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February 2013

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ffmpeg-devel.log.20130207
by burek 08 Feb '13

08 Feb '13

[01:22] <cone-533> ffmpeg.git 03Michael Niedermayer 07release/1.0:8eda88868399: sanm: check image dimensions before using them [01:22] <cone-533> ffmpeg.git 03Xi Wang 07release/1.0:bc58fe0309a0: rtpenc: fix overflow checking in avc_mp4_find_startcode() [01:22] <cone-533> ffmpeg.git 03Xi Wang 07release/1.0:7df878ac1507: rtmp: fix multiple broken overflow checks [01:22] <cone-533> ffmpeg.git 03Xi Wang 07release/1.0:c3744b415a97: rtmp: fix buffer overflows in ff_amf_tag_contents() [01:22] <cone-533> ffmpeg.git 03Michael Niedermayer 07release/1.0:b666debffec1: huffyuvdec: Check init_vlc() return codes. [01:22] <cone-533> ffmpeg.git 03Michael Niedermayer 07release/1.0:db0f7f7394e1: huffyuvdec: Skip len==0 cases [01:22] <cone-533> ffmpeg.git 03Michael Niedermayer 07release/1.0:85a14dbd5dca: adpcm: fix off by 1 error and out of array access in DK4 [01:22] <cone-533> ffmpeg.git 03Michael Niedermayer 07release/1.0:6df0d3e2916c: alsdec: check block length [01:22] <cone-533> ffmpeg.git 03Michael Niedermayer 07release/1.0:2502914c5f8e: atrac3dec: Check coding mode against channels. [01:22] <cone-533> ffmpeg.git 03Michael Niedermayer 07release/1.0:3e196e4def03: error_concealment: Check that the picture is not in a half setup state. [01:22] <cone-533> ffmpeg.git 03Michael Niedermayer 07release/1.0:4cd1dad91ae9: avcodec_decode_audio: do not trust the channel layout, use the channel count. [01:22] <cone-533> ffmpeg.git 03Michael Niedermayer 07release/1.0:89e16e675d3c: sanm: Check decoded_size. [01:22] <cone-533> ffmpeg.git 03Michael Niedermayer 07release/1.0:08e2c7a45f82: vqavideo: check chunk sizes before reading chunks [01:22] <cone-533> ffmpeg.git 03Michael Niedermayer 07release/1.0:c459c7b23eff: aacdec: check channel count [01:22] <cone-533> ffmpeg.git 03Michael Niedermayer 07release/1.0:3f2715804019: h264: Only apply error concealment if theres a frame [01:26] <cone-533> ffmpeg.git 03Michael Niedermayer 07release/1.0:1690c1fa8860: update for 1.0.4 [02:13] <cone-533> ffmpeg.git 03Stefano Sabatini 07fatal: ambiguous argument 'refs/tags/n1.0.4': unknown revision or path not in the working tree. [02:13] <cone-533> Use '--' to separate paths from revisions [02:13] <cone-533> refs/tags/n1.0.4:HEAD: doc: replace '×' symbol with 'x' [03:55] <Compn> ubitux : did you see the opencl filter patch ? [03:55] <Compn> michaelni : any idea if that opencl patch is an acceptable api ? [09:43] <ubitux> <@Compn> ubitux : did you see the opencl filter patch ? // i did, but i'm still not available [10:39] <cone-93> ffmpeg.git 03Tim Walker 07master:08797c599df5: flac: don't check the number of channels before setting the channel layout [12:01] <cone-93> ffmpeg.git 03Michael Niedermayer 07master:199683c898c3: mov: remove unused function [12:44] <cone-93> ffmpeg.git 03Nicolas George 07master:3dea8b375ed5: lavfi: remove disabled code. [12:44] <cone-93> ffmpeg.git 03Nicolas George 07master:b5884db437ee: lavfi: fix missing space in error message. [13:06] <burek> michaelni, i think CVE ids are missing for 1.0.2 at http://ffmpeg.org/security.html [13:29] <cone-93> ffmpeg.git 03Diego Biurrun 07master:79dad2a93253: dsputil: Separate h264chroma [13:29] <cone-93> ffmpeg.git 03Michael Niedermayer 07master:c4e394e46008: Merge commit '79dad2a932534d1155079f937649e099f9e5cc27' [13:41] <cone-93> ffmpeg.git 03Diego Biurrun 07master:82bd04b170de: rv34: Drop now unnecessary dsputil dependencies [13:41] <cone-93> ffmpeg.git 03Martin Storsjö 07master:a846dccb29d2: h264chroma: x86: Fix building with yasm disabled [13:41] <cone-93> ffmpeg.git 03Michael Niedermayer 07master:60a0bc46cdba: Merge commit 'a846dccb29d2bb0798af1d47d06100eda9ca87cc' [13:44] <burek> Forum stats :) Total posts 992 Total topics 245 Total members 215 [13:44] <burek> we are approaching 1000th post :) [13:47] <cone-93> ffmpeg.git 03Tim Walker 07master:41244e13d545: flac: don't check the number of channels before setting the channel layout. [13:47] <cone-93> ffmpeg.git 03Tim Walker 07master:9d083d64172e: flac: add channel layout masks for streams with 7 or 8 channels. [13:47] <cone-93> ffmpeg.git 03Anton Khirnov 07master:ca1fe6c0e608: h263: remove an unused parameter from ff_h263_decode_init_vlc [13:47] <cone-93> ffmpeg.git 03Michael Niedermayer 07master:d5518545d1b6: Merge commit 'ca1fe6c0e60808da45d4dfd8728f45e843b9f9b0' [13:52] <durandal_1707> burek: where is list of stuff fflogger know about? [13:54] <burek> what do you mean [13:55] <burek> list of commands? [13:55] <durandal_1707> exactly [13:55] <burek> well !help should do [13:55] Action: durandal_1707 feels dumb [13:55] <burek> you probably blocked noticies [13:55] <durandal_1707> no, i ignored them [13:56] <burek> thats what i meant :) [13:56] <durandal_1707> i expect it to spam here [13:56] <burek> why [13:56] <burek> you need the help, not everyone else ^^ [13:57] <someone-noone> hello! I'm developing some format for ffmpeg. How should I specify fps value while demuxing? [14:00] <cone-93> ffmpeg.git 03Anton Khirnov 07master:9782c778a205: h264: remove silly macros [14:00] <cone-93> ffmpeg.git 03Anton Khirnov 07master:c5fcdb440237: error_resilience: remove a useless if() and FIXME [14:00] <cone-93> ffmpeg.git 03Michael Niedermayer 07master:7f23b2a9d0da: Merge commit 'c5fcdb440237f06f6c954185ab60970cabf786a2' [14:00] <cone-93> ffmpeg.git 03Michael Niedermayer 07master:b8d6108cfcb8: error_resilience: assert that unsupported B frame interpolation isnt used with h264 [14:00] <michaelni> someone-noone, if fps matches timebase setting the timebase alone should work fine see avpriv_set_pts_info [14:01] <Compn> someone-noone : what format are you working on? :) [14:02] <durandal_1707> funny but ffplay eats 1-3 last lines of image if height is not multiply of 4 [14:03] <Compn> is it sdl bug ? [14:04] <someone-noone> michaelni, actually I don't understand how avpriv_set_pts_info works(is it just settings time_base?). If I set directly time_base it doesn't work (values are 1 and 24000). If I put those values to avpriv_set_pts_info fps got some extraordinary values too. [14:04] <someone-noone> Compn, smth similar to mpegts but with some differences. [14:05] <Compn> michaelni : wow theres a ton of history to look for to find original author copyright in that h264 code :\ [14:05] <Compn> it all gets moved around like crazy [14:05] <Compn> http://git.videolan.org/?p=ffmpeg.git;a=commit;f=libavcodec/x86/dsputil_mmx… [14:13] <someone-noone> I do following: avpriv_set_pts_info(context->video_stream,32,1,24000); [14:13] <someone-noone> but ffmpeg -i testfile shows 3.17 fps [14:16] <michaelni> your timestamps or timebase maybe arent correct, also 24000 is probably not your framerate [14:17] <someone-noone> michaelni, I got those values(1 and 24000) while muxing. Pseudo code: put_be32(time_base.den); put_be32(time_base.num) [14:18] <someone-noone> and then passed this values to avpriv_set_pts_info, while demuxing [14:18] <someone-noone> the original file is in .mp4 format [14:18] <someone-noone> what am I doing wrong? [14:20] <someone-noone> also, original .mp4 file doesn't have r_frame_rate value. So,AFAIK, I should calculate it directly from time_base [14:20] <someone-noone> r_frame_rate = 1/time_base [14:24] <someone-noone> as well as: ffmpeg -i test.mp4 shows 23.94 fps [14:24] <cone-93> ffmpeg.git 03Anton Khirnov 07master:605b047bcc48: rv10: improve buffer size check. [14:24] <cone-93> ffmpeg.git 03Michael Niedermayer 07master:1c65393906ab: Merge commit '605b047bcc48482dc8a356f56629da259fbddbf1' [14:24] <someone-noone> sorry, not 23.94, but 23.98 [14:26] <someone-noone> Another question is, what is tbr, tbn and tbc values? If I remember right, those values are stored in codec->flags and codec->flags2 [14:27] <someone-noone> I thought that providing codec->extradata is enough for correct decoding any codec in ffmpeg [14:33] <cone-93> ffmpeg.git 03Anton Khirnov 07master:12b54a1f39fe: mpegvideo: remove an unused function parameter [14:33] <cone-93> ffmpeg.git 03Michael Niedermayer 07master:c7002e3d3dad: Merge commit '12b54a1f39fee22fa0399825ae47a43e60bad4c5' [14:34] <michaelni> someone-noone, the tbr, tbc, tbn values printed by the command line tool are unrelated to flag fields [14:36] <someone-noone> michaelni, thanks. Then what are they and should I specify them? [14:36] <someone-noone> And any advice about fps? What am I doing wrong? [14:36] <michaelni> someone-noone, i said it already, your values are wrong [14:37] <michaelni> either timebase or timestamps [14:37] <someone-noone> michaelni, timestamps you mean pts? [14:37] <michaelni> pts and dts [14:38] <michaelni> it could be other issues too but these are most likely [14:38] <someone-noone> hm, but I put time_base, pts and dts while muxing directly.... [14:48] <cone-93> ffmpeg.git 03Daniel Kang 07master:659d4ba5af5d: dsputil: x86: Convert h263 loop filter to yasm [14:48] <cone-93> ffmpeg.git 03Daniel Kang 07master:a1d36730342e: dsputil: x86: Fix compile error [14:48] <cone-93> ffmpeg.git 03Michael Niedermayer 07master:54d8322355c0: Merge remote-tracking branch 'qatar/master' [16:41] <cone-93> ffmpeg.git 03Michael Niedermayer 07master:48870853b2b7: x86/dsputil: Fix author attribution after code has been moved/splited around [16:41] <cone-93> ffmpeg.git 03Michael Niedermayer 07master:4e2c63685e03: sws: dont write out of array on bigendian [18:47] <Compn> Fix the value of TWO [18:47] <Compn> haha [19:48] <cone-93> ffmpeg.git 03Michael Niedermayer 07master:9f956611e33f: af_biquad: unroll loop, remove variable copies [19:48] <cone-93> ffmpeg.git 03Michael Niedermayer 07master:a00c4b4db044: af_biquad: avoid subtraction in loop [20:32] <cone-93> ffmpeg.git 03Stefano Sabatini 07master:1afbb4c22620: doc/filters: remove another instance of '×' [21:31] <cone-93> ffmpeg.git 03Stefano Sabatini 07master:872855e2a5ea: lavfi/select: avoid double -> int lossy conversion in debug log [21:31] <cone-93> ffmpeg.git 03Stefano Sabatini 07master:c0a4871c1687: lavfi/select: correctly set prev_t and prev_pts [21:55] <cone-93> ffmpeg.git 03Michael Niedermayer 07master:7c00d853a658: matroskadec: support seek to non keyframes [21:55] <cone-93> ffmpeg.git 03Michael Niedermayer 07master:a8e748eee506: lavf: add seek2any to allow forcing seeking to non keyframes via AVOptions [00:00] --- Fri Feb 8 2013

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ffmpeg.log.20130207
by burek 08 Feb '13

