it known bug? a design choice?) [20:26] <berdario> ok, mail is fine for him, I guess? [20:26] <michaelni> i dont think its a known bug [20:26] <michaelni> yes [20:26] <michaelni> also if the field is optional then its probably a bug and should be reportet on ffmpeg.org/trac [20:27] <berdario> ok, thanks :) [20:27] <michaelni> np [20:48] <berdario> fyi, "c=" isn't really optional: other programs (like vlc) take the connection information from the origin "o=", but a "c=" should be specified either in the session or in each media description (thus, it's not a problem if it's missing from the session description like in my sdp, but since there's no media description the sdp is not actually correct) [20:49] <j-b> mateo`: yes, please use ftp://streams.videolan.org/incoming [20:50] <berdario> I wonder if it could be possible to add an av_log() line inside sdp_probe() to help other people who may have this issue [20:52] <berdario> (mhn, I'll write the patch later) [21:18] <CIA-31> ffmpeg: 03Paul B Mahol 07master * raff9cb8377 10ffmpeg/libavcodec/ (pamenc.c png.c pnmdec.c pnmenc.c): [21:18] <CIA-31> ffmpeg: lavc: remove unused headers [21:18] <CIA-31> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [21:19] <CIA-31> ffmpeg: 03Tomas Härdin 07master * r1de5154dfb 10ffmpeg/libavformat/mxfdec.c: [21:19] <CIA-31> ffmpeg: mxfdec: Consider OPAtom files that don't have exactly one EC to be OP1a [21:19] <CIA-31> ffmpeg: This fixes demuxing of 2011_DCPTEST_24FPS.V.mxf. [21:19] <CIA-31> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [22:16] <CIA-31> ffmpeg: 03Reimar Döffinger 07master * r178f8ea76e 10ffmpeg/libavcodec/utils.c: [22:16] <CIA-31> ffmpeg: Use ff_packet_free_side_data instead of duplicating it. [22:16] <CIA-31> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de> [22:16] <CIA-31> ffmpeg: 03Reimar Döffinger 07master * rd393f41a33 10ffmpeg/ffmpeg.c: [22:16] <CIA-31> ffmpeg: Fix memleak after switch to encode_audio2 API. [22:16] <CIA-31> ffmpeg: Obviously the packet allocated by the encoder must be freed somewhere. [22:16] <CIA-31> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de> [22:52] <durandal_1707> cinepack regression https://trac.videolan.org/vlc/ticket/5872 [23:39] <cbsrobot> durandal_1707: it works wth ffplay .... [23:41] <michaelni> cbsrobot, i can confirm i see nothing odd with ffplay, how do i reproduce this ? [23:53] <CIA-31> ffmpeg: 03Stefano Sabatini 07master * r6c06868e61 10ffmpeg/doc/ffmpeg.texi: doc/ffmpeg.texi: specify symbolic names for the vsync parameter [00:00] --- Wed Jan 18 2012

1 0

ffmpeg.log.20120117
by burek 18 Jan '12

18 Jan '12

[00:00] <JEEBsv> Nah, not only that. I've just gotten used to the fact that most people who think they know something about these things (even people on committees) are people who can't think of anything else to check than PSNR. [00:00] <EOF-sensei> right now nobody seems to know how to make it work for low-bandwidth human-optimized video [00:00] <JEEBsv> yeah [00:00] <EOF-sensei> but that doesn't make it a lost cause [00:00] <JEEBsv> never said it is [00:00] <EOF-sensei> in my opinion it's a diamond in the rough [00:00] <EOF-sensei> err [00:00] <EOF-sensei> better than a diamond [00:01] <EOF-sensei> diamonds aren't worth nearly as much as the diamond cartels would have you believe [00:01] <EOF-sensei> they're pretty easy to dig up [00:01] <JEEBsv> It's an interesting area of development/research as far as I can see. [00:01] <EOF-sensei> well [00:01] <EOF-sensei> the issue is that there's not enough development and research [00:02] <EOF-sensei> because people are all hung on 'which is better?' conversations [00:03] <EOF-sensei> every paper about wavelet compression techniques I've been able to dig up is hiding behind IEEExplore [00:03] <EOF-sensei> which is a very poor way to explore things [00:03] <JEEBsv> Also, IIRC JPEG2K didn't take off very well either, did it? [00:04] <EOF-sensei> as you will bankrupt a fortune 500 company if you do. [00:04] <JEEBsv> for pictures [00:04] <EOF-sensei> well [00:04] <EOF-sensei> it's nice [00:04] <EOF-sensei> though jpeg2k needs quite a bit of refining to be 'better' than jpeg [00:04] <EOF-sensei> it does help for progressive loading though [00:05] <JEEBsv> true [00:05] <EOF-sensei> and if blocking is your main concern then it's a great standard [00:05] <EOF-sensei> I hate going to filter a cheap point-and-shoot camera's photo [00:05] <EOF-sensei> and noticing jpeg artifacts [00:05] <EOF-sensei> it's especially shitty if you do inverting [00:06] <EOF-sensei> or threshold [00:06] <EOF-sensei> because the artifacts just pop right out and spray feces all over your image [00:07] <EOF-sensei> I personally use JPEG2000 frequently [00:07] <EOF-sensei> it does pretty good lossless encoding [00:07] <JEEBsv> too bad it's not widely enough supported for general linkage as far as I know [00:07] <JEEBsv> thus I tend to use PNG [00:08] <EOF-sensei> libopenjpeg isn't so bad [00:08] <EOF-sensei> and if you're used to unix [00:09] <EOF-sensei> it's no issue [00:09] <EOF-sensei> I have a script that handles my lossless transcoding [00:09] <JEEBsv> try linking people lossless pictures in JPEG2000 on the internet :/ [00:09] <EOF-sensei> people go all 'fuck you' mode [00:10] <EOF-sensei> although last I checked my browser supports jpeg2000 [00:10] <JEEBsv> lol, which do? [00:10] <JEEBsv> I haven't even checked [00:11] <EOF-sensei> huh [00:12] <EOF-sensei> I guess not anymore [00:12] <EOF-sensei> or maybe I'm just trippin' [00:13] <EOF-sensei> I think firefox has jpeg2000 support somewhere [00:13] <EOF-sensei> but it doesn't seem to be compiled into my version.. [00:13] <EOF-sensei> perhaps from patent thingies [00:14] <EOF-sensei> hmm [00:15] <EOF-sensei> nope [00:15] <JEEBsv> there's a bug tracker entry at least, and patches there [00:15] <EOF-sensei> jpeg2000 has been cleared by patent owners [00:15] <JEEBsv> https://bugzilla.mozilla.org/show_bug.cgi?id=36351 [00:16] <EOF-sensei> that's convenient [00:16] <JEEBsv> "RESOLVED WONTFIX" <- lulz tho [01:47] <YellowOnion> I'm getting drops with CPU usage every now and then with -threads 6 ( I have 6 cores ) it's only loaded about 85% seems to improve with 12 threads 95% load anyone know how to get anymore without excessive amounts of threads. [01:53] <frighty> could somebody tell me how i get ffmpeg output into a logfile [02:00] <YellowOnion> frighty, ffmpeg > ffmpeg.log?? [02:01] <frighty> do i type that in my command line? [02:01] <frighty> ffmpeg.log [02:02] <frighty> or its there somewhwer [02:02] <YellowOnion> frighty, what system are you using? [02:02] <frighty> windows [02:02] <YellowOnion> oh yep [02:03] <YellowOnion> the ">" outputs the text to a file [02:03] <frighty> example please [02:03] <frighty> soo like [02:03] <YellowOnion> ffmpeg.log is just a text file [02:03] <frighty> ffmpeg -i infile options outfile ffmpeg.log [02:03] <YellowOnion> try "echo blah > test.txt" [02:03] <frighty> wait what lol [02:03] Action: frighty noob sry [02:04] <YellowOnion> the windows command prompt has a feature to redirect any commands output to a file [02:05] <YellowOnion> use the ">" symbol then after it put the name of the output file [02:05] <YellowOnion> ffmpeg -i infile options outfile > ffmpeg.log [02:05] <YellowOnion> will work [02:05] <YellowOnion> ffmpeg -i infile options outfile > ffmpeg.txt [02:05] <YellowOnion> will also work [02:06] <frighty> ok ty ill give that a shot ina few once im done attempting to write a irc bot heh [02:07] <YellowOnion> frighty, http://www.microsoft.com/resources/documentation/windows/xp/all/proddocs/en… [02:08] <frighty> ty [02:11] <YellowOnion> frighty, it's something microsoft pitched from Unix [02:37] <sneakers> How would I convert a WebM contained VP8 encoded video into mpeg4? [02:40] <llrcombs> sneakers: -i webmVideo.webm -vcodec mpeg4 mpeg4video.mp4 [02:40] <llrcombs> throw in some video settings and you're good [02:40] <sneakers> llrcombs, bash: -i: command not found... [02:40] <darkstarbyte> You may also want to use -acodec copy if you don't want to reconvert it [02:40] <llrcombs> sneakers: those are the arguments to ffmpeg [02:40] Action: llrcombs dies a little inside [02:41] <llrcombs> sneakers: sudo shutdown -h now [02:41] <llrcombs> type your password when asked [02:41] <sneakers> llrcombs, Gimme a break. [02:41] <sneakers> been using linux for 10 years. [02:41] <sneakers> >8-] [02:41] <llrcombs> >_> [02:42] <sneakers> and I use Fedora 16 , so it would be su -c 'shutdown -h now' [02:42] <darkstarbyte> I was talking about the audio [02:43] <sneakers> its converting now, thanks. oh, what would be the benefit of NOT reconverting it. [02:43] <llrcombs> sneakers: saves time and keeps quality up [02:43] <llrcombs> remuxing doesn't drop quality, transcoding does [02:43] <sneakers> the file has no audio stream. [02:43] <llrcombs> oh, well then [02:43] <llrcombs> doesn't matter then [02:43] <sneakers> kewl [02:43] <sneakers> thanks again [02:43] <llrcombs> welcome [02:44] Action: llrcombs has never actually had anyone fall for a "type dis cmd it fixs ur problm hrr drr" [02:46] <sneakers> worst thing that would happen is a shutdown