July 2013 - Ffmpeg-devel-irc - lists.ffmpeg.org

ffmpeg-devel.log.20130715
by burek 16 Jul '13

16 Jul '13

[01:03] <durandal_1707> i think mp=dint is ready for removal [02:26] <ilove11ven> is Stefano Sabatini here? [02:27] <kierank> Not at the moment [02:29] <cone-116> ffmpeg.git 03James Almer 07release/1.2:ce11d9490cc2: oggparseskeleton: Fix fisbone header parsing [02:29] <cone-116> ffmpeg.git 03James Almer 07release/1.2:dc7377479226: oggparseskeleton: Replace avpriv_report_missing_feature() with a normal av_log() call [02:29] <cone-116> ffmpeg.git 03Michael Niedermayer 07release/1.2:ba4cb43f0b94: oggparseskeleton: avoid header parsing failure [02:29] <cone-116> ffmpeg.git 03Dale Curtis 07release/1.2:d8538a10024c: avformat/utils: Keep internal and external av_read_frame() packets in sync. [02:29] <cone-116> ffmpeg.git 03Michael Niedermayer 07release/1.2:9a87ba0933c8: update all trac links to use the trac subdomain [02:29] <cone-116> ffmpeg.git 03Luca Barbato 07release/1.2:fffc9316daab: indeo: Properly forward the error codes [02:29] <cone-116> ffmpeg.git 03Luca Barbato 07release/1.2:221bbd002c62: atrac3: fix error handling [02:29] <cone-116> ffmpeg.git 03Michael Niedermayer 07release/1.2:5e3e67f977a3: avcodec/qdm2: store bits in an integer instead of float variable [02:30] <cone-116> ffmpeg.git 03Michael Niedermayer 07release/1.2:72c1de649ae9: avcodec/qdm2: initialize sign_bits [02:30] <cone-116> ffmpeg.git 03Michael Niedermayer 07release/1.2:47de8ccf4266: avfilter/af_earwax: Fix out of array accesses on odd packets [02:30] <cone-116> ffmpeg.git 03Michael Niedermayer 07release/1.2:c9ea1f7fc5e9: Merge remote-tracking branch 'jamrial/release/1.2' into release/1.2 [02:33] <cone-116> ffmpeg.git 03Michael Niedermayer 07master:5fe5f020b6cf: MAINTAINERS: drop 1.1 from the releases that i maintain [03:11] <cone-116> ffmpeg.git 03James Almer 07release/1.1:8695d814e1ad: oggparseskeleton: Fix fisbone header parsing [03:11] <cone-116> ffmpeg.git 03James Almer 07release/1.1:86a816902f36: oggparseskeleton: Replace avpriv_report_missing_feature() with a normal av_log() call [03:11] <cone-116> ffmpeg.git 03Michael Niedermayer 07release/1.1:944c47166d61: oggparseskeleton: avoid header parsing failure [03:11] <cone-116> ffmpeg.git 03Michael Niedermayer 07release/1.1:b0558cd01110: update all trac links to use the trac subdomain [03:11] <cone-116> ffmpeg.git 03Michael Niedermayer 07release/1.1:926071073986: Merge remote-tracking branch 'jamrial/release/1.1' into release/1.1 [03:16] <KGB> 01[13FFV101] 15michaelni pushed 1 new commit to 06master: 02http://git.io/DrOEgg [03:16] <KGB> 13FFV1/06master 1496d3e17 15Michael Niedermayer: ffv1: add missing alpha for line interleaved coding order [03:30] <cone-116> ffmpeg.git 03Timothy Gu 07oldabi:ea4e8e642acc: Deprecate oldabi branch by adding a warning to configure [12:29] <cone-834> ffmpeg.git 03Stefano Sabatini 07master:4aa2874ab1d0: lavc/avcodec.h: extend documentation for AVPicture API [12:46] <cone-834> ffmpeg.git 03Martin Storsjö 07master:1dd1b2332ebb: rtsp: Include an User-Agent header field in all requests [12:46] <cone-834> ffmpeg.git 03Michael Niedermayer 07master:2fe5e3e9702d: Merge commit '1dd1b2332ebbac710d8e0214cec7595e118f2105' [13:02] <durandal_1707> lol, why ffmpeg.org list all news from 2007 ? [13:10] <saste> durandal_1707, because we like to spam users with unrelevant info [13:11] <durandal_1707> iirc there was discussion on ml to resolve this, and iirc it was resolved, but it appears to appear again [13:14] <durandal_1707> and page linkt to censorship something, which can track our pageloads [13:17] <cone-834> ffmpeg.git 03Luca Barbato 07master:25a6666f6c07: indeo: Bound-check before applying motion compensation [13:17] <cone-834> ffmpeg.git 03Michael Niedermayer 07master:1cd74a9b4d6e: Merge commit '25a6666f6c07c6ac8449a63d7fbce0dfd29c54cd' [13:18] <durandal_1707> those requests in news ever did something? [13:21] <durandal_1707> saste: how is going that script filter thing? [13:26] <cone-834> ffmpeg.git 03Luca Barbato 07master:b0eeb9d442e4: indeo5: return proper error codes [13:26] <cone-834> ffmpeg.git 03Michael Niedermayer 07master:e3b13c108160: Merge commit 'b0eeb9d442e4b7e82f6797d74245434ea33110a5' [13:41] <saste> durandal_1707, nowhere, i'm not working on it [13:41] <durandal_1707> :(((( [13:41] <durandal_1707> didn't you at least started something? [13:46] <cone-834> ffmpeg.git 03Luca Barbato 07master:28dda8a691f1: indeo: Sanitize ff_ivi_init_planes fail paths [13:46] <cone-834> ffmpeg.git 03Michael Niedermayer 07master:8e4ecd8774d0: Merge commit '28dda8a691f1c723a4a9365ab85f9625f1330096' [14:04] <cone-834> ffmpeg.git 03Luca Barbato 07master:f9e5261cab06: indeo: Do not reference mismatched tiles [14:04] <cone-834> ffmpeg.git 03Michael Niedermayer 07master:c4b904caeaa7: Merge commit 'f9e5261cab067be7278f73d515bc9b601eb56202' [14:15] <cone-834> ffmpeg.git 03Luca Barbato 07master:5b2a29552ca0: indeo: Reject impossible FRAMETYPE_NULL [14:15] <cone-834> ffmpeg.git 03Michael Niedermayer 07master:fcbf16a76f35: Merge remote-tracking branch 'qatar/master' [14:52] <cone-834> ffmpeg.git 03Michael Niedermayer 07master:3ea765b0331b: avfilter/avf_concat: fix AVOption flags typo [15:47] <michaelni> anyone (BBB ? wbs ?) wants to review "0627 9:53 Ed Torbett ( 11K) [FFmpeg-devel] [PATCH] Joining a SSM multicast group using an SDP (Issue #2171)" ? [17:08] <durandal_1707> saste: whats about sox wrapper? [17:08] <durandal_1707> i may port (m)compand filter to help loudness war [17:09] <saste> durandal_1707, sox wrapper -> still incomplete [17:09] <saste> still has a few bugs i never debugged [17:22] <cone-834> ffmpeg.git 03Michael Niedermayer 07master:6158a3bcdf52: avformat/matroskadec: Detect conflicting sample rate/default_duration [17:27] <cone-834> ffmpeg.git 03Paul B Mahol 07master:5999db97cf26: lavfi/scale: remove av_opt_free() [17:57] <durandal_1707> ubitux: looks like skipping comments in lut3d for 3dl doesn't work [18:06] <cone-834> ffmpeg.git 03Alexis Ballier 07master:69383d055e77: ra144enc: set supported channel layouts. [20:25] <durandal_1707> wm4: send patch for #2179 ? [22:05] <cone-834> ffmpeg.git 03Paul B Mahol 07master:95126728a5bd: mxfenc: use ffio_fill() [23:51] <cone-834> ffmpeg.git 03Martin Storsjö 07master:8f24c12be7a3: ac3dec: Don't consume more data than the actual input packet size [23:51] <cone-834> ffmpeg.git 03Michael Niedermayer 07master:0b95f01140d0: Merge commit '8f24c12be7a3b3ea105e67bba9a867fe210a2333' [00:00] --- Tue Jul 16 2013

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ffmpeg.log.20130715
by burek 16 Jul '13

16 Jul '13

[00:00] <klaxa> no that'd be BTC ;) [00:05] <Macey> :) [00:07] <durandal_1707> Macey: so what is your target fps? [07:04] <tm512> I am having problems with ffmpeg's recorded sounds playing back at slower, and lower pitched [07:06] <tm512> the other day, it was working just fine [16:23] <leandrosansilva> Hello to all. I have an AVFrame with pixel format yuvj420p (read from a jpeg image). How can I access each channel (Y, U and V) from this frame? It's linesize field is {64, 32, 32, 0, 0, 0, 0, 0}, width = 52 and height = 52 [16:24] <leandrosansilva> ops height = 54 [16:26] <Mavrik> that looks planar [16:26] <Mavrik> so data[0] is first plane [16:26] <Mavrik> data[1] second, etc. [16:26] <Mavrik> judging buy the linesizes [16:26] <Mavrik> Y is in data[0] [16:26] <Mavrik> U and V in data[1] and [2] since they're half the first one and you have 4:2:0 [16:33] <leandrosansilva> ok, so as the first plane has size == 64, it means every byte is relative to a pixel? [16:41] <bencoh> linesize is the size (in bytes) of one line of your picture (including padding/alignement) [16:41] <bencoh> which means that line2 begins at start+linesize [16:52] <nb-ben_work> hi :) I am looking for encoding parameters for performance converting a .mov file to .ogv [16:52] <leandrosansilva> thx! [16:52] <nb-ben_work> I want the performance when decoding the file [16:52] <nb-ben_work> don't care if encoding will take a long time [17:04] <durandal_1707> nb-ben_work: and what is in mov? [17:04] <durandal_1707> decoding performance depends on what codec is used [17:04] <nb-ben_work> hd1080 movie, very high quality [17:04] <nb-ben_work> im not sure how it is encoded, is it possible for me to check? [17:06] <nb-ben_work> wonder if this FireFox OS on the raspberry is going to smoothly play this .ogv file I just encoded at full HD [17:06] <nb-ben_work> if it does I'm gonna go to sleep [17:06] <nb-ben_work> :D [17:09] <vincethechimp> If one would want to merge to videos vertically. What would one need to do? [17:09] <vincethechimp> two* [17:10] <saste> vincethechimp, overlay [17:55] <luc4> Hi! When muxing using the libavformat API, is it possible to write to a buffer instead of writing to a file? [17:55] <JEEBsv> yes [17:56] <luc4> JEEBsv: I currently use avio_open to open the output file. Should a use a different function maybe? [17:59] <mapreduce> Do the av* libs give me some hashtable I can use in my own stuff? [18:27] <SKiTZO> In what format is video in just before being displayed on screen by a video player, and would it be possible to cache the video in this format? [19:36] <IchGuckLive> hi i woudt like to shorten my video for uploading to Youtube as HD Reaseon is Capacity of Netline [19:36] <IchGuckLive> http://pastebin.com/6WUkKUh7 [19:36] <IchGuckLive> the file data [19:37] <IchGuckLive> can someone give me a hint half the size of 180MB so around 90 woudt be a perfect meet [19:37] <IchGuckLive> or even shorter if loss is acaptable [19:37] <klaxa> use -t? [19:38] <klaxa> you can use libx264 for starters [19:39] <klaxa> encoding will take longer though, but filesize will decrease [19:39] <IchGuckLive> on ubuntu i think itas not avalilable [19:39] <klaxa> --enable-libx264 [19:39] <klaxa> it's compiled with libx264 [19:39] <IchGuckLive> ok [19:39] <klaxa> try: ffmpeg -i input.mp4 -c:v libx264 -c:a copy output.mp4 [19:41] <klaxa> maybe add -profile:v high [19:41] <klaxa> so: ffmpeg -i input.mp4 -c:v libx264 -profile:v high -c:a copy output.mp4 [19:41] <JEEBsv> the profile thing does something only if you limit it [19:41] <JEEBsv> it doesn't do ANYTHING if you make it higher [19:41] <JEEBsv> (than your current settings) [19:41] <klaxa> ah [19:42] <JEEBsv> as in, if your preset doesn't use high profile settings you will not get a high profile flag [19:42] <JEEBsv> and setting that flag manually won't either [19:42] <klaxa> makes sense [19:42] <klaxa> so the preset is actually the important setting? [19:43] <JEEBsv> preset sets the defaults [19:43] <JEEBsv> those are then limited by the profile, if profile is set [19:44] <JEEBsv> presets control defaults (speed vs compression), tune sets possible tunings regarding source material, profile sets the profile you want to limit your stream to, level sets the level flag (does not enforce it! You have to set the refs yourself!) [19:46] <IchGuckLive> 160MB on 4min [19:46] <JEEBsv> if you are using the default rate control, then the default is -crf 23 [19:46] <JEEBsv> make it higher to make x264 compress more [19:46] <JEEBsv> as well as set a slower preset, perhaps? [19:47] <IchGuckLive> -crf 30 or more [19:47] <JEEBsv> http://mewiki.project357.com/wiki/X264_Settings#preset [19:47] <JEEBsv> list ^ [19:49] <IchGuckLive> Change options to trade off compression efficiency against encoding speed as you said but witch way is what [19:50] <IchGuckLive> Fast speed high compression slow encoding no compression [19:51] <JEEBsv> the faster something goes the generally less things it tries to do [19:51] <JEEBsv> thus worse compression [19:53] <IchGuckLive> ok but on finish now i got a 172Mb file at 4:06min the VLC shows no sound and no picture [19:55] <IchGuckLive> preset is the goal [19:56] <IchGuckLive> JEEB q= crf [19:56] <JEEBsv> no [19:56] <IchGuckLive> its noe at 29 [19:56] <JEEBsv> q=quantizer [19:56] <JEEBsv> crf is crf [19:56] <IchGuckLive> ok [19:58] <IchGuckLive> down to 150MB at ffmpeg -i gps2013_h.mp4 -preset veryfast -c:v libx264 -c:a copy output_test.mp4 [19:59] <JEEBsv> you are not touching the crf value [19:59] <JEEBsv> that controls the general quality level [19:59] <IchGuckLive> no i am [19:59] <JEEBsv> 23 is default [19:59] <JEEBsv> no you are not [19:59] <JEEBsv> I cannot see it [19:59] <JEEBsv> you just posted your command line [19:59] <IchGuckLive> i try now with 30 [19:59] <JEEBsv> also faster presets will NOT make shit compress better [19:59] <JEEBsv> as I said [19:59] <JEEBsv> the faster the preset, the less stuff is done [20:02] <IchGuckLive> ok q now at 36 fps has droped to 32 bitrate at 1900 [20:02] <IchGuckLive> 2min at 36Mb [20:04] <IchGuckLive> finish at 62MB ok so far and quality looks good [20:05] <IchGuckLive> ffmpeg -i gps2013_h.mp4 -preset faster -c:v libx264 -crf 30 -c:a copy output_test.mp4 [20:07] <IchGuckLive> http://www.youtube.com/watch?v=QD9gdCy6mNE this took 4hr to upload on 180MB Now i will be 3hr faster THANKS [20:09] <IchGuckLive> i will overview the kdenlive video output generator to do it as it renders in first place [20:13] <brontosaurusrex> isnt kdenlive using some sort of scripting engine? IchGuckLive [20:13] <brontosaurusrex> for the timeline i mean [20:13] <IchGuckLive> it uses [20:13] <brontosaurusrex> perhaps that is pipe-able to ffmpeg directly? [20:13] <IchGuckLive> no for rendering [20:16] <IchGuckLive> ishoudt try this with only 20sec vid as it takess 5min all the time to generate [20:16] <IchGuckLive> ok it works for me thanks alot to the heölpers and by [00:00] --- Tue Jul 16 2013

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ffmpeg-devel.log.20130714
by burek 15 Jul '13