08 Feb '13

[00:30] Action: klaxa is listening to Louise (CV: Kugimiya Rie) - Gomen ne [07:08] <stephanedev> hi everyone. i wrote some code using ffmpeg C API to stream a video generated in real time over RTP, but on the decoding side, there are ~2s of latency. [07:09] <stephanedev> does anyone know how to get rid of that latency? [07:10] <stephanedev> is it due to my use of avformat_open_input which may scan the received stream to guess the format? [07:10] <stephanedev> avformat_open_input is called with the sdp file generated by the encoding side [07:11] <stephanedev> codec is vp8 [07:12] <stephanedev> i have found several articles related to the ffmpeg command line parameters but i am having a hard time finding anything about how to get rid of this latency when using the ffmpeg C API [07:18] <stephanedev> when streaming the output of the libvpx encoder over tcp and then decoding it, there is no latency [07:20] <stephanedev> from what i have read, i would think the latency occurs on the decoding side which buffers some input to guess the stream format. is it correct? [07:34] <gudy> Hi, I'm running ffprobe on a ts segment, and I'm not seeing the same number of video frames compared to video packets. In particular, the first packet of type video has the "K" flag, but the first video frame is not a key frame (and their pts differ). It seems that the first few video frames are not being considered. What could explain that ? [10:28] <Bor0> hello, I'm experiencing the same exact issue as http://stackoverflow.com/questions/14452817/strange-sound-produced-by-ffmpe…. could anyone give a hint about it? [10:51] <mxmx> who is alive? [10:52] <durandal_1707> noone [10:53] <mxmx> heh [10:59] Action: relaxed kills durandal_1707 [11:02] <mxmx> so i changed my mind, i rather add a png image to the video instead a subtitle [11:03] <mxmx> "movie=logo.png" <-- can logo.png be a path to the image? [11:03] <durandal_1707> yes [11:03] <mxmx> "movie=/path/to/image.png"? [11:04] <durandal_1707> yes [11:05] <mxmx> ok [11:32] Last message repeated 1 time(s). [11:32] <mxmx> what is wrong with this command [11:32] <mxmx> ffmpeg -i video.mp4 -vf "movie=/home/www-data/mysite.com/public_html/include/settings/logo.png [logo]; [in][logo] overlay=main_w:main_h [out]" -acodec libmp3lame -ar 22050 -ab 32 -f flv video.flv"; [11:46] <relaxed> hmm, the bot is gone. [11:46] <relaxed> mxmx: paste your command and all output to pastie.org [11:51] <mxmx> ok sec [11:56] <mxmx> http://pastie.org/6086277 [11:58] <relaxed> that's impossible to read, try again [11:58] <mxmx> i am using putty and thats how it displays it [11:58] <mxmx> sec [11:59] <mxmx> yeah its stupid putty [12:00] <mxmx> these are teh error messges tough [12:00] <relaxed> Then `command` 2>ffmpeg.log (and upload to dropbox or something. [12:00] <mxmx> http://pastie.org/6086299 [13:12] <theos> so i finally compiled x264 and everything else :D [13:13] <theos> now i have to learn how to use it properly for max compression [13:15] <JEEB> 1) encode 2500-5000 frame clips with crf values starting from 23, and going up if it still looks good enough for you, and down if it looks bad for you - until you hit the magical "highest value that still looks good for you" crf value [13:15] <JEEB> 2) pick the slowest -preset you can still manage time-wise [13:15] <JEEB> 3) ??? [13:15] <JEEB> 4) profit [13:15] <JEEB> yes, that's it [13:17] <theos> hi JEEB :) [13:18] <JEEB> and yes, the outlook of the same crf value will be somewhat different between settings (presets), but the tweaking you might or might not need should be relatively small [13:20] <JEEB> and for the ~1-2 minutes of input you'll probably want to learn about -ss and -t , the first one lets you seek and then -t limits how long will be encoded [13:20] <theos> is msmpegv2 better? [13:21] <JEEB> what? [13:21] <theos> i assume h264 is vcodec? [13:21] <JEEB> H.264 is a format, called h264 in the libavcodec/ffmpeg limbo. libx264 is an encoder for it [13:21] <theos> or should i rtfm? [13:21] <theos> oh [13:22] <JEEB> and no, there are no better encoders than libx264 for lossy, and at times nowadays even for lossless encoding [13:22] <theos> wont ffmpeg use x264 by default [13:23] <JEEB> to be honest, no idea -- they have now some defaults depending on the type of container you start outputting to, but I've never trusted ffmpeg's automatic selection of "what you probably want" [13:23] <theos> says Stream #0:0 -> #0:0 (mpeg4 -> mpeg4) Stream #0:1 -> #0:1 (mp3 -> libmp3lame) [13:24] <JEEB> so the mpeg4 (MPEG-4 Visual, what xvid and divx "conformed" to) encoder got used and audio got re-encoded with lame into mp3 [13:25] <JEEB> sounds like you tried to output into the avi container [13:25] <JEEB> at least defaults-wise [13:25] <theos> yes i tried "ffmpeg -i input.avi output.avi" :D [13:25] <JEEB> yes, don't do that, in the worst case you will get mpeg-1 at 200kbps rubbing its balls against your face [13:26] <theos> so what should i use? [13:26] <JEEB> I would prefer you to move to some container that can handle b-frames properly, and then just set video and audio formats to whatever you need [13:27] <JEEB> ffmpeg -i input.derp -c:v libx264 -crf 23 -preset medium -c:a copy out.mkv [13:27] <JEEB> now take a look at the first four lines I listed [13:28] <theos> increase crf and decrease 1 when its not looking good? [13:28] <theos> ah its called libx264 :/ i was using h264 and it was giving error [13:28] <JEEB> also use -ss to seek to some timestamp that has content in it, and -t to then limit it to a couple of minutes so you can actually do some testing [13:29] <JEEB> h264 is the name for the internal decoder methinks, libx264 is the library that handles H.264 encoding, which you can build ffmpeg with [13:31] <theos> can the output be avi too? [13:31] <JEEB> in theory yes, but I will bludgeon you to hell for it. AVI does not take b-frames in well, and that was found out with MPEG-4 Visual already. But at the time we had no alternatives, really -- so people pretty much got used to hacking b-frames into AVI [13:31] <JEEB> let's not continue that shit in the 21st century, thank you [13:32] <theos> haha ok. so mkv is a good option? [13:32] <JEEB> use mkv, mp4 or flv -- whatever floats yer boat. As long as it's not AVI [13:32] <theos> thanks a lot :) [13:32] <JEEB> that said, both mp4 and flv tend to limit audio formats and don't support most commonly used subtitle track formats, so mkv is the current "supermarket of containers" [13:33] <theos> so mkv it is! [13:35] <DEATH> 9 out of 10 pirates use MKV [13:35] <DEATH> Even hardware makers acknowledged it [13:35] Action: theos hides [13:35] <DEATH> Only Apple and Microsoft continue to ignore it with extreme prejudice [13:36] <theos> i dont use either of those ^^ [13:36] <DEATH> Samsung likes MKV [13:36] <burek> can ffmpeg use something like this: ffmpeg -f image2 -i image%03d.tiff -c copy output.WHAT ? [13:36] <theos> i have a samsung android phone if that counts :) [13:37] <burek> meaning is there any video format that stores tiff images in the video, just like jpegs can be saved in mjpeg [13:37] <JEEB> burek, I would guess that's possible, but I have no idea what container takes in tiff [13:37] <JEEB> try mkv? [13:37] <JEEB> lol [13:37] <burek> :) [13:37] <burek> ok, ill just re-encode :) its safer :) [13:37] <JEEB> yeah [13:37] <theos> does h264 have an audio codec? [13:37] <JEEB> it's a video format [13:37] <theos> :( [13:38] <theos> should i use mp3 for audio? [13:38] <JEEB> MPEG-4 family does have an audio format, but to be honest if your input is mp3 you might as well keep it that way [13:38] <JEEB> aka -c:a copy [13:38] <JEEB> aka "just copy the audio from the input" [13:38] <theos> yes i understand copy. thanks :) [13:39] <theos> its 160kbps though. 128kbps is enough for me. but i dont think it will have much difference on the output [13:39] <JEEB> yes, that kind of difference is miniscule compared to the video bit rate [13:39] <JEEB> and it will not make an already highly compressed audio clip any better [13:40] <burek> try aac+ to get as low as 32 kbps ^^ [13:40] <theos> is it lossless? [13:40] <burek> right.. [13:40] <burek> :D [13:40] <theos> :) [13:42] <JEEB> in general I tend to just copy audio tracks over if they're not grossly oversized compared to what they are -- as in, lossy tracks in pretty much all cases will just get copied over, if it's something lossless (PCM from DVDs or BDs, or TrueHD etc. from the BDs) I encode it with whatever I deem good enough for that job [aotuv's vorbis, fdk-aac's AAC, some lossless format possibly?] [13:42] <JEEB> also, depends highly on wtf the stuff will get played on [13:43] <DEATH> Good luck getting "hardware players" to recognize Vorbis or FLAC, heh [13:43] <JEEB> surprisingly both are supported in some things [13:43] <JEEB> but do note that I don't even know what the person is encoding for [13:43] <JEEB> and neither do you [13:46] <DEATH> Yeah [13:46] <DEATH> PC playback with MPC-HC = encode any format you like [13:46] <DEATH> MPC-HC or other good software player [13:46] <DEATH> I'm having great fun with Samsung TV vs "acquired" encodes [13:47] <DEATH> Some H.264 configs just won't decode [13:50] <someone-noone> hello! What is syntax for <rational> values? [13:51] <theos> i use my pc for every multimedia purpose [13:55] <theos> why is the output file longer than the input file? :) conversion isnt complete. maybe it will get fixed [16:55] <Diogo> hi i having problems using ffmpeg and threads.. [16:55] <Diogo> ffmpeg -threads 1 (guarantees that only one thread is running)? [16:56] <Diogo> i'm having one more than one thread [17:05] <Diogo> i'm doing something wrong? [17:33] <kittonian_> morning all [17:35] <kittonian> i am trying to figure out the command line to turn an MKV into an M4V for AppleTV3 while converting DTS surround sound to AC3 surround sound at a bitrate 640 [17:35] <kittonian> i've been using Handbrake and most of the time it works just fine, but there are some files that Handbrake will not work with for some reason [17:46] <kittonian> will this accomplish what I'm searching for? ffpmeg -i input.mkv -c:v libx264 -crf 18 -acodec ac3 -ac 6 -ab 640k output.m4v [17:48] <Mavrik> if you're trying to re-encode video and audio to H.264/ac3 then yes [17:48] <Mavrik> :) [17:48] <Mavrik> why are you reencoding video though? [17:49] <kittonian> trying to convert an MKV with DTS surround sound into an M4V for Apple TV 3 [17:49] <Mavrik> ah [17:49] <Mavrik> there's probably no need to reencode video [17:49] <Mavrik> (afaik AppleTV can handle pretty much any H.264) [17:49] <kittonian> i've been using Handbrake to do it but on some MKV files it doesn't output correctly [17:50] <Mavrik> so replace "-c:v libx264 -crf 18" with "-codec:v copy" [17:51] <kittonian> will that make sure the video quality isn't touched? [17:51] <kittonian> that's my goal [17:51] <kittonian> i'm trying to have the highest quality video and audio possible in an ATV3 compatible file [18:16] <frnk1> does anyone here knows a codec to make ffmpeg _demodulate_ a wav file containing the recorded sound of e.g. a modem connection/fax/datasette [18:17] <frnk1> i'm not that familiar in whistling modem sounds like the terminatrix in T3, but there might only be some parameters, and then some bytestream, txt-file, should result [18:18] <kittonian> Mavrik: hmm, got an error using -codec:v copy but my string seems to be working [18:19] <Mavrik> "an error" [18:19] <Mavrik> ? [18:19] <kittonian> Could not write header for output file #0 (incorrect codec parameters ?): Operation not permitted [18:20] <Mavrik> the actual error is probably written a line or few higher [18:20] <kittonian> Stream #0:0: Video: vc1 (WVC1 / 0x31435657), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 23.98 fps, 90k tbn, 48003.07 tbc (default) [18:20] <kittonian> Stream #0:1(eng): Audio: ac3 (ac-3 / 0x332D6361), 48000 Hz, 5.1, fltp, 640 kb/s (default) [18:20] <kittonian> Stream mapping: [18:20] <kittonian> Stream #0:0 -> #0:0 (copy) [18:20] <kittonian> Stream #0:1 -> #0:1 (dca -> ac3) [18:28] <gudy> Hi, I'm running ffprobe on a ts segment, and I'm not seeing the same number of video frames compared to video packets. In particular, the first packet of type video has the "K" flag, but the first video frame is not a key frame (and their pts differ). It seems that the first few video frames are not being considered. What could explain that ? [18:39] <Mavrik> gudy: it's quite possible the decoder drops input frames until it gets an I-frame [18:40] <Mavrik> are you looking at frames (After decode) or packets (before decode)? [18:52] <casimiro> Hi all, I am trying to broadcast through UDP an mpeg-ts received from TCP. For what I have tested, I need to parse the incoming stream in order to deliver the mpeg packets correctely. I have tried to implement the parse by my self, but it's getting too complicated. I would like to know if anybody could give me some lead in how to use libav to do such parse. [18:53] <Mavrik> casimiro: do you have to do anything else besides pass-through of mpeg-ts? [18:54] <casimiro> No [18:54] <Mavrik> well, you could probably just do [18:54] <Mavrik> ffmpeg -i <your tcp input> -codec copy -f mpegts udp://<output-ip>:<output-port> [18:56] <casimiro> Thank you Mavrik! But actually I am developing a program to do that. Do you know if I could have something similar with some lib ? [18:56] <Mavrik> yeah, you can use libavformat calls to do that [18:56] <Mavrik> casimiro: grab ffmpeg source and check doc/examples/demuxing_muxing.c [18:57] <Mavrik> that's essentialy what you have to do [18:57] <casimiro> Thank you so much for your help! [19:02] <gudy> Mavrik: both [19:03] <gudy> Mavrik: so the first video packet has the "K" flag, but will not show up as a video frame [19:05] <gudy> here are the frames : http://pastebin.com/bBvi7q3X [19:05] <gudy> and the packets : http://pastebin.com/pt1BBAVa [19:24] <kittonian> is there a difference between -codec:v copy and -vcodec copy? [19:27] <JEEB> kittonian, -c:v and -codec:v are a newer way of setting it [19:41] <kittonian> JEEB: cool thanks [19:41] <kittonian> i'm trying to figure out why when i set -crf 18 it still shows q=23 during encoding [19:41] <JEEB> probably unrelated [19:42] <JEEB> as long as libx264 outputs crf=18 , all's fine [19:42] <kittonian> i read through the docs and it said to use -qp instead of -crf [19:42] <JEEB> -qp should never be used by end users [19:42] <kittonian> oh ok [19:42] <JEEB> or well, it can be used with other encoders than libx264 [19:42] <JEEB> it's constant quant [19:42] <JEEB> which is what that q probably shows [19:42] <JEEB> libx264 on the other hand [19:42] <JEEB> has something better [19:42] <JEEB> crf [19:42] <kittonian> gotcha [19:43] <JEEB> before you basically either had 2pass abr or 1pass constant quant [19:43] <JEEB> now you have 1pass crf which takes the better sides of both [19:43] <JEEB> one pass, but still distributes bits better than just setting a constant quantizer \o/ [19:44] <JEEB> thus being the closest to "constant quality" we've ever been [19:44] <kittonian> makes sense [19:44] <kittonian> i'm trying to convert an MKV with DTS surround to an M4V for Apple TV3 but retain the highest possible video and audio quality. [19:44] <kittonian> right now i'm using | ffmpeg -i input.mkv -c:v libx264 -crf 18 -acodec ac3 -ac 6 -ab 640k output.m4v [19:45] <JEEB> if the input is something that the apple tv3 can play back as-is without re-encoding, then that'd be the best way [19:45] <JEEB> but that would depend on what exactly is in there :) [19:46] <JEEB> anyways, I'd recommend going over with -c:v copy once [19:46] <JEEB> (and audio being re-encoded) [19:46] <kittonian> i tried -codec:v copy and it returned an error and wouldn't work [19:46] <JEEB> k [19:46] <kittonian> i would love to not have to transcode the video. that was my goal [19:47] <kittonian> and it's already in h.264 format [19:47] <JEEB> yeah, the problem becoming in what profile/level the apple tv3 likes its stuff [19:47] <kittonian> ffmpeg threw the error - Could not write header for output file #0 (incorrect codec parameters ?): Operation not permitted [19:47] <JEEB> oh [19:47] <JEEB> would probably want to pastebin the whole log there [19:50] <kittonian> http://pastebin.com/wmvZiAd0 [19:51] <JEEB> [ipod @ 0x7feac2843e00] track 0: could not find tag, codec not currently supported in container [19:51] <JEEB> Stream #0:0: Video: vc1 (Advanced) (WVC1 / 0x31435657) [19:51] <JEEB> so yeah, that's VC-1 [19:52] <kittonian> i'm not familiar with VC-1 [19:52] <JEEB> which actually /does/ have mappings for "mp4"-style containers but I'd still be surprised if your Apple thingy could play it :) [19:52] <kittonian> i assume that's a different codec than H.264 [19:52] <JEEB> it's a format from Microsoft that tried to be royalty-less for a moment and got onto both HD-DVD and Blu-ray specs [19:52] <kittonian> ahh [19:52] <JEEB> ...and then promptly got itself dragged into a patent pool of MPEG-LA's [19:53] <JEEB> Warner Bros IIRC have a long history regarding using VC-1 with their discs [19:53] <JEEB> basically VC-1 is Windows Media Video 3 on steroids [19:54] <kittonian> hmm, interesting. alright, well, do you see anything I should be doing to improve video and/or audio quality (as well as speed of transcoding) in my command line? [19:54] <kittonian> oh, and i want to ensure that it doesn't screw with the picture at all (i.e. exact same pixel dimensions, no cropping, etc.) [19:54] <kittonian> in Handbrake i was doing it strict [19:55] <JEEB> ffmpeg doesn't do that by default, unlike many other things it might poke if you don't set them (such as codecs f.ex.) [19:55] <JEEB> crf keeps the quality more or less constant with those settings so quality wise as long as you're around the highest crf value that still looks good to you, you should be fine [19:55] <JEEB> 23 is the default and if it looks good you go higher, and if it looks bad you go lower [19:56] <kittonian> from what i've read 18 is visually lossless [19:56] <JEEB> depends on who is looking at it and what the resolution is etc. [19:56] <kittonian> this is all 1080p movie stuff [19:56] <JEEB> then i'd say you're on the very low end of the line [19:57] <JEEB> as in, you could raise it gratuitiously by a few before you would get anything else but a free minification [19:57] <JEEB> also, regarding compression/speed settings x264 has you covered: http://mewiki.project357.com/wiki/X264_Settings#preset [19:57] <kittonian> ok, so i'm more than safe [19:57] <JEEB> you can set these with -preset [19:57] <JEEB> yes, you're safe but you might be wasting a nice amount of bits [19:58] <kittonian> and do i no longer need the -f mp4 directive? [19:58] <kittonian> i've seen that in a couple of examples but i'm not using it [19:58] <JEEB> if your output file name includes mp4 or m4v, nope [19:59] <JEEB> ffmpeg looks it automagically [19:59] <kittonian> great. that's handy [19:59] <kittonian> the only other thing had to do with the -profile option [19:59] <JEEB> yeah, profile/level (and making sure not too many refs get used) [19:59] <kittonian> i looked at atv3's specs and it says high or main with high quality audio is supported [20:00] <kittonian> but apple's specs don't even mention that it supports AC3 surround at 640k [20:00] <kittonian> and it does just fine [20:00] <JEEB> looking at every apple device since the 3GS (not looking at the apple tvs), they've all actually supported high profile, level 4.1 content [20:00] <kittonian> and is that the default? [20:00] <JEEB> just that itunes usually barfed at it [20:01] <JEEB> libx264's default is high profile, yes -- and the level is set by your preset's settings [20:01] <kittonian> since i didn't use a preset, should i be setting the level directly? [20:01] <JEEB> basically profile = feature set, and level = how much memory the decoder needs to decode, reference frame count [20:01] <JEEB> uhh [20:01] <JEEB> default is preset=medium [20:01] <JEEB> then you can set it to be either faster or slower [20:01] <kittonian> what's the better way to do it? [20:01] <JEEB> ? [20:02] <kittonian> for better looking/sounding output [20:02] <JEEB> huh [20:02] <kittonian> or does that not have anything to do with it [20:02] <kittonian> sorry, trying to wrap my head around all of this [20:03] <kittonian> the reason i asked is that i read that slower yields better results [20:03] <kittonian> but if this literally just has to do with computing power than i guess it doesn't matter [20:03] <JEEB> 1) find your highest, still looking fine crf value 2) pick the slowest preset that is still fast enough for you 3) look at the limitations of your playback device/application and set profile/level accordingly 4) since libx264 doesn't limit the amount of reference frames used to the level's, set it if needed [20:05] <kittonian> i guess the reference frames is what's throwing me off here. not sure what that should be [20:05] <JEEB> 1) sets the visual look 2) sets the compression ratio and encoding speed 3) limits 2's feature set 4) makes you limit the amount of reference frames that can be used to the levels [20:05] <JEEB> reference frames are the amount of frames a frame can reference for predictive encoding, more means better compression [20:05] <JEEB> also more means more memory usage [20:06] <JEEB> for whatever reason libx264 itself doesn't limit itself if you set a level [20:06] <JEEB> (the command line x264 encoder actually does) [20:06] <JEEB> so you have to set the maximum amount of reference frames if you want to actually adhere to the level [20:07] <kittonian> so medium is a good compromise between everything? [20:07] <JEEB> it's a default, offers good enough stuff in general while not being overtly slow [20:07] <JEEB> then the user can easily set it to something either faster or slower while getting more or losing some compression [20:08] <kittonian> isn't less compression always the better quality output? [20:08] <JEEB> no? [20:08] <JEEB> better compression means less size for pretty much the same output [20:09] <kittonian> ahh [20:09] <kittonian> that's cool [20:09] <kittonian> i'm from the audio world and you always want as little compression as possible to retain the highest quality [20:09] <kittonian> just getting into the video side of things [20:09] <JEEB> no, you don't want as little compression as possible, you want the least loss of information [20:10] <JEEB> and that's the same with video, basically. Least loss of information while still compressing things [20:10] <kittonian> ok, so what would your perfect world scenario command line be for Blu-Ray quality outputs? [20:11] <kittonian> without a concern for disk space as I have huge drives [20:11] <JEEB> well since crf sets more or less the quality level, if you have no concern towards the quality level, you might just as well set some faster preset then [20:12] <JEEB> (the quality given by a certain crf value differs between settings because the algorithms that lead to the certain rate factor start differing) [20:12] <JEEB> (but of course the difference usually isn't too big) [20:13] <kittonian> hmm, ok [20:13] <JEEB> the idea really isn't harder than 1) find your highest crf value that still looks good 2) find the preset of your choice regarding the compression/speed ratio 3) check what limitations whatever you're encoding for has 4) set profile/level/x264opts ref=NUMBER as needed per the limitations [20:14] <kittonian> i noticed one example i found referenced -threads 0 as the way to use all the computing power possible and greatly speed up encoding times [20:14] <JEEB> that is the default now [20:14] <kittonian> oh, great [20:14] <JEEB> and it just sets threads to 'auto' in x264 [20:14] <kittonian> though from my pastebin it seems -threads=6 is what is set [20:14] <JEEB> which is 1.5 the amount of virtual cores [20:15] <JEEB> anyways, select crf value and preset as you like (you can encode 2500-5000 frame sequences of actual content to test) [20:16] <JEEB> -ss seeks and -t sets the amount of time to encode [20:16] <JEEB> encode a minute or two for testing a few times and you should find out [20:16] <kittonian> ok [20:16] <kittonian> right now i'm only getting 6FPS on a 4-core mac pro [20:16] <JEEB> then if your thing can handle high profile/level 4.1 you don't really have to set profile as high profile is the default, and level 4.1 is -level 41 [20:17] <kittonian> i think i read that ATV3 can only handle level 4 [20:17] <kittonian> 4.0 [20:17] <JEEB> then -level 4 for that [20:17] <JEEB> and you'll have to limit reference frames to that and the resolution you're using... lemme go look up the numbers [20:19] <JEEB> ok, so level 4 has a MaxDpbMbs of 32768 [20:19] <JEEB> macroblock width and height of 1920x1080 is 120x68 [20:19] <JEEB> thus the amount of macroblocks in a single frame is 8160 [20:20] <JEEB> 32768 / 8160 = 4.something, thus 4 [20:20] <JEEB> so -x264opts ref=4 [20:20] <JEEB> would limit the encoder to four reference frames at a time [20:21] <JEEB> I've been thinking of implementing the automatic limitation of reference frames to the level you select into the libavcodec's libx264 wrapper [20:21] <JEEB> so it does the same as x264's cli encoder [20:23] <kittonian> this is interesting [20:24] <kittonian> just read that even though apple doesn't officially support high profile 4.1 people are playing back without issue [20:24] <kittonian> on both atv2 and atv3 [20:24] <kittonian> but not atv1 [20:25] <JEEB> yes, the first one was atrocious [20:25] <JEEB> it only played baseline profile [20:25] <JEEB> anyways, 4.1 has the same MaxDpbMbs limit [20:28] <kittonian> ok, well i guess i'll see whether or not it'll work when it's finished [20:28] <kittonian> if it doesn't and i need to do it again with different settings for high profile 4, is this the command? ffmpeg -i input.mkv -c:v libx264 -crf 18 -level 4 -x264opts ref=4 -acodec ac3 -ac 6 -ab 640k output.m4v [20:28] <JEEB> yes [20:29] <JEEB> also do note that itunes has its own limitations so if you have to go through it, that's a whole separate can of worms [20:29] <kittonian> i have to import the movie to the itunes library but itunes doesn't play back any of the movies properly. only atv3 plays them fine [20:30] <JEEB> depending on how nazi itunes is about that kind of stuff nowadays (it used to outright block >main profile stuff with the 3GS f.ex., while 3GS and newer phones could play high profile level 4.1 stuff JustFine) [20:30] <kittonian> i'm really hoping the jailbreak comes soon so i can just play the MKV files directly [20:30] <JEEB> you might be fine or might have to use something else to transfer the files over [20:31] <kittonian> you can't use anything else on a non-jailbroken atv. you have to use itunes on a computer and then connect that computer to the atv over ethernet sharing (wi-fi is very choppy) [20:32] <kittonian> i've been considering building a standalone HTPC with the latest XBMC installed to replace atv3 but i can't get netflix unless i run windows and use some crappy add-on [20:33] <kittonian> i really appreciate all the help. thank you [20:40] <gudy> when I run ffprobe on .ts segment with h264 content in it, -show_frames and -show_packets do not see the same number of video frames / packets. What could explain that ? [20:47] <saste> gudy: one packet can contain several audio frames [20:48] <saste> for video it should be the same number [20:48] <saste> unless you have problems decoding them [20:48] <saste> in that case you have frames < packets [20:49] <Hammerhead2011-S> Hi All, I have a rtsp stream coming into VLC and transcoded to rtp(rtsp) out on port "xxxx"/blah.mov [20:49] <gudy> saste: thanks, how can I find out what is wrong with a video frame ? [20:49] <Hammerhead2011-S> totem in Linux plays the stream no problem. Windows(Quicktime player) plays it for about 1 minute then stops. no error just stops. [20:50] <saste> gudy, check the log [20:50] <Hammerhead2011-S> I'm trying to keep the stream size smaller...encoding to FLV triples the size of the stream... [20:50] <gudy> saste: things like this : [h264 @ 0x100bc5280]no picture ? [20:50] <gudy> (that's from mplayer) [20:50] <saste> gudy, yes [20:51] <gudy> is there a way to get more info on what went wrong ? [20:51] <Hammerhead2011-S> transcoded using h264 BTW [20:51] <saste> gudy, -loglevel [20:51] <gudy> with mplayer, of ffprobe ? [20:51] <saste> ffprobe [20:51] <Hammerhead2011-S> The feed is from a video camera....My question....is there an easier way to do this? [20:54] <gudy> saste: "./ffprobe -loglevel 9 -analyzeduration 10000000 -i input.ts" doesn't output anything ... [20:56] <saste> gudy, asking on irc is not a substitute for the fine manual [20:56] <saste> you're not decoding, unless you select -show_frames [20:57] <gudy> ok, thanks [21:06] <gudy> saste: thanks, that helped. I'm now down to "[h264 @ 0x7fd6fb000000] mmco: unref short failure" which I'm trying to find out what it could be [21:06] <saste> gudy, bingo [21:07] <saste> corrupted stream [21:08] <gudy> mmmm [21:09] <gudy> this is an HLS stream [21:09] <gudy> the first problem is that there is a massive audio/video sync problem (segments are 10 seconds long, and audio/video frames are 3.5 seconds appart) [21:10] <gudy> and then, most segments will lack a key frame at the beginning, however, the key frame seems to be there, but simply invalid ... [21:13] <saste> gudy, how are you creating the segments? [21:16] <gudy> I am not creating the segments [21:16] <gudy> I am getting them from a partnet [21:16] <gudy> and my player doesn't like them too much [21:16] <saste> gudy, which player? [21:16] <gudy> the default android hls player [21:17] <saste> did you try with an I* device? [21:17] <gudy> on android 4.0 [21:17] <saste> also i'm curious, which android player? [21:17] <gudy> well, the player will play [21:18] <saste> also you should ask your partner to validate the HLS stream using the apple (proprietary) tools [21:18] <saste> most segments will lack a key frame at the beginning, however, the key frame seems to be there, but simply invalid ... < what do you exactly mean? [21:18] <gudy> here's the output with both packets and frames enabled [21:18] <gudy> http://pastebin.com/D7mgwe1j [21:19] <gudy> I mean that there is a video packet which is supposed to contain a video key frame [21:20] <gudy> if you look at the log I've pasted, first "FRAME" is after a bunch of video packets [21:21] <saste> gudy, a decoding problem [21:26] <gudy> how likely is it a bug in the stream vs a bug in ffmpeg's h264 decoder ? [21:29] <saste> gudy, H.264 decoder is very well tested [21:30] <saste> seems more likely a bug in the stream/encoder [22:07] <stephanedev> hi everybody. do you have any hint on how to reduce latency when using ffmpeg C API? [22:08] <stephanedev> i am talking about latency when streaming using rtp [22:09] <stephanedev> is it due to the encoder, the decoder, or both? [22:10] <stephanedev> from what I have read, i think there may be some option to pass to the encoder so that it does not buffer frames [22:26] <boot> hello, is there another option to record the desktop and games besides x11grab? [22:29] <stephanedev> boot: why? is there anything wrong with x11grab? [22:31] <boot> I'm using KDE and have an ATI with proprietary divers, they don't play well together. So by default KDE disable direct rendering mode. If I have a game in fullscreen ffmpeg only records a black screen... [22:34] <stephanedev> boot: on linux, x11grab is the only option i know of to grab the desktop. but i have very little experience with ffmpeg command-line tool so my answer is not authoritative at all [22:38] <saste> boot, dshow, vfw on windows [22:38] <saste> ah kde [22:38] <saste> no what's wrong with x11grab? [22:38] <boot> explained above [22:39] <saste> i see [22:39] <boot> i'm reading it might be a codec issue, i'll try a different one [22:39] <saste> i think it's related to x11 [22:39] <saste> so nothing related to codec [22:49] <stephanedev> when using avcodec_open2, is there some generic option (i.e. not codec dependent) i can pass in the dictionary that will prevent the encoder from buffering frames? [22:50] <stephanedev> something along ("tune", "zerolatency")? [22:50] <saste> stephanedev, don't think so [22:50] <saste> you need to find a specific solution for each encoder [22:51] <stephanedev> saste: thank you for your answer. does it mean it is impossible to create a rtp video streamer without latency which would be codec-agnostic? [22:52] <saste> stephanedev, in principle, no [22:52] <saste> in practice, yes [22:52] <saste> since you're not going to support all the possible codecs but just a subset [22:52] <saste> so you need to find meaningful options for each different codec [22:53] <stephanedev> yes that's how i understood your "in practice answer" [22:55] <stephanedev> is there any document (other than the source code) describing codec options that can be passed in the dictionary? [22:57] <saste> stephanedev, ffmpeg-codecs(1) [22:57] <saste> and you can do ffmpeg -h fullhelp to get a short description of all the private options [22:59] <stephanedev> saste: thank you again for your help. is ffmpeg-codecs a man page? [22:59] <saste> stephanedev, check also the docs on the website [23:02] <stephanedev> i have been looking at the ffmpeg doxygen doc and manual but maybe i missed it [23:02] <stephanedev> i'll check the output of ffmpeg -h full [23:12] <stephanedev> saste: thank you very much. i found the document providing all the available options so i will try to play with them to hopefully get rid of this latency [23:12] <saste> http://ffmpeg.org/ffmpeg-codecs.html [23:12] <saste> mostly incomplete, but better than nothing [23:12] <saste> (and patches are welcome) [23:12] <stephanedev> yes that's the one i am looking at :) [23:52] <Keshl> Blah. How do I find a video's framerate with ffmpeg? x.x [23:52] <Keshl> I feel like I'm mis-reading the output of "ffmpeg -i video.flv" -- Is "25 tbr" what I'm looking for? [23:55] <stephanedev> Keshl: what about ffprobe <your_video_file> and then looking at the fps value for each stream? [23:58] <Keshl> stephanedev: ... <.< So now I know ffprobe exists. [23:58] <Keshl> stephanedev: It's not saying it. [23:59] <Keshl> Same output as ffmpeg, still no useful information. xwx [23:59] <Keshl> I'm expecting either 15, 24 or 30 and I'm not seeing those numbers anywhere. [23:59] <stephanedev> can you paste the command output to pastebin.com? [00:00] --- Fri Feb 8 2013

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ffmpeg-devel.log.20130206
by burek 07 Feb '13