and..................possible loss of unsaved data!!!! [02:46] <llrcombs> true [02:46] <llrcombs> if I know someone's a huge douchebag troll, I might counter troll with "sudo rm -rf /" [02:46] <llrcombs> or dd'ing /dev/null to /dev/sda1 [02:47] <llrcombs> never had anyone fall for it though [02:48] <sneakers> sudo dd if=/dev/zero of=/dev/sda1 [02:48] <sneakers> lol [02:48] <llrcombs> yep [02:48] <sneakers> messed up [03:17] <darkstarbyte> I will be back after I install slackware 13.37 [03:38] <frighty> YellowOnion, [03:38] <frighty> still here [03:38] <frighty> i did > logfile.txt [03:38] <frighty> but it made it empty [04:25] <frighty> mann somebody help please i make the logfile but it doesnt show nething its just empty [04:25] <frighty> shoutldn it write every line that is encoded or something [04:26] <darkstarbyte> If you want that I think you could just use ffmpeg [options] > file.log [04:27] <frighty> i put [04:27] <frighty> ffmpeg -i infile options outfile > c:\logfile.txt [04:27] <frighty> thats wrong? [04:27] <frighty> it make the logfile txt but its blank [04:28] <darkstarbyte> oh Your using Windows [04:28] <frighty> ya [04:28] <darkstarbyte> I am not much help on this [04:29] <frighty> and ive put file.log now also just spits out blank [04:29] <frighty> id like it to spit out every single info [04:29] <frighty> like basically generate as it encodes [04:29] <darkstarbyte> Someone else please help him [04:30] <frighty> what i say is that what the logfile is suppose to do ? [04:30] <frighty> generate as its encoding [04:34] <frighty> -vstats_file file.txt seemed to work but anyway to make that generate as its encoding [04:34] <flipuhdelphia> hello [04:36] <flipuhdelphia> can you guys see me type? says this nickname is registered, etc. not sure if i am in the room or not. [04:37] <frighty> yep we see [04:40] <flipuhdelphia> cool! i have a small q: I want to send live screen data (fullscreen or portion of screen) via USB to external device. I was told ffmpeg could help in the X11 grabbing, which I tested out and it did. However, it creates a file and grows pretty fast (raw MPEG). Is there a way to write to a file that will keeps its size constant, recycling old frames? I just need to stream the screen to an external device. My approach is to c [04:40] <flipuhdelphia> reate a /proc file entry and have ffmpeg write to it? That way, a /proc module just takes the data and can send it to an external device. Thoughts? Thank you! [07:08] <aaronpeterson> Should one expect -vf "transpose=1" to work with libvpx? I have it working for libx264 but libvpx seems to ignore it. [07:10] <aaronpeterson> Nevermind. I'm a jackass. I had -vf before -i. Must sleep. [07:23] <darkstarbyte> Anyone here use slackware? [10:28] <Ti-CoeuR> hello [10:29] <Ti-CoeuR> quelqu'un parle français ici [12:46] <jsmax> Hi all. [12:46] <jsmax> If i stream to udp with ffmpeg then stream mapping is working correctly [12:46] <jsmax> but if i stream to ffm the mapping is wrong [12:47] <jsmax> for example: [12:48] <jsmax> #/opt/bin/ffmpeg -f mpegts -i udp://@192.168.4.11:1234 -map 0:22 -map 0:23 http://127.0.0.1:8180/comedy.ffm [12:48] <Ti-CoeuR> centos [12:48] <jsmax> yes [12:48] <jsmax> the result is: [12:48] <jsmax> Stream mapping: [12:48] <jsmax> Stream #0:1 -> #0:0 (mp2 -> mp2) [12:48] <jsmax> Stream #0:0 -> #0:1 (mpeg2video -> mpeg1video) [12:49] <jsmax> [root@iptv ~]# cat /etc/redhat-release [12:49] <jsmax> CentOS release 6.2 (Final) [12:51] <jsmax> Can you help me please ? [12:53] <frighty> does anybody know any .tcl scripting and if so how would u make a bot say whats going on in realtime say every 10 mins whats going on with a file encoding on server example bitrate size etc [13:10] <ph1b> hey, I'm trying to convert a video using: [13:10] <ph1b> ffmpeg -i kkurz.avi -vcodec libx264 -acodec copy -b 1024 output.mkv [13:10] <ph1b> but it just says: [13:10] <ph1b> [buffer @ 0xb18b40] w:720 h:576 pixfmt:yuv420p [13:10] <ph1b> [libx264 @ 0xb10040] broken ffmpeg default settings detected [13:10] <ph1b> [libx264 @ 0xb10040] use an encoding preset (e.g. -vpre medium) [13:10] <ph1b> [libx264 @ 0xb10040] preset usage: -vpre <speed> -vpre <profile> [13:10] <ph1b> [libx264 @ 0xb10040] speed presets are listed in x264 --help [13:10] <ph1b> [libx264 @ 0xb10040] profile is optional; x264 defaults to high [13:13] <sacarasc> You're using a really old version, and it tells you what to do. [13:14] <ph1b> how can I update it? [13:15] <sacarasc> What OS/distro are you using? [13:17] <ph1b> ubuntu 11.10 [13:17] <sacarasc> There's a guide in the topic for compiling on Ubuntu. [15:08] <YellowOnion> anyone know how to get best multi processor/core performance? [15:09] <YellowOnion> with Linux [15:10] <YellowOnion> there's some problems with loading all cores, i have -threads 6 set but there's constant movement of load between cores and I'm only getting about 80% total load [15:17] <haakon__> YellowOnion: x264? [15:17] <YellowOnion> haakon__, yeah [15:17] <haakon__> YellowOnion: Tried using -threads 0 ? [15:19] <YellowOnion> haakon__, trying now, doesn't look any better [15:19] <YellowOnion> anywhere between 79 and 95% load [15:20] <YellowOnion> I've noticed that using 12 threads helps quite a lot [15:20] <YellowOnion> but I'm trying to understand why [15:20] <YellowOnion> threads 0 I'm getting 13 fps [15:23] <YellowOnion> hmm, I'm piping the dvd in to ffmpeg via mplayer with fifo's [15:30] <YellowOnion> haakon__, added 6 threads to mplayer [15:30] <YellowOnion> haakon__, no help for -thread 0 [15:31] <haakon__> YellowOnion: http://mewiki.project357.com/wiki/X264_Settings#threads [15:31] <haakon__> YellowOnion: Can you pastebin your complete command? [15:33] <YellowOnion> haakon__, http://paste.pocoo.org/show/536202/ [15:33] <YellowOnion> comments might be a little wrong [15:34] <berdario> Hello, I'm trying to play an rtp stream with ffmpeg (before trying to write a software to do it) [15:34] <berdario> but I have a problem to get it working with ffplay [15:34] <berdario> vlc works fine, but I read that vlc uses live555 for it's rtp client [15:34] <berdario> I have an .sdp that describes the stream [15:35] <berdario> but it complains that the data is invalid [15:35] <berdario> increasing the verbosity doesn't yield further information [15:35] <YellowOnion> haakon__, thing is I noticed this switching balancing problem with when running 3 instances of rubberband (2 threads per app) so I was thinking the Scheduler might be at fault [15:36] <berdario> I don't know if it's possible to connect directly without the .sdp... I looked into the manual but I don't see some options I need [15:37] <berdario> like the payload type: I have in fact read that ffmpeg is hardcoded to use payload type 96 [15:37] <YellowOnion> benpro, specifics or we can't help you, "it deosn't work" doesn't help us help you. [15:37] <YellowOnion> berdario, , specifics or we can't help you, "it deosn't work" doesn't help us help you. [15:37] <YellowOnion> sorry benpro [15:37] <berdario> YellowOnion, what do you mean? [15:37] <YellowOnion> berdario, error message? [15:38] <berdario> YellowOnion, "the data is invalid" is the exact error message outputted by ffplay [15:38] <berdario> I'll paste the whole output now [15:38] <berdario> (ok, not actually the exact error message, 2 seconds and I'll paste it) [15:39] <YellowOnion> berdario, oh whoops I missed the few comments that you said later sorry [15:39] <YellowOnion> berdario, paste the stream URL as well [15:40] <berdario> oh, ok... probably I should write fewer sentences, and longer... It's not the first time I have some problems communicating through irc :) [15:40] <berdario> http://paste.pocoo.org/show/536205/ [15:40] <berdario> the stream is actually local [15:40] <berdario> I can give you the dump I'm using to recreate it if needed [15:41] <berdario> the point is, I'm not actually sure if what I'm planning to do is possible with ffmpeg [15:41] <berdario> like I said, the payload type seems to be hardcoded... if that's the case I'll try to use directly some other library to receive rtp, and leave to ffmpeg only the decoding :) [15:45] <berdario> I'm currently looking at libavformat/rtpdec.c but I never looked at the ffmpeg sources before :) [15:46] <haakon__> YellowOnion: Thanks for the pastebin, see that threads 0 is correct, but unfortunately I do not know what might be the challenge here [15:48] <YellowOnion> x264 is using 9 threads which seems correct [16:00] <YellowOnion> haakon__, not sure why I'm piping from mplayer exactly I'm going to try dump the file and encode that [16:04] <berdario> YellowOnion, ok, I found out that apparently, in september 2011 a private option was added to specify the payload type... do you have any more information on this? [16:06] <haakon__> YellowOnion: Yeah I guess dumping and then encoding should free some resources [16:06] <YellowOnion> berdario, find a tool that can inspect the .sdp file [16:06] <Belial__> I am having problems converting a 3gp video into avi. I've read different posted commands on websites but so far none of them work. The one command that I got to work produced an avi with no sound. What command in ffmpeg will convert a 3gp from a mobile phone into a working avi (with sound!)