15 Jul '13

[00:00] <cone-574> ffmpeg.git 03Paul B Mahol 07master:6347824d5310: lavfi/showwaves: fix floating exception with >8 channels [00:00] <cone-574> ffmpeg.git 03Paul B Mahol 07master:e9678631f16b: lavfi/trim: fix sample copy for >8 channels [00:00] <cone-574> ffmpeg.git 03Paul B Mahol 07master:729709b8904c: lavfi/asetnsamples: unbreak for >8 channels [00:06] <cone-574> ffmpeg.git 03Paul B Mahol 07master:099dfcaa0eef: lavfi/ashowinfo: unbreak for >8 channels [00:09] <durandal_1707> hmm why is stereo3d failing on big endian? [00:25] <cone-574> ffmpeg.git 03Paul B Mahol 07master:f486d7e924ff: lavfi/silencedetect: unbreak for unknown channel layouts [00:29] <durandal_1707> michaelni: are we want to be ABI compatible with Libav lavu ? [00:34] <michaelni> when possible, yes [00:35] <durandal_1707> its not possible at all [00:40] <michaelni> where is the problem? [00:42] <durandal_1707> sizeof(AVFrame) [00:43] <durandal_1707> so no, nonsense combinations of different versions on ffmpeg/libav lavf/lavc/lavu can not work [00:44] <michaelni> sizeof(AVFrame) cannot be used outside the lib defining that struct which is lavu, that has othing to do with ABI compat to the fork [00:45] <michaelni> we add fields to AVFrame ourselfs [00:45] <michaelni> and we need t be compatible to our previous ABI over such changes [00:46] <durandal_1707> i was not talking about our own ABI.... [00:47] <durandal_1707> Do you really think that our lavu and their lavc combination should be supported? [00:48] <michaelni> i dont care much about that, but all our libs together should work with applications build against the forks headers [01:03] <wm4> michaelni: I guess the lack of any reply means libswscale won't get changes wrt. chroma placement ever? [01:03] <michaelni> wm4, ill look into it [01:04] <wm4> michaelni: thanks [01:04] Action: durandal_1707 open bug reports all the time [01:07] <kierank> wm4: where is this chroma placement discussion [01:07] Action: kierank wrote his own chroma placement scaler [01:08] <wm4> kierank: nowhere, a user of my negligible project asked about it [01:08] <wm4> apparently videophiles think chroma placement is the most important thing ever [01:08] <kierank> it is quite important for interlaced video [01:09] <kierank> admittedly i ignored it for ages [01:10] <wm4> interlaced, I see [01:10] <kierank> http://renomath.org/video/linux/interlace/cinelerra.html [01:47] <cone-574> ffmpeg.git 03Paul B Mahol 07master:fc6ca3731638: lavfi/aecho: fix invalid free [03:49] <cone-574> ffmpeg.git 03Michael Niedermayer 07master:d5f5e5166216: swscale: move format handling to its own function [03:49] <cone-574> ffmpeg.git 03Michael Niedermayer 07master:0fc11e7bad71: swscale: make handle_formats() safe to be called multiple times [03:49] <cone-574> ffmpeg.git 03Michael Niedermayer 07master:c75dde607440: swscale: call handle_format() from the functions that need it [05:36] <wm4> michaelni: wow, that was fast! [05:50] <wm4> I wonder why the chroma position is not just specified like the constants in avcodec.h? [05:50] <wm4> it looks like this still requires the user to do some significant (pixel format specific) work to get it right [06:01] <michaelni> a different system could be used to specify them, but it should be kept flexible [06:01] <michaelni> the system in avcodec.h looks like it isnt able to handle some cases [06:03] Action: michaelni falls asleep [11:51] <cone-116> ffmpeg.git 03Nicolas Bertrand 07master:886e1b36f504: jpeg2000: Remove unused passes array in Jpeg200Cblk structure [11:51] <cone-116> ffmpeg.git 03Michael Niedermayer 07master:b24e8f566d0c: Merge commit '886e1b36f5044d3ceccbb01f64619acaf288fb7c' [12:09] <cone-116> ffmpeg.git 03Martin Storsjö 07master:68e57cde68f3: ac3dec: Increment channel pointers only once per channel [12:10] <cone-116> ffmpeg.git 03Michael Niedermayer 07master:198e10b55cc9: Merge commit '68e57cde68f3da4c557ca15491fda74d1ea6321e' [12:17] <cone-116> ffmpeg.git 03Martin Storsjö 07master:031be5b41b54: ac3dec: Consistently use AC3_BLOCK_SIZE and sizeof [12:17] <cone-116> ffmpeg.git 03Michael Niedermayer 07master:ac20ba35e37d: Merge commit '031be5b41b54c3b666f31d83fe3ad41c194af8c5' [12:40] <cone-116> ffmpeg.git 03Luca Barbato 07master:8435bca087c0: indeo4: Do not access missing reference MV [12:40] <cone-116> ffmpeg.git 03Michael Niedermayer 07master:54623619373d: Merge commit '8435bca087c0e79385763c51de009fd89390b6a5' [12:48] <cone-116> ffmpeg.git 03Luca Barbato 07master:6255ccf7d51c: indeo4: Check the quantization matrix index [12:48] <cone-116> ffmpeg.git 03Michael Niedermayer 07master:8c0bb195220d: Merge commit '6255ccf7d51c82ab79bf0cd47a921f572dda4489' [13:22] <cone-116> ffmpeg.git 03Luca Barbato 07master:cd78e934c246: indeo4: Validate scantable dimension [13:22] <cone-116> ffmpeg.git 03Michael Niedermayer 07master:ccb422a69728: Merge commit 'cd78e934c246d1b2510f8fba0abfe40bb75795f6' [14:19] <cone-116> ffmpeg.git 03Luca Barbato 07master:dc79685195a4: indeo: Bound-check before applying transform [14:19] <cone-116> ffmpeg.git 03Michael Niedermayer 07master:a8e5fac1fb3f: Merge remote-tracking branch 'qatar/master' [15:59] <wm4> a user has been pasting this to me, said it happens on freebsd but not on linux using the same ffmpeg version and the same input file, any idea what this could mean? http://sprunge.us/HRLW [16:02] <cone-116> ffmpeg.git 03Michael Niedermayer 07master:76d0a6656bbf: indeo: print errors if transform and block size mismatch [16:02] <cone-116> ffmpeg.git 03Michael Niedermayer 07master:febbddbdd5b2: indeo4: print an error message if ref_mb is needed but unavailable [16:12] <cone-116> ffmpeg.git 03Stefano Sabatini 07master:ab5f5816253a: doc/bitstream_filters: amend name of some bitstream filters [16:26] <Compn> wm4 : ask him to paste larger thing [16:26] <Compn> or submit input file [16:27] <Compn> wm4 : could be mixed mode h264 and theres problem , maybe thread problem ? what ver ffmpeg/libavcodec ? [16:43] <michaelni> wm4 are you working on passing yuv type from decoder to filter ? (if not i might take a look at it) [16:43] <wm4> michaelni: no... I suspect format negotiation in lavfi will be a major mess, and ubitux expressed some concerns about that too [16:44] <wm4> michaelni: but if you do it, IMO the same should be done for chroma location [16:44] <michaelni> yes i know [16:44] <michaelni> iam still a bit unsure how to specify the chroma location best [16:44] <michaelni> the enum looks insufficient though [16:45] <wm4> AVChromaLocation? [16:45] <michaelni> also i dont plan to do anything with format negotiaion just passing the info thrugh for now [16:46] <michaelni> yes AVChromaLocation looks insufficient [16:46] <wm4> AFAIR ubitux was arguing that the jpeg pixel formats make sure that filters which do not support different range are properly negotiated [16:47] <wm4> in which cases is the enum not enough? [16:47] <michaelni> consider croping and updating the chroma position or yuv 410 [16:48] <michaelni> or a single field of a interlaced frame [16:48] <wm4> so what would you argue for? shift values for x/y like in your swscale patch? [16:49] <michaelni> iam not sure but the shft values should be generic enough to handle all these cases [16:50] <michaelni> or rather allow them to be handled if someone implements the logic in the crop, ... filters [18:29] <cone-116> ffmpeg.git 03Paul B Mahol 07master:80c644593223: lavfi: port perspective filter from libmpcodecs [18:29] <cone-116> ffmpeg.git 03Paul B Mahol 07master:ed448efe619d: lavfi/mp: remove mp=perspective [20:05] <cone-116> ffmpeg.git 03Paul B Mahol 07master:d5598c0963d8: lavfi/drawutils: set subsampling for rgb too [21:52] <durandal_1707> michaelni: why all stereo3d tests are failing? [21:54] <durandal_1707> its probably because of 420->444 [22:12] <cone-116> ffmpeg.git 03Michael Niedermayer 07master:17b98c1abded: fate: fix stereo3d [22:13] <durandal_1707> well i was going to add subsampled support to stereo3d .... [22:40] <cone-116> ffmpeg.git 03James Almer 07master:04b9836274f3: oggparsevorbis: Support official chapter extension [22:41] <wm4> michaelni: would it be ok to cancel the genpts loop in libavformat/utils.c after a certain number of retries? [22:56] <michaelni> the problem is the "certain number" [22:57] <michaelni> we could just pick something arbitrary or let the user choose [22:58] <michaelni> or pick something so large it is unlikely to be too small [23:04] <wm4> michaelni: well, as I understand, genpts will make sense only as long as there are possible references between the frames [23:05] <michaelni> yes, you could have a P frame and a billion B frames following it that refer to it in theory, even i mpeg1 [23:08] <wm4> but wouldn't it stop as soon as a recovery point is reached? (or whatever that is called) [23:19] <michaelni> keyframes or recovery points would stop it eventually yes [23:19] <michaelni> if there are any keyframes or recovery points later that is [23:35] <BBB> oh someone is doing yuv type? [23:35] <BBB> _O_ [23:35] <BBB> finally [23:57] <cone-116> ffmpeg.git 03Paul B Mahol 07master:85a22099a7ba: lavfi/stereo3d: subsampled yuv support for non-anaglyph outputs [00:00] --- Mon Jul 15 2013

1 0

ffmpeg.log.20130714
by burek 15 Jul '13

15 Jul '13

[00:07] <defaultro_> hey folks, I'm getting an error invalid option or argument: directpred=3 I got the error when I included -vpre veryslow. This works before [00:10] <llogan> as usual [00:10] <defaultro_> it's ok now. I found a log and I was the same person who asked it few months ago. LOL [00:10] <llogan> use -preset, not -vpre [00:10] <defaultro_> I should use -preset [00:11] <defaultro_> yup [00:11] <defaultro_> i'm still reading the irc log, hehehe [00:12] <defaultro_> it's working now [00:45] <Nunnsy> Hi there, I'm trying to get ffmpeg to encode an mkv file to mpeg2 using a Raspberry Pi GPU. Currently, it's using the CPU for encoding and running around 4fps for a file 700x400 in resolution size. I know using another package called omxtx I can encode the same file at around 150fps to h264 but cannot stream it to another device. So my question is do any of you know id there's a way to interface ffmpeg with the GPU encoding on the Ra [00:45] <Nunnsy> I also have the MPEG2 license keys for it [00:48] <durandal_1707> no gpu encoding [00:49] <Nunnsy> No... it's not using it... [00:50] <llogan> maybe you can pipe from the omxtx thing to ffmpeg [00:50] <Nunnsy> I did try, but it seems it's just for playing on the Pi's video outputs. [00:51] <MrE> Dont ever expect that to work with ffmpeg, those gpu transcoders are pretty basic, x264 is problary a 1000 times better at encoding than the H264 rpi can gpu encode... [00:51] <Nunnsy> Ahh okay... [00:51] <Nunnsy> and I'm guessing there's no way for ffmpeg or mencoder to do that...? [00:51] <Nunnsy> to transcode to x264 [00:52] <llogan> "omxtx uses libavformat for demuxing, and then passes the video stream through an openmax decoder/encoder pipeline and writes the raw encoded video stream to the output file" [00:52] <MrE> there is but not using a gpu [00:52] <llogan> never heard of this omxtx before [00:52] <MrE> Its been arround for some time [00:52] <Nunnsy> Thanks MrE, and yeah, omxtx was able to transcode at around 150fps... [00:53] <Nunnsy> on the Pi [00:53] <Nunnsy> I'm just not sure how to stream the omxtx output to miniDLNA which is streaming the video to my media devices. (Mostly PS3) [00:56] <MrE> Yeah it will do quite some fps, but just dont think that the output of raspi can be compared to x264, like i said x264 will encode much better q at the same rate..... [00:57] <Nunnsy> Hmm, okay, thanks. So I guess I should be looking for ways to pipe omxtx to the media server instead... [00:58] <MrE> uhm [00:58] <MrE> why not pipe omxtx to ffmpeg [00:59] <Nunnsy> That's possible..? o.O [01:01] <Nunnsy> So if I just look into the ffmpeg executing command... I should be able to change it slightly from what the miniDLNA is currently using to pip from omxtx instead...? In theory? [01:01] <MrE> omxtx -r 1280x720 in.mkv - | ffmpeg -i in.mkv -i - -map 1:1 -map 0:1 -vcodec copy -acodec copy -f mpegts out.ts [01:01] <Nunnsy> *pipe [01:02] <Nunnsy> Starting to get this now, thanks :) So currently the command being executed to stream is [01:02] <Nunnsy> -ss $STARTPOSITION -i $SOURCE -t $DURATION -v 0 -loglevel quiet -threads 2 -async 2 -target ntsc-dvd pipe:1 [01:03] <Nunnsy> there'd be an ffmpeg at that start of that... [01:04] <Nunnsy> and then the media server fills the variables with the source file's attributes [01:04] <MrE> you wont be able to seek with pipes [01:04] <Nunnsy> that's alright, if I can get it working barebones, I'll be really happy [01:06] <MrE> doh [01:06] <MrE> omxtx -r 1280x720 in.mkv - | ffmpeg -i - -i in.mkv -map 1:1 -map 0:1 -vcodec copy -acodec copy -f mpegts out.ts [01:06] <MrE> like that or the piped video will be 1:0 [01:08] <Nunnsy> Alright... but if you're piping from omxtx, is it then possible to directly get the output of omxtx to the media server if I execute it from there? [01:08] <Nunnsy> So stream completely via omxtx instead of ffmpeg at all? [01:08] <MrE> omxtx got no audio support from what i can see [01:08] <Nunnsy> Because when I was looking up omxtx usage, I thought it could only output to a file or play onscreen... [01:08] <Nunnsy> really? [01:09] <MrE> and omtx can only output h264 bytestream as faar as i can see [01:09] <Nunnsy> I transcoded a file and it had audio? [01:09] <Nunnsy> ahh [01:09] <MrE> You transcoded a file with audio and got audio on the target file, hmm must have been a copy of the raw audio stream then [01:10] <MrE> Anyway if omtx can include audio and mux as dvd np then, but I doubt it :) [01:10] <Nunnsy> So I'm trying to understand this all... What's the difference between omxtx's bytestream and ffmpeg's output to the media server using "ffmpeg -ss $STARTPOSITION -i $SOURCE -t $DURATION -v 0 -loglevel quiet -threads 2 -async 2 -target ntsc-dvd pipe:1" ? [01:10] <MrE> -target ntsc-dvd pipe:1 [01:11] <Nunnsy> but ffmpeg doesn't output a bytestream? [01:11] <Nunnsy> I'm not exactly sure on what the end result difference is... [01:11] <MrE> That makes ffmpeg mux video as mpeg2 and audio as ac3 or mp2, on a vob container [01:12] <MrE> haha well maybe you need to read up on video encoding then first :D [01:13] <Nunnsy> And it's not possible for either ffmpeg or mencoder to mux to x264 or h264... But omxtx can but only as a bytestream which is usually writes to file. So ffmpeg is needed to pipe the data to the media server... [01:13] <Nunnsy> I really have been looking as hard as I can [01:13] <MrE> the diffrense is ffmpeg with those options encode a dvd target, omxtx will just output a h264 stream :) [01:13] <Nunnsy> this was really a last resort... [01:14] <Nunnsy> okay, but the PS3 can play both formats so it shouldn't matter..? [01:14] <Nunnsy> oh [01:14] <Nunnsy> ohhh [01:14] <Nunnsy> nope, gotcha now [01:14] <MrE> lol please dude read up on the encoding [01:14] <MrE> a h264 bytestream = no audio no nothing just video [01:14] <MrE> :) [01:15] <Nunnsy> yep, I get that now [01:15] <MrE> omxtx -r 1280x720 in.mkv - | ffmpeg -i - -i in.mkv -map 1:1 -map 0:1 -vcodec copy -acodec copy -f mpegts out.ts [01:15] <Nunnsy> sorry, I'll go try some piping now :) thanks for all the help, I'll ask some more dumb questions later if this doesn't work >.> [01:15] <Nunnsy> thank you [01:16] <MrE> np ^^ [01:18] <MrE> im gona try it out myself aswell, if a raspi on the fly encode sd stuff without probs I have wasted seriously manny money on i7 machines todo live transcoding O_o [01:20] <Nunnsy> ahahah, well, omxtx did look promising when I was messing around with it [01:23] <MrE> Depends on how good it is to pack down sd stuff to 1500 KB/s without looking to crappy :) [01:23] <MrE> Faster h264 encoders usualy mean worse quality.. [01:24] <Nunnsy> Too true... it's not great when you're trying to stream to a PS3 over a wireless connection too... [02:38] <defaultro_> can ffmpeg display the total number of frames of a video? [02:39] <defaultro_> maybe i'll just multiply total seconds times 29.97 [03:05] <defaultro_> i'm trying to fade an audio but it's failing, http://pastebin.com/NSWAzkNF [03:12] <beastd> defaultro_: maybe wrong filter? try afade [03:16] <defaultro_> ok [03:16] <defaultro_> i used sox and i got it working [03:16] <defaultro_> i'll try afade later [05:11] <elkng> is it possible to convert video file into a *.png images ? [05:11] <elkng> and at the same time resize them into 192x144 resolution ? [05:14] <elkng> I found that one "ffmpeg -i file -c:v png "%d.png"" but it generates file names like 1.png 20.png 150.png is it possible to have them with leading zero like 001.png 020.png ? [05:18] <elkng> "%3d.png" is working, but its adds two zeros even if I don't have files more than 99 and I don't need 001 just 01 or in case if I have more than 10000 files 3 zeros will not be enough [05:20] <klaxa> just make it %5d then? [05:29] <elkng> I mean you never know how much you need untill you get all frames extracted [05:39] <elkng> when I use options "-s 320x240" can I set only "width" part "-s 320" so it will calculate height on its own ? [05:40] <elkng> I don't want to calculate it but if I set wrong "height" the image will be stretched vertically or horizontally [05:40] <sacarasc> -vf scale=320:-1 or something. [06:32] <fling> I have strange file when joining smaller H246+mp3 mkv files [06:32] <fling> mplayer2 shows total length is 13 hours which is much longer [06:33] <fling> And from some frame I'm not able to rewind [06:33] <fling> Do I need to reencode? [06:34] <elkng> mplayer2 ? [06:34] <fling> elkng: yes. [06:35] <fling> elkng: this is what I do > ffmpeg -f concat -i mylist.txt -c copy output.mkv [06:35] <elkng> no idea [06:36] <fling> tried to remux to ts from the resulting mkv, got the same behavior in mplayer2 [06:37] <fling> elkng: tried to remux to mp4, found this > [mp4 @ 0x13d9b30] Non-monotonous DTS in output stream 0:0; previous: -32, current: -32; changing to -31. This may result in incorrect timestamps in the output file. [06:39] <fling> do I need -debug_ts? [06:46] <fling> I need to fix Non-monotonous DTS somehow [06:50] <fling> reencoded to huffyuw, got 97G output file still have wrong DTS :D [06:52] <fling> avidemux crashed [06:53] <fling> same output with mkvmerge but out of sync [06:58] <fling> fixed DTS with pipe! [06:58] <fling> ffmpeg -i test.mkv -c copy -bsf h264_mp4toannexb -f mpegts - | ffmpeg -i - -c copy test2.mkv [06:58] <fling> now rewind works fine [06:58] <fling> but the total video length is longer, it shows 9 hours now hmm hmhmmhm [07:03] <fling> should I split audio and video, encode to raw audio and video? [07:03] <fling> and then reencode and mux together? [14:50] <nnred73> hello [14:50] <nnred73> how to use ffv1 on MAX compression? [14:51] <nnred73> I exported help full file, but there is only one line about ffv1 [14:52] <nnred73> -slicecrc <int> E..V.. Protect slices with CRCs (from -1 to 1) (default -1) [14:52] <nnred73> there is an avi file uncompressed [14:53] <nnred73> execuse me for my english (if what) [14:54] <fling> nnred73: if what :P [14:54] <fling> Russia? [14:57] <nnred73> yes) [14:57] <nnred73> 40 [15:02] <Fjorgynn> :) [15:19] <nnred73> so, how to use ffv1 on max level compression? there is a file (avi, unpompressed with audio wav PCM) and the goal is to encode video (whithout any resize,fps change, etc) to ffv1 format and "copy" audio without encoding to avi (maybe it must be other container) [15:59] Last message repeated 1 time(s). [16:03] <nnred73> how to use ffv1 on max level compression? there is a file (avi, unpompressed with audio wav PCM) and the goal is to encode video (whithout any resize,fps change, etc) to ffv1 format and "copy" audio without encoding to avi (maybe it must be other container) [16:31] <iive> nnred73: ffv1 is lossless, so there are not many level of compressions. [17:20] <mroman> http://codepad.org/zhKWzfFK <- I'm trying to pipe some png images to ffmpeg to convert them to a video [17:20] <mroman> but it ain't working. [17:40] <fling> Same problem with another file. [17:40] <fling> [matroska @ 0x1ec5ea0] Non-monotonous DTS in output stream 0:1; previous: 3389550, current: 3389545; changing to 3389550. This may result in incorrect timestamps in the output file. [17:40] <fling> So how to fix this dts? [17:41] <mroman> puh [20:37] <joecool> i have a couple of packages failing with this build error after installing ffmpeg 2.0 [20:37] <joecool> error: AVCODEC_MAX_AUDIO_FRAME_SIZE undeclared [20:57] <Eduard_Munteanu> Hi. Is there an option to get more machine-readable output? [20:59] <Eduard_Munteanu> I mainly want to parse the "time=" field of the progress output. [23:03] <Macey> Hi All, i am trying to do a transcode and apply a BITC filter, if the target framerate is different to the incoming the resulting BITC is wrong [23:04] <Macey> how do i make the bitc follow the outgoing fps? [23:05] <Macey> http://pastebin.com/4wJPQP0w [23:05] <durandal_1707> BITC filter? [23:07] <durandal_1707> use fps filter? [23:09] <Macey> durandal_1707, i'm using the drawtext filter [23:09] <Macey> to do Burnt-In TimeCode [23:10] <Macey> unsure as to what i would do with the fps filter.. I've tried it before putting fps=fps=25 in front of the drawtext filter to no avail [23:52] <iive> there is bitcoin filter in ffmpeg :O ;) [00:00] --- Mon Jul 15 2013