07 Feb '13

[00:21] <cone-290> ffmpeg.git 03Michael Niedermayer 07release/1.0:0c2a350762c3: movtextenc: fix pointer messup and out of array accesses [00:21] <cone-290> ffmpeg.git 03Michael Niedermayer 07release/1.0:21185028b4fb: dvenc: dont fail hard if the timecode is invalid [00:21] <cone-290> ffmpeg.git 03Michael Niedermayer 07release/1.1:a55c274f519f: movtextenc: fix pointer messup and out of array accesses [00:21] <cone-290> ffmpeg.git 03Michael Niedermayer 07release/1.1:f6b50924a5fd: dvenc: dont fail hard if the timecode is invalid [00:21] <cone-290> ffmpeg.git 03Michael Niedermayer 07release/1.1:5a97a5291a83: riff: fix infinite loop [00:21] <cone-290> ffmpeg.git 03Michael Niedermayer 07release/1.1:d2c1a8dc2de8: ljpegenc: allocate needed scratch-buffer [00:24] <cone-290> ffmpeg.git 03Stefano Sabatini 07master:58c4b0f99082: ffplay: extend feedback in case of video changes [00:24] <cone-290> ffmpeg.git 03Stefano Sabatini 07master:212bb6490fa6: ffplay: report video frame changes only in debug mode [00:37] <ubitux> llogan: awesome :) [00:40] <cone-290> ffmpeg.git 03Stefano Sabatini 07master:a12787960e47: lavf/segment: extend PTS debugging log with -reset_timestamps [01:31] <cone-290> ffmpeg.git 03Michael Niedermayer 07master:ccf9dd00da05: ffmpeg: dont allow -flags to override -pass [01:47] <cone-290> ffmpeg.git 03Michael Niedermayer 07master:68dfe530e0fb: vdpau: Rename union, avoid name conflict with vdpau [02:27] <cone-290> ffmpeg.git 03Michael Niedermayer 07release/1.1:0e1bb99f26c2: update for 1.1.2 [02:27] <cone-290> ffmpeg.git 03Michael Niedermayer 07release/1.1:5f3fa5f93087: ffmpeg: dont allow -flags to override -pass [03:14] <llogan> michaelni: ping on rmlist [03:15] <michaelni> llogan, iam busy with 1.1.2 release, is it ok i do it tomorrow or later ? [03:16] <michaelni> i dont want to quickly enter a random command as root ;) [03:17] <llogan> fine with me. i also personally archived the archives incase you do blow it up by accident [03:23] <michaelni> llogan, done [03:28] <llogan> michaelni: thanks [04:15] <cone-290> ffmpeg.git 03Michael Niedermayer 07master:127ff88639e7: ffmpeg: do not call exit from exit_program() [04:22] <cone-290> ffmpeg.git 03Michael Niedermayer 07release/1.1:b48cf5412bb4: ffmpeg: do not call exit from exit_program() [04:46] <cone-290> ffmpeg.git 03Michael Niedermayer 07fatal: ambiguous argument 'refs/tags/n1.1.2': unknown revision or path not in the working tree. [04:46] <cone-290> Use '--' to separate paths from revisions [04:46] <cone-290> refs/tags/n1.1.2:HEAD: ffmpeg: do not call exit from exit_program() [10:31] <cone-533> ffmpeg.git 03Michael Niedermayer 07release/0.10:4d9bde86d024: ffmpeg: dont allow -flags to override -pass [10:31] <cone-533> ffmpeg.git 03Michael Niedermayer 07release/0.11:3cecef5b0389: ffmpeg: dont allow -flags to override -pass [10:31] <cone-533> ffmpeg.git 03Michael Niedermayer 07release/0.9:9f0bf48b5ce4: ffmpeg: dont allow -flags to override -pass [10:31] <cone-533> ffmpeg.git 03Michael Niedermayer 07release/1.0:86f072ceb3e9: ffmpeg: dont allow -flags to override -pass [11:08] <durandal_1707> michaelni: there is some talk on HA how ffmpeg native resample is slow and ugly quality [11:18] <LoopHoldYoaBrown> Hi [11:19] <LoopHoldYoaBrown> How can i make sure libavcodec can't call get_format over and over again during hw decoding? [11:30] <durandal_1707> is there way to change audio pitch? changing sample rate without changing audio? [13:22] <cone-533> ffmpeg.git 03Diego Biurrun 07master:25841dfe806a: Use ptrdiff_t instead of int for {avg, put}_pixels line_size parameter. [13:22] <cone-533> ffmpeg.git 03Michael Niedermayer 07master:ede45c4e1da8: Merge commit '25841dfe806a13de526ae09c11149ab1f83555a8' [13:22] <cone-533> ffmpeg.git 03Michael Niedermayer 07master:0ddca7d416cc: dsputil: fixup half a dozen bugs with ptrdiff vs int linesize [13:29] <iive> these commits sound interesting. [13:33] <cone-533> ffmpeg.git 03Diego Biurrun 07master:c9f933b5b62d: Add av_cold attributes to arch-specific init functions [13:33] <cone-533> ffmpeg.git 03Diego Biurrun 07master:70b348ec5b6a: configure: Group all hwaccels together in a separate variable [13:33] <cone-533> ffmpeg.git 03Diego Biurrun 07master:620289a20e02: sh4: Fix silly type vs. variable name search and replace typo [13:33] <cone-533> ffmpeg.git 03Michael Niedermayer 07master:6c3888487686: Merge commit '620289a20e022b9c16c10d546ef86cc0bb77cc84' [13:33] <iive> I actually have 2 questions. Is there any improvement for using ptrdiff over int? is ptrdiff_t signed? [13:34] <durandal_1707> in reality it is just cosmetic change with minor if any gain [13:35] <DEATH> ptrdiff_t is signed size_t pretty much? [13:38] <cone-533> ffmpeg.git 03James Almer 07master:8ab2173ed141: lavc/bink: Chech for malloc failure [13:44] <ubitux> hey there [13:44] <Compn> hey you [13:44] <michaelni> hey [13:44] <ubitux> seems there are some laught in the cvs-log [13:44] <Compn> ours ? [13:45] <ubitux> 25841dfe 0ddca7d4 [13:45] <ubitux> iive: isn't the ptrdiff thing useful for the asm? [13:45] <ubitux> avoid overread etc [13:46] <iive> i don't see how. [13:53] <Compn> devs using unfamiliar calls tend to forget signedness and sizes with the new types. well , afaik anyway [13:58] <iive> i do see now how. some 64bit asm function did sign extend, so it was removed. [13:59] <iive> or rather moved in the C code. So in reality it may even affect execution speed. [14:00] <durandal_1707> at what magnitude? [14:01] <iive> let's hope it makes it faster. [14:01] <cone-533> ffmpeg.git 03Tim Walker 07master:7af876a93fa8: flac: add channel layout masks for streams with 7 or 8 channels [14:09] <iive> but imho it just puts 64bit variable in the stack instead of 32bit. the sign extend should be done somewhere. [14:51] <cone-533> ffmpeg.git 03Luca Barbato 07master:b2a722a87e3b: sh4: drop unused functions [14:51] <cone-533> ffmpeg.git 03Vladimir Pantelic 07master:77bcb89600e7: vc1dec: use codec_id instead of codec_tag for VC1IMAGE [14:51] <cone-533> ffmpeg.git 03Diego Biurrun 07master:48a4ffa722c0: asf: K&R formatting cosmetics [14:51] <cone-533> ffmpeg.git 03Michael Niedermayer 07master:fadf845973ae: Merge commit '48a4ffa722c0874b251de9d201babed52cef0bcb' [14:59] <cone-533> ffmpeg.git 03Anton Khirnov 07master:99162f8d46db: vf_yadif: silence a warning. [15:00] <cone-533> ffmpeg.git 03Anton Khirnov 07master:6ed9fc44badb: svq1: replace struct svq1_frame_size with an array. [15:00] <cone-533> ffmpeg.git 03Anton Khirnov 07master:238614de679a: cdgraphics: do not rely on get_buffer() initializing the frame. [15:00] <cone-533> ffmpeg.git 03Michael Niedermayer 07master:6e9f3f3b6522: Merge commit '238614de679a71970c20d7c3fee08a322967ec40' [15:13] <michaelni> durandal_1707, about ha & ffmpeg resampler, beastd and his friends did some double blind tests to adjust the resampler parameters, so best you ask him [15:14] <michaelni> ill also ping him about it when i see him [15:17] <cone-533> ffmpeg.git 03Anton Khirnov 07master:6a399854517f: cmdutils: remove ansi from the list of broken codecs. [15:17] <cone-533> ffmpeg.git 03Anton Khirnov 07master:9bd6375d5f16: msrledec: check bounds before constructing a possibly invalid pointer, [15:17] <cone-533> ffmpeg.git 03Anton Khirnov 07master:da7baaaae79b: aasc: fix output for msrle compression. [15:17] <cone-533> ffmpeg.git 03Michael Niedermayer 07master:ddc3ca2caa4a: Merge commit 'da7baaaae79b4d7d715d35ea6bcfbdd149edc177' [15:26] <cone-533> ffmpeg.git 03Anton Khirnov 07master:2cd4068071b9: fraps: fix off-by one bug for version 1. [15:26] <cone-533> ffmpeg.git 03Michael Niedermayer 07master:a845ac78c942: Merge commit '2cd4068071b9a8908823a3107f97e938211045ce' [15:31] <cone-533> ffmpeg.git 03Kostya Shishkov 07master:685e6f2e3939: xxan: properly handle odd heights. [15:31] <cone-533> ffmpeg.git 03Michael Niedermayer 07master:68c0144d028b: Merge commit '685e6f2e3939f124b41c7801cc541dad8252af3d' [16:03] <cone-533> ffmpeg.git 03Kostya Shishkov 07master:89f11f498b9c: qtrle: fix the topmost line for 1bit [16:03] <cone-533> ffmpeg.git 03Michael Niedermayer 07master:769eba75eeaa: Merge commit '89f11f498b9c15bc71494a11a7ec560f4adf630d' [16:09] <cone-533> ffmpeg.git 03Anton Khirnov 07master:06cf597c3525: yop: check that extradata is large enough. [16:09] <cone-533> ffmpeg.git 03Michael Niedermayer 07master:3138b158dd4b: Merge commit '06cf597c352519d2b70f293518c61b8c312f5d4f' [16:28] <cone-533> ffmpeg.git 03Anton Khirnov 07master:8136f2344458: yop: check for input overreads. [16:29] <cone-533> ffmpeg.git 03Michael Niedermayer 07master:73562f1f834e: Merge commit '8136f234445862c94d1c081606b2d1e3d44fccf3' [16:40] <aballier> michaelni: you should probably change the comment after the qtrle 1bit merge; row_ptr became a pointer to the current row after the merge [16:41] <michaelni> aballier, will do, thanks [16:42] <michaelni> aballier, otherwise the code is ok after the merge ? I tested it with the files from the ticket both ways produce the same output [16:43] <aballier> wasnt sure if the commit was correct, but it seems so [16:43] <aballier> lemme check again [16:45] <aballier> for what i remember, all the 1st lines have the skip & 0x80 bit set, I have never been sure if that's the spec or just luck [16:45] <cone-533> ffmpeg.git 03Anton Khirnov 07master:39a9fdd00fea: yop: initialize palette to 0 [16:45] <cone-533> ffmpeg.git 03Anton Khirnov 07master:dff6197dfb9b: nuv: do not rely on get_buffer() initializing the frame. [16:46] <cone-533> ffmpeg.git 03Michael Niedermayer 07master:0b74fd113412: Merge commit 'dff6197dfb9b2d145729885c63f865c5fdd184aa' [16:47] <aballier> meaning doing "pixel_ptr = row_ptr;" before the loop is useless [16:50] <aballier> and i'm not sure how to interpret 1st line not having skip & 0x80 bit set; it may mean we should process lines_to_change lines and not lines_to_change+1, but i'm lacking a sample like that [16:50] <michaelni> yes was thinking the same [16:51] <michaelni> either its correct/unused or we have to wait for a bugreport to provide a sample to know which way it should be interpreted [16:52] <cone-533> ffmpeg.git 03Michael Niedermayer 07master:d71f8d74db54: qtrle: fix 1bit row_ptr comment [16:53] <aballier> yep, anyway, if i'm not mistaken, if skip & 0x80 has to be set for 1st lines, then both versions are correct, if it doesnt have to, then its likely both versions are incorrect for that case [17:08] <cone-533> ffmpeg.git 03Anton Khirnov 07master:293065bdb56e: mpegvideo: initialize dummy reference frames. [17:08] <cone-533> ffmpeg.git 03Michael Niedermayer 07master:78dc4cb26d5a: Merge remote-tracking branch 'qatar/master' [18:20] <Compn> seriously? [18:21] <Compn> i cant write -t 36:30 but have to write -t 00:36:30 ? [18:21] <michaelni> 36:30 is 36minutes 30 seconds or 36 hours 30 minutes ? [18:21] <Compn> 36 minutes 30 seconds [18:22] <Compn> if i write 1:30 , 1 minute, 30 seconds [18:22] <Compn> if i write -t 30 , it accepts it as seconds ... how does it know its not hours ? :P [18:22] <michaelni> is this AA:BB thing a regression ? [18:23] <Compn> i'm just used to it working in mencoder :) [18:23] <Compn> the docs mention hh:mm:ss so , its feature request, not regression [18:23] <Compn> -ss behaves the same way [18:23] <Compn> as -t , requiring hh:mm:ss [18:24] <michaelni> probably best to open a ticket so its not forgotten [18:24] <Compn> i guess :) [18:24] <Compn> new question, how to specify container aspect ratio ? [18:25] <cone-533> ffmpeg.git 03Vignesh Venkatasubramanian 07master:579c4add6334: Fate test to print side_data crc if present [18:25] <cone-533> ffmpeg.git 03Michael Niedermayer 07master:b2c443194484: framecrcenc: Also print the size of side data elements [18:25] <cone-533> ffmpeg.git 03Michael Niedermayer 07master:864fdfa0627e: framecrcenc: print all side data elements, not just the first [18:25] <Compn> maybe i should ask in #ffmpeg :) [20:20] <cone-533> ffmpeg.git 03rogerdpack 07master:6d33026668c4: dshow: fix log message [20:20] <cone-533> ffmpeg.git 03rogerdpack 07master:3b9166fc7efb: dshow: better error reporting [20:20] <cone-533> ffmpeg.git 03rogerdpack 07master:c916389bd286: dshow: allow for more codec ID's [20:20] <cone-533> ffmpeg.git 03Michael Niedermayer 07master:474519433634: Merge remote-tracking branch 'rdp/combined2' [20:33] <Zeranoe> Why does FFmpeg have "double versions", such as "libavcodec 54. 91.100 / 54. 91.100" both sides of the / are the same. [20:33] <wm4> isn't that header version vs. library version [20:34] <wm4> they can be different if you use shared libraries and are criminally insane enough to use newer library binaries with programs compiled against older versions [20:59] <michaelni> anyone wants to review the 32bit fixed point FFT patch ? [20:59] <michaelni> Nedeljko Babic ( 46K) [FFmpeg-devel] [PATCH 1/2] libavcodec: Implementation of 32 bit fixed point FFT [21:00] <Compn> thats the mips version ? [21:02] <michaelni> theres generic C code for it and MIPS code [21:03] <michaelni> ive been wrong, rechecking, its just C code no asm/mips code [21:09] <wm4> <Rodeo> FFmpeg applied http://patches.libav.org/patch/33823/ without http://patches.libav.org/patch/33822/ [21:09] <wm4> <Rodeo> I wonder if I should tell them? [21:09] <wm4> <Rodeo> at the very least <= 6 should be <= 8 [21:09] Action: wm4 wonders if he's a spy now [21:12] <llogan> I think the question is: why not tell us? [21:13] <wm4> because dicks? [21:48] <michaelni> beastd, "<durandal_1707> michaelni: there is some talk on HA how ffmpeg native resample is slow and ugly quality" [21:49] <beastd> michaelni: hydrogenaudio? [21:49] <michaelni> yes [21:55] <wm4> is libswsresample better? [22:05] <Compn> michaelni : unless HA did a double blind study like they did for vorbis, opinions dont mean squat :) [22:08] <beastd> the infinitywave graph is done with 0.10.4 from 2012-06 FWIW [22:11] <llogan> anyone try libsoxr? [22:48] <cone-533> ffmpeg.git 03Matthieu Bouron 07master:59d40fc7e66e: lavf/mov: skip version and flags attributes in mov_read_chan function [22:57] <cone-533> ffmpeg.git 03Stefano Sabatini 07master:b3d2c6f8b9d0: lavf/segment: fix DTS inconsistencies with -reset_timestamps [22:57] <cone-533> ffmpeg.git 03Stefano Sabatini 07master:d11ac64ecf03: doc: replace '×' symbol with 'x' [23:37] <cone-533> ffmpeg.git 03Matthieu Bouron 07release/1.0:f3144b0cc694: lavf/mov: skip version and flags attributes in mov_read_chan function [23:37] <cone-533> ffmpeg.git 03Matthieu Bouron 07release/1.1:02d3ad860957: lavf/mov: skip version and flags attributes in mov_read_chan function [00:00] --- Thu Feb 7 2013

1 0

ffmpeg.log.20130206
by burek 07 Feb '13

07 Feb '13

[01:04] <pigoz> what's the channel name for mono in -af pan? trying to map mono to stereo [01:09] <saste> pigoz, does ffmpeg -layouts help? [01:11] <pigoz> saste: dumb me I mispelled it to --layouts. I got what I needed from the source. [01:34] <llogan> cheeseboy: why not just use libx264 with ffmpeg [01:35] <cheeseboy> llogan, too lazy to learn arguments plus x264's progress is prettier [01:35] <llogan> https://ffmpeg.org/trac/ffmpeg/wiki/x264EncodingGuide [01:37] <cheeseboy> llogan, lazy [03:30] <FergusL> Are you around llogan? [03:32] <llogan> FergusL: for a little longer. [03:32] <FergusL> llogan: you helped me yesterday about compilation [03:33] <FergusL> I still have issues, unrelated to ffmpeg itself, rather to how I now do it. I didn't "check install" for all the packages [03:33] <FergusL> could that create problems regarding libraries linking ? [03:34] <llogan> ffmpeg will tell you if it couldn't find things [03:37] <FergusL> hm... [03:38] <FergusL> okay so I _have_ to run checkinstall ? [03:39] <FergusL> I had hoped I could do what I want without touching a single bit of the current packages etc... [03:39] <FergusL> as in, I want the ffmpeg version I compiled to be only used by the program I cmpile, all other programs using ffmpeg should find it in the usual place [03:41] <relaxed> FergusL: I have current static binaries for 64bit linux or you can use a custom prefix and avoid using your package manager. [03:42] <FergusL> I did that, too, relaxed [03:42] <relaxed> and? why would you need checkinstall? [03:43] <FergusL> but the guide I followed ( https://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide) asks to runs checkinstall instead of make install [03:43] <FergusL> btw doesn't "make install" only copy files to the desired location ? [03:43] <relaxed> yes, to your prefix [03:44] <FergusL> exactly, ffmpeg is there, now [03:44] <FergusL> well I'm asking there but my issue now is unrelated to ffmpeg, really [03:45] <relaxed> I don't agree with these guides instructing people to replace software using their package manager. [03:46] <FergusL> well that's the suggestion I was about to do ! [03:47] <FergusL> moreover, someone noted the same thing as you yesterday [03:48] <relaxed> It's trivial to install software using --prefix=/home/blah/ffmpeg-custom, then running "LD_LIBRARY_PATH=/home/blah/ffmpeg-custom/lib /home/blah/ffmpeg-custom/bin/ffmpeg -i ..." [03:49] <FergusL> wait wait wait [03:49] <FergusL> oh okay [03:49] <relaxed> You can have infinite, multiple installs and no conflicts. [03:49] <FergusL> that's like an export $LD_blabla before [03:50] <relaxed> yes, but maybe you don't want it exported. [03:50] <relaxed> anyway, you get the idea. [03:53] <relaxed> install all recent external libs to --prefix=/home/blah/ffmpeg-custom, the configure ffmpeg with --extra-cflags="-I/home/blah/ffmpeg-custom/include" --extra-ldflags="-L/home/blah/ffmpeg-custom/lib" [03:53] <relaxed> There- a one minute guide that's not retarded. [03:54] <relaxed> If yasm is old, install it using the same prefix and then modify your path so the new binary is first. done. [03:58] <FergusL> relaxed: what do you mean by "then configure" ffmpeg ? [03:58] <FergusL> configure the compilation flags of what I'm trying to compile ? [03:59] <FergusL> (yes I'm trying to compile some small program using ffmpeg) [04:00] <relaxed> Install libx264 from git using --prefix=/home/blah/ffmpeg-custom; then when you go to run ffmpeg's ./configure, add --extra-cflags="-I/home/blah/ffmpeg-custom/include" --extra-ldflags="-L/home/blah/ffmpeg-custom/lib" so that the configure script will find the libs/headers of the recent version you just installed to the custom prefix (and not an old system version) [04:01] <FergusL> yes, i've done than... I... I think I've done that [04:01] <FergusL> but I'm pretty sure the issue I have now is not related to that [04:02] <relaxed> I have zero idea what issue you're having. [04:03] <FergusL> I know I know, I didn't mention it ! [04:04] <FergusL> the initial question was just to know if the issue I had could be related to previous omissions [11:17] <LoopHoldYoaBrown> Hello howto make sure libavcodec won't call h264 deocder get_format constantly during HW decoding? [11:49] <sceriffo> hi , I have to build a function that decode an array of AAC data into PCM data. My input is only this array. I've not the header frame. What kind of informations I need to perform this? (bitrate, etc..). Best regards, sceriffo [11:54] <sledges> how can I multiple-trim mp4 without intermediate files (and hell of concatenating them)? something like: ffmpeg -ss 00:00:05 -t 00:00:04 -ss 0:00:20 -t 00:00:04 -i in.mp4 acodec copy -vcodec copy -async 1 out.mp4 [11:55] <sledges> (hello all btw ;)) [11:55] <sledges> *-acodec [12:06] <aleray> hi, I have a set of VOB files and the sound quality is pretty bad and the audio level is low (amateur DV recording of dance performances). Is there any way to improve the sound quality, The files will then be converted to Ogg for being shown on the web. [12:35] <sceriffo> hi , I have to build a function that decode an array of AAC data into PCM data. My input is only this array. I've not the header frame. What kind of informations I need to perform this? (bitrate, etc..). Best regards, sceriffo [14:15] <rcombs> 'ehlo, wondered if anyone could help me track down a possibly-ffmpeg-related bug involving green artifacts around subtitles being burned in [14:17] <rcombs> the issue is present in Plex Media Server, which uses a modified version of ffmpeg (so it might have nothing to do with ffmpeg itself at all) [14:17] <rcombs> the current version, which has the issue, is based on git-2013-01-22-0c46ebe [14:19] <rcombs> issue is demonstrated in screenshots shown here: https://plexapp.lighthouseapp.com/projects/14382/tickets/1414-transcoder-ba… [14:20] <rcombs> once again, not sure if this is an issues with Plex's transcoder modifications, ffmpeg itself, libass, or any number of other things [14:20] <rcombs> but I figured I'd try here to see if anyone had ever heard of a similar issue [14:20] <rcombs> also present here: https://twitter.com/11rcombs/status/297898849245216768/photo/1 [14:22] <relaxed> it's called hardsubbing [14:23] <rcombs> ? [14:23] <relaxed> overlaying subs into the video stream [14:23] <rcombs> yes, I'm aware of the term [14:24] <relaxed> Do you have a command and samples to reproduce the problem? [14:25] <relaxed> If not, that's where you need to start. [14:25] <rcombs> sure, sec [14:25] <rcombs> well, I'll need to cut a sample file [14:26] <relaxed> I don't want to see the result of the problem, just to be clear. [14:26] <rcombs> and the commands (from my logfiles) are for the Plex Transcoder, which has a few differences in input syntax from stock ffmpeg [14:26] <relaxed> ok [14:27] <relaxed> Provide a link to the sample of the input and pastebin the command used. [14:27] <rcombs> OK [14:33] <rcombs> Feb 06, 2013 07:32:54 [0xa85ffb70] DEBUG - Job running: /usr/local/plexmediaserver/Resources/Plex New Transcoder -i http://127.0.0.1:32400/library/parts/4571/file.mkv -threads 0 -y -f mp4 -loglevel quiet -progressurl http://127.0.0.1:32400/video/:/transcode/session/20444012b84d521c6a3737829b… -vcodec libx264 -crf 23 -vsi 640x568 -x264opts ref=3:bframes=3:cabac=1:8x8dct=1:weightp=1:b-pyramid=normal:b-adapt=2:direct=spatial:me=hex:subq= [14:33] <rcombs> 7:analyse=most:trellis=1 -map 0:0 -burnsidx 3 -burnsscale 1 -acodec copy -map 0:1 /mnt/user/Plex-Media/Library//Application Support/Plex Media Server/Cache/Transcode/Sync/35941/103588/4a58b8b31592a6d99f9c.mp4.temp [14:33] Action: rcombs cuts a sample [14:39] <rcombs> will be up on Dropbox momentarily: https://dl.dropbox.com/u/46348978/Baka%20and%20Test%20001%20%281%29.mkv [14:40] <rcombs> (sample is the series' opener) [14:40] <rcombs> up [14:56] <rcombs> relaxed: ping ^ that was for you [15:14] <relaxed> rcombs: Sorry, ffmpeg doesn't have "-burnsidx 3 -burnsscale 1" so I can't reproduce. [15:18] <veleno> hello. is it possible with ffmpeg to receive an UDP stream and, while receiving it, to produce statistics about the quality it ? [15:21] <rcombs> relaxed: yeah, and I'm not sure what options that might correspond to [15:22] <rcombs> relaxed: (-vsi corresponds to -s, for instance, but that's no help with -burnsidx) [16:21] <jeje34> Hi to all ;-) [16:23] <jeje34> I have a problem with the last FFMPEG (1.1.1). I use the libavcodec to decompress H264 stream [16:26] <jeje34> But I'd like to have less CPU usage during decompressing my H264 stream. I want to try Multithreading (m_lpCodecCtx->thread_type = FF_THREAD_FRAME | FF_THREAD_SLICE; m_lpCodecCtx->thread_count = 3;) [16:26] <jeje34> but I have an error The maximum value for lowres supported by the decoder is 0 [16:27] <jeje34> I doesn't reallyunderstand how I can fix my lowres and max_lowres [16:27] <jeje34> if someone can explain me, thks a lot [17:01] <veleno> where do i find some doc about the format of the vstats files ? [18:13] <mm_> Hi there. I'm using ffmpeg as a client for an udp stream ( $ffmpeg -report -vstats -i udp://localhost:60001 output_drops.mpg -loglevel verbose ) and I need to know how can I extract information regarding potential errors from the logs. For instance when packets are dropped I observe errors like "end mismatch left=4677 147B52" and "concealing 520 DC, 520 AC, 520 MV errors in P frame". If you could provide some pointers regarding the semantics of those errors, i [18:13] <mm_> what they actually mean. The reason for this is that I'm developing a network protocol using ffmpeg as a sink and I need to adjust some parameters according to the video quality at the sink, so I need to quantify that quality, even if in a naive way. Thanks for your time. [18:23] <jeje34> no one can give me some reason about the lowres message error I have when I want set multithreading to decode h264 [18:23] <jeje34> ? [18:25] <Compn> anyway to set par/dar in mkv with ffmpeg while -vcodec copy ? [18:28] <Compn> i see -vf setsar but doesnt look like it works for copy.. [18:29] <Mavrik> videofilters get applied only when actually transcoding the steram [18:29] <Mavrik> *stream [18:29] <Mavrik> not while copying [18:29] <Compn> yeah obviously [18:29] <Compn> no way to set container aspect ? [18:29] <Compn> mkvmerge can do this i think... [18:29] <Mavrik> hmm [18:29] <Mavrik> I'm quite sure SAR is set inside H.264 stream, not on the container [18:30] <Compn> pretty sure h264 itself has codec aspect too... [18:30] <Compn> can ffmpeg modify either of them using -vcodec copy ? [18:30] <Mavrik> no because ffmpeg doesn't touch the stream when doing "copy" [18:30] <Compn> so mkvmerge it is , grumble [23:52] <Pinhole> I'm trying to transcode a ts stream from mpeg4 to mpeg2. I'm having trouble with mapping. http://pastebin.com/QkfdB9U4 [00:00] --- Thu Feb 7 2013