? [16:07] <berdario> YellowOnion, an .sdp file is a plain text file... I have it open right now, and vlc parses it fine, any clue about some common problems with sdp files? [16:08] <YellowOnion> Belial__, can you not just do ffmpeg -i file.3gp file.avi [16:08] <Belial__> That apparently is too simple (I'll try it) - every command I found on websites were much more complex. I'll let you know. [16:09] <YellowOnion> Belial__, yeah there's a few settings to improve quality etc, but that should get some baseline working format [16:10] <Belial__> YellowOnion: that worked. Thanks. I guess people posting the command string online must get paid by the character or something. [16:10] <YellowOnion> heh [16:10] <YellowOnion> Belial__, I would just be a bit careful on the quality though and size of the file [16:12] <YellowOnion> Belial__, do you need .avi? you can get better quality compression ratio with a x264 stream in something like a .mkv [16:12] <Belial__> YellowOnion: I want to send the vid to family and they use windows and I wanted something certain to work for them. [16:13] <YellowOnion> Belial__, which version of windows? [16:13] <Belial__> YellowOnion: Not sure, some are Windows 7 but some may be Vista [16:16] <Belial__> YellowOnion: And here's another one. I want to rotate the video clockwise 90 degrees. I see a website with a simple command: 'ffmpeg -i input.avi -vf "transpose=1" output.avi' but naturally this command doesn't work. [16:17] <YellowOnion> Belial__, that'll definitely play a x264 stream in a .mp4 [16:24] <YellowOnion> haakon__, well dumping file first didn't help [16:24] <YellowOnion> default settings I get 11fps [16:24] <YellowOnion> if I use 12threads I get 16-18fps [16:26] <YellowOnion> Belial__, yeah that command should work. what's the error message? [16:26] <Belial__> YellowOnion: Unrecognized option 'vf' [16:27] <Belial__> YellowOnion: I rotated the video with mencoder and it doubled (almost) the size of the video. [16:28] <YellowOnion> Belial__, after using ffmpeg? [16:29] <Belial__> YellowOnion: the unrecognized option is ffmpeg and the video size doubled after rotating the original ffmpeg-produced video using mencoder. [16:30] <YellowOnion> Belial__, yeah you don't want to reencode again I'm wondering why it thinks -vf is missing, can you copy paste the exact command you're using? [16:30] <Belial__> YellowOnion: I would like to try a tighter/better compression video as you suggest but every single complex command string I've tried from other websites don't work (or produce errors about wrong bitrate or frame size). How do I simply convert an avi to a tighter mp4 (with x264 stream)? [16:31] <Belial__> YellowOnion: Sure...here it is: ffmpeg -i input.avi -vf "transpose=1" output.avi [16:33] <YellowOnion> Belial__, I use ffmpeg -i file.ext -thread 0 -vprofile high -preset veryslow -x264opts level=3 -vf 'transpose=1' outfile.mp4 [16:34] <YellowOnion> Belial__, http://en.wikipedia.org/wiki/H.264/MPEG-4_AVC#Levels [16:35] <YellowOnion> that's the only option that probably needs changing [16:35] <Belial__> YellowOnion: Your command fails for me as well: unrecognized option 'thread'. This is Mageia 1 linux with ffmpeg 0.6.4 [16:36] <YellowOnion> Belial__, sorry -threads [16:38] <Belial__> YellowOnion: Thx but now new error: [NULL @ 0xbd6050]Unable to parse option value "high": undefined constant or missing ( followed by "Invalid value 'high' for option 'vprofile'". Am i to assume I have a crippled ffmpeg? [16:42] <YellowOnion> Belial__, there was a recent change I think where they removed a bunch of profiles and started using built in ones for x264 [16:44] <YellowOnion> Belial__, you on linux? [16:44] <Belial__> YellowOnion: It appears that virtually all the options in your command string are defunct. I remove each one in succession to see what message I get and every single option turns up "unknown option" [16:44] <Belial__> YellowOnion: yes [16:44] <YellowOnion> Belial__, some options require others I think [16:45] <Belial__> YellowOnion: -vf is definitely defunct. It always comes up as unrecognized. [16:45] <YellowOnion> $ ls /usr/share/ffmpeg/ see if there's any libx264-high profiles [16:46] <YellowOnion> Belial__, what's your distro? [16:46] <Belial__> Mageia 1, a fork of Mandriva [16:49] <Belial__> YellowOnion: I just did a quick conversion from avi to mp4 with mencoder and it worked but video size is identical and sound is delayed. Cannot use mencoder and so far cannot use ffmpeg. [16:56] <YellowOnion> Belial__, sorry lost internet, did you answer my question? [16:58] <Belial__> YellowOnion: which question? I am on linux - Mageia 1, a fork of Mandriva [16:58] <YellowOnion> Belial__, you got your distro package of ffmpeg? [16:58] <Belial__> YellowOnion: yes [16:59] <YellowOnion> Belial__, try downloading x264 [17:00] <Belial__> YellowOnion: No libx264-high profiles. Up to medium and then lossless [17:00] <Belial__> YellowOnion: it is downloaded and installed [17:02] <YellowOnion> Belial__, I assume you have mplayer? [17:02] <Belial__> YellowOnion: yes [17:04] <YellowOnion> Belial__, well you could try to re compile ffmpeg [17:05] <YellowOnion> or you can decode then rotate with mencoder then encode with ffmpeg [17:05] <Belial__> YellowOnion: Yeah...looking forward to that. Haven't compiled much the last couple years because I inevitably run into a dependency hell. [17:06] <YellowOnion> heh [17:06] <Belial__> A straight convert from avi to mp4 in ffmpeg makes the same size video (double the size of the original unrotated avi). [17:07] <YellowOnion> you were encoding a 4gp right? [17:07] <Belial__> kdenlive is unstable as hell and crashes constantly so trying to edit down the converted video is impossible there. [17:07] <YellowOnion> 3* [17:07] <Belial__> YellowOnion: 3gp [17:08] <YellowOnion> work with the original format [17:08] <YellowOnion> each chain in the process destroys the quality [17:09] <YellowOnion> Belial__, you probably want a format designed for editing [17:11] <YellowOnion> Belial__, decoding frames in an editor can be a serious problem with formats like xvid and x264 [17:15] <berdario> YellowOnion, I'm quite sure that support for payload types has been added both to encoding and decoding [17:16] <berdario> so, I think that the only problems lie to get ffmpeg to accept my sdp [17:16] <berdario> do you know of any other alternative than doing a debug build and step through? [17:18] <YellowOnion> berdario, not sure if it's a text file why not just pull the stream or what ever it is out of the text? [17:18] <berdario> YellowOnion, an sdp contains the address of the server supplying the stream, plus some information on the video/audio codecs, timing and such [17:19] <berdario> this is mine: http://paste.pocoo.org/show/536262/ [17:21] <YellowOnion> berdario, what does 99999 do? [17:23] <berdario> "o" contains the originator and session identifier... I was under the assumption that besides "IP4 <ipaddress>" it wasn't really important [17:23] <berdario> I'll look again at the spec [17:24] <YellowOnion> perhaps the session ID has expired? I don't see a port specified anywhere [17:24] <berdario> ok, it's an identifier... not really important since I'm testing on localhost [17:24] <berdario> <sess-id> is a numeric string such that the tuple of <username>, [17:24] <berdario> <sess-id>, <nettype>, <addrtype>, and <unicast-address> forms a [17:24] <berdario> globally unique identifier for the session. The method of [17:24] <berdario> <sess-id> allocation is up to the creating tool, but it has been [17:24] <berdario> suggested that a Network Time Protocol (NTP) format timestamp be [17:24] <berdario> used to ensure uniqueness [17:24] <berdario> the port is 24104 (for video) [17:25] <berdario> whoops, sorry... I pasted straight from the RFC... if it hadn't any hard newlines it wouldn't have been so messy [17:26] <berdario> btw, ffmpeg doesn't even try to connect [17:26] <YellowOnion> berdario, I like 80 char width limit personally [17:26] <berdario> if I run the stream or not doesn't make any difference: it complains when reading the file [17:27] <YellowOnion> hmm [17:31] <berdario> uhm, I think that stepping through a debug build may be the fastet way to understand what's going wrong [17:32] <berdario> since I've never done it (I used gdb sometimes, but that's it)... I'll look for some documentation [17:33] <berdario> for further questions about debug builds maybe it's better off to ask in ffmpeg-devel ? [17:34] <YellowOnion> yeah ask devel no idea why it's happening [17:35] <YellowOnion> Belial__, report a bug to your distro, about the video filters being missing. [18:35] <Mista_D> 0.9.1 ffprobe has no -show_frames option anymore. is there any way to extract video frames map for gop analysis? [18:37] <dericed> Mista_D: Have you looked at -vf showinfo [18:39] <Mista_D> dericed: same as `-i` by itself. [18:39] <dericed> Mista_D: try `ffmpeg -i file -vf showinfo -f null -` [18:43] <Mista_D> dericed: its a little slow. [18:44] <dericed> Mista_D: it decodes the file