1 0

ffmpeg-devel.log.20130713
by burek 14 Jul '13

14 Jul '13

[00:00] <michaelni> yes, i found no samples that use this codepath when i wrote it [00:12] <durandal_1707> BuxiNess: speed not match libopenjpeg or? [00:12] <durandal_1707> s/not/now [01:47] <durandal_1707> michaelni: try: ffmpeg -f lavfi -i smptehdbars=hd720 -pix_fmt rgb48 /tmp/o.jp2 [01:47] <durandal_1707> i get funny effect with jpeg2000 [01:49] <durandal_1707> ^the native decoder [02:28] <johnsmith> Hey all, has anyone here have had experiance using/playing around with av_write_frame(..)..? [02:28] <michaelni> BuxiNess, if you have time, one more jpeg2000 patch on ffmpeg-devel to review " [FFmpeg-devel] [PATCH] jpeg2000: fix overflow in dequantization" [12:28] <cone-574> ffmpeg.git 03Nicolas Bertrand 07master:f56fe04de3cc: jpeg2000: Initialize only once mqc arrays [12:28] <cone-574> ffmpeg.git 03Michael Niedermayer 07master:dc072c98677f: Merge remote-tracking branch 'qatar/master' [12:40] <saste> wm4, ping on demuxing.c patch [12:41] <saste> i'm not sure I want to add support for resampling though [12:46] <cone-574> ffmpeg.git 03Michael Niedermayer 07master:f57119b8e58c: jpeg2000: fix overflow in dequantization [14:11] <wm4> saste: I'm not sure whether it's worth fixing at all, because at least audio is usually planar, and it doesn't make sense to write planar into a raw file [14:24] <b_jonas> :( [14:25] <saste> wm4, the file is about demuxing, not about writing audio to a raw file [14:33] <b_jonas> argh, I'll have to chime in the mailing list because apparently they didn't read the bug ticket replies [14:33] <b_jonas> let me see where I subscribe [14:36] <b_jonas> though if the patch is safe in cvs and the question is only whether it should be included in the 1.2 branch, then I probably don't much care about that [14:36] <b_jonas> (re https://ffmpeg.org/trac/ffmpeg/ticket/2684 ) [15:09] <saste> wm4, are you going to repost your edited patch, or should I fix and apply it myself? [15:10] <wm4> saste: AFAIK the requested changes were kind of trivial, so feel free to apply them yourself [15:10] <wm4> if it's more convenient for you that I make the changes I can do that too, though [15:11] <wm4> does anyone know whether decoder stereo downmix or libswresample/libavresample is preferable? any quality or correctness differences? [15:11] <saste> wm4, as you prefer [15:11] <saste> i prefer you do the work ;-) [15:12] <wm4> ok [15:13] <cone-574> ffmpeg.git 03Michael Niedermayer 07master:786b0968dd29: ffplay: remove explicit idct option [15:14] <wm4> saste: btw. are you saying that FFMIN(decoded, pkt.size) is unneeded and just decoded should be returned? [15:16] <michaelni> wm4, decoders can in theory do downixing often with less cpu than full decode and downmix outside [15:16] <wm4> michaelni: yes, but are there differences in quality and correctness? [15:16] <saste> wm4, I'm not sure if video_decode() returns the size of the decoded data [15:17] <saste> in that case you don't need to make a distinction, you can assume ret = decoded_data_size [15:17] <wm4> saste: there's some extra code in the video path: "+ decoded = pkt.size;" [15:18] <saste> yes, it's that i don't know which is the meaning of the return value, that's why I asked (and I put a "?") [15:18] <saste> but maybe my remark was not founded [15:18] <wm4> saste: well, the example's decode_packet function returns how much of the packet was decoded [15:19] <wm4> saste: I'll make it a little bit more explicit [15:20] <michaelni> the file might store a downmix matrix, that matrix might be used by the decoders downmix, and the decoder might export that downmix and some application might pass that on to swr. [15:21] <michaelni> depending on which of these "mights" are true one could be better or more correct than the other [15:22] <lukaszmluki> Hi, Is Nicolas George here? [15:23] <saste> lukaszmluki, no, he's almost never on irc [15:23] <lukaszmluki> ok, thanks [15:28] <wm4> also why does demuxing.c allocate a sample buffer? [15:31] <wm4> also maybe this really should be done in libavcodec/utils.c instead [15:36] <michaelni> khali, as you are in MAINTAINERS you should be op in here [15:41] <wm4> so, why is there no simple do-planarize function [15:41] <wm4> this is probably what 99% of libavcodec users going to need [15:41] <wm4> s/do-planarize/de-planarize/ [15:45] <saste> wm4: this is required since rawaudio expects non aligned data [15:46] <wm4> is there a planar raw audio demuxer that can actually read this stuff? [15:46] <saste> wm4, no [15:46] <wm4> then it makes more sense to write the first plane only [15:46] <wm4> which is much simpler [15:47] <saste> can i force the codec to write 1-aligned data? [15:47] <wm4> I mean: fwrite(frame->extended_data[0], 1, frame->linesize[0], audio_dst_file); [15:47] <saste> wm4, yes [15:47] <saste> when i wrote the example, i was striving to keep A/V symmetry [15:47] <saste> there is probably no sense into it [15:48] <saste> also i wanted to avoid to rely on other libraries (lswr, lsws) [15:49] <wm4> I can actually play the video output with mplayer's rawvideo [15:50] <saste> in theory we could pass a linesize argument to the rawaudio demuxer, but that would be rather artificial [15:50] <saste> since there are no such planar rawaudio files on the wild [15:51] <wm4> hm for some reason I can get clear audio with luckynight-partial.shn, but not with some other rather normal mkv file [15:51] <wm4> anyway. [16:04] <khali> michaelni: as you wish :) I don't really care [16:05] <khali> michaelni: I have a fix for the coverity issue, I must go shopping now, I'll post it when I return [16:05] <michaelni> ok [16:05] <khali> michaelni: is there a way to ask Coverity to test with the patch before we apply it? [16:05] <michaelni> AFAIK no [16:26] <cone-574> ffmpeg.git 03Michael Niedermayer 07master:4cdb42b428ef: avcodec/012v: improve stride selection [16:41] <khali> michaelni: should I put any reference in the patch header when fixing a Coverity issue? [16:46] <cehoyos> khali: Yes, please add such references, see for example f73d6d2af7a2b30a647d945b590962a2ab54ff3b [16:51] <khali> done, and sent [17:03] <khali> and now is swimming pool time - -ETOOHOT [17:14] <saste> kierank, ping [17:14] <kierank> saste: pong [17:14] <saste> kierank, hey! [17:15] <saste> i'm doing evil things with H.264+MP4, I want to encode H.264 once, and store to MPEG-TS and to MP4 at the same time [17:15] <kierank> you'll have to either duplicate the sps/pps [17:15] <kierank> or rip it out [17:15] <saste> because of this i hacked the tee_bsfs option, in order to apply separated bitstream filters to the same input [17:16] <cone-574> ffmpeg.git 03Jean Delvare 07master:b6d0bb6086d1: lavfi/delogo: Fix sign extension issue [17:16] <cone-574> ffmpeg.git 03Jean Delvare 07master:9e4bb091401f: lavfi/delogo: Document where parameter show=1 draws its rectangle [17:16] <saste> the problem is that i'm fighting with extradata stuff, I'm using the command: [17:16] <saste> $ffmpeg -i $input -map 0 -flags:v +global_header -c:v libx264 -flags:a +global_header -c:a libvo_aacenc -bsf:v remove_extra -f ssegment -segment_format mpegts out-%03d.ts [17:17] <saste> which is failing in the bitstream filter [17:17] <saste> it seems I cannot set global header (required by MP4) and then strip it with h264_mp4toannexb [17:18] <saste> kierank, >you'll have to either duplicate the sps/pps [17:18] <saste> to your knowledge this can/could be done with some bitstream filter? [17:18] <kierank> i don't see why not [17:18] <saste> my attempts with h264_mp4toannexb and remove_extra failed [17:19] <cehoyos> kierank: Can you analyze the mp4 stream at the bottom of ticket 2777? FFmpeg thinks that recovery points only happen every ~17 seconds. [17:20] <kierank> cehoyos: yes [17:20] <cehoyos> Please do so;-) [17:20] <kierank> 5 mins to download [17:21] <cehoyos> I am afk but I will see your conclusions - than you! [17:24] <saste> kierank, it's always failing at line h264_mp4toannexb.c:84 [17:24] <saste> extradata + 2 + unit_size > avctx->extradata + avctx->extradata_size [17:26] <kierank> saste: not sure about the bsf [17:26] <kierank> i've never used it before [17:26] <saste> kierank, it's required when copying from MP4 -> MPEG-TS [17:35] <kierank> cehoyos: there don't appear to be any recovery points in that file [17:37] <kierank> there are points which probably are recovery points but none are flagged correctly [17:46] <cehoyos> Does that mean that FFmpeg behaves correctly or incorrectly? [17:47] <cehoyos> by skipping several hundred frames before showing anything [18:01] <kierank> Correctly I guess [18:09] <saste> kierank: SPS/PPS -> these are H.264 specific terms, not directly related to MP4/MPEG-TS, right? [18:09] <kierank> Yes [18:10] <saste> too many unknown terms in my equation, i'll probably give up... [18:10] <kierank> The storage of them is different [18:10] <kierank> Mp4 had global header [18:10] <kierank> TS has repeated on keyframes [18:11] <JEEB> in-band vs out-of-band [18:11] <saste> i suppose this is stored in the codec extradata, and updated during the encoding process [18:11] <cone-574> ffmpeg.git 03Carl Eugen Hoyos 07master:149def7781fd: configure: Only build ffserver if SA_RESAMPLE is available. [18:11] <cone-574> ffmpeg.git 03Carl Eugen Hoyos 07master:8e3c5c70ca9f: g726: Do not try to decode more than one channel. [18:11] <cone-574> ffmpeg.git 03Carl Eugen Hoyos 07master:aecb9d39bc66: rmdec: Forward error messages from rm_assemble_video_frame() to the caller. [18:11] <saste> but then the encoder needs to be aware of the format container, since different format will require different headers [18:12] <saste> so basically I don't know how I am supposed to interpret extradata (and debug related problems) [18:14] <saste> cehoyos, no commit hash? [18:14] <cehoyos> For a ticket that nobody is interested in? [18:14] <cehoyos> I will close the rv ticket later with all three hases. [18:14] <cehoyos> hashes. [18:14] <saste> cehoyos, how do you know people is not interested? [18:15] <cehoyos> That qnx compilation was broken for a long time and nobody reported it and that nobody set the ticket to "reproduced" (I cannot reproduce it) [18:15] <cehoyos> (It was broken because of a change in qnx) [18:16] <cehoyos> But I sent an email to the mailing list, that is important imo;-) [18:21] <cehoyos> If anybody wants to comment on ticket 2777, please do so! [18:22] <cone-574> ffmpeg.git 03Carl Eugen Hoyos 07master:289737c8be28: configure: Disable avresample if an old version of yasm is used. [20:16] <durandal_1707> michaelni: see 374 line of libswresample/swresample.c [20:17] <durandal_1707> it asserts if layout is unset [20:17] <durandal_1707> and there is probably lavfi thing that resets channel count using layout 0 [20:17] <durandal_1707> which is retarded [20:18] <durandal_1707> for testing remove check at 128 of libavfilter/asrc_aevalsrc.c [20:19] <durandal_1707> and than try to create 9 channe; audio: ffmpeg -f lavfi -i aevalsrc=0|0|0|0|0|0|0|0|0 [20:22] <durandal_1707> also try this: cat /dev/zero > /tmp/out.pcm [20:23] <durandal_1707> and abort after file reaches few MBs [20:23] <durandal_1707> ffmpeg -channels 22 -f s16le -i /tmp/out.pcm /tmp/o.wav [20:24] <cehoyos> ffmpeg -ac 22 -f s16le -i /dev/zero ? [20:24] <durandal_1707> produced wav can't be filtered, afade,aecho,volume gives "Error while decoding stream #0:0: Cannot allocate memory" [20:24] <durandal_1707> cehoyos: yes [21:14] <cone-574> ffmpeg.git 03Michael Niedermayer 07master:0a06e6e49b9b: av_frame_get_buffer: Do not fail when the channel layout is unknown [22:18] <cone-574> ffmpeg.git 03Paul B Mahol 07master:673d4e82d4c5: fate: add tests for stereo3d filter [22:33] <durandal_1707> when will it be possible to burn subtitles stream? [22:33] <cehoyos> works fine here [22:35] <durandal_1707> with filters? [22:35] <cehoyos> I suspect it does not work without filter or do I misunderstand? [22:35] <durandal_1707> subtitle filter takes files [22:36] <wm4> ubitux kept talking about that [22:36] <cehoyos> You mean text subtitles have to be written to an ass file first? [22:36] <wm4> you'd need to add a subtitle path to lavfi [22:36] <wm4> cehoyos: any file type should work [22:37] <durandal_1707> wm4: then add it :) [22:37] <wm4> it doesn't have to be ass, as long as there's a lavc decoder AFAIK [22:37] <wm4> where decoder means converter [23:17] <cone-574> ffmpeg.git 03Michael Niedermayer 07master:f3d138ab2292: avdevice/lavfi: fix input with unknown channel layout [23:17] <cone-574> ffmpeg.git 03Michael Niedermayer 07master:aac57c765ff4: avfilter/asrc_aevalsrc: fix cases of unknown channel layout [23:50] <wm4> michaelni: any chance to get correct chroma placement in libswscale? currently it seems to always use mpeg1 placement, but mpeg2/h264 placement would be better [23:55] <Compn> chroma placement [23:56] <wm4> Compn: ? [23:57] <durandal_1707> michaelni: bunch of filters use av_get_channel_layout_nb_channels(), this just does not work for >8 channels [00:00] --- Sun Jul 14 2013