1 0

ffmpeg-devel.log.20130205
by burek 06 Feb '13

06 Feb '13

[00:02] <cone-831> ffmpeg.git 03Stefano Sabatini 07master:2d22e97f6d73: doc/filters: fix @item syntax in showspectrum docs [00:13] <cone-831> ffmpeg.git 03Stefano Sabatini 07master:0e2b0033f161: doc/filters: apply various rendering changes to the showspectrum docs [00:37] <cone-831> ffmpeg.git 03Michael Niedermayer 07master:b77d94dc6cc6: ffv1enc: check for malloc failure [00:37] <cone-831> ffmpeg.git 03Michael Niedermayer 07master:5a8311513091: ffv1: check for malloc failure [00:41] <llogan> saste: any comments (since you were the only one with input on ML)? http://pastebin.com/raw.php?i=6B1ege28 [00:44] <saste> llogan, i find annoying when I'm subscribed to a new ML without my explicit approval [00:45] <llogan> one reason i'm not going to auto-subscribe [00:45] <saste> so I'd suggest to advertise the current ffserver-user subscribers of the move [00:45] <saste> you could keep the ML active, but tell all the posters to move to ffmpeg-user [00:46] <saste> yes an option would be to forward ffserver-user mails to ffmpeg-user [00:46] <saste> i don't have a strong opinion on that [00:46] <Compn> i wasnt suggesting subscribing them w/ email subscriptions [00:46] <llogan> i already made two announcements on ffserver-user. [00:46] <Compn> only putting their emails in so they could post, automatically [00:46] <saste> you could leave ffserver-user for a month or so [00:47] <saste> i don't think there is much traffic anyway [00:47] <saste> and send a mail every week advertising that the list will be closed in X days [00:47] <saste> and providing instructions/link to subscribe to ffmpeg-user [00:48] <llogan> i already said today is it's last day (on jan 27 and again today). [00:51] <saste> llogan, i think it's safe to remove the list anyway [00:52] <saste> it would be a bit annoying for the few subscriber to move to an high traffic list [00:52] <saste> but should be allow a better support, since there will be more feedback [00:52] <llogan> and they can always disable delivery on their choice. [00:54] <saste> i'm fine with it [01:06] <llogan> michaelni: please rmlist ffserver-user at your convenience [01:45] <michaelni> llogan, ill wait 24h to give people a last chance to raise objections and then delete it [02:03] <llogan> ok [03:09] <cone-831> ffmpeg.git 03multiple authors 07master:708ed15d8ccd: libmpcodecs/vf_stereo3d: update to latest version from mplayer [03:32] <Compn> ehehe [03:43] <cone-831> ffmpeg.git 03multiple authors 07master:103a2c2b1757: libmpcodecs: update img_format.* & mp_image.* to latest from mplayer [10:36] <j-b> morning [11:54] <michaelni> morning [11:55] <TimNich> moening [11:58] <durandal_1707> its 12 [12:02] <cone-290> ffmpeg.git 03Diego Biurrun 07master:e66240f22e24: avfilter: x86: consistent filenames for filter optimizations [12:02] <cone-290> ffmpeg.git 03Michael Niedermayer 07master:0b6f34cc9f28: Merge remote-tracking branch 'qatar/master' [12:16] <TimNich> not in my timezone [12:20] <kierank> it is always morning on irc [12:23] <michaelni> that proofs the earth cant be flat [12:36] <TimNich> sometimes feels like a moebius loop! [13:07] <cone-290> ffmpeg.git 03multiple authors 07master:a54737f7ead0: libmpcodecs/cpudetect: update to latest from mplayer [13:07] <cone-290> ffmpeg.git 03multiple authors 07master:745cb39d0558: libmpcodecs/help_mp: update to latest from mplayer [13:07] <cone-290> ffmpeg.git 03multiple authors 07master:99a0813a7e0b: libhmpcodec/mp_msg: update to latest from mplayer [13:09] <michaelni> TimNich, a moebius loop is unlikely because if it was that way that moebius loop could be seen by simply looking up from some places [13:49] <cone-290> ffmpeg.git 03multiple authors 07master:5d0115118df4: libmpcodecs/vf_divtc: update to latest version in mplayer [13:49] <cone-290> ffmpeg.git 03multiple authors 07master:e02c713fb149: libmpcodecs/vf_filmdint: update to latest version from mplayer [13:49] <cone-290> ffmpeg.git 03multiple authors 07master:2b9eb6167b32: libmpcodecs/av_helpers: update to latest version from mplayer [13:49] <cone-290> ffmpeg.git 03multiple authors 07master:0b36a05c1752: libmpcodecs/vf_fspp: update to latest version from mplayer [14:12] <cone-290> ffmpeg.git 03multiple authors 07master:011702f5a934: libmpcodecs/vf_ivtc: update to latest version from mplayer [14:12] <cone-290> ffmpeg.git 03multiple authors 07master:25ed6e99d789: libmpcodecs/vf_ilpack: update to latest version from mplayer [14:12] <cone-290> ffmpeg.git 03multiple authors 07master:b7e6622971eb: libmpcodecs/vf_mcdeint: update to latest version from mplayer [14:23] <cone-290> ffmpeg.git 03multiple authors 07master:fdd574ed3f60: libmpcodecs/vf_qp: update to latest version from mplayer [14:23] <cone-290> ffmpeg.git 03multiple authors 07master:3ca46d0c10aa: libmpcodecs/vf_pp7:update to latest version from mplayer [14:23] <cone-290> ffmpeg.git 03multiple authors 07master:8f7e3e678b58: libmpcodecs/vf_noise: update to latest version from mplayer [14:41] <juanmabc> mmm, mplayer pulseaudio got excellent, not even 1% [14:41] <juanmabc> cpu [14:59] <cone-290> ffmpeg.git 03multiple authors 07master:fe8bc6ddfcd2: libmpcodecs/pullup: update to latest version from mplayer [14:59] <cone-290> ffmpeg.git 03multiple authors 07master:14132599a372: libmpcodecs/vf_sab: update to latest version from mplayer [14:59] <cone-290> ffmpeg.git 03multiple authors 07master:7a4885600ff0: libmpcodecs/vf_spp: update to latest version from mplayer [15:48] <cone-290> ffmpeg.git 03multiple authors 07master:3ce3d4cc3c0a: libmpcodecs/vf_unsharp: update to latest version from mplayer [15:48] <cone-290> ffmpeg.git 03multiple authors 07master:13afcdd00eec: libmpcodecs/vf_uspp: update to latest version from mplayer [15:48] <cone-290> ffmpeg.git 03multiple authors 07master:81e357db6259: libmpcodecs/vf_pullup: update to latest version from mplayer [16:11] <cone-290> ffmpeg.git 03Michael Niedermayer 07master:2b20397e1fbe: libmpcodecs/vd_ffmpeg.h: remove, its unused and removed upstream [16:25] <wm4> lol old broken mplayer sources lingering in ffmpeg [16:27] <cone-290> ffmpeg.git 03Xi Wang 07release/1.1:8d0631c8fa89: mpegvideo: fix loop condition in draw_line() [16:27] <cone-290> ffmpeg.git 03Luca Barbato 07release/1.1:01050448cfbc: get_bits: change the failure condition in init_get_bits [16:27] <cone-290> ffmpeg.git 03Janne Grunau 07release/1.1:1a28948eb3e5: videodsp_armv5te: remove #if HAVE_ARMV5TE_EXTERNAL [16:27] <cone-290> ffmpeg.git 03Martin Storsjö 07release/1.1:62de693a17f9: rtp: Make sure priv_data is set before reading it [16:27] <cone-290> ffmpeg.git 03Michael Niedermayer 07release/1.1:85e94a30eeb9: Merge commit '62de693a17f9b107be7867d822d5accacd4be544' into release/1.1 [16:33] <cone-290> ffmpeg.git 03Martin Storsjö 07release/1.1:348cd84fc82e: rtp: Make sure the output format pointer is set [16:33] <cone-290> ffmpeg.git 03Martin Storsjö 07release/1.1:a856623e87d3: rtpenc_chain: Use the original AVFormatContext for getting payload type [16:33] <cone-290> ffmpeg.git 03Justin Ruggles 07release/1.1:d7e7e12abc7e: lavr: call mix_function_init() in ff_audio_mix_set_matrix() [16:33] <cone-290> ffmpeg.git 03Reinhard Tartler 07release/1.1:8d55c2441c67: Prepare for 9.2 Release [16:33] <cone-290> ffmpeg.git 03Martin Storsjö 07release/1.1:3d67f52f9dcc: arm: vp8: Fix the plain-armv6 version of vp8_luma_dc_wht [16:33] <cone-290> ffmpeg.git 03Michael Niedermayer 07release/1.1:8a4464514fac: Merge remote-tracking branch 'qatar/release/9' into release/1.1 [17:01] <cone-290> ffmpeg.git 03Piotr Bandurski 07release/1.1:22561bc0e963: aasc: fix 16bpp on big-endian [17:01] <cone-290> ffmpeg.git 03Michael Niedermayer 07release/1.1:6baa54924980: huffyuvdec: Check init_vlc() return codes. [17:01] <cone-290> ffmpeg.git 03Michael Niedermayer 07release/1.1:75e88db33013: huffyuvdec: Skip len==0 cases [17:01] <cone-290> ffmpeg.git 03Diego Biurrun 07release/1.1:0e68b6ddce29: Use proper "" quotes for local header #includes (cherry picked from commit 6c1a7d07eb41b4cf36e48ebb560fecd9504fd968) [17:01] <cone-290> ffmpeg.git 03Michael Niedermayer 07release/1.1:62c9beda0c18: sanm: Check decoded_size. [17:01] <cone-290> ffmpeg.git 03Michael Niedermayer 07release/1.1:9547034f9120: gifdec: gif_copy_img_rect: Fix end pointer [17:01] <cone-290> ffmpeg.git 03Michael Niedermayer 07release/1.1:f3d16706060a: vqavideo: check chunk sizes before reading chunks [17:01] <cone-290> ffmpeg.git 03Michael Niedermayer 07release/1.1:47e462eecc0a: aacdec: check channel count [17:01] <cone-290> ffmpeg.git 03Michael Niedermayer 07release/1.1:3ef1538121fa: h264: check the pixel format directly and force a reinit on mismatches. [17:01] <cone-290> ffmpeg.git 03Michael Niedermayer 07release/1.1:d3bec2473972: h264: Only apply error concealment if theres a frame [17:01] <cone-290> ffmpeg.git 03Michael Niedermayer 07release/1.1:eaa9d2cd6b8c: h264: skip error concealment when SPS and slices are mismatching [18:26] <wm4> how do I use avcodec_copy_context() when I want to use a codec different from the source context? maybe call avcodec_get_context_defaults3() with the wanted AVCodec after that? [18:36] <cone-290> ffmpeg.git 03Michael Niedermayer 07master:a8343bfb6a3f: riff: fix infinite loop [19:01] <cone-290> ffmpeg.git 03Michael Niedermayer 07master:c2dd5a18b27b: ljpegenc: allocate needed scratch-buffer [21:05] <Zeranoe> Could anyone tell me the current status of GPU encoding/decoding? I see NVIDIA provides a H.264 encoder/decoder that works off of CUDA. I'm interested in starting a LGPL GPU encoder [21:08] <nevcairiel> in general? only fixed presets, quality below software encoders without a significant speed advantage in many cases [21:08] <nevcairiel> for encoding, that is [21:09] <nevcairiel> decoders are usually working quite well, provide bit-exact output, speed is usually below any modern CPU, only real use-case is on embedded/mobile [21:10] <nevcairiel> and limited to mainstream profiles of course, for ex. h264 only 8-bit 4:2:0 baseline/main/high, both for enc/dec [21:12] <Zeranoe> nevcairiel: Is the speed below CPU encoding because of limitations of the GPU? Is there a possibility to make an encoder/decoder that would out preform CPU encoders/decoders? [21:12] <nevcairiel> it always depends what quality you're looking for [21:13] <nevcairiel> and what kind of gpu you have =p [21:13] <nevcairiel> latest nvidia models for example have a fixed function encoder, earlier models implemented it in cuda [21:13] <nevcairiel> the tests i remembered, running x264 on settings that resulted in similar quality and filesize, the speed was pretty similar [21:13] <wm4> what's the use case for GPU encoding? cameras? [21:14] <nevcairiel> video conferencing, i guess [21:14] <Zeranoe> nevcairiel: [21:14] <nevcairiel> or if you go into the phone market, camera recordings as well, i suppose [21:15] <Zeranoe> nevcairiel: Does FFmpeg support any GPU features? [21:15] <nevcairiel> ffmpeg supports gpu decoding [21:15] <nevcairiel> technically, avcodec does, you need some user code to make it work [21:15] <Zeranoe> nevcairiel: Is it automatically used, or is there special way to test it? [21:15] <nevcairiel> which the ffmpeg application does not implement [21:16] <nevcairiel> you could run vlc, it uses it [21:16] <Zeranoe> nevcairiel: Is there a way to test it on the command line? [21:16] <nevcairiel> is mplayer considered command line? =P [21:17] <Zeranoe> nevcairiel: is it only available in other applications? [21:17] <nevcairiel> gpu decoding is not useful for transcoding because the gpu decoders are so slow in comparison [21:17] <nevcairiel> the ffmpeg applications dont support it [21:18] <Zeranoe> nevcairiel: Interesting, so it only really helps with playing a steam? [21:18] <cone-290> ffmpeg.git 03rogerdpack 07master:a50049a197a5: dshow: use standardized raw pixel format lookup [21:18] <cone-290> ffmpeg.git 03rogerdpack 07master:47e88486b4b3: hdyc col [21:18] <cone-290> ffmpeg.git 03rogerdpack 07master:c540312ac3b5: Merge remote-tracking branch 'origin/master' into combined [21:18] <cone-290> ffmpeg.git 03Michael Niedermayer 07master:f7b7f021e528: Merge remote-tracking branch 'rdp/combined' [21:19] <llogan> "hdyc col". not very descriptive. [21:20] <Zeranoe> nevcairiel: How many encoders support CPU threading in FFmpeg? [21:20] <nevcairiel> no idea, i dont encode much [21:22] <J_Darnley> Wasn't a flag added to codecs to state whether they supported frame and/or sliced threads? Does ffmpeg -codecs tell you? [21:28] <J_Darnley> Answer: My (slightly) old ffmpeg doesn't but a search shows 59 matches in 39 files. [21:35] <llogan> ubitux: how is the adventure? [22:02] <cone-290> ffmpeg.git 03Michael Niedermayer 07master:b0635e2fcf80: movtextenc: fix pointer messup and out of array accesses [23:04] <cone-290> ffmpeg.git 03Michael Niedermayer 07master:f0eacbc760c8: dvenc: dont fail hard if the timecode is invalid [00:00] --- Wed Feb 6 2013