so yeah [18:45] <Mista_D> dericed: I don't need the entire frame decode, just the frame headers and extract frame type. [18:45] <saste> Mista_D: -show_frames was recently added in ffmpeg master [18:46] <saste> -show_frames was an option of ffprobe in sourceforge [18:46] <dericed> saste: that's what i thought [18:47] <Mista_D> saste: compiling rigth now. thanks! [18:48] <Mista_D> dericed: thanks for the alternative, it actually has some additional data that I need as well. [19:28] <terinjokes> I'm trying to -acodec copy to an RTMP server the following stream "Stream #0:1[0x1100]: Audio: aac ([15][0][0][0] / 0x000F), 48000 Hz, stereo, s16, 256 kb/s" but I get an error "[flv @ 0x7fc5f9024000] malformated aac bitstream, use -absf aac_adtstoasc av_interleaved_write_frame(): Operation not permitted" [19:51] <Bombo> how does ffmpeg calculate the SAR? or is it a fixed value in the header somewhere? like DAR? [19:53] <ayaka> does vdpau only can be used in decode, not in encode? [20:03] <Bombo> when i do "scale=640:ih*ow/sar/iw" i get 640x287 format, but when i use bc: "scale=640:455*640/(64/45)/710" (its 720x576, but with crop) i get echo "scale=2;455*640/(64/45)/710"|bc -> 288.83 [20:05] <Bombo> so it should be 289 if you round up [20:06] <Bombo> i'm not sure if something is wrong here or i'm just missing something [20:09] <saste> well actually the decimal part is just discarded when converting to int [20:11] <Bombo> then it should be 288 [20:13] <Bombo> when i'm taking an image from the mpg, png, and resize it to correct aspect (16:9, it would be 1024x576 (anamorph)) remove the borders (crop) i get an image in 1014x455, and when i resize it with [x] keep aspect i get 640x287 too, so whats going wrong? [20:24] <zap0> Bombo, whats the ratio of your crop? [20:31] <Bombo> zap0: SAR before crop is 64:45 (720x576) i do crop top 60 bottom 61, left 4 right 6, not sure how to calculate the ratio then [20:34] <cbreak> if you crop, you should always only crop to even numbers [20:34] <zap0> for non-rgb that is requirement, but RGB too? [20:35] <cbreak> no [20:35] <cbreak> it's a requirement of video codecs (most have even more requirements) [20:35] <cbreak> and the 4:2:0 or 4:2:2 color subsampling [20:38] <Bombo> hmmmkay when i crop 60-60-4-6 i get 640x289 [20:58] <Bombo> so then ffmpeg adds something or subtracts when i crop odd crop numbers? [21:02] <Bombo> its 455 with 61, so 455*640/(64/45)/710 == 288.83 and 453*640/(64/45)/710 == 287.56 so ffmpeg subtracts 2 pixels from the height hmmm [21:08] <grimmace> does anyone know how to set the output video bitrate to the same as the input video bitrate. I can't use "-sameq" or "-vcodec copy" because some other parameters have to be different. thanks for any help [00:00] --- Wed Jan 18 2012

1 0

ffmpeg-devel.log.20120116
by burek 17 Jan '12

17 Jan '12

[00:24] <ubitux> saste: you have a small typo in the ifzero patch: "togheter" [00:25] <saste> ubitux: that's one of my favourite typos [00:26] <ubitux> if you love them, then i guess you have to keep it [00:26] <ubitux> but just add a comment so it isn't fixed later [00:26] <saste> i should change the dictionaries instead [00:27] <ubitux> i'm sure the higher institutions will agree with this [00:27] <ubitux> ffdictionary [00:28] <ubitux> let's also rewrite dictionnaries in a better way [00:28] <ubitux> heh, we already have AVDictionary. [00:32] Action: iive handles the price "Best Joke Of The Year" to ubitux [00:33] <ubitux> sorry :( [00:33] <ubitux> i won't do it again. [00:34] <iive> why, it was really nice joke. [00:35] <ubitux> i'm not the original author of the AVDictionary joke [00:35] <ubitux> :-° [00:40] <ubitux> btw, am i the only one to be ignored by most of the libav people? [00:42] <michaelni> ubitux, no [00:42] <ubitux> ok [00:42] <ubitux> i guess it's because i'm definitely part of the dark side [00:53] <beastd> It is not dark. It is just FFmpeg, fast and forward you know :D [00:54] <ubitux> i like when it's dark [00:54] <ubitux> it feels safer. [00:55] <ubitux> saste: i like michael' suggestion about the if() instead of ifzero [00:58] <saste> ubitux: +1 [01:06] <Compn> ubitux : i am also ignored by most of libav [01:07] <ubitux> it feels like they want us to despise them so they could have a reasons to be [01:08] <ubitux> it sucks :( [01:08] <Compn> i would recommend dropping those patches [01:08] <ubitux> yes, i'll ping them another time and then drop them [01:44] <CIA-31> ffmpeg: 03Laurentiu Ion 07master * r4fccc7c46c 10ffmpeg/libavcodec/vb.c: [01:44] <CIA-31> ffmpeg: vb: Use bytestream2 functions [01:44] <CIA-31> ffmpeg: Signed-off-by: Justin Ruggles <justin.ruggles(a)gmail.com> [01:44] <CIA-31> ffmpeg: 03Kostya Shishkov 07master * re6e7bfc11e 10ffmpeg/libavcodec/wavpack.c: [01:44] <CIA-31> ffmpeg: wavpack: fix clipping for 32-bit lossy mode [01:44] <CIA-31> ffmpeg: Reference decoder clips data before shifting it to final range and also [01:44] <CIA-31> ffmpeg: forces 32-bit lossy mode to be actually 24-bit lossy mode in order to be [01:44] <CIA-31> ffmpeg: able to perform proper clipping. [01:44] <CIA-31> ffmpeg: 03Daniel Huang 07master * rbcd5d979aa 10ffmpeg/libavformat/ (mov.c mxfdec.c): [01:44] <CIA-31> ffmpeg: mov, mxfdec: Employ more meaningful return values. [01:44] <CIA-31> ffmpeg: Signed-off-by: Diego Biurrun <diego(a)biurrun.de> [01:44] <CIA-31> ffmpeg: 03Diego Biurrun 07master * r2df5f59ad0 10ffmpeg/tests/ (Makefile fate/adpcm.mak fate/dpcm.mak): fate: split ADPCM and DPCM test references into separate files. [01:44] <CIA-31> ffmpeg: 03Michael Niedermayer 07master * r4a4c4278b7 10ffmpeg/: (log message trimmed) [01:44] <CIA-31> ffmpeg: Merge remote-tracking branch 'qatar/master' [01:44] <CIA-31> ffmpeg: * qatar/master: [01:44] <CIA-31> ffmpeg: fate: split ADPCM and DPCM test references into separate files. [01:44] <CIA-31> ffmpeg: mov, mxfdec: Employ more meaningful return values. [01:44] <CIA-31> ffmpeg: lavc: Relax API strictness in avcodec_decode_audio3 with a custom get_buffer() [01:44] <CIA-31> ffmpeg: wavpack: fix clipping for 32-bit lossy mode [01:44] <CIA-31> ffmpeg: 03Reinhard Tartler 07master * re2ff436ef6 10ffmpeg/libavcodec/ (avcodec.h utils.c): (log message trimmed) [01:44] <CIA-31> ffmpeg: lavc: Relax API strictness in avcodec_decode_audio3 with a custom get_buffer() [01:44] <CIA-31> ffmpeg: Do not fail audio decoding with avcodec_decode_audio3 if user has set a [01:44] <CIA-31> ffmpeg: custom get_buffer. Strictly speaking, this was never allowed by the API, [01:44] <CIA-31> ffmpeg: but it seems that some software packages did so anyways. In order to [01:44] <CIA-31> ffmpeg: unbreak applications (cf. http://bugs.debian.org/655890) this change [01:44] <CIA-31> ffmpeg: clarifies the API and overrides the custom get_buffer() with the defaults. [03:40] <CIA-31> ffmpeg: 03Jean First 07master * r0c10885c39 10ffmpeg/libavformat/movenc.c: [03:40] <CIA-31> ffmpeg: movenc: indent cosemtics [03:40] <CIA-31> ffmpeg: Signed-off-by: Jean First <jeanfirst(a)gmail.com> [03:40] <CIA-31> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [03:40] <CIA-31> ffmpeg: 03Jean First 07master * r60168f9319 10ffmpeg/libavformat/movenc.c: [03:40] <CIA-31> ffmpeg: movenc: add FIXME to add alis and rsrc atoms [03:40] <CIA-31> ffmpeg: Signed-off-by: Jean First <jeanfirst(a)gmail.com> [03:40] <CIA-31> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [03:40] <CIA-31> ffmpeg: 03Jean First 07master * rfa169d5650 10ffmpeg/libavformat/mov.c: [03:40] <CIA-31> ffmpeg: mov: cosmetics - move a line to a better position and add a comment [03:40] Last message repeated 1 time(s). [03:40] <CIA-31> ffmpeg: Oana Andreea Stratulat submitted a similar patch to trac, but forgot [03:40] <CIA-31> ffmpeg: to notify the ML about it. [03:40] <CIA-31> ffmpeg: Signed-off-by: Jean First <jeanfirst(a)gmail.com> [03:41] <CIA-31> ffmpeg: This is required by the spec and fixes video-1frag.ogg.48.ogg. (FPE) [03:41] <CIA-31> ffmpeg: Based on the debuging work of Oana Stratulat and ubitux. [03:41] <CIA-31> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [03:53] <CIA-31> ffmpeg: 03Panagiotis H.M. Issaris 07master * r811de351ea 10ffmpeg/libavformat/applehttp.c: [03:53] <CIA-31> ffmpeg: Enhance HLS seeking. [03:53] <CIA-31> ffmpeg: Enhances seeking by demuxing until the requested timestamp is reached within [03:53] <CIA-31> ffmpeg: the segment selected by the seek code using the playlist info. [03:53] <CIA-31> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [04:13] <troy_s> Quick question... according to the transfer enum in avcodec.h, which would be the proper bt601.7 implementation? [04:16] <troy_s> Am I to assume it is 2.2 bt470m? [10:33] <ubitux> cbsrobot: i will send the first draft of the new public timecode api today, or tomorrow [10:34] <ubitux> so you can move on with your mov work [10:34] <ubitux> feel free to comment it on the mailing list (or here) when i post it btw [10:59] <CIA-31> ffmpeg: 03Carl Eugen Hoyos 07master * r203ea375ca 10ffmpeg/libavcodec/dpxenc.c: Support rgb64 dpx encoding. [11:00] <CIA-31> ffmpeg: 03Carl Eugen Hoyos 07master * ra8b117f800 10ffmpeg/tests/ (lavf-regression.sh ref/lavf/png): Add