1 0

ffmpeg.log.20130713
by burek 14 Jul '13

14 Jul '13

[00:20] <Wulf> Hi [00:21] <Wulf> first time trying to use ffmpeg. I want to convert a part of a video to h.264, but not with high quality. I tried "-q:v 31" and "-q:v 1" and the resulting file size is the same. [00:21] <Wulf> What am I doing wrong? [00:21] <sacarasc> Try -crf [00:22] <sacarasc> And don't specify a bitrate. [00:23] <Wulf> sacarasc: I can't find "-crf" in the manual, and it doesn't work. "Trailing options were found on the commandline" [00:24] <sacarasc> -crf is x264's quality control thing. Goes from 0 (lossless) to 51 (worst quality). [00:26] <Wulf> sacarasc: okay, got it working. 51 looks really bad. [00:27] <Wulf> 35 looks fine [00:28] <Mavrik> that's probably why 23 or so is the default :P [01:32] <roboman2444> so... opencl support... [01:36] <llogan> used by deshake and unsharp filters. im not sure what else does [01:38] <llogan> "highgod" would know more, but he's not here. [02:44] <roboman2444> llogan, so no opencl for any encoding? [02:44] <roboman2444> i would DEFINATELY love to encode 1080p video at like 200fps [04:59] <uersbob> hi. im writing a simple screen capture program. i'm using the following line in ffmpeg ' -f image2pipe -i pipe:.bmp -c:v libx264 -y "output.mp4" 'this works well, but the output video plays at 25fps (or whatever i set it to). Is there a way to have the output play at the same speed as the capture speed? the frame rate changes constantly during each image save [07:48] <blehtm> I'm trying to slice up a 60 second flac file into 15 second segments. To get the first segment I use the following command: "ffmpeg -i infile.flac -acodec copy -t 15 -ss 15 outfile.flac". But I'm having a problem. When I load outfile.flac into VLC, the track slider still thinks it is 60 seconds long. Any idea what gives? [07:48] <blehtm> By the way once it reaches 15 seconds playback does stop as intended. [07:49] <blehtm> (also, the command I showed would give the second segment, not the first, sorry) [07:51] <roboman2444> uersbob, this is what i use... ffmpeg -threads 0 -i hw:0,0 -f x11grab -framerate 30 -s 1366x768 -r 30 -i :0.0+0,0 -vcodec libx264 -crf 16 output.mp4 [07:52] <mark4o> blehtm: what version? [07:52] <roboman2444> -framerate 30 and -r might be the answer [07:54] <blehtm> mark4o: "ffmpeg version N-54643-g15cee5e" on Windows 8, build from http://ffmpeg.zeranoe.com/builds/ 64-bit Shared version [08:13] <mark4o> blehtm: I see the same issue [08:15] <blehtm> interesting [08:15] <mark4o> blehtm: re-encoding works, and is of course lossless [08:15] <mark4o> looks like an issue/limitation with flac & -acodec copy [08:16] <mark4o> may want to file a ticket; not sure if it is a known limitation [12:08] <MojoRisin> hello there people [12:08] <MojoRisin> Its been years since I've come in here :) [12:09] <MojoRisin> I need some help with ffpeg streamng multicast... Been developing a small server for a lot of things, one of them being multicast streaming. I thought I could use vlc for it but the command line interface lacks information and i just simply cant do it [12:10] <MojoRisin> so, i've turned to ffmpeg. I thought of a simple shell script to launch 2 streams over multicast. What is your recommendation for this? Should I use flv format and libx264 vcodec and mp3 as audio? [12:11] <MojoRisin> i dont need quality - just need to make it work for a prototype [12:34] <MojoRisin> nevermind, vlc appears to be working [12:34] <MojoRisin> ffmpeg and ffserver wold be better, but i need this to work asap [12:34] <MojoRisin> cheers! [12:47] <Russel-Athletic> hiho, i am trying to convert a vob file to mkv with this snd pass cmd: ffmpeg -i out.vob -c:v libx264 -preset slow -b:v 900k -pass 2 -c:a:1 libvorbis -qscale:a 6 output.mkv [12:47] <Russel-Athletic> but somehow i get audio out of sync in the mkv [12:47] <Russel-Athletic> in the vob everything is fine [18:13] <EugenA> ffmpeg -threads 4 -i 00003.MTS -b:v 900k -vcodec mpeg2video -sameq -s 720x576 00003.mpg [18:13] <EugenA> 1) why the bitrate is still about 7000 kbits/s ? [18:13] <EugenA> 2) running only on 1 core, why? [18:14] <sacarasc> You put the threads part in the decoding section, and you used -sameq. [18:14] <klaxa> you are using -sameq, this is a) probably not what you think, b) discouraged c) messes with your quality settings [18:15] <JEEB> 1) you set two differing rate control modes (bit rate and sameq which copies over quantizers from the origin [AKA not "same quality" if you thought it was that]) [18:15] <JEEB> 2) you most probably will need to try setting threads on the encoding side (after -i) [18:15] <JEEB> if you want to set a bit rate, you set a bit rate. If you want to set a quality level, you set a quantizer or with libx264 a crf value. [18:17] <EugenA> thank you, understood [18:20] <EugenA> cpu is still not completly loaded: ffmpeg -i 00003.MTS -threads 4 -b:v 6000k -vcodec mpeg2video -s 720x576 00003.mpg [18:21] <JEEB> then the encoder just isn't that well threaded or whatever :D [18:21] <JEEB> what else can I say, really [18:21] <EugenA> ok :-) [22:03] <miguel_> Im trying to record my screen with ffmpeg, but the audio gets delayed [22:03] <miguel_> is there an option I can set to change the delay? [22:04] <miguel_> or to make sure the screen and audio recording start simultaneously? [22:22] <MrE> Can ffmpeg overlay text based dvb subs on a live feed, its a live feed so I cant convert its subs to srt or ass first to use with the subtitles or ass filter, it has to be a oneliner :) [22:23] <MrE> And why does copying the dvbsub track or encoding it as dvdsub result in a target stream/file that no player will open ? Is the dvbsub muxer just totaly broken ? [22:24] <Mavrik> no to the first point [22:24] <Mavrik> and yes to the second ;) [22:26] <MrE> Haha ok fair enough :D [22:31] <MrE> Mavrik Do you know about the status of overlaying picture based dvb subs then, seems to work with filter complex blah blah, but I just cant get those damn subs to be in the right place using anything larger than a sd source and sd target, on hd i end up with this : [22:31] <MrE> http://10only.explode.dk/screenshot003.png [22:32] <Mavrik> is that VLC? [22:33] <MrE> The player is xbmc, its for a rather complex iptv system, but for that picture ffmpeg is the encoder using the overlay filter, but the sub location is quite odd [22:33] <MrE> It gets worse when using a 1080i source [22:33] <MrE> http://10only.explode.dk/screenshot004.png [22:34] <Mavrik> yep [22:34] <Mavrik> that's because you have picture subtitles [22:34] <Mavrik> and those are always rendered at SD resolution [22:34] <MrE> Aha so atm no way to fix that ? :) [22:34] <Mavrik> and the DVB standard demands that TVs render them at 720x576 and then upscale [22:34] <Mavrik> MrE, fix the filter ;) [22:34] <Mavrik> send a patch. [22:34] <Mavrik> :P [22:35] <Mavrik> I wrote my own burn-in subtitle renderer for those cases so I have no idea what the state of current ffmpeg implementation is [22:36] <MrE> VLC really got this working, just add soverlay to the transcode line, and --sub-language=dan [22:37] <Mavrik> VLC has a failing subtitle renderer which doesn't upscale subtitles [22:37] <Mavrik> at least it had in 2.0 [22:37] <MrE> But their encoder have all kinds of other bugs that makes it crap to use in my situation, we build it arround ffmpeg so it really only works good with that [22:38] <Mavrik> MrE, what's your command line_ [22:38] <MrE> might have to find a solution to using vlc as the decoder and overlay, and then somehow pipe it as yuv to ffmpeg, with audio!!! thats the hard part hoho :) [22:39] <MrE> for ffmpeg subtitle overlay with picture based subs ? [22:40] <MrE> Pretty default the usual vcodec libx264 and all its setting and shit, and then this part for burning the subs [22:40] <MrE> -filter_complex "[0:v][0:s]overlay[v]" -map [v] -map 0:a [22:41] <Mavrik> MrE, well, did you try scaling the subtitle input before overlaying it? [22:41] <MrE> No cant find any documentation on it [22:42] <Mavrik> on what? [22:42] <MrE> scaling subs [22:42] <Mavrik> um [22:42] <Mavrik> it's just a stream [22:42] <Mavrik> apply scale filter to size of video [22:43] <Mavrik> check the filter_complex syntax, it's not that hard [22:43] <Mavrik> would be probably something like [22:43] <Mavrik> [0:s]scale=1920:1080[scaled_subs],[0:v][scaled_subs]overlay[v] [22:50] <MrE> Hmm that did something but not what was intended [22:52] <Mavrik> well I did write it out of my head [22:52] <Mavrik> :P [22:53] <MrE> True but it should work, its still in the middle of the screen [22:54] <MrE> This is my complete command [22:54] <MrE> ./ffmpeg -re -threads 4 -analyzeduration $analyzeduration -y -i "$link" \ [22:54] <MrE> -vcodec libx264 -preset $x264preset -vb $x264vb -minrate $x264minrate -maxrate $x264maxrate -bufsize $x264bufsize -r 25 -g 25 -acodec copy -vf yadif=0,scale=$res1:$res2 -f mpegts \ [22:54] <MrE> -filter_complex "[0:s]scale=1920:1080[scaled_subs],[0:v][scaled_subs]overlay[v]" -map [v] -map 0:a \ [22:54] <MrE> "udp://$2:5501?ttl=1&pkt_size=1316&buffer=262140" & [22:54] <MrE> $res1 is = 1280 $res2 is = 720 [22:56] <MrE> Heh my vf line is not evalutated with filter complex on also, gotta put that shit inhere also [22:58] <Mavrik> doh [22:58] <Mavrik> :) [23:00] <MrE> [dvbsub @ 0x2979480] Invalid object location! 78-720 38-38 11 [23:05] <vhann> Hi, is there a way to normalize audio in a .avi,.mp4,etc. file so the max volume is 0 dB with ffmpeg? The manpage and Google don't seem to help me here [23:06] <Mavrik> vhann, ffmpeg processes files sequentally so normalization really can't be done easly [23:06] <Mavrik> (since ffmpeg doesn't know what max volume is) [23:06] <MrE> Looks like they are located just where you would expect on a 720x576 resolution now but the size of the subs got better, how do i force the location to the bottom ? [23:07] <Mavrik> MrE, overlay filter almost certanly has positioning parameters. [23:07] <vhann> Mavrik: Oh ok. Thanks [23:07] <Mavrik> vhann, if it helps, there is a volume audio filter (http://www.ffmpeg.org/ffmpeg-filters.html#volume) which allows you to change volume [23:08] <MrE> Yeah but the man page its quite cryptic about em, oh well maybe i can find an example somewhere with google :p [23:08] <Mavrik> vhann, there is also "volumedetect" filter which outputs volume stats [23:09] <Mavrik> so using a script you can run ffmpeg first with volumedetect and use that output with "volume" filter to normalize it [23:09] <Mavrik> MrE, don't read the man page [23:09] <vhann> Mavrik: Yes, I'll experiment with this. If I'm lucky and it works, I might be able to run volumedetect, then calculate the needed amplification and rerun ffmpeg [23:09] <Mavrik> MrE, read documentation pages I'm linking -_- [23:10] <MrE> ye that was also what i ment [23:10] <MrE> this page [23:10] <MrE> http://ffmpeg.org/ffmpeg-all.html#overlay-1 [23:11] <Mavrik> yep [23:11] <MrE> But already the first example [23:11] <MrE> overlay=main_w-overlay_w-10:main_h-overlay_h-10 [23:11] <Mavrik> yes? [23:11] <MrE> Does absolutely nothing to my subs, should locate then in the bottom right corner [23:11] <MrE> them* [23:12] <Mavrik> why are you just blindly typing random stuff from the internet without reading what it does? :P [23:12] <MrE> Ehh [23:12] <MrE> Draw the overlay at 10 pixels from the bottom right corner of the main video: [23:13] <MrE> That shoud count for all overlays also subs ? [23:13] <Mavrik> remember [23:14] <Mavrik> you're incoming frames are 1920x1080 because you're scaling them beforehand [23:19] <MrE> Hmm yeah a 1920x1080 image can only be overlayed at one fixed pos over another 1920x1080 image, so why do the subs end up in the middle of the screen :D [23:19] <Mavrik> well [23:19] <Mavrik> from which size are the subs resized from? [23:22] <MrE> Hm no idea, my guess would be 720x576 [00:00] --- Sun Jul 14 2013

1 0

ffmpeg-devel.log.20130712
by burek 13 Jul '13

13 Jul '13

[00:43] <michaelni> durandal_1707, it sounds usefull [02:27] <cone-947> ffmpeg.git 03Carl Eugen Hoyos 07master:0fff7f039c14: Supply a User-Agent header when opening rtsp streams. [02:58] <cone-947> ffmpeg.git 03Thierry Foucu 07master:38ce775eead7: libavformat/mov.c: Skipped duplicated MOOV atom [03:24] <michaelni> aballier, if you want to maintain 1.0, please send a patch for MAINTAINERs and a link to the git repo where you (will) maintain 1.0 [04:27] <cone-947> ffmpeg.git 03Piotr Bandurski 07master:161047f010b1: rmdec: fix crash in case of oom [09:16] <xlinkz0> I'm reading from a rtsp stream and want to save to a different file every 60 seconds so every 60 seconds I create a new output context and save offsets for pts and dts so the new file starts from pts and dts 0 [09:17] <xlinkz0> what are some other corrections i need to do to the packets that come from the input context? [09:17] <xlinkz0> is packet->pos important? [09:17] <av500> no [09:18] <av500> pos would refer to the byte offset in the rtsp stream [09:18] <xlinkz0> where is ffprobe getting coded_picture_number ? [09:20] <xlinkz0> also, ffprobe complains about the file that is created after the first file, it says : Missing reference picture, default is 0 [09:20] <xlinkz0> does it identify the ref picture by dts, pts = 0 ? [09:20] <av500> well, if you cut a stream at a random point, of course there will be erros [09:20] <av500> errors [09:21] <xlinkz0> why? [09:21] <av500> because frames are not independent [09:21] <xlinkz0> ah i should cut only on iframes [09:22] <xlinkz0> but how do i get that information without decoding? [09:23] <xlinkz0> ffprobe decodes every frame? [09:29] <av500> no [09:29] <av500> there is a key frame flag [09:29] <xlinkz0> in AVFrame [09:29] <av500> in avpakchet [09:29] <av500> avpacket [09:29] <xlinkz0> http://ffmpeg.org/doxygen/trunk/structAVPacket.html [09:29] <xlinkz0> can't see it :\ [09:29] <xlinkz0> ooooh [09:30] <xlinkz0> flag [09:30] <xlinkz0> thanks, will try it [09:53] <funman> hello [10:00] <xlinkz0> av500: so i need to cut on packets that have the AV_PKT_FLAG_KEY flag right? [10:01] <xlinkz0> do i need to include that packet in both the previous and the next file? [10:03] <av500> in the next file [10:03] <av500> but in case of mpeg2 that is not enough [10:03] <xlinkz0> h264 mp4 [10:03] <xlinkz0> thanks [10:03] <av500> you need a "closed gop" [10:27] <wm4> who really uses the 0.5/0.7 branches? [12:13] <cone-138> ffmpeg.git 03Martin Storsjö 07master:ef5b70affc63: configure: Check for support for labels in the inline assembly [12:13] <cone-138> ffmpeg.git 03Michael Niedermayer 07master:c6953589f9c4: Merge commit 'ef5b70affc6376bfeadd1ff649b79bad9a124fa8' [12:41] <cone-138> ffmpeg.git 03Luca Barbato 07master:170fb593c617: pcx: K&R formatting cosmetics [12:41] <cone-138> ffmpeg.git 03Michael Niedermayer 07master:54bbb9056801: Merge commit '170fb593c617cd797d00cf05bfb8d8c21d612893' [12:55] <Guest80280> I'm not sure if there's anyone responsible for the homebrew formula here, but does anyone know if there's an updated version for ffmpeg 2.0 in the works? [12:56] <av500> 2.1? [13:01] <michaelni> Guest80280, whats "homebrew formula" ? [13:01] <michaelni> and 2.0.1 is planed in 1-2 weeks [13:03] <relaxed> michaelni: I think it's like FreeBSD's ports for OS X. [13:03] <Guest80280> Mac users install ffmpeg via homebrew, but the formula "ffmpeg" links to is still on 1.21 https://github.com/mxcl/homebrew/commits/master/Library/Formula/ffmpeg.rb [13:04] <michaelni> ahh, understand [13:07] <Guest80280> yeah, see here http://ffmpeg.org/trac/ffmpeg/wiki/MacOSXCompilationGuide Homebrew just makes it easy to install, but it hasn't been updated yet [13:08] <cone-138> ffmpeg.git 03Luca Barbato 07master:3abde1a3b49c: pcx: Do not overread source buffer in pcx_rle_decode [13:08] <cone-138> ffmpeg.git 03Michael Niedermayer 07master:50617fce073c: Merge commit '3abde1a3b49cf299f2aae4eaae6b6cb5270bdc22' [13:17] <cone-138> ffmpeg.git 03Luca Barbato 07master:2b379a925162: mlpdsp: x86: Respect cpuflags [13:17] <cone-138> ffmpeg.git 03Michael Niedermayer 07master:9d47333e3eee: Merge commit '2b379a925162b6783bd9a81dc03e647e8b65494c' [13:36] <cone-138> ffmpeg.git 03Luca Barbato 07master:e9d394f3fad7: mlpdec: Do not set invalid context in read_restart_header [13:36] <cone-138> ffmpeg.git 03Michael Niedermayer 07master:e9348e107baf: Merge commit 'e9d394f3fad7e8fd8fc80e3b33cb045bbaceb446' [13:36] <durandal_1707> a warning should be printed if user give multiple -vf/-af [13:48] <cone-138> ffmpeg.git 03Luca Barbato 07master:c0d973c41b45: vdpau: use the correct namespace for the union [13:48] <cone-138> ffmpeg.git 03Michael Niedermayer 07master:0da0caa9a9ea: Merge commit 'c0d973c41b4568d5bad1295879e35cfa611bdcf2' [14:13] <cone-138> ffmpeg.git 03Luca Barbato 07master:c82da343e635: pcm: always use codec->id instead of codec_id [14:13] <cone-138> ffmpeg.git 03Michael Niedermayer 07master:73f940e4d94c: Merge commit 'c82da343e635663605bd81c59d872bee3182da73' [14:32] <cone-138> ffmpeg.git 03Luca Barbato 07master:f261e508459e: dca: Error out on missing DSYNC [14:32] <cone-138> ffmpeg.git 03Michael Niedermayer 07master:107a56c1ed8c: Merge commit 'f261e508459e28beca59868a878e1519a44bb678' [14:50] <cone-138> ffmpeg.git 03Luca Barbato 07master:3802833bc1f7: dca: Respect the current limits in the downmixing capabilities [14:50] <cone-138> ffmpeg.git 03Michael Niedermayer 07master:cce254373c41: Merge commit '3802833bc1f79775a1547c5e427fed6e92b77e53' [14:58] <cone-138> ffmpeg.git 03Sean McGovern 07master:52ea29867070: fate: use current syntax for the scale filter [14:58] <cone-138> ffmpeg.git 03Michael Niedermayer 07master:bc8240518830: Merge commit '52ea29867070243385220a223d7512ea7d4ae2d2' [15:03] <aballier> michaelni: (re 1.0 maintainers) thanks for the offer but before claiming to be the maintainer id like to have started to do somthing about it, which I have not yet [15:24] <michaelni> aballier, ok [15:26] <cone-138> ffmpeg.git 03Michael Niedermayer 07master:bec509db4364: vf_interlace: better handling of odd video size [15:26] <cone-138> ffmpeg.git 03Michael Niedermayer 07master:86d1a5d80f16: Merge remote-tracking branch 'qatar/master' [16:16] <BBB> I don't think mac users necessarily use homebrew, a regular git checkout + xcode + yasm works [17:20] <mateo`> hello, anyone to review the "lavf/movenc: support iTunes cover art" patch ? [17:28] <b_jonas> I have verified, and it seems that https://trac.ffmpeg.org/ticket/2684 is indeed fixed, fix works for me. Thanks for all the developers. [17:52] <cone-138> ffmpeg.git 03Paul B Mahol 07master:937cfebd72d3: lavfi/aevalsrc: remove hard limit on number of channels [17:53] <cone-138> ffmpeg.git 03Michael Niedermayer 07master:8aea97a49d19: avcodec/jpeglsdec: Implement the 4th (lossy) transform [17:59] <michaelni> mateo`, dont ask a group for help, ask individuals. If you ask "anyone help me iam bleeding to death" people will think "why me, there are other who could help, and i am in a hurry and have to catch that bus" [18:05] <mateo`> michaelni: sure :) [18:07] <mateo`> michaelni: is there an official maintainer of movenc ? [18:07] <cone-138> ffmpeg.git 03Paul B Mahol 07master:b3405b1bdacf: lavfi/trim: use AV_OPT_TYPE_DURATION [18:10] <michaelni> i think no active one, are you interrested ? [18:15] <mateo`> michaelni: right now, i'm not [18:17] <durandal_1707> this channel layout thing nonsense know nothing about >8 channels [18:17] <durandal_1707> saste: how should i resolve this? [18:17] <saste> a channel layout can specify more than 8 channels [18:18] <saste> but i don't know which problem you're tackling [18:18] <durandal_1707> how? [18:19] <durandal_1707> try to create 9 channel audio with aevalsrc [18:19] <wm4> there are no standard layouts for > 8 channels [18:19] <saste> also there is a flag to tell "unknown" layout [18:19] <wm4> but you should be able to create a layout by using a list of speakers [18:20] <saste> see the work on nicholas, for specifying a generic channel count [18:20] <durandal_1707> is it in master? [18:23] <mateo`> michaelni: would you be interested in reviewing my patch then ? [18:24] <saste> durandal_1707, sure [18:28] <michaelni> mateo`, if noone else volunteers then sure (ping me if i forget, i probably will forget) [18:29] <durandal_1707> saste: thing is if one does not specify layout, but just enters 9 expr it get error [18:30] <mateo`> durandal_1707, ubitux: would you review the lavf/movenc: support iTunes metadata patch ? [18:31] <durandal_1707> mateo`: only if you send me gift [18:34] <durandal_1707> mateo`: you mean cover art? [18:37] <mateo`> durandal_1707: yeay :) [18:37] <mateo`> durandal_1707: another "evil" patch ^^ [18:46] <durandal_1707> saste: doing something lavfi related? [18:50] <durandal_1707> if i remove that error check. than swr complains with silly error [18:50] <durandal_1707> [SWR @ 0x2a04e000] Input channel count and layout are unset [18:50] <durandal_1707> which is lie [18:50] <durandal_1707> as channel count is set [18:50] <durandal_1707> and layout if not set should not be problem [18:50] <durandal_1707> libswresample is just useless [18:51] <durandal_1707> perhaps devs only have 8 fingers [19:07] <durandal11707> huh, why ticket get owned by michael? I never set it. [19:13] <durandal11707> this pullup filter is one big mess [19:17] <wm4> vf_pullup? [19:20] <durandal11707> yes [19:24] <saste> durandal11707, same limitation in libavresample? [19:25] <durandal11707> don't know [19:35] <durandal11707> this perspective filter is very interesting, gonna port it in 5 minutes [19:38] <durandal11707> michaelni: ok to move it to LGPL? [20:19] <michaelni> mateo`, do you want to review "0711 22:02 Matthew Heaney (6.4K) [20:22] <durandal11707> ok, silence means it will stay GPL [20:32] <wm4> so, how do I actually know whether av_probe_input_format3 did a match by file extension? [20:33] <wm4> is there any guarantee that the exact value of AVPROBE_SCORE_EXTENSION is used in this case, and never used for content based matching? [20:33] <michaelni> wm4 isnt guranteed [20:34] <wm4> so better do it manually by looping over all formats and do the matching yourself? [20:34] <michaelni> what are you trying to do ? [20:34] <wm4> doing matching by file extension as last fallback [20:35] <michaelni> the probe code should already favor other means than file extension when possible [20:36] <wm4> the probe function knows only the libavformat formats though [20:36] <wm4> I might have my own code for certain things [20:37] <michaelni> you could simply not tell the probe code the extension then it cant use it [20:38] <wm4> that might work [20:38] <michaelni> but that might end up with you doing probing twice first without extension then with [20:38] <wm4> I first looked at the function av_guess_format() which sounds ideal, because it can handle mime types too [20:39] <wm4> but it's for output formats only [20:39] <wm4> (detection by mime types is missing from the probing API btw.) [20:39] <wm4> well, probably not important at all for most people [20:39] <michaelni> or just do your detection thing first and only when it fails use lavf probe [20:40] <wm4> well, lavf content based probing should always be preferred [20:40] <michaelni> or you could just "integrate" your detection by also picking scores and comparng againt the score fro lavf [20:46] <michaelni> durandal11707, you broke trim [20:46] <michaelni> ./ffmpeg -i matrixbench_mpeg2.mpg -ss 5 -t 1 out.mp4 [20:48] <durandal11707> how could i broke it? [20:48] <michaelni> before it was 1 second now its 6 [20:48] <michaelni> out.mp4 that is [20:49] <durandal11707> but why ffmpeg use trim? [20:49] <durandal11707> trim stuff from fate did not complain at all [20:50] <michaelni> ffmpeg uses trim because of: a83c0da539fb07260310bc3b34056239d2b138b2 [20:53] <durandal11707> if thats true fate would all be red [20:55] <durandal11707> so that is not covered at all ... [20:55] <durandal11707> thing is it code that insert trim filter need updating.... [20:55] <durandal11707> it still sets double instead of ints [20:56] <durandal11707> michaelni: could you change it and see if it fixes it? [20:56] <michaelni> it would be nice if ffmpeg didnt need to be chnaged because it could break other applications [20:57] <michaelni> too [20:57] <durandal11707> this thang is introduced recenly [20:57] <durandal11707> and ffmpeg does not change [20:58] <durandal11707> actually the double thing that changed in 2.0 is one that is "problematic" [21:01] <durandal11707> i fixed it locally [21:02] <michaelni> if you change ffmpeg to match trim it will break --enable-incompatible-fork-abi [21:02] <durandal11707> fuck this [21:04] <wm4> is anyone actually using this fork-abi thing [21:06] <michaelni> if noone uses it we could drop it but still filters arguments should be compatible where possible [21:07] Action: michaelni unhappy that durandal left [21:07] <michaelni> nothing particularely hard to solve here [21:07] <michaelni> just a bit entangled [21:12] <llogan> who wants some ffmpeg stickers? email me. if you're in europe then thilo will send them to you (but i need to send him a bunch of stickers first). [21:12] <llogan> lou at lrcd.com [21:13] <llogan> michaelni: ^ that should make up for the unhappiness [21:24] <michaelni> llogan, yes i want some stickers [21:25] <llogan> i'll try to send them to thilo today. [21:28] <cone-492> ffmpeg.git 03Carl Eugen Hoyos 07master:6476967f21dc: Fix compilation with --disable-everything --enable-demuxer=aac. [21:28] <cone-492> ffmpeg.git 03Michael Niedermayer 07master:9219ec93b145: avfilter/trim: add compatibility layer to not break ABI used by ffmpeg [21:28] <cone-492> ffmpeg.git 03Andrey Utkin 07master:1e85b5e077e7: ff_network_wait_fd_timeout(): check for interrupt before operation [21:28] <cone-492> ffmpeg.git 03Andrey Utkin 07master:34fd21120d86: retry_transfer_wrapper(): check for interrupt before operation [21:54] <cone-492> ffmpeg.git 03Paul B Mahol 07master:7f4a1fddd39e: lavf: add missing build dependency for tta demuxer [21:55] <mateo`> michaelni: i might be able to review it tomorow [21:55] <michaelni> mateo`, ok thanks [23:52] <durandal_1707> michaelni: your 9-16 jpeg2000 patch fixes rgb48 pixfmt or not? [23:54] <michaelni> durandal_1707, i still have to test [23:56] <cone-492> ffmpeg.git 03Nicolas Bertrand 07master:dd1382ac9534: jpeg2000: Initialize only once mqc arrays [23:57] <durandal_1707> michaelni: sending untested patches? [00:00] --- Sat Jul 13 2013