1 0

ffmpeg.log.20130205
by burek 06 Feb '13

06 Feb '13

[00:26] <llogan> FergusL: https://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide [00:27] <llogan> blitz: https://ffmpeg.org/trac/ffmpeg/wiki/Encoding%20VBR%20%28Variable%20Bit%20Ra… [00:35] <FergusL> thanks llogan [00:36] <FergusL> but ... erh... haven't tried it yet but I suspect the first command will ask for the removal of like tens of packages [00:39] <llogan> you could compile and not install the new ffmpeg but still use it [00:40] <FergusL> okay, that's what i hought [00:43] <FergusL> llogan: I just do make and then I should have everything in the output dir ? [00:45] <llogan> FergusL: the examples here may help http://ffmpeg.org/pipermail/ffmpeg-user/2012-August/008552.html [00:46] <FergusL> perfect llogan I would have been looking for someting like this indeed :) [00:47] <FergusL> what about /include llogan ? will follow the same --prefix ? [00:47] <FergusL> well I'll just look at ./configure anyway first [00:47] <llogan> you will have to refer to ./configure --help for each thing you're compiling [00:49] <FergusL> yes, a bit used to this, thanks again for your help llogan [00:49] <llogan> a "local install" version of the guide would be good [01:12] <saste> llogan, what do you mean by "local install"? [01:13] <saste> i have some notes about general configuration/compilation/installation in *NIX, I plan to put them on the wiki soon or later [01:16] <llogan> saste: using --prefix=$HOME for dependencies and then pointing ffmpeg to those with cflags, ldflags, etc. [01:17] <llogan> would be useful for users with no superuser/root access, and could be good to keep system "ffmpeg" from repo for other repo shit and a compiled ffmpeg for command-line usage [01:21] <llogan> repo package conflict has always been the biggest issue with the ubuntu guide. or a simple copy binary to ~/bin or whatever in $PATH might suffice. [01:21] <saste> llogan, I see [01:21] <saste> i'll try to write it (hardly before the weekend) [01:22] <llogan> i actually haven't done much of that myself though since i use a more flexible (IMO) distro [01:26] <FergusL> llogan which is... ? :) [01:37] <llogan> FergusL: arch linux [01:38] <FergusL> I knew it would be that or Gentoo :) [02:00] <ldionmarcil> a mp4 is like an container right? it has a sound track + Video? then how can I see what type the video is? like h.264 etc [02:03] <ldionmarcil> ffmpeg -i on file, got my annswer, thanks guys [02:04] <llogan> FergusL: gentoo? i'm not that neckbeard. [02:04] <FergusL> Haha [02:35] <quup> can I make ffprobe just print dimensions '%dx%d' for first video stream? [02:38] <sacarasc> With grep/sed/awk I am sure you can! [02:40] <quup> sacarasc· no doubt, I just don't have a shell :p [02:40] <quup> oh well, I'll just bite the sour apple and regex it [03:11] <FergusL> trying to compile, compiled yasm to /opt/yasm, now x264 doesn't find it, how can I specify yasm location ? [03:12] <FergusL> although it correclty isntalled with checkinstall... [03:12] <FergusL> aptitude shows the right version and location [03:44] <FergusL> compiling... still expecting it to break at some point... haha [06:07] <relaxed> FergusL: PATH=/path/to/prefix/bin:$PATH; ./configure ... [09:59] <gotwig> Hey [09:59] <gotwig> please help me, why does -ss option to cut some seconds from the beginning make the audio so bad? the audio offset is awful [10:00] <gotwig> in combo with a video [10:10] <relaxed> gotwig: -async 1 [10:10] <gotwig> relaxed, thank you reall much ^_^ [10:10] <gotwig> ^really :/ [10:17] <gotwig> can I add audio pitches with ffmpeg? [10:19] <leo2013> How to add filters for ffmpeg? [10:20] <leo2013> add one new filter or another functions based on old filters. [10:22] <gotwig> Is there anything I can do with ffmpeg, so the audio music files dont get recoginzed by youtube, with ffmpeg? [10:22] <gotwig> like filters (^) ? [10:23] <gotwig> because in Germany there are big problems with that... [10:36] <gotwig> somehow -t does not work for me... [10:36] <gotwig> what is 2:35:53 - 2:29:27 :/? [10:36] <gotwig> 6: [10:36] <gotwig> oups... [10:45] <gotwig> can I cut the endning of the video with the help of -t ? [10:45] <gotwig> *ending [10:45] <gotwig> like -t 350s ? or -t 350 ? [11:00] <gotwig> Any ide how to add audio pitch? [11:11] <durandal_1707> gotwig: isn't that just changing sample rate? [11:13] <gotwig> yeah I guess so [11:13] <gotwig> I just have to find a trick to get my video widly available on youtube... [11:16] <gotwig> and somehow I am not able to use -t in combination with -ss [11:16] <durandal_1707> sox have pitch effect that just change sample rate of audio, and audio is apparently unchanged [11:16] <gotwig> why does -t not work in combo with -ss? [11:17] <durandal_1707> -ss just seek [11:17] <gotwig> -ss cuts the specified seconds before the video, -t before the end [11:17] <gotwig> oh damn... [11:17] <gotwig> do you use avconv? [11:18] <gotwig> copy does not work for me... [11:18] <durandal_1707> avconv is another project, man [11:18] <gotwig> yes I know... [11:18] <gotwig> but its somehow the same application... [11:18] <durandal_1707> not really .... [11:18] <gotwig> from the syntax :X [11:18] <durandal_1707> not really .... [11:19] <gotwig> I always lose quality, I cant find a way to keep it... [11:19] <gotwig> durandal_1707, can I use -ss in combo with -t? [11:19] <durandal_1707> what you want to archieve? [11:19] <gotwig> or reach my goal in another way [11:19] <gotwig> cut a bit from the beginning [11:19] <gotwig> and a bit from the end [11:20] <gotwig> with one run [11:20] <durandal_1707> with -c copy? [11:20] <gotwig> no, I talk about cutting here... [11:20] <gotwig> with -ss and -t [11:20] <gotwig> but you told me -ss is just seaking.. [11:21] <durandal_1707> you want to cut file without transcoding? [11:22] <gotwig> yes [11:22] <gotwig> without [11:22] <gotwig> just cut the thing [11:23] <gotwig> can I do that? [11:25] <durandal_1707> thing is you can't precisely cut with -c copy, it is simply not possible [11:25] <gotwig> :( [11:25] <gotwig> can I use -sameq? [11:26] <gotwig> the thing is, that my video file shrinked down to 300 mb, from 900 mb... and I only cut a few minutes [11:26] <durandal_1707> because compressed bitstream packets of audio can have several samples... [11:26] <gotwig> can I try with -c copy and -sameq? [11:26] <gotwig> btw, is avconv faster or ffmpeg..? [11:27] <durandal_1707> video patckets should be fine except you cant cut in middle of non-keyframe because you will get nothing for few frames.... [11:27] <durandal_1707> gotwig: avconv is slower and buggy [11:27] <gotwig> ubuntu ships it... [11:27] <durandal_1707> gotwig: try for yourself [11:27] <durandal_1707> gotwig: so what? [11:28] <gotwig> durandal_1707, so I am going to do two runs? [11:28] <durandal_1707> -sameq is removed, and it worked only for transcoding.... [11:28] <durandal_1707> gotwig: i can help you only if you use ffmpeg, you are on #ffmpeg after all [11:28] <gotwig> yeah [11:28] <gotwig> I use ffmpeg now xD [11:28] <gotwig> but -c copy does not work for me... [11:29] <gotwig> http://paste.ubuntu.com/1611935/ [11:30] <durandal_1707> gotwig: do you understand what word complete means? [11:30] <durandal_1707> i'm just wasting my time here [11:32] <gotwig> durandal_1707, what? [11:33] <DEATH> gotwig: Complete console output? [11:33] <gotwig> :) [11:34] <gotwig> http://paste.ubuntu.com/1611946/ [11:39] <durandal_1707> gotwig: you are using ffmpeg from project that is not FFmpeg [11:39] <gotwig> it seems... [11:40] <durandal_1707> gotwig: put -c copy after -i ... [11:41] <gotwig> Failed to set value 'copy' for option 'c' [11:43] <gotwig> it seems my version is too old? [11:43] <durandal_1707> you are using wrong version [11:43] <durandal_1707> use ffmpeg [11:43] <gotwig> -sameq works for me [11:43] <gotwig> can I use that? even if its an older version [11:45] <durandal_1707> -sameq is not supposed to be used at all with -c copy [11:45] <durandal_1707> it does nothing [11:45] <durandal_1707> -sameq is not same quality [11:45] <durandal_1707> -sameq is same quantizier [11:45] <durandal_1707> it works when transcoding by pure luck [11:46] <durandal_1707> it sets quality to very high, so get that feeling you did something good when if fact you did nothing [11:46] <gotwig> :| [11:47] <gotwig> http://paste.ubuntu.com/1611965/ [11:48] <durandal_1707> gotwig: this is not avconv/Libav support channel [11:48] <durandal_1707> either leave or install ffmpeg [11:51] <gotwig> that is ffmpeg, isnt it? :( [11:51] <durandal_1707> no it is not, it is ffmpeg from Libav [11:51] <gotwig> I installed ffmpeg with apt-get install.. from my software resources [11:51] <durandal_1707> that is not ffmpeg but libav [11:51] <durandal_1707> ubuntu is bad [11:52] <gotwig> :( ! [11:52] <JEEB> basically it's an old ffmpeg from the libav project, the up-to-date binary in libav is called 'avconv' and works like a more up-to-date ffmpeg, also libav channel is #libav [11:52] <DEATH> Libav situation is annoying, too confusing for new users [11:52] <DEATH> Shot myself in the foot as well trying libav first not realizing what it really was [11:52] <DEATH> Well lesson learned [11:52] <gotwig> when I install just avconv, I dont am able to use ffmpeg [11:52] <JEEB> I think the worst part is how they kept the ffmpeg binary there without updating it, throughout at least 0.8 [11:52] <gotwig> when I install ffmpeg, I can use ffmpeg.. [11:53] <gotwig> I better should ask in #ubuntu... [12:01] <durandal_1707> gotwig: afaik ubuntu do not yet provide ffmpeg package [12:02] <gotwig> I grab the binary... [12:02] <durandal_1707> gotwig: you can download ffmpeg from www.ffmpeg.org ... [12:02] <gotwig> :) https://launchpad.net/~jon-severinsson/+archive/ffmpeg [12:04] <dentex> hi everyone. [12:05] <dentex> Just a little question on compiling flags... [12:05] <gotwig> actually ffmpeg in the newer version does give me the same results... [12:05] <dentex> what would be a minimum flag set to have just the ability to extract audio from mp4, webm, flv, 3gpp containers? [12:06] <dentex> --enable-decoders and --enable-demuxers ? [12:06] <durandal_1707> gotwig: i dont see full uncut output [12:06] <durandal_1707> and -c copy just works [12:07] <gotwig> http://paste.ubuntu.com/1612197/ [12:09] <durandal_1707> gotwig: 10.6 is very old version [12:09] <durandal_1707> dentex: and store it where? [12:10] <durandal_1707> you need muxer [12:10] <gotwig> its just 16 weeks old [12:10] <gotwig> I mean 12 weeks [12:10] <JEEB> yes, but 0.10.* as a version is already unfortunately old :P [12:11] <durandal_1707> gotwig: build date? [12:11] <dentex> you're right. I want to strip audio only. so "-vn -acodec copy $AUDIO_CODEC" [12:11] <gotwig> I get the static builds.. [12:11] <durandal_1707> dentex: in what container? [12:13] <gotwig> mummy, it works... [12:13] <gotwig> is that the right command? ./ffmpeg -ss 152 -i OUTPUT.mkv -c copy output3.mkv [12:14] <JEEB> yes, if you want to leave stream selection to ffmpeg (one video, one audio and not sure about subtitles) [12:14] <gotwig> lol, finally I see a nice output from ffmpeg... [12:14] <JEEB> anyways, that just means that it was fixed later than either libav 0.8 or ffmpeg 0.10 :3 [12:14] <gotwig> avconv does not offer that, just without any time formats [12:17] <gotwig> and ffmpeg is really very fast :( [12:17] <JEEB> remuxing generally is :) [12:46] <gotwig> And now I no longer have problems with audio offset [12:46] <gotwig> even when I dont use async 1 [12:47] <JEEB> yes, generally async 1 shouldn't be needed [12:49] <gotwig> I guess because of -c copy [12:49] <JEEB> I would rather guess at the much newer code that you are now testing :D [12:50] <FergusL> relaxed: thanks for your tip about PATH. I think I used a symlink instead [13:15] <_raven_> hi [13:15] <_raven_> how to merge some mp3-files together? [13:16] <gotwig> ffmpeg -i MP3_FILE -i MP3_FILE OUTPUT ? [13:17] <gotwig> no, sry, I am a noob... [13:17] <_raven_> i need any option with * [13:17] <gotwig> oh ok [13:17] <gotwig> so you need a loop? [13:17] <_raven_> not sure perhaps ffmpeg is able to handle* [13:17] <gotwig> I dont think so [13:19] <gotwig> something like that? http://ubuntuforums.org/showthread.php?t=2005963 [13:20] <gotwig> but that is for converting each one as individual... [13:20] <JEEB> that just re-encodes all files separately, yeah [13:20] <gotwig> how can I cut the end of the video? [13:20] <gotwig> like remove the last 20 seconds? [13:20] <JEEB> have fun calculating the right timestamp for -t [13:21] <JEEB> I don't think you can set it to minus-something to just cut stuff off the end [13:21] <JEEB> (mostly because in general ffmpeg doesn't know or care about the "length" until it hits the end of stream) [13:21] <_raven_> cat *.mp3 | ffmpeg ? [13:22] <JEEB> I would not recommend that, at least use mkvmerge to append them or something and then demux the stream :P [13:22] <JEEB> although depending on your luck that might or might not work [13:23] <DEATH> While on the subject: it would be handy to have an alternate -t syntax that specifies end offset instead of duration (same as -t if no -ss specified, subtracts -ss time from the specified value if -ss is specified). Less math to do in my brain. [13:24] <dentex> Hi. I wanted to re-issue my previous question, if I may. [13:47] <WillieNL> ola i have something weird with ffserver version 0.7.1 theire is no data send on the output stream untill i stop the input stream [13:47] <WillieNL> is this a known problem ? [13:48] <WillieNL> next to what is best advice for streaming audio only (aac) mp4 seems not always working without video [14:55] <WillieNL> wb burek [15:44] <makara> racking my brains with this math. I have 1920x1080 and want to author DVD (PAL 720x576). What steps to take? [15:44] <Mavrik> makara: first of all, your aspect ratio is problematic :) [15:45] <Mavrik> makara: widescreen DVDs are usually anamorphic [15:46] <WillieNL> or place the 4:3 content in a 16:9 frame with subtitle bar [15:47] <Mavrik> yeah, you can pad it aswell. Not as optimal of a situation :) [15:47] <WillieNL> well indeed even when my description is way wrong [15:47] <WillieNL> here i still do 16:9 -> 4:3 [15:47] <WillieNL> but visa versa is also possible [15:48] <makara> which does the DVD expand to reach 16:9 aspect? Height or width? [15:49] <Mavrik> makara: width [15:50] <Mavrik> makara: you just "squish" the 16:9 video to 720x576 [15:50] <Mavrik> then set SAR so it tells the decoder the video is 16:9 [15:50] <Mavrik> and it'll stretch it when playing back [15:50] <WillieNL> makara: an example http://forum.videohelp.com/images/guides/p1538554/aspectratio.jpg [15:52] <WillieNL> personal keep the aspect ratio as is else you endup with really tall or fat people onscreen :) [15:52] <Mavrik> WillieNL: that's why you set the SAR. [15:53] <WillieNL> well i'm no expect on ffmpeg since i can't make rtmp/rtsp restreaming functional for audio only [15:54] <makara> WillieNL, that easy? [15:54] <WillieNL> makara: well WowzaMediaServer aac audio fetch works but restream part stalls untill i stop the feeding then vlc starts playing the buffered content :S [15:54] <WillieNL> i render clueless at that stage [15:54] <makara> just resize to 720x576; create mpeg-ps; author with widescreen on (16:9) [15:55] <Mavrik> makara: just make sure your SAR is correct when resizing [15:55] <makara> Mavrik, what is SAR [15:55] <Mavrik> sample aspect ratio [15:55] <Mavrik> basically, it tells how to stretch each pixel [15:57] <Mavrik> makara: SAR has to be set so that "720 * SAR = 1024" [15:57] <Mavrik> (because your video should be 1024x768) [15:58] <Mavrik> er [15:58] <Mavrik> 1024x576, sorry :) [16:01] <JEEBsv> something munch, I am expecting that your HD source is BT.709 and with a 1:1 aspect ratio: -vf colormatrix=bt709:bt601,scale=width=702:height=576,pad=width=720:height=576,setsar='16:11' [16:01] <JEEBsv> *something to munch at [16:05] <JEEBsv> (this contains some of the 'dirty secrets' of certain things that are not exactly how they look like) [16:06] <Mavrik> hmm, so colormatrix conversion is necessary? [16:07] <JEEBsv> well, yes -- naturally most people just wouldn't care about most of this stuff 8) After all, ffmpeg does have a PAL DVD -target option, which would take care of most of it all 8) [16:10] <Mavrik> JEEBsv: btw, why the black padding with anamorphic video? [16:11] <JEEBsv> http://www.x264bluray.com/home/576p-pal (DVD uses the same aspect ratios) [16:11] <JEEBsv> also a calculator that uses the "exact values" instead of their simplifications http://ps-auxw.de/cgi-bin/ar-calc.pl [16:12] <JEEBsv> basically the 702x576 part of the picture is going to be widened to 1024x576, and the rest goes further [16:13] <brontosaurusrex> JEEB; continued from yesterday: the player (tv) does play "bluray" vc1 mkvs, only it cant read the "bluray" subs ... [16:13] <JEEBsv> so if you want to hifist, you can do it like this (resize to 702x576, and then pad), if you just want to get off easily, you can just resize to 720x576 :) [16:13] <brontosaurusrex> and the bluray subs are always bitmaps? [16:13] <JEEBsv> that is, if your source content is 16:9 [16:13] <JEEBsv> brontosaurusrex: not always, but pretty much nothing handles the few cases of text-based subtitles [16:14] <JEEBsv> I think only some French BDs actually used the text-based subtitles [16:14] <brontosaurusrex> so is there a way to hardburn those on a non-windows machine, using ffmpeg only perhaps? [16:16] <JEEBsv> as I said, pretty much nothing except for payware players support that part of blu-ray subtitles thanks to the fact that they are barely used [16:16] <brontosaurusrex> no, i mean bitmap based [16:17] <JEEBsv> not that I know [16:17] <JEEBsv> ffmpeg can render them, but I don't think it could hardcode them [16:20] <brontosaurusrex> i mean transcoding from vc1+subs > h.264 [18:13] <serard> hello [18:13] <serard> I'm looking to replace mp4split use using ffmpeg, any howto or link ? :) [18:27] <Fjorgynn> serard: for what? [18:29] <serard> to get some examples of command line arguments [18:31] <Fjorgynn> for what? [18:31] <Fjorgynn> I don't know what mp4split does or what you're using it for [18:31] <Fjorgynn> My car doesn't work. It is red. [18:35] <juanmabc> split mp4s? [18:36] Action: juanmabc guesses hard [18:44] <WillieNL> Fjorgynn: well more examples on how to stream only audio in a couple formats will be welcome and solve the 1 hr trial and error [18:44] <WillieNL> here ffmpeg feeding to ffserver still is not sending any data untill stop the feed [18:44] <WillieNL> or only to the first client [18:45] <WillieNL> highly frustrating if unsure how to do it right :) [18:46] <WillieNL> dunno if these bugs are fixed afther version 0.7.1 but dinner time here [18:46] <Fjorgynn> aha [19:20] <pskosinski> ffmpeg -f x11grab -r 25 -i :0.0 file.mpg [19:20] <pskosinski> Will it give me "raw" video? [19:21] <pskosinski> Or it will be compressed using a codec. How to get as raw video as I can? [19:22] <pskosinski> Because I want to "modify" it a bit, and if I compress it to webm and then edit result is awful [19:22] <klaxa> pskosinski: -c:v rawvideo [19:23] <klaxa> i'm not sure the mpeg container supports it though, i think avi can handle it [19:23] <pskosinski> ok, thanks, I will try :) [19:24] <klaxa> you will need a reasonably fast disk though (ram, ssd?) otherwise you can use a lossless videocodec for compression, needs more cpu though [19:25] <sweb> http://superuser.com/questions/547027/ffmpeg-multiple-video-filter [19:25] <pskosinski> Unrecognized option 'c:v' [19:26] <klaxa> pskosinski: update your ffmpeg :P (or use -vcodec) [20:01] <pskosinski> So, I would like to cut beginning and end of video. I used: ffmpeg -i in.mkv -ss 8 -t 541 -vcodec copy -an oot.mkv and it is cutting but there is one problem: when I play video in mplayer it shows that video starts from 8th second. Can I do something with that? [20:02] <pskosinski> Without re-encoding. [20:03] <pskosinski> Well, looks like for those 8 second video is staying in one place& [20:03] <pskosinski> seconds * [20:15] <llogan> pskosinski: you mean the image is still for 8 seconds and then plays? [20:15] <pskosinski> Yes [20:15] <pskosinski> But I just reencoded and I don't see quality lose, so kinda no problem now. :p But would be nice to know if it's possible anyway [20:17] <pskosinski> Ah& It seems that I have a bit old ffmpeg, I have Debian Wheezy& Maybe first I [20:17] <pskosinski> I'll try to update * stupid enter [20:31] <llogan> pskosinski: this could help https://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuideHardy [20:31] <llogan> or a static build is an option [20:32] <pskosinski> Yeah, thanks. :) I see that Debian switched to libav& [20:38] <llogan> that has caused much confusion for users [21:16] <creep> hey http://www.androidinabox.com/mini-joystick-capacitive-touch-screen-joystick… what do you think about this stuff? [22:26] <taylanub> The ffplay, ffserver etc. executables aren't supposed to be very big, right ? 140M here, I'm guessing dynamic linking was disabled because I'm on OpenBSD. [22:28] <cheeseboy> [image2 @ 0xbe2d00] Could not get frame filename number 2 from pattern '/var/log/encodes/file.png' [22:29] <cheeseboy> i want to write to same file every time. how I accomplish this? [22:32] <cheeseboy> http://pastie.org/6074719 [22:40] <beastd> cheeseboy: not sure i understood you correctly. but please look at this http://ffmpeg.org/ffmpeg-formats.html#image2-1 and check updatefirst option [22:40] <saste> cheeseboy, -updatefirst [23:06] <cheeseboy> ffmpeg -f yuv4mpegpipe - -i - -updatefirst 1 -f image2 "/srv/http/file.png" | x264 - why isnt x264 recivng pipe? [23:09] <klaxa> because ffmpeg isn't writing to stdout [23:10] <klaxa> the first '-' is somewhat misplaced and incorrect completely i think [23:10] <klaxa> what are you trying to do even? [23:12] <sacarasc> cheeseboy: ffmpeg [input options] -i input [output options] output [23:14] <cheeseboy> ah [00:00] --- Wed Feb 6 2013

1 0

ffmpeg-devel.log.20130204
by burek 05 Feb '13

05 Feb '13

[00:02] <burek> michaelni, these powerpcs also need a different timeout value? [00:04] <michaelni> burek, probably [00:10] <burek> judging by this history log, it seems powerpcs are also stuck or something: http://fate.ffmpeg.org/history.cgi?slot=powerpc-linux-gnu-gcc-4.3.5 [00:11] <burek> they used to provied results every couple of hours [00:11] <burek> provide* [01:30] <michaelni> burek, maybe, i have no access to them so i cant tell ... [04:19] <cone-298> ffmpeg.git 03Michael Niedermayer 07master:91debec976d2: swr/resample: fix filter rounding and cliping for s32 [04:19] <cone-298> ffmpeg.git 03Michael Niedermayer 07master:b8c55590d54d: swr/resample: fix integer overflow, add missing cast [04:19] <cone-298> ffmpeg.git 03Michael Niedermayer 07master:24244cec2fdd: swr-test: Fix clip to 32bit [04:19] <cone-298> ffmpeg.git 03Michael Niedermayer 07master:be989f923678: swr: reorder init code to make rematrix status available earlier [04:19] <cone-298> ffmpeg.git 03Michael Niedermayer 07master:b5f544a0f998: swr: Dont use floats for S32->S32 when possible [11:11] <elk> I'm trying to get ffmpeg to read from a file which is growing (ongoing recording) but to not exit when it reaches the files EOF, I have tried retuning AVERROR(EAGAIN) instead of a AVERROR_EOF when it reaches the EOF.. [11:12] <elk> Because I'm reading from a mov-file i looked at mov.c in libavformat but I can find where the actual file is read [11:12] <elk> I was thinking of peeking into stream and wait if the next data is a eof [11:58] <ubitux> JEEB: arg @captioning :) [12:02] <mateo`> ubitux: \o/ [12:02] <ubitux> :) [12:16] <JEEB> ubitux, ARIB captions are quite fabulous, are they not 8) [13:04] <cone-831> ffmpeg.git 03Anton Khirnov 07master:d9df93efbf59: mpegvideo: simplify REBASE_PICTURE [13:04] <cone-831> ffmpeg.git 03Michael Niedermayer 07master:a2bff49e26a9: Merge remote-tracking branch 'qatar/master' [13:31] <cone-831> ffmpeg.git 03Michaël Cinquin 07master:60f06f986f81: libopenjpegenc: make dci compliant j2c [13:31] <cone-831> ffmpeg.git 03Michael Niedermayer 07master:c67ddb56da55: Merge remote-tracking branch 'mjbshaw/master' [13:32] <ubitux> JEEB: well, it's not surprising to see such thing [13:38] <JEEB> it's been around for quite a while but I finally found a kind of open source renderer for it [13:39] <JEEB> caption2ass has been around for a while, but the conversion is shoddy at best [14:37] <cone-831> ffmpeg.git 03Michael Niedermayer 07master:63c61f0be036: vc1dec: Fix 2 "warning: may be used uninitialized in this function" [14:37] <cone-831> ffmpeg.git 03Michael Niedermayer 07master:353bd158f5d4: sws: Fix "warning: comparison of distinct pointer types lacks a cast" [14:41] <cone-831> ffmpeg.git 03Rudolf Polzer 07master:81f2549ec9ab: lavfi/showspectrum: display multiple channels in separate row [15:01] <durandal_1707> michaelni: what patch is supposed to fix #2227 [15:05] <ubitux> divVerent: you should add yourself in the showspectrum filter copyright [15:26] <ubitux> showspectrum=mode=separate:color=intensity:slide=1:scale=cbrt i e this [15:26] <ubitux> thanks divVerent :) [15:27] <ubitux> we should find a friendly way of making the playback fluid& [15:49] <michaelni> durandal_1707, [FFmpeg-devel] [PATCH 2/2] ffmpeg: Avoid matching attachment pics to video on wildcard maps. [15:57] <durandal_1707> what is policy about pathes that quatar folks just forget to apply? [15:57] <durandal_1707> *patches [17:07] <michaelni> durandal_1707, i dont think theres a policy about that, but which patch ? [17:43] <burek> can anyone please help me to properly answer this question about security vulnerabilities in ffmpeg 1.0.2: http://ffmpeg.gusari.org/viewtopic.php?f=16&t=818 [17:47] <burek> also, if someone knows what is a NEON simd, I would appreciate any help on that too: http://ffmpeg.gusari.org/viewtopic.php?f=8&t=819 [17:48] <JEEB> NEON simd is just simd for armv7 cpus [17:48] <iive> burek: simd - single instruction multiple data. NEON is used on arm cpu's [17:48] <JEEB> he is expecting some kind of hw H.264 encoding/decoding or whatever, and the reality hits him :P [17:48] <JEEB> yay armv7 [17:48] <burek> i see, how can someone check if ffmpeg is using simd or main cpu for encoding [17:49] <burek> that's assentially what the guy is asking [17:49] <JEEB> SIMD is main cpu [17:49] <burek> oh [17:49] <JEEB> it's just instructions that deal with "multiple data" at the same time (like SSE/MMX and their derivatives) [17:50] <burek> yes, i understand, i guess the guy thought it was some kind of additional gpu or something [17:50] <JEEB> yeah [17:50] <iive> MMX and SSE are SIMD [17:50] <JEEB> yup [17:50] <JEEB> for IA32 and x86_64 [17:50] <burek> ok, at least now i know what to answer there :) thank you both :) [17:52] <wm4> burek: didn't we have this question here on this channel recently [17:52] <JEEB> yeah [17:52] <JEEB> I remember a similar guy coming around here [17:53] <burek> probably, i didnt follow the channel in detail, but this one was from the forum, so it might be the same guy perhaps [18:04] <wm4> which mp3 decoder is actually preferred, "mp3" or "mp3float"? [18:08] <wm4> I would have thought it's mp3float, but both ffplay and ffmpeg seem to select "mp3" by default [18:09] <wm4> avcodec_find_decoder(AV_CODEC_ID_MP3) returns "mp3" too [18:12] <burek> if anyone knows this thing about security vuln. in ffmpeg 1.0.2 ( http://ffmpeg.gusari.org/viewtopic.php?f=16&t=818 ), just ping me and I'll see the message later :) and thanks in advance :) [18:23] <burek> one more thing :) is there any relation between "FFserver version SVN-r25767" and some git version currently hosted on ffmpeg.org ? [18:24] <burek> i.e. is there a way to find out what is the git commit hash of that svn version? [18:26] <wm4> burek: it was 47d2ddca802f4c1bc4b454c5ac40f06f79b740a0 [18:27] <wm4> the svn commits imported in git always have this in their log message: "Originally committed as revision 25767 to svn://svn.ffmpeg.org/ffmpeg/trunk" [18:27] <wm4> so you can just do git log and search by text [18:27] <burek> oh great [18:27] <wm4> and that version is from 2010 (lol) [18:27] <burek> thanks [20:31] <michaelni> burek, ffmpeg 1.0.2 security fixes are listed at ffmpeg.org/security.html [21:13] <llogan> michaelni: i'd like to remove ffserver-user today: rmlist ffserver-user [21:13] <llogan> that should keep the archives [21:13] <llogan> actually i'll send out one more notice beforehand... [21:15] <michaelni> llogan, assuming theres consensus on this, is there anything i will need to do ? [21:16] <Compn> move all ffserver-user subscribers to ffmpeg-user , but turn off messages, so they are allowed to post, but dont recieve ffmpeg-user mails [21:16] <Compn> ? [21:17] <llogan> michaelni: i asked on -devel if there were no objections. only saste replied saying he supported this. i made an [announce] on ffserver-user describing the situation on jan 27. [21:18] <llogan> Compn: i am not migrating users due to the volume of messages in ffmpeg-user. they will have to do that themselves as mentioned in the announce. [21:19] <michaelni> so i dont have to do anything ? :) [21:19] <llogan> michaelni: you will have to do the actual list removal with "rmlist". [21:19] <michaelni> work :( [21:19] <llogan> i know...i would do it if i could. [21:20] <llogan> you should check first with "rmlist --help" [21:20] <llogan> in case the defaults/syntax differs from what i refered to [21:21] <llogan> i also already updated contact [21:22] <michaelni> isnt it better to leave the list and forward mails to ffmpeg-user ? [21:23] <michaelni> that is if someone from somewhere uses the ML address to post a mail [21:23] <michaelni> or reply to an old mail [21:23] <llogan> if you prefer that method you can do it that way [21:23] <michaelni> you can do that yourself i think :) [21:23] <llogan> i didn't see an option to set that up in the admin panel, but maybe i missed it [21:24] <michaelni> remove all subscribers, put the ffmpeg-user list as only subscriber to ffserver-user then it would get mails sent to it [21:27] <llogan> there did not seem to be a trivial method to disable subscribers. [21:28] <llogan> and considering the low-volume of ffserver i thought rmlist was the best tradeoff of work vs potential user convienence [21:28] <llogan> *disable subscribing [21:29] <michaelni> well i can rmlist it if thats what the ffserver-user people want [21:35] <llogan> alternatively, you can setup an alias in postfix to forward ffserver-user mail to ffmpeg-user. [21:36] <llogan> but truthfully i don't see much "reply to old mail" happening. [23:05] <cone-831> ffmpeg.git 03Michael Niedermayer 07master:96a08d862730: wmavoice: silence may be used uninitialized warnings [23:05] <cone-831> ffmpeg.git 03Michael Niedermayer 07master:fc0d069feb69: flacdec: silence several "warning: X may be used uninitialized in this function" [00:00] --- Tue Feb 5 2013