gray16 and rgb48 png encoding regression tests. [11:10] <cbsrobot> ubitux: great [11:10] <cbsrobot> btw here's the WIP http://pastebin.com/Pm6JaG1Q [11:11] <cbsrobot> it's not worth sending to the ML because it's rather basic [11:11] <ubitux> the new API should be way more consistent and flexible [11:11] <cbsrobot> ok [11:11] <ubitux> and i'm introducing the flags indeed [11:12] <cbsrobot> i'm not sure if we should add metadata to ff_timecode .... [11:14] <ubitux> good point [12:01] <kriegerod> hi, devs - in libav-user(a)ffmpeg.org there's interesting question about usage of context pointer argument for logging callback function. Title is "[Libav-user] Questions about using libav in a multi-threaded application". Could anybody shed some light? [12:02] <kriegerod> I mean answering in that thread [12:57] <CIA-31> ffmpeg: 03Carl Eugen Hoyos 07master * rd93b6340d7 10ffmpeg/tests/ (lavf-regression.sh ref/lavf/dpx): Add dpx encoding regression test. [16:38] <durandal_1707> how to handle formats that put data in single one big chunk after headers? [16:38] <av500> durandal_1707: run from them as fast as possible [16:39] <durandal_1707> our api cant handle them? [16:39] <Tjoppen> what format? [16:39] <Tjoppen> mov would fit that vague description [16:39] <durandal_1707> brstm [16:40] <durandal_1707> http://wiibrew.org/wiki/BRSTM_file [16:41] <av500> ah, so it's like WAV [16:41] <Tjoppen> write a demuxer for it? should be less than 200 lines [16:41] <Tjoppen> yeah, could probably share code with wav.c [16:42] <av500> refactor into header_and_large_chunk.c [16:42] <Tjoppen> wav.c ought to be split into wavdec.c and wavenc.c btw [16:42] <merbanan> durandal_1707: just return a suitable amount of data [16:44] <merbanan> durandal_1707: do you have complete specs for the adpcm format ? [16:45] <durandal_1707> you have implementation in vgmstream [16:45] <merbanan> ok [16:50] <merbanan> durandal_1707: yupp, it's all there, write demuxer and pass the adpcm coeffs as extradata to the decoder [17:48] <CIA-31> ffmpeg: 03Clément BSsch 07master * r5fc51c3155 10ffmpeg/libavutil/audioconvert.c: lavu: add 2.1 channel layout. [19:35] <CIA-31> ffmpeg: 03Reimar Döffinger 07master * r6003ec880b 10ffmpeg/libavformat/asfdec.c: [19:35] <CIA-31> ffmpeg: asfdec: Move variable declarations into the blocks they are used in. [19:35] <CIA-31> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de> [19:35] <CIA-31> ffmpeg: 03Reimar Döffinger 07master * r203f94fd7c 10ffmpeg/libavformat/mxfdec.c: [19:35] <CIA-31> ffmpeg: mxfdec: Add missing free. [19:35] <CIA-31> ffmpeg: Should fix valgrind FATE. [19:35] <CIA-31> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de> [19:35] <CIA-31> ffmpeg: 03Reimar Döffinger 07master * r7594791541 10ffmpeg/libavcodec/vp8.c: (log message trimmed) [19:35] <CIA-31> ffmpeg: Rework VP8 decoder frame skipping code. [19:35] <CIA-31> ffmpeg: This reworks the frame skipping code such that the reference [19:35] <CIA-31> ffmpeg: buffers are still updated according to the header. [19:35] <CIA-31> ffmpeg: However it also ensures that the current frame will not end [19:35] <CIA-31> ffmpeg: up in any reference buffer. [19:35] <CIA-31> ffmpeg: Also fixes a hang with frame-multithreading, probably because [19:45] <CIA-31> ffmpeg: 03Reimar Döffinger 07master * rc203044f37 10ffmpeg/tests/ref/lavf/dpx: [19:45] <CIA-31> ffmpeg: Remove extraneous newline from dpx reference which breaks FATE. [19:45] <CIA-31> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de> [20:43] <CIA-31> ffmpeg: 03Maksym Veremeyenko 07master * r94bf9ac473 10ffmpeg/libavcodec/parser.c: [20:43] <CIA-31> ffmpeg: fix av_dlog call with non-AVClass struct [20:43] <CIA-31> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [00:00] --- Tue Jan 17 2012

1 0

ffmpeg.log.20120116
by burek 17 Jan '12

17 Jan '12

[00:00] <JodaZ> hmm, i see [00:00] <JodaZ> so whats that fancy new 10bit stuff ? [00:01] <Mavrik> good question... I think the guys on #x264 are better equipped to answer that, since x264 is prbably the only one that supports 10bit :) [00:32] <Mavrik> man, ffmpeg makes a mess of VP8 encoding [00:35] <darkstarbyte> I would say so [00:35] <darkstarbyte> I wanted to use it once [00:35] <darkstarbyte> I decided to just use x264 [00:35] <darkstarbyte> since they are related [00:36] <Mavrik> x264 is better in every aspect [00:36] <Mavrik> but... clients wanna have what they wanna have -_- [00:36] <darkstarbyte> yeah [00:38] <teratorn> in every aspect except that it's proprietary [00:38] <darkstarbyte> High school of the dead [00:39] <darkstarbyte> The encoder is free [00:39] <darkstarbyte> You just have to pay royalties for the converted content after 2016 [00:39] <Mavrik> teratorn, that argument is only valid once we see what vp8 is. [00:39] <Mavrik> teratorn, plus, x264 is GPL ;) [00:39] <darkstarbyte> vp8 and h264 are related [00:40] <darkstarbyte> The techniques are patented [00:40] <teratorn> GPL is a problem for a lot of folks [00:40] <JEEB> x264 has a commercial license if one is needed >_> [00:40] <darkstarbyte> frame prediction stuff is patented [00:40] <teratorn> darkstarbyte: you're saying vp8 is patent-encumbered? [00:41] <darkstarbyte> There are patents on it, but google bought them [00:41] <JEEB> orly [00:41] <JEEB> I think no-one actively said that it's "safe to use" [00:41] <JEEB> it's taken for granted :) [00:41] <darkstarbyte> Hi JEEB [00:42] <cbreak> it's software [00:42] <darkstarbyte> and? [00:42] <cbreak> which countries accept software patents? [00:42] <darkstarbyte> US [00:42] <cbreak> just only sell in the others [00:43] <darkstarbyte> Japan might, but I have never looked into it. [00:43] <JEEB> anyways, IANAL and if you're thinking about officially using pretty much any tech I recommend you visit a lawyer >_> [00:43] <cbreak> anyway, patents are a national thing anyway [00:43] <cbreak> they are only worth anything in the country that granted them. [00:43] <Mavrik> of course [00:43] <Mavrik> but if you get killed off of a 200 mil market online... that's quite a hit [00:44] <darkstarbyte> I think copyrights work the same way [00:44] <Mavrik> especially since covering rest of the world is annoying [00:44] <cbreak> copyrights aren't granted on request [00:44] <cbreak> they are granted for everything [00:44] <Mavrik> darkstarbyte, I think most of the world consders copyright a default [00:44] <cbreak> and they are recognized with some agreements between countries [00:44] <ptriller> is it possible that libvo_aacenc is unable to encode 5.1 6 channel audio, I can't get conversion from AC3 5.1 to AAC 5.1 to work [00:45] <cbreak> if I write a Poem or make a picture, then those have my copyright, and most countries accept that [00:45] <darkstarbyte> vorbis might [00:45] <JEEB> ptriller, I think I remember someone saying that it has limitations with >2ch [00:45] <ptriller> JEEB, thanks, at least now I know I am not just incompetent :) [00:45] Action: cbreak would not compress ac3 further [00:45] <JEEB> yeah [00:45] <JEEB> diminishing returns [00:46] <cbreak> at least not without converting to stereo or something [00:46] <JEEB> aye [00:46] <darkstarbyte> I am sure vorbis will do it, but I will check [00:46] <cbreak> (for devices like iThings) [00:46] <cbreak> then you could hard code something like hrtf [00:47] <darkstarbyte> It says vorbis is not good for multichannel [00:47] <darkstarbyte> Its good for [00:47] <darkstarbyte> <3ch [00:48] <darkstarbyte> I for that from a form 6 years old [00:48] <darkstarbyte> for = got* [01:26] <EOF-sensei> I built ffmpeg against libx264 [01:26] <EOF-sensei> but it doesn't seem to be able to find my libx264.so.120 [01:27] <EOF-sensei> even though it's clearly in /usr/local/lib [01:27] <EOF-sensei> what might I[t] be doing wrong? [01:38] <EOF-sensei> bumpe [01:39] <EOF-sensei> ffmpeg: error while loading shared libraries: libx264.so.120: cannot open shared object file: No such file or directory [01:46] <Mavrik> hmm [01:46] <Mavrik> EOF-sensei, either you didn't run "ldconfig" yet or you're trying to load a x86 lib to a x64 process [01:46] <Mavrik> EOF-sensei, do run "ldd ffmpeg" and see if it finds the .so [01:48] <relaxed> EOF-sensei: sudo echo "/usr/local/lib" >> /etc/ld.so.conf; sudo ldconfig [01:49] <EOF-sensei> needed to run ldconfig [01:49] <EOF-sensei> remembered a few minutes ago [01:49] <relaxed> ok [01:49] <EOF-sensei> I wish that sort of thing were automated [01:50] <EOF-sensei> caching and indexing daemons should really try to be transparent if they exist at all [01:50] <EOF-sensei> in my humble opinion... [01:50] <relaxed> well, your package manger does that when installing libs [01:51] <relaxed> manager* [01:52] <EOF-sensei> I guess I haven't used shared libraries right after compiling them frequently enough to remember to run ldconfig [03:40] <chcha> Is there anyone has some information about build ffmpeg with stagefright? [03:40] <chcha> I don't know how to do that exactly... [05:53] <EOF-sensei> chcha: ./configure --enable-libstagefright [07:21] <chcha> <EOF-sensei> chcha: ./configure --enable-libstagefright is that all? [07:32] <EOF-sensei> if libstagefright is all you want [07:33] <EOF-sensei> if you don't know what I'm talking about then yes [08:11] <chcha> Actually I was using ffmpeg on android phone to use custom codec based on h.264 codec over rtsp. [08:16] <chcha> For the performance issue, we decided to use standard h.264 codec but, some Android phone don't show rtsp stream with Android Multimedia Framework. [08:17] <chcha> So, I'd like to know It's possible switching between ffmpeg and stagefright according to the phone's capability. [08:17] <chcha> Anyway, libstagefright is not all I want :) [08:18] <EOF-sensei> hmm [08:18] <EOF-sensei> well [08:18] <EOF-sensei> rather [08:18] <EOF-sensei> what I meant is [08:18] <EOF-sensei> of the libraries that ffmpeg is capable of linking in [08:18] <EOF-sensei> s/libraries/extra libraries/ [08:19] <EOF-sensei> is libstagefright the only one you want? [08:19] <EOF-sensei> considering you could be wanting to use say, x264 [08:19] <EOF-sensei> or work with jpeg2000 sequences [08:19] <EOF-sensei> ;p [08:20] <EOF-sensei> chcha: there should be a way to switch between libstagefright and the standard h.264 codec [08:20] <EOF-sensei> but I don't know it [08:20] <EOF-sensei> ;p [08:20] <EOF-sensei> I'm a noob [08:20] <darkstarbyte> ¿pyPoqÝ o e1 uoy s1 PeM [08:22] <EOF-sensei> darkstarbyte: yeh, good luck... [11:19] <burek> halp :) [11:20] <burek> http://pastebin.com/S7yYzq3G [11:20] <burek> libass says it cannot create track from .ass file [11:21] <burek> I've created .ass file with subtitleeditor, using File - Save As... (and selected .ass extension) [11:22] <burek> this is the .ass file: http://pastebin.com/3iB9HsL9 [11:27] <cbsrobot> burek: this does not look like an ass file. [11:29] <burek> hm, ok [11:29] <burek> thx [11:32] <cbsrobot> burek: http://en.wikipedia.org/wiki/SubStation_Alpha [11:32] <sacarasc> Yeah, that looks like basic SRT. [11:33] <burek> yes, i was suspecting on that :) [11:33] <burek> i've used subtitle composer to save as .ass and now it works :) [11:33] <burek> thanks guys :beer: :) [11:57] <eakin> hi all, I have a windows media player plugin embedded in my html, I am referencing an mp4 video from an external website, the video plays but the progress bar doesn't move, any idea why or how to resolve this? [12:04] <jermy> Can I typically speed up decoding of interlaced H.264 content that I intend to then scale anyway by only decoding a single field? [12:05] <jermy> Or are fields often dependent on each other? [12:05] <Tjoppen> I susepct the answer to that question is "it depends" [12:05] <JEEB> other field can very well be dependant on the other one [12:05] <JEEB> thus, no [12:06] <jermy> fair enough [12:06] <jermy> I really do despise interlacing [12:06] <nifan> When i try to use "-strict experimental" ffmpeg says "Invalid value 'strict' for option 'experimental'" using ffmpeg version 0.8.9 any idea what i'm doing wrong ? [12:07] <jermy> Tjoppen/JEEB: I guess it depends on if the top field ever has dependencies on the bottom field of the previous frame [12:08] <JEEB> jermy, or the other way, H.264 can have biprediction [12:08] <jermy> Indeed [12:10] <jermy> In any case, if I am going to throw away half the picture, is there anything less work I could make a decoder do? [12:11] <jermy> (this is more of a theoretical question than a ffmpeg-specific one) [12:12] <JEEB> not really [12:12] <JEEB> calculating if a frame's parts depend on other parts will mostly end up making you do MORE work >_> [12:17] <jermy> I guess I'll have to see if I can make sure I'm not passing anything I don't need into the filtering stage [12:21] <jermy> any also try to work out why the h264 decoder segfaults when we ask it to run with multiple threads [12:22] <jermy> s/any/and/ [12:33] <isset_> why doesn't ffmpeg show me the duration while converting [12:34] <jlue> what's the syntax for image2 input, please? Using "ffmpeg -y -an -i "Plate28_1/Plate28_1.0000%d.jpg" -f image2 -vcodec libx264 -b:v 1024k -vpre slow Plate28_1.mp4" fails in version 0.9.1. [12:38] <eakin> ok the seek bar works for my mp4 video via VLC but not with windows media player, any idea why? [12:39] <cbreak-work> what are plausible pts/dts values for passing into an AVPacket? [12:39] <cbreak-work> 0, 1, 2, 3, 4, 5 and so on? [12:50] <Tjoppen> any values where dts monotonically increases and pts >= dts [13:35] <ph1b> hey. I wanna crop a video using -vf but how can I find out the right coordinates of the area to be cropped? Just made a screenshot using totem but now the screenshot is bigger than the original video (764 vs 720) [13:36] <Mavrik> is it anamorphic? [13:36] <ph1b> what that means? [13:37] <Mavrik> ph1b, do a ffprobe on the video and see if PAR/DAR/SAR values are different than 1:1 [13:37] <Mavrik> that means that pixels aren't square in video [13:37] <ph1b> 16:15 [13:38] <ph1b> so i just have to correct my cropping results by this factor? [13:38] <Mavrik> ph1b, hmm, probably not [13:38] <Mavrik> you should just use the coordinates from Totem screenshot (the bigger one) [13:38] <Mavrik> if I remember correctly the decoded images aren't anamorphic and video filters work on decoded images [13:40] <ph1b> That what leads me to this forum. I tried a [13:40] <ph1b> ffmpeg -i kkurz.vob -vcodec libx264 -acodec libvo_aacenc -b 1024 -ab 128k -vf '749:574:13:0' -ss 50 -t 30 output.mkv [13:40] <ph1b> but result is: [13:40] <ph1b> [buffer @ 0x1b20ec0] w:720 h:576 pixfmt:yuv420p [13:40] <ph1b> No such filter: '749:574:13:0' [13:40] <ph1b> Error opening filters! [13:40] <Mavrik> ph1b, yeah, you didn't specify the video filter [13:40] <Mavrik> -vf crop=749:574:13:0 [13:41] <Mavrik> the -vf is in form of [13:41] <Mavrik> -vf [filter1]=[filter1parameters],[filter2]=[filter2parameters]... [13:41] <ph1b> Ah, thanks. [13:41] <ph1b> [buffer @ 0xc3fec0] w:720 h:576 pixfmt:yuv420p [13:41] <ph1b> [crop @ 0xc41380] w:720 h:576 -> w:748 h:574 [13:41] <ph1b> [crop @ 0xc41380] Invalid too big or non positive size for width '748' or height '574' [13:41] <ph1b> Error opening filters! [13:42] <Mavrik> um yea [13:43] <Mavrik> why are you trying to crop an image from coordinates 749,574 when your video is only 748x574 big? :) [13:44] <Mavrik> ph1b, http://ffmpeg.org/libavfilter.html#crop [13:44] <ph1b> the screenshot says:768x576 [13:45] <Mavrik> hmm [13:45] <Mavrik> ph1b, you'll have to multiply by 15/16 then to get right size [13:45] <Milos> If I have a stream at http port 4242 at URI /herp/derp how can I grab this and restream it on another port etc? [13:45] <Milos> Is this possible with ffmpeg alone? [13:46] <Mavrik> ph1b, I guess I was wrong about decoder fixing anamorphic images [13:46] <Mavrik> Milos, ffmpeg can grab UDP and simillar streams, I'm not sure if it can do HTTP downloads [13:46] <Milos> I see [13:47] <Mavrik> but I somehow doubt you have a HTTP stream on 4242 port ;) [13:48] <Milos> I do [13:48] <Milos> that's why I asked [13:48] <Milos> o-o [13:48] <Milos> fail [19:27] <DrSlony> Hey, which do you recommend I use when ripping a DVD, MPEG-1 layer III, or AC3? [19:29] <JEEB> for LCPM DVDs and stuff with ridiculously high bitrate DTS/AC3 (2ch with 400+kbps f.ex.) there's neroaacenc/aotuv's vorbis, I'd usually leave 5.1ch AC3 stuff as-is [19:29] <llrcombs> AC3 [19:30] <llrcombs> you might be able to get away without transcoding at all [19:30] <llrcombs> just reencapsulate [19:30] <newl> use vobcopy -> avi [19:30] <llrcombs> that also works [19:30] <llrcombs> what do you intend to play this on, anyway? [19:31] <llrcombs> if you can get away with just reencapsulating and/or merging VOBs, do that. Otherwise, there are a few things that might have to be done here... [19:32] <DrSlony> llrcombs probably not on dvd players, just computers [19:33] <llrcombs> DrSlony: what player, though? Not all players like MPEG2/AC3 AVIs [19:33] <JEEB> why the avi love? [19:33] <llrcombs> also, I usually prefer MKVs [19:33] <DrSlony> i want to deinterlace, fix colors, etc [19:34] <JEEB> you can properly do deinterlacing/IVTC and re-encode to H.264 in mkv, audio codecs by what I said first [19:34] <llrcombs> yep [19:34] <llrcombs> if you want to adjust colors and deinterlace, H.264->MKV is the way to go [19:34] <JEEB> over-high-bitrate stuff you can deal with neroaacenc or aotuv's vorbis, and then the normal-bitrate stuff you can just remux as I said [19:34] <llrcombs> usually if it's AC3 audio, you should just remux into the MKV [19:34] <llrcombs> JEEB: ^5 [19:35] <JEEB> llrcombs, unless it's 400kbps+ AC3 for stereo [19:35] <DrSlony> why mkv and not mp4? [19:35] <llrcombs> JEEB: what then? [19:35] <JEEB> DrSlony, more freedom of choice with video/audio codecs [19:35] <JEEB> llrcombs, then re-encoding it to vorbis/aac with a good encoder is somewhat viable [19:35] <JEEB> as well as subtitles [19:36] <JEEB> oh, and chapters [19:36] <llrcombs> MKV's overall a better encapsulation format, as you can pack just about anything into it, and it hints well for web. Also, subs. Still, if you can't play MKVs in your player, then MP4s and MOVs are still an option [19:38] Action: newl plays whatever he gets [19:38] <newl> my tv card gives me avi so i like avis [19:39] <newl> the best from youtube is webm/mkv so