1 0

ffmpeg.log.20130712
by burek 13 Jul '13

13 Jul '13

[00:40] <juanmabc> hi, since recently someone work on adpcm ima radical, i wonder if some ffmpeg developer would like, or get interested, on work on adpcm ima amv encoder, else i'll keep waiting for another year until someone (if ever) does (before devices die would be better ;)), there is some info, first even a ticket time ago (known for me from ages): http://trac.ffmpeg.org/ticket/747 , second a working code to base: http://code.google.com/p/amv-codec- [00:40] <juanmabc> tools/ and third a working, in trunk, adpcm ima amv decoder to test the results (cause not everyone got the gadget ;)), so the workflow is far from scratch, and far from hard to test. Discalimer: this is just a push, ;) [00:41] <juanmabc> to summarize, adpcm ima amv enc, now [00:41] <juanmabc> ;P [04:31] <Lobs> is there a way i can force an output resolution in a mov container? I.E the encoded resolution differs from the playback resolution and isnt directly linked to the aspect ratio. [05:26] <calcifer_> is an updated homebrew formula for ffmpeg 2.0 in the works? [05:29] <llogan> calcifer_: i don't think anyone here is responsible for that [05:29] <llogan> i mean i [05:29] <llogan> 'm not sure who updates that (damned wrist brace) [05:33] <calcifer_> looking at the history, it's been updated by a few different people. hmm. :/ [09:51] -:#ffmpeg- [freenode-info] why register and identify? your IRC nick is how people know you. http://freenode.net/faq.shtml#nicksetup [10:30] <t4nk488> hi, how to mux h264 video and pcm_s16le by ffserver [10:30] <t4nk488> I try to output a rtp stream that include both video and audio. the config I use: <Stream test1-rtsp.mpg> Feed feed1.ffm Format rtp AVOptionVideo flags +global_header VideoSize 1280x720 VideoFrameRate 30 VideoCodec libx264 AVOptionAudio flags +global_header AudioCodec pcm_s16le AudioChannels 2 AudioSampleRate 48000 </Stream> [10:31] <t4nk488> I can NOT play by ffplay and VLC. [10:31] <t4nk488> Does anyone have idea how to mux h264 video and pcm_s16le audio? [10:39] <Mavrik> hmm, I think the raw audio is the problem [10:39] <Mavrik> why aren't you encoding it? [11:06] <sweb> how can i strip all metatags from file with ffmpeg and then add some tags ... [11:07] <sweb> ffmpeg -i 1.mp3 -map_metadata -1 2.mp3 will strip all tags. but how can i add some tags after then [11:22] <t4nk488> hi [11:22] <t4nk488> what is the error:ffserver: libavformat/asfenc.c: 642: flush_packet: Assertion `asf->packet_timestamp_end >= asf->packet_tim estamp_start' failed. [11:22] <t4nk488> does anyon know the cause of the ffserver fail [11:46] <t4nk488> hi, any ideas about how to mux h264 video + pcm audio? [11:47] <t4nk488> Can anyone help? [11:50] <Mavrik> mux to what. [11:53] <t4nk488> mux to network stream [11:53] <t4nk488> I try to mux asf stream, but VLC can not play [11:54] <t4nk488> and I mux tp rtp stream , VLC can not play ,too [11:54] <t4nk488> both VLC and ffplay can not play the stream [11:54] <Mavrik> streaming raw anything is a problem [11:54] <Mavrik> encode it to AAC and it'll work. [11:56] <t4nk488> I also try to encode to AAC , but when I try to mux to rtp stream [11:56] <t4nk488> VLC can not play,too [11:57] <t4nk488> I got the error message:Application provided invalid, non monotonically increasing dts to muxer in stream [11:57] <t4nk488> do you have idea about :Application provided invalid, non monotonically increasing dts to muxer in stream [11:57] <t4nk488> hence VLC can not play asf stream [11:59] <Mavrik> well if you have messed up timestamps [11:59] <Mavrik> then no player will be able to play it no matter ne mux [12:00] <t4nk488> do you have idea about tiemstamps issue? [12:03] <t4nk488> do you have any idea: which encode mthod is better for audio other than aac? [14:30] <FunnyLookinHat> Hey guys - So I've got some scripts that generate a bunch of thumbnail preview images from a video using something like this: ffmpeg -i input.mp4 -ss 45 -f image2 -vframes 1 out.png ... but I've noticed it takes a significantly long time as the timestamp gets further out ( several minutes sometimes to get something just 3 or 4 minutes into the video ) [14:30] <FunnyLookinHat> Is there any way to do this that's faster using ffmpeg? [14:33] <Mavrik> move -ss before -i. [14:40] <FunnyLookinHat> Mavrik, That will seek faster ? [14:41] <JEEB> yes [14:41] <JEEB> it will, though, only be able to seek to points that it can start decoding from straight away [14:42] <JEEB> seek on the container level (before decoding) vs seek on the stream level (with decoding) [14:42] <JEEB> latter is more accurate, but will decode everything, first is faster [14:43] <Mavrik> also on some files it might not work [14:49] <FunnyLookinHat> Interesting... ok I'll do a bunch of testing. Much appreciated, thanks guys. [14:56] <i0scribe> hello everyone. [14:56] <i0scribe> I just followed the UbuntuCompilationGuide at the Wiki and wasn't sure if it "took" successfully. [14:58] <i0scribe> At the end of the guide, is directed me to give the following command "ffmpeg 2>&1 | head -n1" and my output is "ffmpeg version 0.8.6-4:0.8.6-0ubuntu0.12.04.1, Copyright (c) 2000-2013 the Libav developers." however the guide (which I believe is current) says "If you do not see FFmpeg developers in your output then something went wrong and you're probably using the fake "ffmpeg" from the repository.". [15:00] <i0scribe> So, my question is, what is the difference between "Libav developers" and "FFmpeg developers" when it comes to knowing if my compilation was completed successfully. [15:03] <JEEB> libav and ffmpeg are separate projects [15:03] <JEEB> debian-based distros use binaries built from the libav project's source code by default [15:03] <JEEB> so if you see "ffmpeg developers" there, it means you are indeed using a binary that is built using the ffmpeg project's source code [15:19] <i0scribe> Thanks JEEB ... So I would see "FFmpeg developers" if I was using Redhat (or some other distro) and "Libav developers" is just the expected result on all Ubuntu and other deb based distros ... ok, thanks. [15:25] <i0scribe> Since I've confirmed that my ffmpeg is current and unrestricted, can anyone offer me any tips as to why my AVI to MPG conversions aren't going well? I've tried three good working AVIs coming from two different sources, and each one gives me sound only. Well, that is except for one "still image" snapshot. I think the "still image" is the first image from the AVI file. [15:26] <i0scribe> I'm using WinFF which does a fine job of converting FLV to MPG but the AVI to MPG is not working. [15:44] <i0scribe> I tried "ffmpeg -i file.avi -vcodec mpeg4 -f mpeg output.mp4" and that is working. Also tried "ffmpeg -i file.avi -vcodec h263 -acodec aac -f mpg output.mpg" ... Am looking for some other options on the net hoping to get this AVI converted successfully to an MPG. [15:47] <Mavrik> uh [15:47] <Mavrik> ok. [15:47] <Mavrik> that I can see why it's going wrong >D [15:47] <Mavrik> i0scribe, what are you trying to achieve? why (the horror) MPG? [15:49] <i0scribe> Maverick|MSG: Trying to convert it to MPG because it's the only thing supported by my DVD player (USB port). [15:49] <Maverick|MSG> Mavrik ^^ [15:49] <Mavrik> ;) [15:49] <Mavrik> i0scribe, well, MPG is just the container [15:50] <Mavrik> first you MUST check which video and audio formats your DVD player supports [15:50] <Mavrik> and then we can talk [15:50] <Mavrik> stuffing random stuff into MPG just won't work ;) [15:50] <i0scribe> I also tried "ffmpeg.exe -i input.avi output.mpg" just now and that's not working. [15:50] <ilove11ven> i0scribe: you may want to use something like dvd author, which use FFmpeg internally to do it. [15:51] <i0scribe> Well, for now I'm just trying to get the MPG file to play on my computer and I can't even get that far. [15:51] <i0scribe> Trying to convert a working AVI file to a working MPG file. [15:51] <ilove11ven> i0scribe: Mavrik is correct. To rip DVD, not only did you have to use -f dvd, but you should also take into account of codec problem and many other complicated stuff. [15:52] <ilove11ven> MPEG-2 video and MPEG audio could be used with '-f dvd' [15:54] <Mavrik> i0scribe, which DVD player is it? [15:54] <i0scribe> ilove11ven: I think I confused the issue by bringing up my DVD player ... My DVD player has a USB port where I can play normal MPG files via a USB stick. ... So nothing special is required on that side. ... I can take 'any' working MPG file that I pull off my computer and play it on my DVD player. [15:55] <i0scribe> Mavrik: I've used both "Mobile Media Converter" and "WinFF" to successfully convert FLV files to MPG, but when that didn't work I decided to drop down to the CLI and see what was going on with FFmpeg. [15:55] <Mavrik> dude. [15:55] <Mavrik> mpg is just a container [15:55] <Mavrik> you need to put a correct video and audio format in there. [15:55] <Mavrik> that's why we're asking, WHICH DVD player do you have [15:55] <Mavrik> so we can help you [15:56] <Mavrik> running random crap won't just make it work except by blind luck [15:59] <ilove11ven> i0scribe: try this: ffmpeg -i .avi -f dvd -vcodec mpeg2video -acodec mp2 output.mpg [16:02] <ilove11ven> or take any working mpg file, use FFtools to analyze its container/video codec/audio codec. Then use ffmpeg to do the conversion. [16:03] <i0scribe> Maverick|MSG: It's a Xenon DVD player. [16:04] <i0scribe> ilove11ven: thanks, I'll try that. [16:06] <i0scribe> ilove11ven: oh wait, what part of that string is the input.avi ? [16:06] <i0scribe> ilove11ven: I'm guessing to ".avi" part of it. [16:06] <ilove11ven> your avi file [16:07] <Mista_D> Anyway to force "-vf scale=-1:480" to use nearset disible by 16? [16:08] <Mavrik> Mista_D, I think you can use expressions [16:10] <Mavrik> so you could do something like [16:10] <Mavrik> scale=round((oh * a) / 16) *16:480 [16:10] <Mavrik> or something like taht [16:10] <Mavrik> :) [16:10] <i0scribe> ilove11ven: same problem as before. Audio only as the output. The 28 MB AVI file always comes out as a half meg MPG file that is audio only. [16:11] <i0scribe> ilove11ven: Don't know if this is useful for you, but every time I do this one of the error lines says "[buffer @ 0x87a99c0] Changing frame properties on the fly is not supported.". [16:11] <Mavrik> Mista_D, I doubt there's a simpler ffmpeg-only way [16:12] <Mavrik> sorry, it's floor() not round() [16:12] <Mista_D> Mavrik: Doesn't work - "Unknown function", using 5 hr old master. [16:12] <Mavrik> https://ffmpeg.org/ffmpeg-utils.html#Expression-Evaluation [16:12] <Mavrik> writing out of my head ;) [16:13] <Mavrik> anyway, bbl [16:13] <Mista_D> Mavrik: Will test a few, thanks. [16:15] <ilove11v1> i0scribe: would you mind uploading the avi file? [16:15] <i0scribe> no prob. [16:15] <i0scribe> ilove11v1: no prob. [16:18] <ilove11v1> ok [16:20] <i0scribe> ilove11v1: just started the file transfer. [16:21] <i0scribe> ilove11v1: it's says it is waiting for you. [16:54] <Mista_D> Why would this scale folmula be rejected: -vf 'scale=round((iw*480/ih)/16) *16:480'? seems ok to me. [17:00] <saste> Mista_D, what is "round"? [17:01] <Mista_D> @saste: Tried celi too. [17:01] <Mista_D> Sorry ceil [17:03] <Mista_D> @saste: stand corrected, -vf 'scale=ceil((iw*480/ih)/16) *16:480' works. Had a differnt error, didn't noticed at first. [17:04] <ilove11v1> i0scribe have upload the original file: http://www.fileconvoy.com/dfl.php?id=g0a0c8b0c8b7f9a9d999327977b1aa086cf5a5… [17:05] <ilove11v1> I have uploaded the transcoded MPEG-PS/MPEG2 video/MP2 audio: http://www.fileconvoy.com/dfl.php?id=ge1604f2b7b8f9e6b999327998151be79b266f… [17:05] <ilove11v1> i0scribe: please check it. time to go to bed. good night [19:29] <teratorn> what aac codec to folks normally recommend these days? [19:30] <Mavrik> fdk_aac is probably still the best [19:30] <Mavrik> otherwise internal is good enough [19:30] <teratorn> cool. now to figure out how to use it from xuggle :-( [19:30] <Fjorgynn_> Mavrik: I heard different [19:30] <Fjorgynn_> that it's not good. The default [19:30] <teratorn> or that. we're getting sporadic SIGFPE exceptions from the lovely libvo-aacenc that xuggle uses by default [19:30] <Mavrik> Fjorgynn_, none of the others are any better. [19:31] <Mavrik> teratorn, yeah, voaacenc is probably the worst of the bunch [19:35] <JEEB> <teratorn> what aac codec to folks normally recommend these days? <- do you mean encoder or decoder wise? [19:36] <teratorn> encoder [19:36] <JEEB> ok, then fdk-aac [19:36] <JEEB> do note that you will have to compile fdk-aac yourself, it can't be distribute [19:36] <JEEB> *distributed [19:36] <JEEB> as binary [19:36] <teratorn> ah [19:36] <Fjorgynn_> I have the binary for windows :) [19:37] <JEEB> yes, but according to the license you cannot distribute it :) [19:38] <JEEB> VideoLAN tried to find some ways around it [19:38] <JEEB> but in the end just gave up [19:39] <teratorn> what is the license? according to the .c file it says "is part of FFMPeg... FFMpeg is distributed under LGPL blah blah" [19:41] <JEEB> teratorn, that's the license for the wrapper in ffmpeg [19:41] <JEEB> teratorn, https://github.com/mstorsjo/fdk-aac/blob/master/NOTICE [19:41] <JEEB> this is the license for fdk-aac [19:41] <teratorn> yeah I just realized 411 lines can't be enough for the full codec [19:41] <teratorn> cool, thanks [19:53] <eggyknap> If ffprobe tells me "Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 48000 Hz, 4.0, fltp, 51 kb/s", the "4.0" means I have four channels of audio in one stream, right? [19:54] <JEEB> I would think so, yes [19:54] <eggyknap> (follow-up questions: how can I extract one of those channels into a separate mp3, and second, I created that file with ffmpeg and would really like one stereo and two mono streams... how can I do that?) [19:56] <JEEB> there are IIRC options to use certain channels and such [19:56] <JEEB> I don't remember them all myself, but looking at the documentation for the ffmpeg command line tool on ffmpeg.org should be useful [19:56] <eggyknap> Yeah, I've been trying to find examples, because I haven't been able to make those options work (disclaimer: the ffmpeg version I've got on my ubuntu box seems hopelessly disabled -- old version, with few enabled filters) [19:57] <eggyknap> IOW, I've found the documentation, and have read about the options, but am getting something wrong. [19:57] <JEEB> with ubuntu you will most probably want to use the avconv binary as you're using code from the libav project [19:57] <JEEB> in libav avconv is the more up-to-date binary [19:58] <JEEB> while in ffmpeg (project), ffmpeg is [19:58] <JEEB> (also #libav is the IRC channel for the libav project) [19:58] <eggyknap> I had the impression (based only on manpages and my inept blundering around) that on my box they were the same binary, essentially. I'll look more closely into it. [19:58] <eggyknap> In any case, my avconv doesn't support -filter_complex any better than my ffmpeg does [19:59] <JEEB> I know that at least until version 0.8 of libav there was still a separate ffmpeg binary, I think? [19:59] <JEEB> after the 0.8 release it got finally removed [19:59] <JEEB> and the ffmpeg binary, while it was there, was pretty much untouched [19:59] <JEEB> also if you're looking for the avconv documentation, I recommend you look at libav.org's documentation :) [20:01] <eggyknap> So, say I have a camera and multiple microphones, and I say something like "ffmpeg <options to capture video> <options to capture mic 1> <options to capture mic 2> outputfile.mp4"... [20:01] <eggyknap> ...I'd like to end up with multiple audio tracks, rather than multiple channels on one track. Suggestions? [20:02] <Mavrik> just map them accordingly to different tracks and set encoding options to each [20:03] <eggyknap> Mavrik: while that sentence makes sense, I'm too much of a n00b to see how to do that :) [20:04] <Mavrik> I suggest reading the documentation then [20:04] <Mavrik> you can provide multiple inputs (one -i for each input) and then use -map parameter to map inputs to a single output file with different streams [20:05] <eggyknap> Clearly I need to grok -map better than I do. I'll reread its docs. Thx [20:05] <Mavrik> you provide encoding parameter with each stream as well [20:05] <Mavrik> and you give only one output [20:05] <Mavrik> in the docs it's explained how to use the -map parameter and how to number encoding parameters (codec:a, codec:v etc.) for each output stream :) [20:06] <Mavrik> eggyknap, also by default each input will be a separate stream [20:06] <Mavrik> so if I read the docs correctly you shouldn't need a -map call ata all [20:07] <eggyknap> Hrm. I didn't include any options to my capture command and ended up with two streams, one in 4.0 audio [20:08] <Mavrik> does your audio input record 4 channels? :) [20:08] <Mavrik> eggyknap, maybe you should pass parameter for only 1 output channel for that stream then :) [20:08] <Mavrik> especially if your microphone records mono [20:08] <Mavrik> eggyknap, it's worth checking what ffmpeg thinks the INPUT is [20:10] <eggyknap> There are four separate mono mics, if that's what you mean. How do I get ffmpeg to cough up more data on what it thinks its INPUT is? [20:10] <Fjorgynn> soon it's the 27th... [20:11] <Mavrik> eggyknap, um, it always outputs what it thinks the input is [20:13] <eggyknap> Oh, I just need to pay more attention to the STDOUT stuff. Gotcha :) [21:09] <llogan> eggyknap: try -map_channel (or pan audio filter) to monkey with the 4 audio channels [21:10] <eggyknap> Ooo... that could be helpful [21:10] <eggyknap> Right now I'm recording from four mics simultaneously, and the result I get is a file with one mono output stream. :( [21:13] <eggyknap> e.g. http://pastie.org/8135391 [21:17] <eggyknap> I can add "-filter_complex '[0:0] [1:0] [2:0] [3:0] amerge=inputs=4'" and get a single audio stream in 4.0 layout... but that's still not four separate streams. [21:17] Action: eggyknap keeps reading [21:18] <eggyknap> OIC... there's where I need a -map command. [21:18] <eggyknap> ...perhaps [21:19] <eggyknap> w00t! That works. Thanks, all. [22:12] <nuizzy> does ffmpeg support h.265? [22:14] <Sean__> I want to analyze a source video file and determine 1. At What frames each "scene change" occurs. 2. Given the information from #1 and a min and max key frame interval , what frames will need to be I-frames in a h264 encode.. [22:14] <Sean__> I know I can do this by doing a "ultrafast" pass=1 encode with x264 and analyzing the .stats file afterwards. However , This method seems a bit heavy for the simple information I'm trying to obtain [00:00] --- Sat Jul 13 2013