1 0

ffmpeg.log.20130204
by burek 05 Feb '13

05 Feb '13

[00:01] <taylanub> Is Diego Biurrun in this channel ? [00:02] <taylanub> I'll mail 'em I guess. [01:05] <pskosinski> Hi. Can I increase/decrease video speed while converting it? [01:07] <pskosinski> Oh, I was goind to ask what I should use as PTS& Now I see that it's variable, no need to replace. >.> Ok, nvm, thanks. [01:08] <pskosinski> Oh, but result for webm video is very bad& [01:42] <Jiyuu|C2> [00:40:15] <@saste> burek, can't be hader than subtitles=in.mkv [01:42] <Jiyuu|C2> tried that [01:42] <Jiyuu|C2> doesnt work [01:42] <Jiyuu|C2> in the documentation i dont see any mention of something like that [02:09] <glisignoli> Hello, I'm trying to use avconv to convert a mp3 to a 64k ogg. I'm doing "avconv -i somefile.mp3 -b 64k output.ogg" except the file output ends up as a flac, and much larger [02:10] <sacarasc> #libav is the avconv channel. [02:10] <glisignoli> ah sorry [02:10] <sacarasc> Hint: -c:a libvorbis [02:12] <glisignoli> Nice! [02:28] <Animawish> so I just transcoded an mkv file to mp4, and the video was fine, but the audio went to shit. I used the command "ffmpeg -i file.mkv newfile.mp4". What should I do to make the sound survive the conversion? [02:28] <sacarasc> -c:a copy [02:29] <sacarasc> If you have an MP4 compatible audio stream, anyway. [02:29] <Animawish> how do I know if I have an mp4 compatible audio stream? [02:30] <sacarasc> ffprobe -i file.mkv [02:30] <sacarasc> Then paste to a pastebin type site and someone will tell you. [02:31] <Animawish> http://pastie.org/6035603 [02:32] <kcm1700> I want to know the required lifetime of the 'extradata' field of AVCodecContext [02:32] <kcm1700> Is it okay to throw away after avcodec_open2? [02:33] <sacarasc> Animawish: I don't think AC3 can go in MP4... [02:33] <Animawish> hm [02:33] <Animawish> well it doesn't have to be mp4 [02:33] <Animawish> idek if i'm doing the right thing [02:33] <Animawish> i'm just trying to get these video files into a burnable format [02:34] <sacarasc> Why is it not burnable now? Or do you mean playable in some kind of device? [02:34] <Animawish> yeah. i want to burn them to DVDs to put in a DVD player to play on an awful TV that doesn't have HDMI in :P [02:35] <Animawish> hate that TV but whatareyagonnado [02:35] <sacarasc> What does the DVD player support? [02:36] <Animawish> idk. I was thinking I'd just use like, iDVD or equivalent to burn it [02:36] <kcm1700> I'm talking about the situation of decoding. [02:38] <sacarasc> Animawish: What is iDVD? :D [02:39] <Animawish> a first party DVD burning program in mac os [02:39] <sacarasc> Does it just burn, or does it convert? [02:41] <Animawish> i'm not sure [02:41] <Animawish> like [02:41] <Animawish> i can't use mkvs with it for sure [04:04] <Soho_> hi all [04:05] <Soho_> I am new in ffmpeg field [04:05] <Soho_> I have question about :how to get video frame from usb web cam then straming to remote client [04:05] <Soho_> the video format is h.264, [04:06] <Soho_> is there any example about this kind of application [04:07] <Soho_> it is not necessary to encode for video frame, because the content has been encoded by webcam [04:07] <Soho_> Does anyone have idea ? [04:15] <Soho_> Can I send video frame from usb webcam to network without transcode ? [04:18] <Soho_> And [04:18] <Soho_> if the video is H.264 and audio is PCM from USB web CAm, [04:19] <Soho_> which container is suitable for streaming by ffserver? [04:19] <Soho_> Does anyone have idea about my question? [04:20] <Soho_> I would like to send video stream to remote client by ffmpeg, is there any exmaple for reference? [08:08] <elkng> I use command ffmpeg -i "1.mp4" -vcodec mpeg4 -vb 1500k -s 650x276 -ab 128k -ar 44100 -ac 2 -acodec libmp3lame -y -f avi "1.avi" to convert video from mp4 to avi and thats the error output: http://sprunge.us/ZdbW what is wrong ? [08:35] <relaxed> elkng: replace "-s 650x276" with "-filter:v scale=650:-1" [08:36] <relaxed> you may need to specify an output frame rate too. [08:50] <SubJunk> How can I load videos with commas in the name? I've tried using -i file:'filename.mpg' with no luck [09:02] <relaxed> SubJunk: -i ./input,name.ext [09:02] <elkng> or './input,name.ext' ? [09:12] <SubJunk> Thanks, got it working by escaping the comma with a backslash :) [09:12] <SubJunk> Is it normal for subtitles to ignore seeking? Like if I use -ss 100 then the video and audio start 100 seconds in but the subtitles start at the beginning [09:31] <XATRIX> Guys, can you advice me what do i do wrong ? I'm tryting to glue/concat 3 mp3 files with voice [09:31] <XATRIX> Into a one, result.mp3 [09:32] <XATRIX> Encoding does good, but the result mp3 is unstable on a few browsers + it has broken pitch rate [09:32] <XATRIX> The command i issue to endoe it is : ffmpeg -i "concat:en1.mp3|ru1.mp3|ru2.mp3" -acodec copy /mnt/www/rf.ua/result.mp3 [09:32] <XATRIX> You can check the result mp3 by http://rf.ua/result.mp3 [09:33] <XATRIX> Also you can check the initial (good ones) mp3s by http://rf.ua/en1.mp3 http://rf.ua/ru1.mp3 http://ru2.mp3 [09:33] <XATRIX> English voice is ok, but the next 2 mp3s pitch rate failed [09:34] <XATRIX> What can be wrong ? [09:34] <XATRIX> http://fpaste.org/gxcT/ - the log of encoding process [09:52] <Soho> hi [09:52] <Soho> I got the trouble, when I build ffmpeg 1.1.1 [09:52] <Soho> ffmpeg is not build [09:53] <Soho> but fferver and fforibe are built [09:53] <Soho> the config.h shows #define CONFIG_FFMPEG 0 [09:53] <Soho> does anyone have idea? [09:53] <Soho> that I can use ffmpeg [09:53] <Soho> that I canNOT use ffmpeg [09:55] <Soho> Does someone help me? [10:51] <XATRIX> Sorry guys, were network connection problems... [10:52] <XATRIX> So any ideas on my question of ffmpeg and concat a few files ? [11:14] <elk> I'm trying to get ffmpeg to read from a file which is growing (ongoing recording) but to not exit when it reaches the files EOF, I have tried retuning AVERROR(EAGAIN) instead of a AVERROR_EOF when it reaches the EOF.. [11:14] <elk> Because I'm reading from a mov-file i looked at mov.c in libavformat but I can find where the actual file is read [11:14] <elk> I was thinking of peeking into stream and wait if the next data is a eof [11:25] <XATRIX> I have a trouble with ffmpeg encoding of a few files, can you look at ? http://fpaste.org/O8cf/ [12:20] <XATRIX> Guys, please help me with my trouble, i'm really stucked here ... :( http://fpaste.org/O8cf/ [13:08] <XATRIX> As long as i see my troubles is that en1.mp3 has 16000Hz freq, and ru1.mp3&ru2.mp3 has 22050 [13:09] <XATRIX> So when it does concat of en1.mp3+ru1.mp3+ru2.mp3 it prolong 16000 freq along the whole resulting mp3 [13:09] <XATRIX> Can't change it :( [13:09] <XATRIX> Any ideas ? [14:07] <sacarasc> XATRIX: You'd have to resample before concatenating... [14:14] <XATRIX> sacarasc: Any possibility to resample while concat ? [14:15] <sacarasc> Only if you want to re-encode all of them, I think. [14:16] <XATRIX> Yes, i tried to reencode all of them to -ab 32kb -ar 22050 [14:16] <XATRIX> But the resulting mp3 has the same shit [14:16] <p4plus2> If I had a set of 6 videos and wanted to arrange them in a 3x2 grid how would one go about this in ffmpeg? [14:17] <XATRIX> I can only fix it out with recoding the 'broken' mp3s from 16000 -> 22050 and then apply concat [14:17] <p4plus2> (optionally there can be a 7th video as a background generated black if needed -- not a problem) [14:17] <XATRIX> But isn't there any possibility to resample/recode it on fly with only one command ? [14:58] <Zeeflo> hi guys [14:58] <Zeeflo> the -sn option leaves out subtitles in mkv's [14:58] <Zeeflo> what the option if I dont want chapters as well? [15:00] <JEEB> see the documentation on ffmpeg.org about metadata, I think chapters are included in that part of it [15:00] <JEEB> since it's not really a "codec" I don't think you have a simple option like an/vn/sn for it [15:00] <Zeeflo> hmm [15:00] <JEEB> as those are what "codecs" handle [15:00] <JEEB> there was something to null all metadata or not copy it around [15:00] <JEEB> that might be what you want [15:01] <Zeeflo> that would be useful as well [15:03] <JEEB> oh [15:03] <JEEB> there's something separate [15:04] <JEEB> -map_chapters [15:04] <JEEB> http://ffmpeg.org/ffmpeg.html [15:04] <JEEB> -map_chapters input_file_index (output) [15:04] <JEEB> "Copy chapters from input file with index input_file_index to the next output file. If no chapter mapping is specified, then chapters are copied from the first input file with at least one chapter. Use a negative file index to disable any chapter copying." [15:04] <JEEB> see the last sentence [15:17] <Zeeflo> JEEB, use a negative file index? [15:17] <Zeeflo> what would that look like? [15:18] <JEEB> -map_chapters -1 ? [15:19] <happarappa> Hello. I am trying to convert some .wavs to .flacs to save storage space with ZERO loss of quality. Is ffmpeg.exe -i test.wav -sameq test.flac safe for doing this? [15:20] <sacarasc> No need for -sameq. [15:20] <Zeeflo> ahh like that [15:20] <happarappa> I cannot HEAR a difference to the resulting file, but Winamp says "14H" in the case of the WAV and 925 kbps for the FLAC. [15:20] <sacarasc> Flac is lossless. [15:20] <D0minat0r> ffmpeg -vcodec copy -acodec copy -ss 0 -t 00:01:00 -i VIDEO_SOURCE.mp4 VIDEO_PART.mp4 That command will be the easiest way for me if i want to 1min split video_source.mp4 to video_part.mp4, right? [15:21] <happarappa> sacarasc: Well, it does say it ignores the "-sameq". [15:21] <happarappa> But if FLAC is lossless, why doesn't Winamp say "14H" in the FLAC file? [15:21] <sacarasc> happarappa: If you have a text file and you put it into a zip file, does it not get smaller without losing anything? FLAC is the same thing, but for music. [15:21] <Zeeflo> because winamp sucks [15:21] <happarappa> sacarasc: I know what lossless means... [15:21] <D0minat0r> And i will be able to split everything that ffmped supports? [15:21] <sacarasc> It reduced the bitrate. [15:21] <D0minat0r> I will get audio and everything? [15:22] <sacarasc> So it won't show the same one. [15:22] <happarappa> So basically, you are saying that it reduced the bitrate to the smallest possible WITHOUT losing a single bit of info? [15:22] <sacarasc> Yes. [15:22] <happarappa> Ah. Makes sense. [15:22] <happarappa> So CD quality is in other words 925 kbps. [15:22] <sacarasc> Well, maybe not the smallest, as you didn't tell it which compression setting to use. [15:22] <happarappa> I who thought 320 kbps sounded great... [15:23] <sacarasc> And it varies between tracks. [15:23] <happarappa> That last thing you said confused me entirely. [15:23] <happarappa> "as you didn't tell it which compression setting to use." [15:23] <happarappa> Why would it not use the maximum? [15:23] <sacarasc> It can compress lightly or heavily. [15:23] <sacarasc> Because ffmpeg's defaults aren't always what you want. [15:23] <happarappa> What possible reason could there be to not do it as efficiently as possible? [15:24] <sacarasc> It uses SLIGHTLY more CPU time to encode/decode higher compressed audio. [15:24] <happarappa> I suppose... in 1994. These days, the computers barely need to grasp for air to do it. [15:24] <happarappa> What option would that be, BTW? I would like to make them as small as possible. [15:25] <sacarasc> -q 9, I think, but I am unsure. [15:25] <happarappa> :( [15:25] <happarappa> Will the difference be noticable? [15:25] <happarappa> I mean in resulting file size. [15:25] <durandal_1707> happarappa: what is your issue? [15:25] <happarappa> Or is this an exercise in futility? [15:26] <sacarasc> happarappa: You could get it down another few hundred kbps... [15:26] <sacarasc> Depending on track. [15:26] <happarappa> durandal_1707: Um... I guess I'm trying to figure out how to convert a .wav to .flac with the best possible compression without losing any data. [15:26] <sacarasc> It's variable bitrate, so could be anywhere. [15:26] <durandal_1707> -q is for lossy not for lossless [15:26] <durandal_1707> happarappa: -h full and look for flac [15:27] <durandal_1707> there are options that control compression [15:27] <happarappa> But... [15:28] <durandal_1707> what? [15:29] <durandal_1707> use -prediction_order_method search [15:29] <durandal_1707> other options like -q do nothing for lossless codecs [15:29] <happarappa> I am confused about what you mean. [15:29] <happarappa> "use -prediction_order_method search" is not English to me. [15:30] <sacarasc> ffmpeg -i foo.wav -prediction_order_method search out.flac [15:32] <happarappa> Can't find a mention of prediction_order_method in http://ffmpeg.org/ffmpeg.html [15:32] <sacarasc> ffmpeg -h full [15:32] <sacarasc> That would have told you about it. [15:32] <sacarasc> ffmpeg -h full > blah.txt #so you can read it after. [15:33] <happarappa> It's just that it's not PATHed, so it's very annoying to call. [15:33] <happarappa> But why is the HTML manual not complete? [15:34] <happarappa> I have it now. Reading. [15:36] <sacarasc> The HTML docs aren;'t complete because just a text output of the full help is over 90kB long. :D [15:37] <happarappa> Hehe. 8 kB larger was the file that was not -prediction_order_method searched. [15:38] <happarappa> At least for the test track. [15:38] <happarappa> I would never in a million years have guessed that I would need to use that parameter/argument. [15:38] <happarappa> Why can't the defaults be optimized? [15:40] <durandal_1707> because defaults are to be fast [15:41] <durandal_1707> and flac can be documented in encoders, but it just is not.... [15:41] <durandal_1707> could file bug report [15:42] <happarappa> "flac can be documented in encoders"? [15:46] <durandal_1707> happarappa: encoders is file that document encoders present in ffmpeg [15:47] <durandal_1707> see http://ffmpeg.org/ffmpeg-codecs.html#Audio-Encoders [15:54] <Youka> How i can extract attachments and subtitles as raw data with the API? It seems that AVPacket doesn't contain attachments and subtitles just as event lines (which seems to be buggy because in an example the first 15 lines are missing) [15:57] <happarappa> durandal_1707: That is confusing. [15:58] <durandal_1707> what is confusing? [16:01] <happarappa> "encoders is file that document encoders present in ffmpeg" [16:01] <durandal_1707> ok full name is doc/encoders.texi [16:02] <happarappa> Sigh... [16:02] <happarappa> Are people getting more vague and confusing? [16:02] <happarappa> It feels like it to me. [16:05] <durandal_1707> if you are not interested in helping you should not care at all for "confusing lines" [16:06] <durandal_1707> actually the easiest way for you to help is filling bug report for anything you feel should be improved/documented/etc... [16:07] <happarappa> I don't understand what you are saying. [16:07] <happarappa> That's why I ask. [16:07] <retard> if you have a general feeling of other people getting more vague and confusing the root cause might not be the other people [16:07] <happarappa> Or it might actually be just that. [16:08] <retard> or it might not [16:08] <retard> like i said [16:08] <happarappa> But it is. [16:08] <retard> not necessarily [16:08] <happarappa> Not necessarily, but it is. [16:10] <durandal_1707> funny [16:13] <retard> it is... but is it really? [16:50] <D0minat0r> Got so many errors when i try to use ffmpeg ;< [16:50] <D0minat0r> isnt it just to apt-get install ffmpeg [16:58] <D0minat0r> configuration: --extra-version='4:0.7.6-0ubuntu0.11.10.3' --arch=amd64 --prefix=/usr --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-vaapi --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --enable-shared --disable-static [16:59] <D0minat0r> did i miss'd something or wtf? ;< [17:05] <D0minat0r> cmon.... someone? [17:33] <D0minat0r> still no one here? [17:49] <Youka> ask in stackoverflow, ffmpeg forum or google for it when nobody here can help you [17:50] <burek> D0minat0r probably because [17:51] <burek> you didnt read the topic [17:51] <burek> and also didnt provide log [17:51] <burek> -ffmpeg command +configure command [18:07] <WillieNL> ola i wonder what happend with libva ? [18:07] <WillieNL> original project host seems to be offline [18:07] <burek> which host [18:08] <WillieNL> burek: http://www.splitted-desktop.com/~gbeauchesne/libva/ [18:09] <burek> did you try google [18:09] <burek> for example http://cgit.freedesktop.org/libva/ [18:09] <WillieNL> yes found via other sources but unsure if the project is still maintained [18:09] <burek> btw, isn't there some kind of irc channel for that or ML or something [18:09] <burek> i mean, is libva related to ffmpeg in some way or what [18:10] <WillieNL> dunno ffmpeg costh me half my day here due a old bug in old release... [18:10] <WillieNL> ffserver from 0.6.x -> Aspect ratio mismatch between encoder and muxer layer [18:10] <WillieNL> known issue [18:10] <burek> oh man... 0.6... [18:11] <WillieNL> well before we argue i only need to test something on this old rig [18:11] <WillieNL> somehow ffserver is having a issue wich made update ffmpeg more usefull then reinstall a complete machine [18:11] <burek> well i can understand you need that exact version, i just feel sorry, knowing what all does it bring with itself :S [18:11] <WillieNL> well it sucks if need to debug a remote rtmp host like this yes [18:11] <WillieNL> :S [18:12] <WillieNL> well it do work but no meta data [18:18] <WillieNL> burek: do ffmpeg hold old releases on a ftp or some like that ? [18:18] <D0minat0r> I did try everything that the wiki said, still nothing so [18:19] <D0minat0r> Got alot of errors thar i havent enable everything [18:21] <burek> WillieNL, what do you mean by "old releases" [18:21] <burek> you do use git, right? [18:21] <burek> D0minat0r, right above what I already answered you earlier [18:21] <WillieNL> -> ffmpeg-0.6_p28715.tar.bz2 [18:21] <burek> read that [18:22] <burek> WillieNL, when you type ffmpeg in your shell [18:22] <burek> what is the version string that you get in return [18:22] <WillieNL> FFserver version SVN-r25767, [18:22] <burek> oh, i see.. well, it's just too old [18:23] <burek> i was hoping it was a version after the transition to git [18:23] <burek> so you could simply checkout that version [18:23] <burek> wait, let me ask other people [18:23] <WillieNL> burek: this is my problem -> http://roundup.libav.org/issue2467 [18:24] <D0minat0r> burek http://pastie.org/6044359 [18:24] <D0minat0r> that what i got [18:24] <burek> D0minat0r, you are not using ffmpeg [18:24] <burek> you are using a fork of ffmpeg [18:24] <burek> and it's a very old one [18:25] <burek> either ask your question in a support channel for that fork project [18:25] <burek> or use real ffmpeg [18:25] <D0minat0r> Oh, shit, sry [18:25] <D0minat0r> apt-get install ffmpeg was the wrong one then [18:26] <burek> yes it was [18:27] <D0minat0r> Sry, [18:27] <burek> there was an issue with ubuntu/debian community which they didnt properly resolve [18:27] <burek> causing all that mess you are experiencing right now [18:27] <D0minat0r> Oh i see, so i have to build it from the source? [18:27] <beastd> D0minat0r: Debian does not have FFmpeg anymore because the maintainer of FFmpeg Debian packages is one of the forkers. [18:28] <D0minat0r> But i use ubuntu server [18:28] <burek> ubuntu is based on top of debian [18:28] <beastd> it is related. but look here [18:28] <beastd> D0minat0r: http://ffmpeg.org/download.html [18:28] <burek> WillieNL, good new [18:28] <burek> s [18:28] <burek> you can find a corresponding git version based on your svn version [18:28] <D0minat0r> Hm i see [18:29] <D0minat0r> But do you guys know any other program like ffmpeg? [18:29] <D0minat0r> I just need to make samples [18:29] <burek> if you check out the official ffmpeg's git log, you'll see something like this in one of the commits: "Originally committed as revision 25767 to svn://svn.ffmpeg.org/ffmpeg/trunk" [18:29] <burek> which happens to be this one [18:29] <burek> 47d2ddca802f4c1bc4b454c5ac40f06f79b740a0 [18:30] <WillieNL> burek: ahh but if can upgrade to 0.7.1 then also ensured not bumping into the same bug i hope while unsure if can play/stream/transcode mp4 without h264 [18:30] <D0minat0r> Ah okey [18:31] <burek> WillieNL, using git you can always move forward/backward with different ffmpeg versions and test your issues [18:31] <burek> with tar.bz2 file, you just have one version [18:32] <WillieNL> burek: jup but gentoo is not always using git in theire ebuilds [18:32] <burek> D0minat0r, it's actually very easy to compile your own ffmpeg [18:32] <burek> did you check out our wiki for example [18:33] <WillieNL> burek: manual plugin versions in a outdated gentoo setup can be tricky :) [18:33] <D0minat0r> Yepp, tried this one: http://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide [18:33] <D0minat0r> Didnt work [18:33] <WillieNL> but thanks do have a running build gona get a smoke and look if can make the ffserver functional [18:34] <burek> D0minat0r, what exactly didnt work [18:34] <WillieNL> ubuntu and compilation sounds wrong :S [18:34] <burek> WillieNL ok :) good luck with all that [18:35] <WillieNL> ffserver version 0.7.1 ffserver.conf:22: Unknown VideoCodec: libx264 [18:35] <WillieNL> well looks much better :) [18:35] <WillieNL> Mon Feb 4 18:35:40 2013 FFserver started. [18:35] <WillieNL> ;-) [18:54] <JEEBsv> ls [18:54] <JEEBsv> ugh, did the usual [19:27] <quup> hi, can I export all frames as PNG back-to-back to the same (fifo) file? [19:29] <quup> or just to stdout [20:34] <DimStar> Good evening... do you know of issues with ffmpeg when playing .wma / .wmv files, as described in http://trac.videolan.org/vlc/ticket/1817 (the 'solution' seems to be to revert to ffmpeg 1.0) [20:51] <juanmabc> http://www.youtube.com/watch?v=BHPzjpojlz0 [20:52] <juanmabc> mistery intended [20:52] <shevy> hmm can .lpd files be converted via ffmpeg? [20:52] <shevy> ik.lpd: Invalid data found when processing input [21:11] <llogan> DimStar: can you duplicate the issue in current ffmpeg? one of the samples in that report sounds fine to me with ffplay. [21:15] <DimStar> llogan: I tried with vlc 2.0.5 linked against ffmpeg 1.1.1 (which is the latest release) [21:15] <WillieNL> burek: well quick testing looks like even working :) but no audio due my own misconfig somewheire so i have something todo :p [21:46] <llogan> DimStar: if it worked for me in ffplay then i would guess it is a VLC issue (unless the file was created by ffmpeg and it also does not work in WMP). [21:46] <DimStar> llogan: seems to be the interaction VLC / ffmpeg 1.1... whereas it was fine with 1.0 [22:09] <veleno> hello. for a research project, i'm trying to convert an input video file into so that i have several versions of it at various bitrates. i need the output files to have a constant bit rate. [22:11] <veleno> now, this is the coomand-line that i use http://pastebin.com/k9pUH1cR and then i play with VLC to see what the media information statistics report [22:27] <relaxed> veleno: for cbr you need to set bitrate, minrate and maxrate to the desired bitrate [22:28] <veleno> relaxed: are those encoder-specific options ? [22:28] <veleno> relaxed: i was using already " -x264opts bitrate=256:vbv-maxrate=256:vbv-bufsize=166 " [22:29] <relaxed> -b:v 256k -minrate 256k -maxrate 256k [22:29] <relaxed> I guess you can do it with -x264opts too [22:35] <veleno> so the muxrate is something not related [22:35] <relaxed> correct [22:42] <blitz> if I wanted to encode something to vbr mp3 can I do that through ffmpeg? I'm trying to use subsonic and transcode anything that isn't mp3 over to v0, right now it uses ffmpeg -i %s -ab 320k -v 0 -f mp3 - where %s is the filename [22:44] <blitz> or do I need to use lame [22:53] <veleno> mmm does ffmpeg also provide some tool to check the bitrate (average, instantaneous) of a given video file ? [23:40] <brontosaurusrex> would vc1 in mkv be something common/supported, or ....? [23:46] <JEEB> brontosaurusrex, not sure if VC-1 has an official mapping in matroska [23:47] <brontosaurusrex> hmm, mkvmerge just removes a ts to mkv with vc1 with no hickups, lets see if my tv can play those ... [23:47] <brontosaurusrex> removes = remuxed [23:48] <JEEB> don't find it in http://www.matroska.org/technical/specs/codecid/index.html at least [23:48] <JEEB> wouldn't stop it from muxing it some other way of course [23:49] <FergusL> Hi, I'm on Ubuntu, trying to compile something coded on Debian, I'm running into an error with libavutil-dev... which is not being updated on Ubuntu in favor of Libav. What could I do to compile ? [23:51] <veleno> is it possible to tell ffprobe to use the same size unit for all the frames? sometimes it uses byte, others Kibyte [00:00] --- Tue Feb 5 2013