i play that :) [19:39] <DrSlony> do you know of any GUIs that would let me use x264 and deinterlace and optionally show a preview of the video so that i can crop? [19:39] <newl> kino? [19:39] <JEEB> I would probably use avisynth + AvsPmod [19:39] <JEEB> and then command line x264 [19:39] <JEEB> of course, this is in no way the easy path :P [19:40] <JEEB> but usually quite worth it if you want to do something "properly" [19:40] <DrSlony> using avisynth in liux is meh [19:40] <JEEB> well, yes [19:40] <newl> why the x264 love? [19:40] <JEEB> newl, I would rather ask why not [19:40] <JEEB> > easy to use > currently the best compression ratio and speed available [19:40] <DrSlony> why not the x264 love [19:40] <DrSlony> heh exactly [19:41] <DrSlony> hm [19:41] <JEEB> avs on linux via wine is a bit of a meh, but IIRC AvsPmod has been fixed [19:41] <JEEB> or well [19:41] <DrSlony> methink i'll use avidemux2 [19:41] <JEEB> "suddenly started working after we updated the base" [19:41] <newl> webm is best size and quality ? [19:41] <JEEB> lol [19:41] <JEEB> webm is a container format [19:41] <JEEB> vp8 is a video codec [19:41] <JEEB> vorbis is an audio codec [19:42] <JEEB> and libvpx that encodes vp8 is unfortunately A) slow B) worse off than baseline x264 still [19:42] Action: newl still has no clue why they make this container format distinction - just to separate us who know little from those who actually know [19:42] <llrcombs> new1: actually, YouTube's highest-quality file is currently an MP4 with H.264 [19:43] <llrcombs> http://en.wikipedia.org/wiki/YouTube#Quality_and_codecs [19:43] <newl> mp4 with h.264 is the best but on a 650Mhz it don't play [19:43] <newl> it is huge too [19:43] <llrcombs> the MPEG4 goes up to 2304p [19:43] <llrcombs> (AAC audio) [19:43] <JEEB> the libavcodec vp8 decoder is newly coded and somewhat faster than the H.264 one, yes [19:44] <llrcombs> the WebMs go up to WUXGA (1080p), though [19:45] <JEEB> but vp8 does not compress better than x264 generally, youtube does use crappy settings so the comparison isn't exactly valid by just checking youtube stuff >_> (and still H.264 usually ends up looking better) [19:45] <JEEB> newl, they pretty much called a subset of matroska "webm" to limit the features people would have to support + to have their own buzzword .-. [19:47] <DrSlony> which fine overpaid body of people come up with these resolution combination names like WUXGA? [19:47] <llrcombs> that's pretty much it, yeah [19:47] <llrcombs> I think there are like 2 added features, both related to web streaming hinting [19:47] Action: newl doesn't want to actually admit that he get the crap youtube flvs and only better formats if he wants to save something like http://www.youtube.com/watch?v=OYpwAtnywTk [19:48] <llrcombs> Widescreen Ultra Extended Graphics Array [19:52] <DrSlony> how do i do video stabilization with ffmpeg, is there an official version with that filter or is it only in git? [19:55] Action: newl the only video he knows that was stabilized was the zapruder film [19:55] <DrSlony> avidemux shows "mpeg-4 avc", is that an alternate name for H.264? [19:56] <JEEB> yes [19:56] <JEEB> H.264/AVC [19:56] <JEEB> mpeg-4 part... 10 IIRC [19:56] <DrSlony> thx [19:56] <DrSlony> newl ... [20:16] <isset_> why doesn't ffmpeg show the duration in the status prompt? [20:37] <khali> does anyone know what output method ffplay uses? xv? opengl? [21:17] <DrSlony> I have the following choice of audio codecs: mp3 (LAME), AC3 (Aften), AC3 (Lav), AAC (Faac). Which of those do you recommend? [21:22] <khali> DrSlony: really depends on your (hardware) target and goals [21:22] <khali> and video codec [21:23] <khali> it's usual to go for mp3 with xvid, ac3 with mpeg2 and AAC with x264 [21:23] <khali> for hardware playback compatibility [21:24] <DrSlony> h.264 for computer playback [21:24] <DrSlony> mkv or mp4, not sure which [21:24] <DrSlony> was told earlier mkv offers more options [21:25] <khali> mkv is certainly better but mp4 is more portable in my experience [21:25] <khali> with h.264 I'd do AAC for audio and mp4 for container [21:26] <khali> this should play on pretty much everything [21:26] <DrSlony> is AAC from FAAC better than the others? [21:27] <DrSlony> i know nero is recommended [21:29] <j-b-r> You don't really ever want to encode AC3 unless it's hardware specific [21:29] <j-b-r> The only reason you see encodes with AC-3 is because the AC-3 audio came directly from a source, and it was better to keep it intact than to re-encode it [21:30] <j-b-r> You don't really want to use mp3 if you have a more modern option like AAC, but Faac really is pretty bad (you're correct, Nero would be the best choice, if possible) [21:31] <j-b-r> I'd just use Faac and not care too much, unless you're working in a situation where you're transcoding some other audio source, in which case I'd see about keeping the original audio intact [21:31] <DrSlony> the original is AC-3 48000Hz 224kbps [21:32] <DrSlony> not sure converting it to AAC using the nero encoder @ say 160kbps 41000 would be worth it [21:34] <j-b-r> If I was encoding for personal use I'd probably just go with the AAC, but if it was for distribution I'd probably keep the original audio unless it has way too many channels or something [21:34] <newl_> DrSlony: no ogg? [21:34] <JEEB> DrSlony, 224kbps is so low that you might as well remux it [21:34] <JEEB> as-is [21:35] <DrSlony> well this is for personal use, but im also wondering what works best for when i do screencasts for youtube [21:35] <newl_> 224kbps low? - i guess you don't want to listen to my 32kpbs :) [21:35] <DrSlony> JEEB low? [21:35] <DrSlony> newl_ no, shit support [21:35] <JEEB> DrSlony, "low enough that you won't be getting major gains from re-encoding" [21:36] <newl_> mp3 is kinda universal [21:36] <JEEB> 1400kbps LPCM -> ~200kbps AAC is still quite figurable f.ex. [21:36] <newl_> DrSlony: ^^ [21:39] <DrSlony> btw is the term muxing reserved for DVDs/BluRays etc, or is it used simply for putting an audio and video stream into mkv or avi as well? [21:41] <JEEB> muxing is generally used to refer putting streams into a container [21:42] <DrSlony> thx [21:56] <newl_> oh muxing i thought it said muffing nvm [22:54] <GuySoft> hi all, i need to encode a blender video i just made, its 1920x1080 , and needs to stay high quality. does anyone have a way to encode it? [22:56] <khali> GuySoft: well... ffmpeg is exactly what you need, and obviously you know that, otherwise you wouldn't be there? [22:57] <GuySoft> khali, yes, but i am not sure what flags to give, so something that large would stay in a decent quality [22:58] <JEEBsv> is this for end users? [22:58] <JEEBsv> or is this just you wishing to compress a lossless more for archival? [22:59] <JEEBsv> :3 [23:02] <khali> GuySoft: just pass high quality flags to ffmpeg (there are good suggestions in the FAQ) [23:03] <JEEBsv> khali: maybe you should actually wait until he tells you whatever he really wants [23:03] <khali> GuySoft: if you do bitrate-based encoding, use a high bitrate; if quality based, use a high quality setting [23:03] <JEEBsv> .-. [23:03] <khali> JEEBsv: you're right, GuySoft did not provide enough information for me to reply anything useful [23:03] <khali> that's what I'm writing bullshittish generalities for now ;) [23:04] <JEEBsv> in any case, GuySoft -- make sure you have as new as possible ffmpeg built with libx264 and possibly with some other aac encoder than ffaac (internal) [23:04] <JEEBsv> and do answer my question about what exactly will this encode be fore [23:04] <khali> JEEBsv: is there anything else than faac? I didn't know [23:05] <JEEBsv> libvo_aacenc [23:05] <JEEBsv> and libaacplus, but that is for HE-AAC only IIRC [23:05] <JEEBsv> libaacplus and faac are both non-distributable [23:05] <JEEBsv> libvo_aacenc is a'OK as long as you build ffmpeg with GPLv3 [23:05] <JEEBsv> :) [23:06] <khali> it's getting late, good nigh guys [23:06] <GuySoft> khali, JEEBsv i have a 1920x1080 mjpeg avi that takes up 2GB, and need to compress is somehow [23:07] <newl_> boy those guys at fsf legal sure have suceeded putting the fear of legal into people without even one lawsuit [23:07] <JEEBsv> GuySoft: which once again says nothing [23:08] <newl_> 2GB = 2hours? [23:08] <JEEBsv> is it for "end users" AKA "for people to watch on their PCs"? Or do you just want to compress that as much as possible losslessly? [23:08] <GuySoft> newl_, no its made up of mjpeg files from blender [23:09] <GuySoft> its around 800 frames [23:09] <JEEBsv> Also, I do want to remind you that you'd probably want to use ffvhuff instead of MJPEG [23:09] <JEEBsv> because MJPEG is not lossless the last I checked [23:09] <GuySoft> JEEBsv, with 2GB i need to be losy now [23:10] <JEEBsv> why can't you just say what exactly are you aiming at .