1 0

ffmpeg-devel.log.20130711
by burek 12 Jul '13

12 Jul '13

[00:04] <Daemon404> figures the first avreample failure is on an OS nobody uses or will ever use [00:04] <Daemon404> :V [00:04] <Daemon404> wait a pain to fix [00:04] <Daemon404> what* [00:16] <durandal_1707> what OS? plan9? [00:16] <Daemon404> haiku [00:18] <durandal_1707> that is really nice one [00:20] <michaelni> Daemon404, wait till the next run of haiku before you investigate [00:20] <Daemon404> ? [00:20] <michaelni> i think its missing a file [00:20] <Daemon404> oh [00:25] <Guest5607> damned phone calls at the wrong time... [00:50] <cone-59> ffmpeg.git 03Stefano Sabatini 07master:e7a6962e7650: doc/developer: update sections about external development [00:50] <cone-59> ffmpeg.git 03Stefano Sabatini 07master:b3ac8a2a606c: doc/formats: add section describing stream specifiers [00:50] <cone-59> ffmpeg.git 03Stefano Sabatini 07master:19ea08a11a8e: lavf/segment: add initial_offset option [00:50] <cone-59> ffmpeg.git 03Stefano Sabatini 07master:459dc81a64a3: lavf/segment: reindent after last commit [00:50] <cone-59> ffmpeg.git 03Stefano Sabatini 07master:71a2e5245ca5: lavf/segment: use AV_OPT_TIME_DURATION for time_delta [01:06] <durandal_1707> TFTR? [01:10] <saste> durandal_1707, Too Fast To Read [01:11] <saste> durandal_1707, "time duration specification" or "duration specification" works for me [01:12] <saste> users will have to look for the specification in man ffmpeg-utils [01:13] <durandal_1707> so just add: specify time of ... see time duration specification. ? [01:13] <saste> or ... the argument is a duration specification. [01:13] <saste> bonus if you provide a reference to the spec [01:16] <durandal_1707> i fill just link time to duration specification thing... [01:20] <durandal_1707> its @anchor{time duration syntax} [01:20] <durandal_1707> but its in another file [01:25] <durandal_1707> michaelni: hmm, but atrim filters really trim frame samples [01:27] <durandal_1707> saste: muxers mentions time duration specification, but there is no such thing [01:28] <durandal_1707> but there is 'duration syntax' [02:40] <cone-59> ffmpeg.git 03Michael Niedermayer 07master:94e86ae15a43: mjpegdec: initialize source variables before gbr remap [02:40] <cone-59> ffmpeg.git 03Michael Niedermayer 07master:15cee5e5628a: mjpegdec: Fix used quant index for gbr [11:21] <cone-135> ffmpeg.git 03Luca Barbato 07master:3865ba7b21ae: iff: K&R formatting cosmetics [11:21] <cone-135> ffmpeg.git 03Michael Niedermayer 07master:56090b5b8043: Merge commit '3865ba7b21aef5d60183719e238361ec8797ab5a' [11:33] <cone-135> ffmpeg.git 03Luca Barbato 07master:7d65e960c72f: iff: Do not read over the source buffer [11:33] <cone-135> ffmpeg.git 03Michael Niedermayer 07master:2efc2b53be96: Merge commit '7d65e960c72f36b73ae7fe84f8e427d758e61da9' [11:41] <cone-135> ffmpeg.git 03Luca Barbato 07master:04e9853a210e: wmavoice: return meaningful error codes [11:41] <cone-135> ffmpeg.git 03Michael Niedermayer 07master:5698f6bd7334: Merge commit '04e9853a210e07b1a150aa2b8e9af0aa3601ff21' [11:50] <cone-135> ffmpeg.git 03Luca Barbato 07master:f5c48f5adaea: wmavoice: use init_static_data [11:50] <cone-135> ffmpeg.git 03Michael Niedermayer 07master:45cc62f0a099: Merge commit 'f5c48f5adaead5259660d4eaab2e695d1bac3a08' [11:58] <cone-135> ffmpeg.git 03Luca Barbato 07master:d14a26edb7c4: wmavoice: conceal clearly corrupted blocks [11:58] <cone-135> ffmpeg.git 03Michael Niedermayer 07master:531f0169444c: Merge remote-tracking branch 'qatar/master' [12:13] <cone-135> ffmpeg.git 03Michael Niedermayer 07release/2.0:9711b527394c: avfilter/af_earwax: Fix out of array accesses on odd packets [12:13] <cone-135> ffmpeg.git 03Piotr Bandurski 07release/2.0:1cda4aa1e0e1: avformat/utils: avformat_find_stream_info fix a crash in case of oom [12:13] <cone-135> ffmpeg.git 03Paul B Mahol 07release/2.0:56bf38859b93: lavfi/aconvert: unbreak [12:13] <cone-135> ffmpeg.git 03Piotr Bandurski 07release/2.0:c1c84f0a55f8: avformat/utils: avformat_find_stream_info set value for ret in case of oom [12:13] <cone-135> ffmpeg.git 03Michael Niedermayer 07release/2.0:ea28e7420517: avcodec/qdm2: store bits in an integer instead of float variable [12:13] <cone-135> ffmpeg.git 03Michael Niedermayer 07release/2.0:37268dcc863e: avcodec/qdm2: initialize sign_bits [12:13] <cone-135> ffmpeg.git 03Luca Barbato 07release/2.0:a7315116dd4a: wmavoice: conceal clearly corrupted blocks [12:13] <cone-135> ffmpeg.git 03Carl Eugen Hoyos 07release/2.0:fc3dec8b626f: lavf/utils.c: Avoid a null pointer dereference on oom after duration_error allocation. (cherry picked from commit c9eb5c9751c88caaed62af5ffe908fe545022e7e) [12:13] <cone-135> ffmpeg.git 03Carl Eugen Hoyos 07release/2.0:fd2cf9c45d3b: Suggest recompilation with openssl or gnutls if the https protocol is not found. [12:13] <cone-135> ffmpeg.git 03Michael Niedermayer 07release/2.0:6127f792f900: mjpegdec: initialize source variables before gbr remap [12:13] <cone-135> ffmpeg.git 03Michael Niedermayer 07release/2.0:7740e36a89cd: mjpegdec: Fix used quant index for gbr [12:15] <ubitux> is Luca re-doing every j00ru fixes? [15:14] <durandal_1707> can @ref{} point out of doxy file? [15:41] <durandal_1707> why people thing place for bug reporting is support forum? [15:42] <durandal_1707> *think [15:42] <durandal_1707> ubitux: did you read this http://forum.doom9.org/showthread.php?t=168185 ? [15:47] <durandal_1707> http://forum.doom9.org/showthread.php?t=143624&page=17 [15:48] <durandal_1707> michaelni: yuv444 -> rgb issues [16:07] <wm4> how do you write picture attachments to a file, e.g. mp3? -attach Doesn't Work [16:10] <durandal_1707> iirc you do it as normal video stream [16:10] <durandal_1707> michaelni: did that swf issue got finally fixed? [16:16] <michaelni> durandal_1707, why do you never ask a clear question and why do you always ask me ? [16:17] <durandal_1707> michaelni: there was only one security issue with swfdec for over 6 months [16:17] <durandal_1707> its about not really fixed security issues [16:18] <durandal_1707> michaelni: is there some more experienced dev for asking about swscale conversion issues? [16:22] <michaelni> durandal_1707, we fix all security issues for which sufficient information is provided to reproduce them or that contain a detailed description of the issue [16:23] <durandal_1707> we? [16:24] <michaelni> yes, you dont want to fix security issues ? [16:27] <michaelni> about swf, iam not aware of any open security issues [16:37] <cone-135> ffmpeg.git 03Michael Niedermayer 07master:7afa68d37df9: mjpegdec: parse app-colr [16:37] <cone-135> ffmpeg.git 03Michael Niedermayer 07master:96747e6cbe75: mjpegdec: parse app-xfrm [16:37] <cone-135> ffmpeg.git 03Michael Niedermayer 07master:8024b4880ccc: avcodec/jpeglsdec: support xfrm 1-3 [16:37] <cone-135> ffmpeg.git 03Michael Niedermayer 07master:0ef40446a864: mjpegdec: print length too in APPx debug code [18:27] <Compn> michaelni : i think durandal is talking about the securina / vlc spat (which is about the swf parser in libavformat) [18:27] <Compn> which hit slashdot and others [18:30] <Compn> michaelni : http://seclists.org/fulldisclosure/2012/Dec/96 [18:30] <Compn> :P [18:33] <Compn> michaelni : heres the .swf file that crashed vlc . http://www39.zippyshare.com/v/91522221/file.html [18:34] <Compn> from 2012 so it might be fixed since then [18:34] <Compn> dont know if anyone tested ffmpeg [18:34] <Compn> j-b : did you test swf with ffmpeg? (i know you want to talk about swf and securnia more :D) [18:34] <j-b> Compn: fixed since a long time, you know... [18:35] <ubitux> Compn: j-b linked to the fix in ffmpeg upstream in his blog post [18:35] <j-b> http://git.videolan.org/?p=vlc/vlc-2.0.git;a=blob;f=contrib/src/ffmpeg/swfd… [18:56] <Compn> j-b : oh ok :) [18:56] <Compn> i saw that you linked to vlc git , i wasnt paying attention if it were a fix for ffmpeg or vlc's ffmpeg .... [18:56] <Compn> and isnt vlc using libav ? so confused [19:07] <j-b> Compn: vlc compiles with both [19:10] <Compn> very professional answer that didnt quite answer the question ;P [19:10] <Compn> my question wasnt precise either of course [19:11] <j-b> Compn: it depends on the maintainer [19:11] <j-b> Compn: hence I cannot answer the question [19:11] <wm4> what are you using for development? [19:12] <j-b> I use FFmpeg, because my distro is Arch [19:22] <durandal_1707> VideoLAN is paying trips to Paris? [19:30] <Compn> for Videolan dev days, i think they have invited ffmpeg devels in the past [19:30] <Compn> dunno if they currently have funds for it [19:30] <Compn> best ask j-b or in #videolan [19:40] <j-b> durandal_1707: of course. [21:53] <cone-135> ffmpeg.git 03Dan Flett 07master:43d36599fe33: lavd/fbdev.c: Support RGB565 input colour space. [22:23] <durandal_1707> should i write adelay filter? [22:37] <Compn> durandal_1707 : you mean for offsetting audio ? like mplayer -delay or -af delay ? [22:37] <saste> durandal_1707, time to move on with the framework? [22:38] <saste> what's going on on the l. side with regards to API/framwork? [22:38] <saste> anybody knows? So we don't need to redo from scratch after they NIH [22:39] <saste> but probably the best thing to do at this point is to work on bindings [22:40] <durandal_1707> saste: framework what? [22:41] <saste> durandal_1707, reconfiguration for example [22:41] <saste> extend input movie, add output movie [22:41] <durandal_1707> you just call config_props for each link [22:41] <durandal_1707> i don't touch movie filters as i consider them evil [22:42] <saste> durandal_1707, why? [22:42] <saste> why people hate movie [22:42] <saste> ? [22:42] <durandal_1707> becaue it does not play movie you want [22:42] <saste> especially since the alternative (-filter_complex) is a huge application-level hack [22:43] <saste> durandal_1707, what do you mean?? [22:44] <durandal_1707> it doesn't play well [22:45] <durandal_1707> it doesn't work with http/ etc... [22:45] <saste> can't you specify http://... [22:45] <saste> ? [22:46] <durandal_1707> iirc it never worked for me [22:46] <saste> it's the usual pain to escape, but what does it prevent to work as expected? [22:46] <saste> "it doesn't play well" => too vague [22:46] <durandal_1707> it doesn't expand ~ [22:47] <durandal_1707> i need to give absolute path [22:51] <durandal_1707> also isn't (a)movie cloning ffmpeg funcionality and is missing some hacks? [22:51] <saste> durandal_1707, the plan is having movie substitute ffmpeg [22:52] <saste> so you can implement ffmpeg functionality in your program, with (relatively) few lines of code [22:52] <durandal_1707> interesting idea, dunno if its doable in reasonalbe time frame [22:53] <durandal_1707> there are other filters that ffmpeg could be using, but does not [22:54] <durandal_1707> and instead do it on its own, like setfps [22:54] <durandal_1707> and trim, which is used by avconv, but not ffmpeg [22:55] <durandal_1707> i think with my patch, ffmpeg could use it though [22:55] <durandal_1707> michaelni: ^ [22:55] <saste> not everything can be done with filters [22:55] <saste> for example for manipulating packets timestamps you need essentially - a packet filter [22:56] <durandal_1707> yes, so why not write one? [22:56] <saste> because you need a whole packet filtering framework [22:56] <saste> also maybe we should merge it somehow to lavfi [22:57] <durandal_1707> merge AVPacket with AVFrame [22:57] <saste> for example we could add data filtering to lavfi [22:57] <saste> but this requires careful design [22:57] <saste> and I have no more energy for it [22:57] <durandal_1707> what? you are retiring? [22:58] <saste> http://sophia.estec.esa.int/socis2013/?q=node/13 [22:58] <saste> still waiting (after almost two weeks) [22:58] <durandal_1707> others got approved or? [22:59] <saste> no, only thinks we know since a week is that the list will be published "soon" [22:59] <saste> it's a bit frustrating [23:00] <durandal_1707> well, ESA Summer of Code in Space becomes ESA Autumn of Code in Space [23:05] <saste> durandal_1707, because in space, no one can hear when you wait [23:29] <nevcairiel> you just have to wait long enough, then summer is on the southern hemisphere [23:51] <Daemon404> g 42 [23:54] <durandal_1707> so should i do adelay filter, the only way currently one can do delay, is by splitting channels, and joining them again after processing [00:00] --- Fri Jul 12 2013