1 0

ffmpeg-devel.log.20130203
by burek 04 Feb '13

04 Feb '13

[00:43] <Compn> whats this image_to_j2k thing in the libopenjpeg patch ? [00:43] <Compn> is it another project? [00:43] <Compn> is it lgpl ? [00:45] <Compn> oh i guess its not much code [03:12] <cone-106> ffmpeg.git 03Michael Niedermayer 07master:0cecaa9e2edf: ffmpeg: free attachments, fix memleak [09:43] <burek> are these two image-based? dvb_teletext, dvb_subtitle [09:51] <nevcairiel> dvbsubs are bitmaps, teletext is text based [09:52] <burek> ok, thanks [09:53] <burek> just one more thing, is it possible to copy dvb_subtitle from a random input file into another output file (with completely different video) ? [09:53] <burek> assuming that it makes sense to do such thing [09:53] <nevcairiel> i suppose thats possible [09:54] <burek> cool :) [09:54] <burek> did i mention lately that ffmpeg is great? :) [10:42] <wm4> what's with fate [11:20] <cone-9> ffmpeg.git 03Stefano Sabatini 07master:c91f1f3f926d: ffplay: drop redundant NULL sws_freeContext() check in stream_close() [11:20] <cone-9> ffmpeg.git 03Carl Eugen Hoyos 07master:90020d7f4bf2: lavf/segment: use correct spelling and value for EXT-X-ALLOW-CACHE tag [11:50] <DEATH> Is --disable-mmx meant to disable all SSE etc optimizations too? Because --enable-runtime-cpudetect --disable-mmx apparently means that no CPU specific optimizations are in use [11:51] <nevcairiel> its not necessarily meant to do this, but in practive it sometimes does [11:51] <nevcairiel> practice* [11:51] <DEATH> Also my build only needs audio decoders, build issues that I've been having affect video specific code, configure option to disable all video code could be nice (smaller output). [11:52] <Skyler_> I think --disable-mmx disables all mmx-and-above [11:52] <beastd> DEATH: did you look at current configure options? you should be able to disable most video stuff [11:52] <DEATH> Yeah, I do --disable-everything then enable specific decoders I need [11:53] <DEATH> I got this yesterday: [11:53] <DEATH> libavcodec/x86/dsputil_mmx.c:2309:14: error: 'gmc_mmx' undeclared (first use inthis function) [11:54] <DEATH> As far as I can tell it's for motion compensation, so while I don't really have clue about FFmpeg internals, it does not concern my usage case as I'm using FFmpeg to decode AAC+MP3+Vorbis audio only. [11:55] <DEATH> Tried to remedy by --disable-mmx but that disabled more stuff than I'd like to disable. [11:55] <beastd> DEATH: ok, i see. [11:57] <DEATH> With all instruction set extensions disabled, FFmpeg is only as fast as most of the competing decoders :) [11:57] <DEATH> (On MP3/AAC/Vorbis audio) [11:57] <beastd> DEATH: your problem looks like a bug to me. so it probably should be fixed. [11:58] <DEATH> OK, thanks for acknowledging [11:58] <beastd> maybe the assignment at dsputil_mmx.c:2309 needs just more #if protection ? [12:00] <beastd> DEATH: AFAICT the sse init has "&& CONFIG_VIDEODSP" . would that fix your compile problem? and hopefully run too ;) [12:02] <beastd> i am just poking around though. i do not know this code well at all. so please do not take my suggestions too serious. [12:05] <saste> beastd, are you going to work on kerndeint? [12:05] <saste> ubitux mentioned that you wanted optimize it, so we can drop mp=kerndeint [12:05] <beastd> saste: i think this is a misunderstanding. i have shown no interest in kerndeint [12:06] <saste> beastd, ok [12:06] <beastd> saste: an lavfi version was posted? it was slower thant mpcodecs wrapper? [12:06] <saste> i think i'll wait for ubitux then, it's not anything like high priority to get rid of mp=kerndeint [12:06] <beastd> agree [12:06] <saste> beastd, check the archive [12:06] <saste> kerndeint is already in [12:07] <beastd> saste: oh ok. sorry, for my ignorance. [12:10] <DEATH> Hmm yes looks like it's missing CONFIG_VIDEODSP check in the #if line, compiles fine after adding it: [12:10] <DEATH> #if (ARCH_X86_32 || !HAVE_YASM) && CONFIG_VIDEODSP [12:10] <DEATH> c->gmc = gmc_mmx; [12:10] <DEATH> #endif [12:10] <DEATH> Thanks for the input. [12:11] <beastd> DEATH: ok, i will sent on ffmpeg-devel list for discussion/approval [12:12] <beastd> thanks for reporting and testing [12:15] <michaelni> burek, can you do something about the FATE-IDLE logger thing ? [12:15] <michaelni> its 99% nonsense ATM [12:15] <michaelni> we have only 1 client (haiku) that actually is stuck [12:16] <michaelni> well, was stuck, i hope it works now again [12:17] <michaelni> the duron box needs >2days for a cycle, blame valgrind & amd if you like but it is that slow it was not stuck [12:19] <nevcairiel> imho it should also say the names of boxes which are newly added to the stuck list, the number is not really useful [13:14] <cone-9> ffmpeg.git 03James Almer 07master:d1511c02f49e: proresdsp: Fix checkheaders [13:14] <cone-9> ffmpeg.git 03James Almer 07master:13eb9fcf5686: build: Remove superfluous MAKE variable for the build suffix [13:55] <beastd> DEATH: Just found out that the problem you described was fixed in exactly that way on Fri Jan 25 by Carl Eugen. So no problem with current git master. [13:55] <beastd> I was actually looking at an older version because I did not notice I was on a topic branch... [16:11] <saste> michaelni, how do we handle mid-stream frame parameter changes before feeding the filtergraph in ffmpeg? [16:11] <saste> or don't we handle them at all? [16:14] <wm4> saste: would it be possible to let the lavfi glue code reconfigure the immediate following part of the filter chain if the frame fed to a filter differs from the config of the previous frame? [16:14] <wm4> or would that be too complex, or impossible? [16:17] <saste> wm4: http://thread.gmane.org/gmane.comp.video.ffmpeg.devel/153146 [16:20] <wm4> saste: I'd say always cal config, and filters which can handle the change don't reset themselves - does this make sense? [16:20] <wm4> *call [16:21] <michaelni> saste, also see -reinit_filter [16:21] <saste> wm4, please comment on the thread [16:21] <wm4> marking filters with bit flags seems fragile and complex [16:21] <saste> wm4, that's why it is an RFC [16:36] <cone-9> ffmpeg.git 03Michael Niedermayer 07master:cb573f7fbca1: avcodec/x86: Add daniels copyright to the recent gcc->yasm convertions he did. [16:46] <wm4> how do you know if new metadata has been added during streaming? [16:52] <JEEB> ubitux, would you have interest in looking at the source code of a renderer for the ARIB B-24 captioning system? :D https://dl.dropbox.com/u/175558/screenshots/tvtest_caption_lol_positioning.… [16:58] <wm4> how's the source? is it full of japanese comments? [17:02] <JEEB> haven't really looked at it yet, but it consists of a captions parsing library (that's used by caption2ass as well f.ex.), and then the actual renderer [17:02] <JEEB> I would guess Japanese Shift-JIS comments tho :) [17:02] <wm4> lol [17:09] <cone-9> ffmpeg.git 03Stefano Sabatini 07master:6e32b377bbb5: ffplay: remove pts_ptr argument from audio_decode_frame() [17:09] <cone-9> ffmpeg.git 03Stefano Sabatini 07master:02af4e9a97cc: ffplay: rename audio_decode_frame() variable "pts" to "audio_clock0" [17:09] <cone-9> ffmpeg.git 03Stefano Sabatini 07master:c44281906a87: ffplay: remove misleading comment from audio_decode_frame() [17:09] <cone-9> ffmpeg.git 03Stefano Sabatini 07master:1897109c00a0: lavfi/pad: add support to named options [17:09] <cone-9> ffmpeg.git 03Dave Rice 07master:3bba91d32d86: doc/syntax: fix frame rate documentation [17:09] <cone-9> ffmpeg.git 03Dave Rice 07master:fe63f3d301cc: doc/syntax: add missing frame sizes [17:34] <cone-9> ffmpeg.git 03Michael Niedermayer 07master:4d37d2bfc53c: put_vp_no_rnd_pixels8_l2_mmx: fix type [17:57] <durandal_1707> divVerent: mutt messed up patch somehow, you should use git send-mail instead [17:59] <wm4> so I heard mp4 file always have a perfect index that contains the timestamp for each frame (I don't know mp4,s o this might be wrong) [18:00] <wm4> doesn't that mean libavformat could set the duration for each packet exactly? [18:02] <nevcairiel> there is still complications with frame reordering [18:02] <wm4> oh, right [18:02] <nevcairiel> but it does provide some duration, not sure where it comes from in mp4 [18:03] <wm4> ah well, I'm mostly interested in an API that doesn't force the player to go haywire when video frames are very sparse [18:03] <wm4> like it happens with some "slide" videos containing audio and mjpeg frames [18:03] <wm4> the duration field seems to be mostly unreliable, whatever it is [18:04] <nevcairiel> most containers just have timestamps, not duration info [18:08] <cone-9> ffmpeg.git 03Rudolf Polzer 07master:773fc6e0c712: lavfi/showspectrum: set default height to 512 [19:42] <divVerent> 17:57:01 @durandal_1707 | divVerent: mutt messed up patch somehow, you should use git send-mail instead [19:43] <divVerent> I cannot add my own text when using git-send-email [19:43] <divVerent> also, it'd use the wrong from address [19:43] <divVerent> next time I'll just attach the patch as a file then [19:48] <divVerent> durandal_1707: wtf... [19:48] <divVerent> [rpolzer@grawp ffmpeg (git)-[master]-]$ git apply ../0001-lavfi-showspectrum-display-multiple-channels-in-sepa.patch [19:48] <divVerent> error: patch failed: Changelog:10 [19:48] <divVerent> git-am also fails it [19:50] <durandal_1707> changelog changed [19:50] <durandal_1707> the @ somehow become at in your patch .... [19:56] <durandal_1707> saste: i sent new patch, anxiously expecting your fine review [20:00] <saste> durandal_1707, thanks [20:00] <divVerent> 19:50:19 @durandal_1707 | the @ somehow become at in your patch .... [20:00] <divVerent> durandal_1707: very sure mutt didn't do that [20:00] <divVerent> maybe the mail archive did? [20:01] <divVerent> durandal_1707: ah, thanks... this solved the missing link [20:01] <divVerent> didn't know how to make a named choice [20:02] <divVerent> and yes, I can still fully sign off on the new patch, great work [20:08] <saste> bah ffplay -af is growing complicate... [20:11] <cone-9> ffmpeg.git 03Michael Niedermayer 07master:ccfd8cffe867: qdm2: Fix data type used in multiplication. [20:11] <cone-9> ffmpeg.git 03Michael Niedermayer 07master:d106679fb526: qdm2: disable superblocktype_2_3==0 code [20:14] <michaelni> burek, i suggest you disable the FATE-IDLE thing until its fixed, its of no use like this [20:15] <michaelni> it just prints a random meaningless number every hour [20:23] <Daemon404> \o/ someone broke ffmpeg's configure [20:23] <Daemon404> in such a suble, hidden way [20:23] Action: Daemon404 guess he does a git bisect [20:30] <durandal_1707> what got broken in configure? [20:33] <Daemon404> every single math check function [20:33] <Daemon404> e.g. lrint [20:33] <Daemon404> instea of: foo(floata,floatb) [20:33] <Daemon404> int main() { return (int) foo(...): } [20:33] <Daemon404> it has return 0 [20:33] <Daemon404> which does fuck all [20:34] <Daemon404> if the compiler has DCE [20:34] <Daemon404> and i have a feeing the breakage is hidden in a merge commit... [20:34] <Daemon404> behind wallas of shit [20:35] <wm4> lol [20:36] <wm4> maybe ffmpeg devs should send patches to libav instead to make merging easier [20:36] Action: wm4 runs [20:41] <divVerent> 20:08:33 @saste | bah ffplay -af is growing complicate... [20:41] <divVerent> saste: yes [20:42] <divVerent> ffplay -i lavfi 'amovie=filename.mp3, showspectrum' [20:42] <Daemon404> what if i have a file named lavfi? ;) [20:42] <divVerent> we also need a better way for this eventually... preferably by (later) getting rid of ffplay's own copy [20:42] <divVerent> sorry, -f lavfi ;) [20:42] <Daemon404> :P [20:42] <saste> divVerent, not that convenient [20:43] <divVerent> I agree [20:43] <divVerent> but that's lavfi [20:43] <divVerent> that's something to be fixed separately [20:43] <saste> ffplay IN -af volume=+2dB [20:43] <divVerent> yes, but an -af cannot create a video stream... or can it? [20:43] <saste> divVerent, no [20:43] <divVerent> then maybe that's what is missing ;) [20:45] <divVerent> ffplay filename.mp3 -filter_complex "[0:0] asplit [a] [va]; [va] ashowspectrum [v]" -map "[a]" -map "[v]" [20:45] <divVerent> would be great ;) [20:45] <divVerent> in other words: "consistency with ffmpeg" [20:46] <divVerent> at least that can be hidden in a shell alias then [20:51] <saste> divVerent, that's correct in principle [20:52] <saste> but it's hard to implement in practive, because of the high inconsistencies between the tools [20:53] <saste> ffmpeg/avconv has lots of specific hacks which have never been ported to the other tools [20:54] <teratorn> wm4: i think its pretty clear whos making whos job harder, as an outside observer of sorts :)) [20:54] <wm4> teratorn: the interesting question would be whether libav makes merging intentionally harder [20:55] <teratorn> my answer is to care less i think :/ [20:55] <teratorn> and also to maybe put pressure on debian to repackage ffmpeg [20:55] <teratorn> since that is just silly [20:56] <divVerent> saste: I know [20:56] <divVerent> that this is the issue [20:56] <divVerent> sort of like mplayer/mencoder, just other way round [20:57] <divVerent> ffplay has become the neglected stepchild [20:57] <divVerent> to a point where I wonder if it'd be better to make ffmpeg learn what ffplay can do - i.e. seek [20:57] <divVerent> anything else already can be done by lavfi :P [20:58] <wm4> wut, so video playback with ffmpeg+lavd? [20:58] <divVerent> i.e. if the console, and the video out window of the "sdl" output device would take keyboard input [20:58] <divVerent> and allow seeking [20:58] <divVerent> then basically ffplay could be killed [20:59] <divVerent> oh, possibly AV sync too, possibly in communication between video and audio driver... that sounds ugly ;) [20:59] <divVerent> but MAYBE both video and audio out syncing to pts would work well enough (for audio this is tricky, though :P) [20:59] <wm4> how would you do user input and seeking? [21:00] <divVerent> that's the thing I don't know [21:00] <divVerent> currently both ffmpeg and lavfi movie source lack seeking support [21:00] <divVerent> it's just wishful thinking [21:00] <saste> wm4, input and seeking: you program events (e.g. in lua) which are sent to the filtegraph as events [21:00] <divVerent> yes [21:00] <divVerent> THAT half is the easy half ;) [21:00] <wm4> divVerent: avf_videoplayer? xD [21:00] <divVerent> we already have this event handling [21:00] <divVerent> as in, sending commands to the filter graph [21:01] <saste> divVerent, in the meanwhile i'll add -af to ffplay ;-) [21:01] <wm4> I'd be careful with this commands and filters thing [21:01] <divVerent> sure :P [21:01] <divVerent> not a bad idea either way [21:01] <wm4> that'll become a mess in no time [21:01] <divVerent> just, in the long run, PROBABLY ffplay should be killed [21:01] <divVerent> or maybe replaced by a hardlink to ffmpeg that detects how it should output by binary name ;) [21:02] <divVerent> but talking the REALLY long run here [21:02] <saste> and i'm still awaiting for glplay (or in general an opengl output device) [21:02] <wm4> divVerent: actually I'm glad if "they" can't change their own APIs so easily (due to the code duplication in ffmpeg/ffplay) [21:02] <divVerent> saste: you SORT of could have it by changing the API from SDL 1.2 o SDL 2 ;) [21:03] <divVerent> but... don't [21:03] <wm4> saste: that'd be a good thing, since SDL sucks for video [21:03] <divVerent> I know how to use SDL 2 so it actually works [21:03] <divVerent> but it was tricky [21:03] <divVerent> well, actually, SDL 1.2 also has video surface support [21:03] <divVerent> but by Xvideo [21:03] <wm4> saste: you could also display all sorts of pixfmts natively (think 10 bit 444...) [21:03] <saste> divVerent, what's the status of libsdl2? [21:03] <divVerent> while SDL 2.0 throws that away and introduces a simplified common 2D interface for OpenGL and D3D [21:03] <divVerent> can't recommend it yet [21:03] <saste> what would be the gain for switching from 1.2 -> 2.0 [21:04] <divVerent> I implemented sdl2 output for mpv, and it works [21:04] <divVerent> but it was tricky [21:04] <divVerent> OpenGL output, mainly [21:04] <divVerent> basically, if you DO use sdl2, you WILL hit bugs [21:04] <saste> the last time I tried libsdl dev tree it was buggy as hell [21:04] <divVerent> exactly [21:04] <wm4> they're still changing APIs [21:04] <divVerent> yes, but not much any more [21:04] <divVerent> I am reading the ML [21:04] <divVerent> but, yes, I can't recomemdn it yet [21:05] <divVerent> I will recommend SDL2 as only output device for ffmpeg the day SDL2 enters Debian stable [21:05] <divVerent> and that's still like 2, 3 years in the future [21:05] <wm4> lol [21:05] <divVerent> I assume SDL2 will be stable by end of the year [21:05] <wm4> so long [21:05] <divVerent> given it's now being pushed by app developers for iOS, Android [21:05] <wm4> they sure are taking their time [21:05] <divVerent> indeed [21:06] <divVerent> if the iOS/Android guys wouldn't be already using it for apps they sell [21:06] <divVerent> we could assume SDL2 will NEVER finish [21:06] <divVerent> you know, SOMEONE has to push them ;) [21:06] <divVerent> saste: also, as for SDL2 bugs [21:06] <divVerent> the HW accelerated outputs are mostly bugfree [21:06] <wm4> hm does that mean they can use native code easily on android? [21:06] <divVerent> i.e. opengl, direct3d [21:06] <divVerent> but the SW output... will cause nightmares ;) [21:06] <divVerent> if you REALLY want to try tzhat [21:06] <divVerent> look in mpv source for the APIs I used, and use ONLY these [21:07] <divVerent> wm4: "sort of" ;) [21:07] <divVerent> not sure if it's for rooted devices only [21:07] <wm4> I remember point scaling... [21:07] <divVerent> just know people ARE making money with it [21:07] <divVerent> wm4: that's the missing features part [21:07] <divVerent> but the bugs when you tell SDL a texture is "static!" [21:07] <divVerent> and it then converts the alpha channel to 1bit RLE... [21:07] <divVerent> that's awful [21:08] <divVerent> that's why all textures in mpv are marked and used as dynamic ones [21:08] <divVerent> even the OSD ones which COULD be seen as kinda static [21:08] <divVerent> given they tend to update only once per sec [21:08] <divVerent> (IMHO that's already static enough :P) [21:09] <divVerent> note: these bugs ONLY happen with the SW renderer. Nevertheless awful. [21:09] <divVerent> these bugs ALSO quite likely affect the old SDL_Surface API too [21:09] <divVerent> wonder if SDL 1.2 has these bugs too ;) [21:09] <wm4> divVerent: the worst for video output, though, is libva [21:09] <divVerent> given the SW renderer is just a wrapper for SDL_Surface [21:09] <wm4> well, at least for OSD [21:10] <Daemon404> oh fuckin boy.... [21:10] <Daemon404> the configure / return 0 problem only happens on msys [21:10] <Daemon404> FML [21:11] <Daemon404> and only in ffmpeg [21:16] <durandal_1707> so when new fate look will finally come in? [21:18] <michaelni> Daemon404, theres no return 0 in ffmpegs check_mathfunc [21:19] <Daemon404> yet someone it ends up there [21:19] <Daemon404> it's mind boggling [21:19] <Daemon404> somehow* [21:22] <durandal_1707> so any new filter comming in? [21:23] <saste> durandal_1707, yours ;-) [21:24] <saste> you're getting a filter addicted [21:24] <saste> who's your favourite (git-)pusher? [21:26] <durandal_1707> i dont have favorite (git-)punisher [21:32] <durandal_1707> to download vapoursynth source code i need install svn - no way [21:33] <wm4> use git-svn? [21:33] <durandal_1707> that needs svn too [21:34] <wm4> but you don't need to use svn directly [21:35] <Daemon404> you see, durandal_1707 uses an antiquated OS called BSD [21:35] <Daemon404> which uses an antiuated system called ports [21:35] <Daemon404> and this to install svn he needs to invest time [21:35] <Daemon404> compiling... [21:35] <Daemon404> compiling.... [21:35] <Daemon404> compiling..... [21:35] Last message repeated 1 time(s). [21:35] <Daemon404> etc [21:35] <durandal_1707> ^troll [21:36] <durandal_1707> there are nice and smart people with clones on github [21:36] <Daemon404> none of which are up to date i suspect [21:36] <durandal_1707> what, why this depens on lav* stuff? [21:37] <Daemon404> what? [21:37] <wm4> lav* actually contains a useful video decoder [21:37] <durandal_1707> i read INSTALL [21:37] <Daemon404> vapoursynth uses swscale [21:37] <durandal_1707> ignores lavfi [21:37] <Daemon404> i think i just died of laughter. [21:37] <durandal_1707> should be perished into vapour [21:39] <durandal_1707> now i need to install waf.... [21:39] <wm4> ouch [21:39] <durandal_1707> this is just too complicated for me.... [21:48] <durandal_1707> michaelni: is it really true that S32 when going to/from planar is converted to float? [21:48] <durandal_1707> that does not make any sense... [22:03] <michaelni> durandal_1707, no silly convertions should be done, in what case exactly does this happen? [22:04] <durandal_1707> dunno, that i read from nicolas reply about reordeing in lavd.... [22:23] <durandal_1707> saste: actually i need to do this from my histogram filter: http://avisynth.org/oldwiki/index.php?page=histogram [22:30] <saste> durandal_1707, vectorscope? [22:33] <durandal_1707> all [23:02] <cone-9> ffmpeg.git 03Dave Rice 07master:71956371a7bb: lavu/parseutils: add digital cinema frame sizes [23:17] <cone-9> ffmpeg.git 03Stefano Sabatini 07master:cf7b71b0b812: doc/filters: reformat various filter tables items [23:26] <burek> michaelni, what should i fix about the fate-idle? [23:26] <burek> do you want to have the names of machines always displayed or something? [23:27] <nevcairiel> personally i think it would be great if it just names the machine which was just added to the list once, and doesnt list all machines all the time [23:31] <burek> hmh, i could make a list i guess, but i think it would be easier to have idle time defined for each machine [23:31] <burek> for example, the default of 24h would be applied to each machines except those configured with a different idle value [23:32] <burek> that way it wouldnt spam too much [23:32] <burek> and it would always provide a list of all stuck machines, so you don't have to scroll back and calculate which one is new in the list and if previous ones in the list got unstuck or not, etc [23:40] <burek> also, which one of these is that "duron box" ? I see a lot of durons in the list, so I don't know for which one to set the idle timeout to > 48h [23:41] <michaelni> burek, all that say duron [23:41] <michaelni> they all run on the same box [23:41] <michaelni> thats part of the reason why it takes lots of time to finish a cycle [23:42] <burek> i see [23:42] <burek> ok, how many hours should i set for timeout then? 72 hrs perhaps? [23:43] <michaelni> yep [23:44] <michaelni> or maybe you can automate it [23:44] <michaelni> like using 24h+average past update time or so [23:46] <burek> i'll try first 72hrs, just in order not to complicate things if i dont need to [23:51] <saste> nevcairiel, why RGB -> YUVJ in the showspectrum patch? [00:00] --- Mon Feb 4 2013