-. [23:10] <GuySoft> JEEBsv, er because i was hoping technical info would help.. but sure [23:11] <GuySoft> JEEBsv, I have a blender animation, its 1920x1080 and takes up 2GB for 800 frames, and i need to sent it to the BBC [23:11] <JEEBsv> to be further processed there? [23:11] <GuySoft> JEEBsv, I don't know [23:12] <GuySoft> JEEBsv, I think it wont be, they didn't specify a format [23:12] <JEEBsv> so do they just want to take a look at it, or put it through their broadcast and thus re-encode it again? [23:12] <cbreak> x264 has a lossless mode [23:12] <GuySoft> JEEBsv, put it though a broadcast [23:13] <JEEBsv> ok [23:13] <cbreak> (or some very small loss settings) [23:13] <cbreak> expensive to encode though [23:13] <JEEBsv> so you want it to be as good as possible [23:13] <GuySoft> JEEBsv, yep [23:13] <JEEBsv> I'd just recommend you take that MJPEG then [23:13] <JEEBsv> and put it onto a USB stick [23:13] <JEEBsv> and take it there [23:13] <JEEBsv> A) there's a big chance their systems will be able to read it and B) you certainly don't want to make it become any more lossy [23:14] <GuySoft> JEEBsv, yes, but they wanted me to drop it in their dropbox.. now 2GB is a problem :) [23:14] <GuySoft> JEEBsv, hmm, i just ran ffmpeg -i /mnt/sdc1/hawking_bbc0001_0800.avi -s 1920x1080 final_video.mp4 and it does not look like it was that lossy [23:14] <GuySoft> its 20MB now [23:15] <cbreak> 2gb is a problem? [23:15] <cbreak> takes a few hours at most [23:15] <JEEBsv> GuySoft: it is... it is very lossy :< [23:15] Action: cbreak would not rescale [23:15] <JEEBsv> also, seriously... dropbox for master clips... [23:15] <JEEBsv> wtf [23:15] <JEEBsv> (IIRC it had lulzy limits file size wise) [23:16] <JEEBsv> of course, they might be paying for more space [23:16] <JEEBsv> in any case, it's only 2GB and it's already lossy [23:16] <GuySoft> JEEBsv, ah, didnt think of that [23:16] <JEEBsv> general thought in a case like this would be: leave it as it is [23:17] <JEEBsv> you've already raped it as MJPEG [23:17] <JEEBsv> (it's not lossless) [23:17] <cbreak> jpeg is quite lossy [23:17] <cbreak> especially if your source was RGB 4:4:4 [23:17] <GuySoft> JEEBsv, its also not well compressed [23:17] <JEEBsv> yes [23:17] <GuySoft> JEEBsv, my PC can hardly display it like that [23:18] <JEEBsv> well, I just wonder what exactly the BBC will be able to read from the lossless compression formats [23:18] <JEEBsv> of course if they have ffmpeg or something available there they'd be capable of then converting to whatever they want on their side [23:18] <JEEBsv> lossless H.264 with libx264 would most probably compress that thing nicely [23:19] <JEEBsv> but that isn't supported by anything else but libavcodec [23:19] <JEEBsv> (and CoreAVC) [23:19] <GuySoft> JEEBsv, I am not sure how to get blender to produce that kind of stuff [23:19] <cbreak> just tell them to get mplayer :) [23:19] <JEEBsv> cbreak: seems like it's going into airing [23:19] <JEEBsv> so it needs to be possible to get into their workflow [23:19] <cbreak> well... mplayer can pipe out yuv4mpeg [23:20] <JEEBsv> and ffmpeg can do a lot as well [23:20] <cbreak> and ffmpeg can pipe out anything [23:20] <JEEBsv> no need to bring mplayer out [23:20] <JEEBsv> that's what I meant [23:20] <JEEBsv> one tool needed only [23:20] <newl_> a bbc technician getting advice here ? [23:20] <JEEBsv> more like someone who was told to send his stuff up there [23:20] <GuySoft> newl_, no, i got approached by them [23:21] <JEEBsv> in any case, unless your internet really sucks [23:21] <JEEBsv> I'd probably stick to just uploading the 2GB [23:21] <JEEBsv> if you know they have some saner way of opening whatever you send there [23:22] <JEEBsv> then you'd have some options for extra compression [23:23] <JEEBsv> (basically if they don't have anything sane, the pretty much only sane'ish compressed formats would be exactly what you chose, which is, I think, MJPEG in AVI with WAV audio, or PNG in MOV for the quicktime users >_>) [23:26] <cbreak> quicktime supports motion jpeg [23:26] <cbreak> (3 versions even, last time I checked) [23:27] Action: cbreak would compress the audio at least with something like FLAC, if they can decode that [23:27] <JEEBsv> yeah, that's the problem [23:27] <cbreak> didn't BBC develop their own video codec? Something super wavelet based? Snow :) [23:27] <JEEBsv> we have no idea how they're equipped there [23:27] <JEEBsv> nah, snow was separate [23:27] <JEEBsv> BBC has Dirac [23:27] <cbreak> right [23:27] <JEEBsv> but even that has highly limited usage [23:28] <JEEBsv> anyways, gonna say that sending out that MJPEG is probably currently the best idea if it's meant to go to some pipeline [23:28] <JEEBsv> we have no idea what they support on the other side [23:28] <JEEBsv> BBC seems like a capable house [23:28] <JEEBsv> but derp [23:28] <JEEBsv> we have NFI if we're dealing with the capable people or the button-pushers [23:38] <newl_> JEEBsv: impossible for me to get any downloads from them [23:38] <newl_> their site is horrendous too [23:43] <EOF-sensei> JEEBsv: I wish dirac were worked on [23:45] <EOF-sensei> though Dirac is nice for going between filtering setups [23:45] <EOF-sensei> because it prefers correctness over perceived quality [23:46] <EOF-sensei> (right now at least) [23:46] <JEEBsv> is that... another weird way of saying 'it only has a lossless mode' [23:46] <EOF-sensei> hmm? [23:47] <EOF-sensei> it has lossy modes [23:47] <EOF-sensei> they're just more psnr than ssim [23:47] <EOF-sensei> which is probably not the best way to tune things for encoding consumer final render video [23:47] <JEEBsv> you really do make no sense [23:47] <JEEBsv> it's not the best way to tune ANY encoder [23:47] <JEEBsv> that is lossy [23:47] <JEEBsv> period [23:48] <JEEBsv> PSNR = blurriness galore [23:48] <EOF-sensei> it's the best way to tune a codec if you're going between mathematical operations that benefit from correctness and consistency more than perceived visual 'quality' [23:48] <JEEBsv> SSIM has been found to be a much more better metric generally [23:48] <JEEBsv> ok, you're making no sense so I'll just stop here [23:49] <EOF-sensei> ssim has been found to be a better metric for perceived quality [23:49] <JEEBsv> because if you think that PSNR is better than SSIM you're already having pants on your head [23:49] <EOF-sensei> if you think either is 'better' you're obviously in way over your head [23:49] <JEEBsv> they're just different ways of calculating "how close to original" [23:50] <EOF-sensei> psnr is the overall correctness [23:50] <JEEBsv> and PSNR has generally been shown to just love blurriness [23:50] <EOF-sensei> it's a measure of error for well-aligned pixels [23:50] <JEEBsv> which isn't gonna help with your mezzanine either [23:50] <EOF-sensei> ssim is a measure of Structural SIMilarity [23:51] <JEEBsv> I don't see any reason why I'd rather use a PSNR-optimized codec for a mezzanine than a SSIM optimized [23:51] <EOF-sensei> you're thinking that all processes are used for the same things [23:51] <JEEBsv> I just can't think why I'd rather have a blurred off picture [23:51] <EOF-sensei> bbc needed a lossless or slightly lossy codec for production [23:51] <EOF-sensei> and they work with some loss in their studios [23:51] <JEEBsv> VS a picture with more details in it [23:52] <EOF-sensei> it's better to have slightly blurrier images with better-aligned pixel values [23:52] <EOF-sensei> than one with misaligned pixels and sharper details that may or may not be there [23:53] <EOF-sensei> it's not PSNR versus SSIM [23:53] <EOF-sensei> it's a matter of situation [23:54] <JEEBsv> Yet encoder developers always cry at projects that have optimized for PSNR only and still do [23:54] <JEEBsv> I really don't see why exactly in the end a PSNR-optimized one would be better off [23:54] <JEEBsv> even for that limited use case [23:54] <EOF-sensei> for distributing massive amounts of video that looks to the human eye 'sharp' over ultra-cheap consumer lines in a country with poor infrastructure (i.e. the U.S.) you generally use a DCT-based codec with optimizations targeting general sharpness rather than correctness [23:55] <EOF-sensei> it's not about which one is universally better [23:55] <EOF-sensei> psnr is a really good metric for correctness of pixels [23:55] <EOF-sensei> ssim is a really good metric for perceived similarity when focusing on contrast [23:56] <EOF-sensei> it's not about favoring one or the other either [23:56] <EOF-sensei> because if you just optimized for ssim [23:56] <EOF-sensei> that's all you'll get [23:56] <EOF-sensei> higher ssim numbers [23:57] <EOF-sensei> Dirac doesn't optimize for PSNR [23:57] <EOF-sensei> it has good psnr by nature [23:57] <EOF-sensei> because of the nature of the discrete wavelet transform [23:57] <JEEBsv> glad to hear that they weren't completely bonkers :P [23:58] <EOF-sensei> they wouldn't be if they went after psnr [23:58] <EOF-sensei> if they optimized the hell out of psnr [23:58] <EOF-sensei> there's a chance their ssim would be better as well [23:58] <JEEBsv> AQ and similar things will lower SSIM, of course I have no idea how this applies to wavelets [23:59] <EOF-sensei> you've been reading too much of Dark_Shikari 's blog by the way [23:59] <EOF-sensei> you sound like a fucking parrot [23:59] <EOF-sensei> try exploring the world of the discrete wavelet transform [00:00] --- Tue Jan 17 2012

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