1 0

ffmpeg.log.20130711
by burek 12 Jul '13

12 Jul '13

[00:07] <sybariten> oh hai [00:08] <sybariten> i'm on ubuntu and installed this package: chromium-codecs-ffmpeg-extra , because my chromium wouldnt play Soundcloud music. Now it plays, but its super high pitched.... any ideas? [00:20] <sybariten> that is a slightly more flattering nick, where i come from [02:46] <acovrig> When I use -filter:v 'setpts=0.5*PTS' I loose video, the stream doesn't end up in the output, why? [03:24] <LordThumper> Hi, can encoding problems (graphical artifacts) be fixed using ffmpeg? [03:42] <LordThumper> Re-encoding did not fix the issue, i think I need another source [03:49] <Kuukunen-> LordThumper: if the artifacts are present in the source, very likely you can't fix them with anything [03:50] <klaxa> edit the frames by hand, lol [03:54] <LordThumper> The artifact is the object that moved between one frame and another [03:54] <LordThumper> I have no idea how h264 is encoded [03:54] <Kuukunen-> wuh? :P [03:55] <klaxa> i meant extract the frame, edit with an image editor, glue it all back together [03:55] <LordThumper> can that be done? [03:55] <LordThumper> What software should I use? [03:55] <Kuukunen-> sure [03:55] <Kuukunen-> photoshop? :P [03:56] <klaxa> it's more of a troll suggestion, but it is not impossible [03:56] <LordThumper> The problem is not the image itself [03:56] <klaxa> it depends on how well you can reconstruct the image [03:56] <LordThumper> It's like the transition between one frame and another [03:56] <Kuukunen-> LordThumper: take a picture, stick it on imgur or something :P [03:56] <LordThumper> Ok sure [03:56] <Kuukunen-> but is it present on the source? [03:57] <klaxa> his source is b0rked already [04:02] <LordThumper> http://i.imgur.com/V23dD2T.jpg [04:02] <Kuukunen-> right, looks like there's some corruption on the bit level or something [04:02] <LordThumper> The objects that moved between one frame and another appear glitched [04:03] <LordThumper> I tried reencoding, but the glitches remained [04:03] <klaxa> LordThumper: check your torrent-file [04:03] <Kuukunen-> LordThumper: well, if they're in the source, then reencoding will do nothing [04:03] <klaxa> let your torrent software verify the local data, it should then re-download broken chunks [04:03] <LordThumper> Yes it is valid [04:03] <klaxa> torrent link? [04:03] <LordThumper> I will try to get it from another source [04:03] <Kuukunen-> then it's messed up when it was made [04:03] <klaxa> and timestamp of that scene [04:04] <klaxa> (not watching attack on titan i guess i'm prepared to be spoilered :P) [04:04] <LordThumper> You recognized it from the image? :) [04:04] <klaxa> well still want torrent + timestamp [04:05] <klaxa> it also has distinctive character design [04:05] <Kuukunen-> klaxa: yea, the faces are all blocky and stuff [04:05] <klaxa> ;) [04:06] <LordThumper> hehe indeed [04:06] <LordThumper> one sec [04:06] <LordThumper> http://thepiratebay.sx/torrent/8575596/Attack_on_Titan_S01E02_That_Day_(192… [04:06] <LordThumper> 7m00sec [04:06] <klaxa> wtf you're downloading anime from the pirate bay? [04:07] <LordThumper> Ye :-/ Friend dloaded them [04:07] <LordThumper> Just finished downloading Horriblesubs sub and it's perfect [04:07] <klaxa> given size, resolution, encoding and matroska streams that's the horriblesubs 1080p release [04:07] <LordThumper> mmm [04:07] <klaxa> nobody else releases TV anime in 1080p with high profile [04:07] <LordThumper> That's strange [04:07] <LordThumper> Maybe its a reencoding? [04:08] <LordThumper> Or corrupt download? [04:08] <klaxa> oh re-encoding might be [04:08] <LordThumper> [HorribleSubs] Shingeki no Kyojin - 02 [1080p] [04:09] <klaxa> hmm audio isn't 128 kbps AAC so it is NOT horriblesubs [04:09] <LordThumper> Is perfect at 7m00 sec [04:09] <klaxa> but seriously, don't download anime from thepiratebay :X [04:09] <klaxa> unless it's an older series you can't find elsewhere [04:09] <LordThumper> Indeed [04:09] <LordThumper> Well thanks alot for the help, and I suggest watching the anime, it's very good so far [04:10] <LordThumper> I think it competes with some of the top anime [04:10] <LordThumper> And it's a nice change from mechs (Code Geass, Evangelion etc etc etc) [04:11] <klaxa> i've heard a lot and see a lot of people that hate it [04:11] <klaxa> haven't seen either of those :P [04:11] <klaxa> haven't watched most of the "you have to have seen this show"-shows [04:11] <LordThumper> Oh, what is your favourite? [04:11] <klaxa> tough question, probably ika musume [04:12] <klaxa> the world god only knows, bakemonogatari and C are close behind [04:12] <tm512> have you watched Now and Then, Here and There? [04:12] <klaxa> i've heard of it somewhere, can't remember where, but i haven't seen it [04:12] <LordThumper> Ah you're definately not a mainstream guy then [04:13] <klaxa> yeah, no mainstream here :P [04:13] <LordThumper> anyways, going to watch the rest, thanks again [04:13] <klaxa> np :) [04:16] <klaxa> haha lol LordThumper: that file is broken [04:18] <klaxa> even the subtitle file seems to be incorrect [04:18] <klaxa> or rather not perfectly compliant with aegisub [05:17] <t4nk201> hi, does anyone know the purpose about "presets" file in the folder /ffmpeg/presets? [05:17] <t4nk201> how to use the presets files? [05:17] <t4nk201> if the video codec is libx264, how to use "presets" file for reduce latency? [05:18] <LordThumper> klaxa: Yeah, Subtitle Workshop opens the subtitle but it does not support subtitle position, had to use Subtitle Edit [05:18] <LordThumper> SubtitleEdit even [05:18] <LordThumper> Was trying to repair some c [05:18] <LordThumper> some broken characters [05:18] <LordThumper> Horriblesubs release is fine though [06:03] <praveenmarkandu> Hi. When I try to transcode a Apple ProRes file to H.264, it tells me vprofile main is not available [06:03] <praveenmarkandu> it only accepts high422 and high444? [06:03] <praveenmarkandu> is that expected behaviour? [06:30] <mark4o> praveenmarkandu: use -pix_fmt yuv420p [06:31] <mark4o> main profile does not support yuv422p or yuv444p [06:44] <tm512> I've been trying to figure this out for some time with no luck, does anyone have a way to stream with ffmpeg that doesn't have the audio out of sync with the video? I'd really appreciate that [07:16] <praveenmarkandu> mark4o, thanks! [09:50] <praveenmarkandu> someone answered my question on #ffmpeg, today was a good day [09:57] <Mavrik> :)) [09:57] <Mavrik> well, it IS early morning ;) [10:22] <MrE> Can ffmpeg overlay text based dvb subtitles ? [10:41] <killown> how about this security issue https://secunia.com/blog/372/ has it been fixed? [10:52] <killown> dis-consider my question, I know now these secunia guys are a liars [13:01] <xyz_> what's the reason for licensing the library under the restrictive LGPL? [13:02] <JEEB> because that's what people chose to be the closest thing for their goal at various points of time, I would guess? [13:03] <JEEB> and as such, LGPL really isn't *that* restrictive [13:03] <JEEB> FSF actively wants people to use GPL for libraries as well :D [13:04] <JEEB> I mean, am I the only one who thinks that presenting the source code of the library you used if you used shared linking, and source code of the library + object files for the rest of it if you used static linking not too bad a limitation of you wanted to use it in a closed source application? [13:05] <xyz_> well, i don't know [13:05] <xyz_> but i prefer non-copyleft permissive licenses [13:06] <xyz_> does there exist any permissive alternative for ffmpeg/libav? [13:06] <JEEB> I prefer that for most of the random stuff I do, but personally I really don't see the problem with libav* being LGPL [13:06] <JEEB> and no [13:07] <JEEB> no such a wide and feature-filled library exists [13:08] <JEEB> the biggest problem for corps regarding libav* generally tend to be the encoders that are GPL anyways (third-party libs used via libavcodec) [13:09] <xyz_> well, i've tried to start such a project on sourceforge... but i could myself write propably just some wrappers for permissive libraries like Ogg, which is BSD [13:09] <xyz_> or was it MIT? i don't remember [13:11] <JEEB> for whatever reason I really feel like such a project wouldn't get much wind under itself :/ Because ffmpeg/libav by themselves are not GPL, just LGPL. And that already makes them usable in closed-source applications [13:11] <JEEB> you just have to give the user the chance to swap the libraries should they really, really want to do it [13:12] <JEEB> basically there's just so much stuff done already in libav* that I'd be surprised if someone tried to rewrite it all [13:12] <JEEB> or even most used features of it [13:14] <xyz_> well... the gpl and lgpl require 'derivative works' to be under the same license. that sucks :( Because an (L)GPL project can itself be a derivative work of some more permissively licensed software, but not the other way around. [13:14] <xyz_> but if I decide to distribute my object code, could I just pack all of the .o files into a single .a archive? [13:15] <JEEB> whatever way lets people link it all together [13:16] <JEEB> that's in the case of static linking anyways, shared linking lets you have much more leeway [13:16] <xyz_> well... maybe I shouldn't argue about this license thing any more. [13:17] <JEEB> well, it's good to bring things up, but LGPL just isn't that restrictive in the end [13:17] <JEEB> the only semi-good argument I've heard is from Japanese developers, since in Japan you can tell people not to RE your stuff [13:17] <JEEB> and LGPL lets people RE your stuff [13:17] <xyz_> what is RE? [13:18] <JEEB> "Our bosses put that clause in our EULA so we can't really use LGPL stuff" [13:18] <xyz_> reverse engineer? [13:18] <JEEB> reverse engineering [13:18] <JEEB> yes [13:18] <JEEB> in most countries you can't even limit it [13:18] <xyz_> but actually, I now have a problem compiling libav for my Nokia N9 cell phone. [13:18] <JEEB> libav or the libav* libraries from ffmpeg? [13:18] <JEEB> two separate projects, after all :) [13:18] <xyz_> it's not actually ffmpeg, but well... [13:19] <JEEB> just noting that it's better to ask the related party :) [13:19] <xyz_> there's a less populated libav channel [13:19] <JEEB> elenril and spaam have already been active there today [13:19] <JEEB> and I am there, too [13:19] <JEEB> so it's just a better idea to push your questions to the related party :) [13:22] <kamalbanga> when compiling a basic code that includes libavformat.h, i get the error: fatal error: libavcodec/avcodec.h: No such file or directory compilation terminated [13:22] <JEEB> you used something that also needed libavcodec around? [13:22] <JEEB> wait... isn't the header supposed to be avformat.h or so? [13:22] <kamalbanga> no i just did av_register_all() [13:23] <kamalbanga> the header i put is: #include <libavformat/avformat.h> [13:23] <JEEB> and yeah, that's it [13:23] <JEEB> also I wouldn't be surprised if that one actually needed libavcodec as well [13:23] <JEEB> if nothing else, the header [13:24] <kamalbanga> i hav actually put all the three headers [13:24] <kamalbanga> avcodec, avformat and avswscale [13:24] <JEEB> also f.ex. the raw AVC parser as far as I can tell uses some libavcodec utilities [13:24] <JEEB> well then they are not in your search path [13:24] <kamalbanga> exactly i found this stacoverflow: http://stackoverflow.com/questions/12127397/fatal-error-libavcodec-avcodec-… [13:25] <kamalbanga> as it says i should give search path [13:25] <JEEB> have you in any way or form actually installed libav* ? [13:25] <JEEB> or are you just trying to copy the random headers off to somewhere and trying to use it like that? [13:25] <kamalbanga> but my problem is this : when i do "find . -name avformat.h", i get this path:..ffmpeg_sources/ffmpeg/libavformat/avformat.h [13:26] Action: JEEB takes a deep breath [13:26] <kamalbanga> avformat.h and avcodec.h are there but not in any folder called "include" [13:26] <JEEB> of course [13:26] <JEEB> you are supposed to install the headers and libraries you have built [13:27] <JEEB> that process puts stuff into the generally used paths along a prefix you have set (or, by, default /usr/local ) [13:27] <kamalbanga> i followed the complete instructions given here: http://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide [13:28] <JEEB> yes, this uses a prefix within your home directory [13:28] <JEEB> so that it works on machines where you don't have root rights [13:28] <JEEB> ls ${HOME}/ffmpeg_build/include/libavformat/avformat.h should show you something :P [13:28] <kamalbanga> thanks for your response <JEEB>, i wish you help me, this problem is frustrating me since days... [13:29] <JEEB> you seem to have been executing things blindly and have no idea about things work, I'm not surprised at all :P [13:30] <kamalbanga> $HOME/ffmpeg_build/include has only these things: fdk-aac libyasm libyasm.h libyasm-stdint.h opus vpx x264_config.h x264.h [13:31] <JEEB> then you have not installed ffmpeg itself [13:32] <JEEB> and then yes, you will need to later add ${HOME}/ffmpeg_build/include|lib to your header and library search paths [13:33] <JEEB> do note though that if you build your ffmpeg with fdk-aac, your compile will become non-distributable [13:33] <kamalbanga> i had installed it and am reinstalling it [13:33] <kamalbanga> yes, i understand that [13:34] <JEEB> if you don't have the headers in there, you haven't followed the damn instructions you have linked, btw :) Since the ffmpeg compilation step a) has the --prefix set and b) contains a step to install the thing [13:38] <kamalbanga> after following the installation steps i also run the command "ffmpeg 2>&1 | head -n1" and it gave me the version of the ffmpeg i installed [13:38] <kamalbanga> so i had thought that it has successfully installed [13:39] <JEEB> the binary actually goes to another place, ${HOME}/bin (you can see the --bindir option being set) [13:40] <JEEB> but that configuration line as far as I can tell dos not disable installation of the libraries [13:40] <JEEB> so if the installation went fine you should have had stuff under ffmpeg_build/include|lib [13:42] <kamalbanga> yes ffmpeg shows in $HOME/bin/ along with ffplay ffprobe and ffserver [13:45] <kamalbanga> i just installed it again and if you (JEEB) permit, i will show the output to you [13:45] <JEEB> just pastebin it and link here [13:47] <kamalbanga> http://pastebin.com/qL44QzbH [13:48] <JEEB> yes, the things seem to have been installed [13:48] <JEEB> now check whatever got set as the --prefix [13:48] <JEEB> and what the include/lib directories there contain [13:48] <kamalbanga> my working directory while doing "sudo make install" was $HOME/ffmpeg_sources/ffmpeg [13:49] <JEEB> you don't need the sudo there [13:49] <JEEB> unless you weren't following the instructions [13:49] <JEEB> sudo is only needed when installing into somewhere where your user has no rights to things [13:51] <kamalbanga> the ffmpeg_build/include contains the same thing as before, no change [13:51] <kamalbanga> i mean no avcodec and avformat libraries [13:51] <JEEB> try running install without the sudo [13:51] <kamalbanga> ok [13:52] <kamalbanga> then it gives some errors - permission denied [13:53] <kamalbanga> http://pastebin.com/Xb8gzuac [13:53] <JEEB> yes [13:53] <JEEB> since you used sudo [13:54] <JEEB> congratulations [13:54] <kamalbanga> oh.. [13:54] <JEEB> you might want to clean up ffmpeg_build and do it again [13:54] <JEEB> without sudo [13:54] <kamalbanga> actually i used sudo make install out of habit but that seems to be a problem now... [13:55] <JEEB> sudo make install is correct if you are installing into a place where you have no rights as a normal user, it runs stuff as a priviledged user [13:55] <JEEB> so if you were for example installing into /usr/local you would need it [13:55] <kamalbanga> ok, clean as in "make clean" or "make distclean" or something else like remove or uninstall [13:55] <JEEB> remove the ffmpeg_build directory structure [13:55] <JEEB> and start again :P [13:55] <kamalbanga> yes i got that point now and will take care next time [13:56] <kamalbanga> ok [13:56] <JEEB> you might need sudo for it [13:56] <JEEB> because you started creating new directories and all as such, so