1 0

ffmpeg.log.20130203
by burek 04 Feb '13

04 Feb '13

[00:28] <Youka> could it be that ASS subtitle decoding is buggy? i tried it, but it always skip first 15 lines [00:29] <Youka> https://github.com/Youka/NyuFX/blob/master/src/lva.cpp#L493 [01:46] <taylanub> Today I compiled from git and ffplay (and mplayer) give corrupted output. ffplay error output: http://sprunge.us/gOBS [01:46] <taylanub> Weird thing is, libav has the same issue (tested with mplayer, since libav has no ffplay). [03:09] <pietro10> is ther ea way I can capture the audio from either my speakers only (not the mic) or from a specific program? -f alsa -i pulse records from my mic. Thanks. [03:09] <klaxa> pietro10: open pulseaudio volume control and move the record stream to the monitor of your soundcard [03:11] <pietro10> I could do that but I mean with ffmpeg, so I don't have to do that each time if I actually want to use my mic [03:14] <klaxa> you could script it with pacmd or something, i think there is a way to choose that with ffmpeg itself though i don't remember it [03:25] <pietro10> ok thanks [03:59] <blue_misfit> anyone have experience with Smooth Streaming? [06:05] <makara> I want to check that my m4a's aren't ALAC. Why doesn't this work: ffprobe *.m4a | grep alac [06:20] <drv> makara: ffprobe output is on stderr, you could use e.g. ffprobe *.m4a 2>&1 | grep alac [06:23] <makara> drv, could you please explain "2>&1" [06:23] <drv> it redirects stderr to stdout [06:33] <makara> drv, it doesn't work [06:33] <makara> it will output on screen, but can't pipe to grep or a file [06:42] <makara> drv, it works, thanks [14:18] <paste> hi, is there an ffmpeg player? reason is when using yadif with mplayer clearly field 1 is incorrect, but when encoding with ffmpeg I get jutters regardless if I choose top (0) field or bottom (1) field. Or I'm DIW? [14:19] <saste> paste: ffplay [14:19] <paste> cool thanks. [14:20] <paste> hmm. [14:20] <paste> looks like its not available. [14:21] <paste> ugh [14:21] <paste> nvm [14:21] <paste> bed time. [14:21] <paste> thanks. [14:30] <jastr> hi.. can anyone help me with a raw video format issue? [14:30] <jastr> when i run this ./ffmpeg -i MyStream_0.f4v -vb 1200k -f rawvideo -t 00:05:00 -pix_fmt yuv420p -s 832x468 - | ./ffmpeg -f rawvideo -r 30 -pix_fmt yuv420p -s 832x468 -crf 0 -i - out.mp4 the output keeps being in 200kb/s i've tried -qscale and -b but either helps [14:31] <jastr> now it detects the input to be 1200kb after i set the -vb on the raw input but the output it writes is always approx 260kb/s no matter what option i give the output [14:32] <jastr> here is the pastebin http://pastebin.com/hz27cXEj [14:33] <spyvingen> anyone know how to set swe as first choice in winff and if swe dont exist then eng should be choosen? [14:34] <spyvingen> audio that is [14:35] <Mavrik> jastr: why are you trying to set bitrate on input? [14:35] <Mavrik> jastr: input has a bitrate of what it has, forcing anything won't change things [14:35] <Mavrik> plus... why two ffmpeg processes? [14:36] <Mavrik> what are you trying to achieve? [14:36] <jastr> well when i didn't use the -vb it detected the input at 116812 [14:37] <Mavrik> it doesn't matter [14:37] <Mavrik> you can't change what you already have :P [14:37] <jastr> Mavrik i have a movie that looks fine when played back in quicktime but it's a movie recorded in 25fps on a 30fps canvas so it looks a bit stuttering when played back in flash. So what i wanted was ffmpeg to kind of playback the movie and reencode the video stream. If i reencode from file to file in ffmpeg either at 25fps or 30fps it's the same. [14:38] <jastr> i know but the input is 1200kb [14:39] <Mavrik> again, last time: detection doesn't friggin matter [14:39] <Mavrik> the decoder reads stream as it is [14:40] <Mavrik> plus, you're probably looking at totally wrong output anyway [14:40] <Mavrik> now. [14:40] <Mavrik> WHY are you using 2 ffmpeg processes that essentially do the same a simple one-process command line would do? [14:40] <Mavrik> also, crf 0? losless? [14:40] <jastr> thought i had to. that's how i did it in the old days. [14:41] <Mavrik> hmm :) [14:41] <Mavrik> jastr: so if I understand this correctly, you're just trying to convert 25fps video to 30 fps video? [14:41] <jastr> the reason for the two processes was that i just wanted ffmpeg to playback the movie and recapture it hoping to get a little more smooth result [14:42] <jastr> no i'm trying to smooth it. It is 25fps and should be 25fps BUT the program that encoded the file was working with 30fps so it's not smooth when it's played back in flash but quicktime handles the file fine [14:44] <Mavrik> I see. [14:44] <Mavrik> well, your command line won't have the result you seek [14:44] <jastr> i see.. :D any other commandlines that could have? [14:45] <Mavrik> it's equivalent of just reencoding video with parameters with added overhead of piping :) [14:45] <spyvingen> is it possible to choose what audio languege to choose [14:45] <spyvingen> *to rip [14:45] <Mavrik> jastr: well, you can try just dropping frames [14:45] <Mavrik> down to 25fps [14:45] <jastr> also when the file is 25fps? [14:46] <Mavrik> oh doh [14:46] <Mavrik> hmm [14:46] <Mavrik> now that's quite a problem [14:46] <jastr> hehe exactly.. that's what the program did and now i just want to smoothen it as much as possible [14:48] <Mavrik> ffmpeg -r 30 -i yourfile.flv -codec:v libx264 -crf 20 -codec:a copy -r 25 output.mp4 [14:48] <Mavrik> try this :) [14:55] <jastr> that dropped a lot of fps but didn't really get more smooth. more like the other way around. [14:55] <jastr> maybe -f 25 and exporting a 30fps? [14:56] <Mavrik> probably, but I'm afraid you already lost frame info :\ [14:59] <jastr> yeah i know.. that's why people who don't know about video encoding shouldn't do it.. arhh [15:34] <Guest17665> Hi I'm using this options to convert avi to mp4: [15:35] <Guest17665> "-i $INPUT -vcodec libx264 -threads 0 -r 25 -g 25 -b 500k -bt 500k -acodec libvo_aacenc -ar 44100 -ab 64k -y $OUTPUT" [15:35] <Guest17665> But I get: [15:35] <Guest17665> Error while opening encoder for output stream #0:1 - maybe incorrect parameters such as bit_rate, rate, width or height [15:36] <Mavrik> Guest17665: there has to be a message actually explaining what's wrong. [15:37] <Guest17665> Mavrik: you mean, to pastebin it? [15:37] <Mavrik> well [15:38] <Mavrik> you can read it as well :P [15:40] <Guest17665> ok [15:41] <TheLink> if I set hqdn3d=:::0, does this leave the first 3 parameters at default and the fourth at 0? [15:41] <Guest17665> http://pastebin.com/k8t58KGi [15:42] <Guest17665> What are advantages of avconv vs ffmpeg [15:42] <Mavrik> *shrug* [15:43] <Mavrik> TheLink: hmm, I'm not sure that's even valid syntax [15:43] <Mavrik> TheLink: newer ffmpegs prefer hqdn3d=chroma_tmp=0 type syntax [15:43] <Mavrik> Guest17665: [libvo_aacenc @ 0x1ba5b60] Unable to set encoding parameters [15:43] <Mavrik> is seems your audio encoder doesn't initialize properly [15:44] <Mavrik> ah [15:44] <Mavrik> Guest17665: you have 5.1 audio [15:44] <TheLink> well, ffmpeg didn't throw an error but I guess your syntax is much cleaner [15:44] <TheLink> thanks [15:44] <Mavrik> Guest17665: libvo_aacenc doesn't support 5.1 audio [15:48] <Guest17665> so i should pick some other acodec? [15:48] <Mavrik> yep, you'll need another aac encoder if you want to keep 5.1 audio [16:14] <Guest17665> well what's 5.1 audio? [16:16] <TheLink> multi channel audio [16:16] <TheLink> in your case 6 channels [16:21] <Guest17665> why it says: Unknown encoder 'libfaac' [16:25] <Guest17665> I get now: [16:25] <Guest17665> [ac3 @ 0x2684e60] frame sync error [16:25] <Guest17665> Error while decoding stream #0:1 [16:25] <Guest17665> Application provided invalid, non monotonically increasing dts to muxer in stream 1: 22906 >= 22871 [16:25] <Guest17665> av_interleaved_write_frame(): Invalid argument [16:25] <Guest17665> I change acodec to libfaac [16:34] <LithosLaptop> ewww :) [16:34] <Guest17665> and what is that: [16:34] <Guest17665> encoder 'aac' is experimental and might produce bad results. [16:34] <Guest17665> Add '-strict experimental' if you want to use it. [16:34] <Guest17665> Or use the non experimental encoder 'libfaac'. [16:35] <Guest17665> literally -strict experimental ? [16:36] <LithosLaptop> the quality of the experimental aac encoder is worse than libfaac [16:36] <LithosLaptop> and libfaac aint that great [16:36] <Guest17665> I've built it with [16:36] <Guest17665> ./configure --enable-libx264 --enable-libvo-aacenc --enable-libfaac --enable-gpl --enable-nonf [16:36] <Guest17665> ree --enable-version3 [16:36] <Guest17665> why say not libfaac? [16:37] <LithosLaptop> not really transparent at 128Kbit/s for 2 channel audio [16:37] <LithosLaptop> thats why I don't use it [16:37] <LithosLaptop> even libmp3lame will give you better quality than libffac [16:38] <LithosLaptop> try to use fdkaac in ffmpeg [16:39] <LithosLaptop> libvo-aacenc is the worst btw [16:39] <Guest17665> LithosLaptop: well, I need whatever gets the job done [16:40] <LithosLaptop> --enable-libfdk_aacenc [16:40] <LithosLaptop> oh [16:41] <LithosLaptop> well I haven't really tried more than 2 channels before [16:42] <Guest17665> Unknown option "--enable-libfdk_aacenc". [16:43] <Guest17665> oh it's libfdk_aac [16:44] <Guest17665> oops not found [16:54] <taylanub> So is ffmpeg from git broken for anyone else ? Doesn't play H.264, gives corrupted video. [16:58] <Guest17665> taylanub: it isn't suppoused to play, it's for converts [16:58] <taylanub> Eh ? Doesn't mplayer use libav to decode H.264 for playing ? What does ffplay do ? [17:02] <taylanub> I'll be compiling old commits and see what commit broke it. Assuming the problem isn't elsewhere. [17:03] <Guest17665> LithosLaptop: how do I use libmp3lame? [17:03] <Guest17665> instead of aac [17:03] <LithosLaptop> -acodec libmp3lame [17:04] <Guest17665> Unknown encoder 'libmp3lame' [17:05] <LithosLaptop> strange [17:05] <LithosLaptop> maybe it is not built with it [17:07] <Guest17665> oh [17:20] <Guest17665> I cloned git://github.com/mstorsjo/fdk-aac.git how to build it? [17:22] <Guest17665> after I install libmp3lame do I rebuild avconv? [17:24] <saste> Guest17665, avconv != ffmpeg [17:25] <saste> and we can't support avconv here [17:25] <Guest17665> saste: sorry [18:18] <Zeranoe> Is qt-faststart now part of FFmpeg? If so, how do I use it? If not, are they any other ways to move the moov atom? [18:19] <JEEB> there's a movflag for that IIRC [18:19] <beastd> Zeranoe: movflags IIRC [18:20] <JEEB> movflags +faststart was it? [18:20] <beastd> "-movflags faststart", according to the fine manual [18:21] <JEEB> :) [18:24] <taylanub> OK, I'm back to some December commit, and ffmpeg works again. :\ [18:24] <taylanub> This is OpenBSD 5.1 by the way. [18:25] <beastd> taylanub: i hope you did ffbisect. might have been lots of work otherwise [18:26] <taylanub> What's that ? I use the magit Emacs library to visualize the log and reset from there. [18:28] <beastd> taylanub: ffbisect is a script to ease biscecting ffmpeg code (because of merges). bisecting is the process of choosing an old (working commit) can be way back in time. and than doing a binary search on the interval [18:29] <beastd> anyway could you pinpoint exact commit that is causing you trouble? [18:36] <taylanub> I have yet to find that out .. Indeed, traversing the history seems hard. [18:36] <taylanub> I'm on some December 22 commit now, but the log also shows commits from Jan 23. :\ [18:36] <taylanub> (Not very good at git yet.) [18:39] <beastd> taylanub: maybe you should try ot bisect from now on [18:40] <beastd> i assume you have git clone of ffmpeg and found a commit in the past that worked your you (e.g one in the beginning of december) [18:44] <beastd> taylanub: are you testing for the regression with ffplay? [18:47] <taylanub> Yes. [18:51] <taylanub> Is there a way to not re-compile everything when changing commits ? A full compilation takes 5-10 minutes. [18:51] <beastd> taylanub: not sure that is a goode idea [18:51] <taylanub> OK then, will just take my time. [18:51] <beastd> taylanub: try it this way: [18:52] <beastd> taylanub: remember the bad revision or just checkout the bad revision [18:53] <beastd> then execute tools/bisect-create [18:53] <beastd> tools/ffbisect start [18:54] <beastd> tools/ffbisect need ffplay [18:54] <beastd> tools/ffbisect bad # marks current revision as bad [18:54] <beastd> oop wrong [18:54] <beastd> tools/ffbisect good # assuming you checked out a revision where it was working for you [18:55] <beastd> tools/ffbisect bad master [18:55] <beastd> then it should chose a commit for you to test [18:56] <beastd> make disclean; ./configure; make # use -j4 or whatever brings speed up on your systems, i assume you have multiple cores [18:56] <taylanub> -j4 tends to randomly break; or is that just due to bad Makefiles ? [18:57] <beastd> taylanub: too bad it shouldn't :( [18:57] <taylanub> I mean in-general .. dunno about ffmpeg. [18:57] <beastd> test if it works. depending if it works/or doesn't work you shoult then mark the revision as good or bad with: tools/ffbisect good/bad [18:58] <beastd> continuing that cycle you should arrive at the offending commit in i guess about 10 steps [18:58] <taylanub> OK [18:59] <beastd> taylanub: i don't know. GNU make with -j<n> usually works for me. i remember that on windows i had some warnigns from make that some assertions weren't met. on linux it always worked for me so far. and i use it on ffmpeg all the time. [19:00] <taylanub> I'll just use it then. [19:01] <beastd> taylanub: good luck! if you found the offending commit please read http://ffmpeg.org/bugreports.html and report it as regression [19:02] <taylanub> OK [20:49] <taylanub> On another note, the configure script bails out with "too many arguments" for me. Bash works. I guess I should just report a bug. [20:55] <taylanub> The first bad commit could be any of: [20:55] <taylanub> 528878ee7b377e23a194d7c801571d97793047e0 [20:55] <taylanub> 9c555bca68cf3f168c2e4ffcb5b4dacbda8366bf [21:00] <taylanub> [528878ee7b377e23a194d7c801571d97793047e0] openbsd: configure: Stop enabling PIC by default [21:00] <taylanub> Since it mentions OpenBSD, I'm betting on that. [21:00] <taylanub> Oh, in fact the other one just merges that. [21:05] <durandal_1707> taylanub: you reverted that commit and confirm it fixes your problem? [21:05] <durandal_1707> or you are just using old release with broken shell? [21:06] <taylanub> durandal_1707: Excuse me ? The configure problem is something unrelated; I've always had it and just change the script to use bash. [21:06] <taylanub> The PIC commit is likely what introduced corrupt H.264 video playback. [21:07] <taylanub> (And re. configure, OpenBSD's ksh is a POSIX-compliant sh, so it would be configure script that's somehow broken.) [21:07] <taylanub> (Assuming that this ksh is not simply buggy itself.) [21:08] <durandal_1707> taylanub: not really, there is ksh bug that got fixed in later version [21:08] <taylanub> Oh, when exactly ? I'm using OpenBSD 5.1. [21:08] <durandal_1707> how old is that? [21:08] <taylanub> The previous release .. they release approximately every 6 months, so it would be maximally about a year old. [21:10] <taylanub> May 1 2012 apparently. [21:26] <Jiyuu|C2> is it possible to use the libass subtitles filter so it will use an mkv for the source? [21:27] <Jiyuu|C2> i.e. avoiding the need to use mkvextract to get a subtitles file out of the mkv file [21:27] <Jiyuu|C2> also [21:27] <Jiyuu|C2> is is possible to use attached font files or do i have to extract and install them? [21:34] <trosh> i am on linux, and when i use ffmpeg to convert a part of a movie to still frames, using -ss for starting time, i have to wait for a hell of a long time until it reaches the desired time and then convert normally. is this normal behaviour? thanks. [21:34] <Fjorgynn> nah [21:35] <nronksr> trosh: try placing the -ss before the input file for a faster seek. add -ss for exact accuracy after the input file for a slower seek once you are in the general timeframe vicinity. [21:35] <trosh> oh thanks [21:35] <trosh> is -ss relative ? [21:36] <Fjorgynn> yeah -ss hh:mm:ss -i file.avi -t hh:mm:ss [21:37] <bkmz> hi guys, what is the exactly computation complexity of AAC-LC ?? where i can find complexity function for it [21:37] <nronksr> Ideas on extracting 6 still images every second from a 59.96fps clip? [21:42] <durandal_1707> nronksr: -vf select? [21:56] <nronksr> durandal_1707: thanks, I thought that too; I was experiencing a typo that boggled me. XD [22:02] <trosh> nronksr: just to be sure, -r 6 works in your case ? [22:05] <nronksr> trosh: for 6 frames extracted every second? I used this: -vf 'select=not(mod(n\,6))' [22:05] <vlt> Hello. Does anyone know what pkg I need to install additionally to ffmpeg to be able to use -vcodec libx264 on Ubuntu 12.04? [22:05] <JEEB> vlt, libavcodec-extra [22:06] <nronksr> vlt: ubuntu uses avconv by default, but yeah, libavcodec-extra and if I recall, something with 53 in it? [22:06] <JEEB> and yeah, if you want to use the packaged libav (which is not ffmpeg, long story), you should use the avconv command instead of the ffmpeg one :) [22:06] <JEEB> just a newer binary in that project [22:07] <trosh> nronksr: how come? [22:08] <vlt> JEEB, nronksr: Thank you! [22:11] <nronksr> trosh: I'm not an expert by any stretch of the imagination and that was the way I've done it and it works? [22:17] <vlt> :-( Installing libavcodecs-extra-53 breaks my whole pkg system on ubuntu o_O [22:17] <vlt> I should try a chroot env & [22:18] <JEEB> more like it wants to install packages that use it instead of the standard libavcodec package [22:18] <JEEB> it's OK [22:18] <JEEB> unless you have something weird in there [22:18] <JEEB> if you look at the list of installed and removed packages you should see it :P [22:19] <vlt> JEEB: (In the chroot env) apt is removing like 280 pkgs now. [22:20] <JEEB> and it should be installing something instead of them, no? [22:21] <vlt> JEEB: I dont know (and care) exactly now. I only know I cant do this on a production machine. [22:21] <vlt> Hence the chroot. [22:21] <JEEB> uhh [22:21] <JEEB> it should just remove the packages that are configured and linked against the standard (without x264 etc.) libavcodec packages [22:22] <JEEB> and then it should at the same time replace them with packages that are configured with x264 etc. enabled in the libavcodec [22:22] <JEEB> you should look at what it says exactly [22:22] <JEEB> because that is what it should do [22:22] <JEEB> if it does something else then something's wrong [22:22] <JEEB> but I do know that in the cases I've done it, it has worked rather nicely [22:23] <vlt> JEEB: Yes, Ill have a look at the apt log files later and show them here in case youre interested. [22:23] <JEEB> no, I'm not interested. I just want to tell you why it looks like it's going to "break" something [22:23] Action: JEEB sighs [22:27] <JEEB> vlt, although it might be apt failing to see the resolution, too [22:28] <JEEB> http://askubuntu.com/questions/132164/ubuntu-restricted-extras-wants-to-rem… [22:28] <JEEB> looking at this [22:28] <JEEB> if this is what happens then take a look at the answer [22:30] <relaxed> there are static ffmpeg binaries available with libx264 support [23:28] <Jiyuu|C2> no help for [22:26:47] <Jiyuu|C2> is it possible to use the libass subtitles filter so it will use an mkv for the source? [23:28] <Jiyuu|C2> ? [23:29] <burek> Jiyuu|C2, you could use -vf subtitles [23:30] <Jiyuu|C2> would that work with a .mkv file as the source? [23:31] <Jiyuu|C2> or do i still have to extract the subtitles from it? [23:33] <burek> hmh, did you read the docs about that filter? [23:33] <burek> it's kinda new, so i didnt have time to read it thoroughly [23:38] <saste> burek: should be possible, since subtitles uses libavformat to read the sub stream from the file [23:39] <burek> saste, do you have any example/sample cmd line to showcase how to tell it to ffmpeg? [23:40] <saste> burek, can't be hader than subtitles=in.mkv [23:41] <burek> oh right [23:41] <burek> thanks [00:00] --- Mon Feb 4 2013

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