you just might not have the rights for it as a normal user :D [14:00] <kamalbanga> yes sudo rm -r did it [14:10] <kamalbanga> when i build yasm, it gives error while running "make install", the error being: cannot create regular file '$HOME/bin/yasm': permission denied [14:10] <kamalbanga> so it seems i need sudo for that [14:16] <JEEB> kamalbanga, that just means that it got messed up as well [14:16] <JEEB> remove both the bin and the ffmpeg_build directories [14:17] <JEEB> and then let the make install process re-create them without sudo [14:19] <kamalbanga> yes, now it's fine [14:28] Action: ffmpegNoob learning ffmpeg [14:40] <kamalbanga> while configuring ffmpeg as here http://pastebin.com/ABBxk3QQ i got error "ERROR: opus not found" although opus is present in $HOME/ffmpeg_build/include/ [14:42] <JEEB> better to pastebin config.log [14:44] <kamalbanga> http://pastebin.com/3UU5fUtT [14:48] <JEEB> ok [14:49] <JEEB> add before your configure command PKG_CONFIG_PATH=${HOME}/ffmpeg_build/lib/pkgconfig [14:50] <JEEB> so it becomes PKG_CONFIG_PATH=${HOME}/ffmpeg_build/lib/pkgconfig ./configure --your-options [14:50] <kamalbanga> yes i had done that already [14:50] <kamalbanga> since it was in the instructions [14:50] <JEEB> then the pkg-config check should go through fine [14:52] <kamalbanga> pkgconfig folder contains these: fdk-aac.pc opus.pc vpx.pc [14:52] <JEEB> yup [14:52] <JEEB> which the pkg-config binary will then check [14:54] <kamalbanga> any idea what i should do next... [14:55] <kamalbanga> shall i leave opus and continue with the other options as i may very well never user opus, i primarily need to deal with aac ac3 and mp3 [14:56] <JEEB> what [14:56] <JEEB> what I said should fix it, unless you weren't showing me the whole config.log [14:56] <JEEB> and yes, if you don't need opus then don't compile it in [14:58] <kamalbanga> i thought i would show relevant section of config.log, this contains more: http://pastebin.com/KvfyHzSS [14:59] <JEEB> and you had the PKG_CONFIG_PATH thing set when configuring? [15:00] <kamalbanga> yes [15:01] <kamalbanga> i may need opus sometimes as i hav to deal with internet videos and ogg format may use it beside vorbis [15:12] <kamalbanga> oh no: i left opus and then configured and now same with x264 "ERROR: libx264 not found" http://pastebin.com/9KPmiizN [15:12] <JEEB> well that one is a simple one [15:13] <JEEB> you've forgotten your --extra-cflags="-I${HOME}/ffmpeg_build/include" [15:13] <JEEB> in the configure :P [15:13] <JEEB> wait it's there [15:13] <JEEB> then you just haven't installed the x264 headers [15:14] <kamalbanga> i did install [15:15] <kamalbanga> god knows what problem is there... [15:15] <kamalbanga> i followed the complete instructions\ [15:15] <JEEB> /home/user06/ffmpeg_build/include -- try ls'ing this directory and see if there is a x264.h :P [15:16] <kamalbanga> yes i did ls it and it wasn't there [15:16] <kamalbanga> although it should have been there [15:16] <JEEB> thus you have not compiled/installed x264 properly! [15:17] <JEEB> And just to note, these things aren't that fragile. You must have just forgotten or mistaken something. [15:17] <JEEB> so back to compiling x264 you go :) [15:33] <kamalbanga> JEEB: thanks a lot dude, ffmpeg compiled and now it shows in ~/ffmpeg_build/include/ [15:36] <kamalbanga> but now when i compile my sample av_register_all() code it gives this:undefined reference to `av_register_all' collect2: ld returned 1 exit status [15:36] <kamalbanga> seems to be some linking error [15:37] <JEEB> yes, because you also need to -L${HOME}/ffmpeg_build/lib when compiling if you want to link against those libraries [15:38] <JEEB> and if you have that already, then you lack the "link this!" option for the library that contains that function [15:38] <JEEB> s/function/export/ [15:40] <kamalbanga> i did this: cc tut.c -I/home/user06/ffmpeg_build/include -L/home/user06/ffmpeg_build/lib/ and same error [15:40] <JEEB> you are then not actually linking against anything [15:41] <JEEB> you need to tell it what libraries you want to link against, you know [15:41] <kamalbanga> like -lavformat and -lavcodec [15:41] <JEEB> yes [15:41] <kamalbanga> ok [15:43] <kamalbanga> i linked it and now i get this royal error: http://pastebin.com/5z2WLUrX [15:43] <JEEB> -lavutil ? [15:44] <kamalbanga> i didn't include avutil.h in my source code [15:45] <JEEB> yes, but those exports are from avutil [15:45] <kamalbanga> and the error i pasted was only part of the error as it was very big [15:45] <kamalbanga> ok i will link avutil too [15:45] <JEEB> yes, but those are all as far as I can see from avutil [15:45] <JEEB> so link avutil [15:46] <kamalbanga> same type of error again: http://pastebin.com/xdNcJ5hw [15:47] <kamalbanga> i linked avutil avcodec avformat and swscale [15:49] <JEEB> try switching the linked libraries' order because that should either be in util or codec [15:49] <kamalbanga> any orders you recommend becoz there will be 4! permutations... [15:52] <kamalbanga> i tried many orders but the same error [15:56] <kamalbanga> JEEB: no luck here, btw do you think this error http://ffmpeg.gusari.org/viewtopic.php?f=8&t=942 is relevant to me [16:08] <kamalbanga> i wish somebody helps me.. while compiling a sample code that just has av_register_all() code by command "cc tut.c -I/home/user06/ffmpeg_build/include -L/home/user06/ffmpeg_build/lib/ -lavutil -lavformat -lavcodec -lswscale" i get the error http://pastebin.com/47npnJ1P [16:12] <kamalbanga> . [16:12] <kamalbanga> 3 [16:12] <kamalbanga> . [16:17] <sussman> We conjure the spirits of the computer with our spells. [16:23] Last message repeated 1 time(s). [16:28] <Mavrik> g'day [16:32] <kamalbanga> JEEB: can you suggest anything i can do.. i am very desperate to make it work or in other case give link to any resource or tutorial that explains these building stuff in detail like config or linking fundamentals [16:42] <kamalbanga> well.. no help.. thanks to JEEB though.. [16:48] <taqattack> hey everyone [16:48] <taqattack> I'm wondering if ffmpeg 2.0 does colorspace conversion with opencl [16:55] <Mavrik> taqattack, nop. [16:55] <taqattack> Mavrik, so what is opencl used for atm? [16:55] <Mavrik> if you grep the code [16:56] <Mavrik> you'll see it's used in deshake and denoise filters [16:56] <Mavrik> + the framework for more was prepared [16:56] <taqattack> oh ok [16:57] <taqattack> thanks Mavrik. Are there any plans to implement colorspace conversions for gpu? [16:57] <Mavrik> no idea [16:57] <Mavrik> probably if anyone remembers& even though, colorspace conversions are even now accelrated with MMX/SSE ASM [16:58] <Mavrik> so I'm not sure using OpenCL would provide huge performance gains [17:00] <taqattack> hmm [17:20] <DannyZB> "Input #0, mov,mp4,m4a,3gp,3g2,mj2" [17:20] <DannyZB> how do I extract the codec and not just .. potential filenames ? [17:21] <JEEB> that's the type [17:21] <JEEB> not a file name or extension [17:21] <JEEB> it should after that output the streams in that file [17:21] <DannyZB> how can it have 5 types ? [17:21] <JEEB> except [17:21] <JEEB> DannyZB, because the demuxer supports all of those and they overlap at parts [17:22] <DannyZB> ohh so it says it's compatible with all those container types .. cool thanks [17:22] <JEEB> MPEG-4's container was based on MOV, and 3GP's container was based on MPEG-4's container [17:22] <JEEB> and so forth and so forth [17:22] <JEEB> the way it presents them is extension-like tho [17:22] <JEEB> lol [17:23] <DannyZB> FFMPEG is wierd haha [17:23] <JEEB> basically it means "I've got 'mov-like' stuff coming in" [17:23] <DannyZB> btw any of you guys related to the development team? [17:23] <JEEB> depends on how you mean that, I've done a video encoder once for libavcodec, but that's more or less it [17:23] <JEEB> but yeah, some active devs are afaik around [17:24] <JEEB> although development related talk is generally on the -devel channel [17:25] <JEEB> anyways, if you wanted to know the stream types within the file [17:25] <DannyZB> do you know when they intend to integrate quicksync? [17:25] <JEEB> it should output that after that [17:25] <JEEB> quicksync as in what exactly? [17:25] <JEEB> it has the ASIC-using decoding part [17:25] <DannyZB> as in super-fast intel transcoder [17:25] <DannyZB> it's only that fast when used both for encoding and decoding [17:26] <DannyZB> just decoding kinda misses the point [17:26] <JEEB> no-one has worked on that yet, and so far it seems to be more or less on the same level as x264's superfast [17:26] <JEEB> speed-wise [17:26] <JEEB> so it's not really anything SuperSpecial in that sense [17:26] <DannyZB> thats nonsense written by people that tested encoding/decoding separately [17:26] <DannyZB> I benchmarked myself [17:26] <DannyZB> and it's considerably faster [17:27] <DannyZB> + the quality is higher than superfast , it's closer to "faster" [17:27] <JEEB> how did you test btw? [17:27] <JEEB> also what kind of CPU did you test it with? [17:27] <DannyZB> dual core Intel I5 Ivy Bridge over windows .. ffmpeg commandline for software vs. MediaEspresso for ffmpeg [17:28] <DannyZB> quicksync converted Gladiator Full HD in 12 minutes ... [17:28] <taqattack> ffmpeg has quicksync? [17:28] <JEEB> what metric did you use and what was the general testing scenario [17:28] <DannyZB> Full HD AVI -> HD mp4 [17:28] <JEEB> taqattack, there's a decoder patch incoming from libav afaik [17:29] <JEEB> DannyZB, no -- I meant the actual testing procedure [17:29] <JEEB> what metric, what did you test etc. [17:29] <DannyZB> stopwatch -> convert [17:29] <JEEB> ugh [17:29] <DannyZB> took a bunch of HQ movies and converted with FFMPEG superfast [17:29] <Paranoialmaniac> ... [17:29] <JEEB> since it doesn't seem to be going through, let me put up an example [17:29] <JEEB> you have two encoders [17:29] <DannyZB> you can say I could use more accurate settings .. but the difference was x2.5 [17:29] <JEEB> no [17:30] <JEEB> I'm just wanting you to note your methods, let me basically write up an example :P [17:30] <taqattack> Thanks JEEB [17:31] <DannyZB> ok are you just against my methods or plain saying I'm wrong? [17:31] <JEEB> no [17:31] <JEEB> you haven't even said your methods, as I said I will give an example [17:31] <JEEB> so you can understand what I mean [17:32] <DannyZB> k [17:33] <DannyZB> "Video: h264 (High) (avc1 / 0x31637661)," [17:34] <DannyZB> does ffmpeg have (High) or whatever after the encoder name in all cases? [17:34] <JEEB> not all [17:34] <JEEB> it's the profile [17:34] <JEEB> of the AVC/H.264 stream [17:34] <DannyZB> how do I reliably extract the decoder name then ? [17:34] <DannyZB> *encoder [17:34] <JEEB> uhh [17:34] <JEEB> you don't know what encoder was used for the file unless the encoder writes it somewhere [17:35] <JEEB> for x264 you can `strings file.welp | grep "x264"` [17:35] <JEEB> but even that is not 100% correct because in theory the AVC/H.264 stream could be cut in the middle [17:35] <DannyZB> one sec .. the encoder doesn't always write it ? [17:35] <JEEB> write what? [17:35] <DannyZB> this is messed up haha [17:35] <DannyZB> I need the codec ugh .. I didn't mean encoder [17:36] <DannyZB> I need to extract all codecs used [17:36] <JEEB> it's the thing that comes after Video: - h264 is AVC/H.264, the thing within () is then the profile and the () after that is the ID as ASCII and hex that is noted in the container [17:37] <JEEB> most probably ffprobe would be simpler for you to parse, it can output json and stuff even :P [17:37] <DannyZB> cool [17:38] <DannyZB> why do online sites never know useful stuff like that haha [17:38] <JEEB> One example is to set both encoders up to eleven, and making sure you're not making any set-up misses regarding the encoder optimizing or not optimizing to your metric, and then encode to a set bit rate (say, with two-pass), and then compare the results with your set metric. Another one is to [17:38] <JEEB> match the encoders in speed and bit rate [17:38] <JEEB> and then see with your metric which is better [17:38] <DannyZB> I used the same bitrate and resolution in both [17:39] <DannyZB> used the ffmpeg superfast preset (optimized to SSE3 etc') [17:39] <DannyZB> 756kb/s [17:39] <DannyZB> Quicksync quality was higher and speed was more than twice as fast [17:40] <DannyZB> ffmpeg superfast has a tendency to blur action scenes .. which is why I usually stay away from it [17:41] <DannyZB> preg_match("/Video: (.*?) (/", $content, $matches); - > should do the trick for extracting video name ? [17:41] <DannyZB> *codec name [17:41] <JEEB> just use ffprobe :P [17:41] <JEEB> it's made for stuff like that [17:42] <JEEB> Also when comparing encoders, you in general want to make sure that the encoders are on the same ground, adding the decoder + encoder combo in is of course an interesting extra. In the best case you should then also test just decoding on the other side. Also you do see how I note that you should in one way or another match something except for the VeryObvious stuff, right? [17:42] <DannyZB> yea I get what you mean .. but quicksync performance is damn impressive in any case [17:42] <JEEB> anyways, let's just say that while your results seem interesting, I wouldn't publish them without further proper testing :) [17:43] <JEEB> I have an ivy bridge here home [17:43] <JEEB> i5 3570K [17:43] <DannyZB> using QS for decoding is pointless since you lose the performance gain by pushing data back and forth between the graphics core and the processor [17:43] <DannyZB> you lose the graphics processor memory bandwidth advantage [17:43] <JEEB> I wouldn't dismiss it completely, you know. It could still be faster than other alternatives [17:44] <JEEB> that said, in general when testing encoders [17:44] <JEEB> you use raw video as input [17:44] <DannyZB> I have a server which needs to convert around 40 hour videos per hour [17:44] <DannyZB> that would kill software encoders if you want good quality [17:45] <DannyZB> what I'm trying to do now is parse video files and just copy video/audio if it is close enough to what I want .. hopefully most people encode with h264 [17:45] <JEEB> well, yeah -- ffprobe for that [17:46] <DannyZB> yea looking into it .. looks good [17:46] <DannyZB> you guys should make ffprobe for ffmpeg logs lol [17:48] <JEEB> anyways, I might one day look into QS encoding again [17:49] <DannyZB> you can do it right now [17:49] <DannyZB> intel are about to release an official linux sdk ( with commandline transcoders ) [17:49] <DannyZB> I just don't like using two different tools [17:50] <JEEB> well, I'm on Windows anyways, and I think the QS avcodec decoder will IIRC be at least at first windows-only [17:50] <DannyZB> as a decoder handbrake already integrated it in windows [17:51] <JEEB> too much fluff around, and I don't know the line stuff goes through with handbrake too well :) [17:52] <DannyZB> i don't get something .. how do you set ffprobe writer ? [17:52] <DannyZB> ps .. I'm pretty sure ffmpeg -i is just an alias for ffprobe [17:52] <JEEB> nah [17:52] <JEEB> they're more different now [17:52] <JEEB> they're separate apps to begin with tho :) [17:53] <JEEB> also I'm pretty sure even with the possible transfer latency between the ASIC and the CPU with decoding stuff isn't too bad of a problem with QS -- I've seen numbers >200fps even with that going on [17:54] <DannyZB> how do I make ffprobe use json ? [17:54] <JEEB> -print_format json ? [17:55] <JEEB> http://ffmpeg.org/ffprobe.html [17:56] <DannyZB> yea the doc is a bit chinese to me haha [17:56] <DannyZB> thanks [17:58] <DannyZB> ok maan .. the "json" output just puts braces around the same output lol [18:00] <JEEB> looks quite more parse'able for me http://stackoverflow.com/a/8191228/1620828 [18:00] <JEEB> haven't used it myself tho :P [18:00] <DannyZB> wtf lol [18:00] <DannyZB> how did you do that [18:01] <DannyZB> okokokok got it .. [18:03] <Fjorgynn> Öka [18:03] <Fjorgynn> Sorry wrong channel [18:36] <DannyZB> whats the difference between using -c:a and -acodec ? [18:36] <JEEB> latter is an older way, first is the newer way [18:37] <JEEB> for now at least the old way is still there as well [18:37] <DannyZB> and what happens if I use -vcodec copy and -x264opts in the same link ? are x264 options just ignored ? [18:37] <DannyZB> *command [18:39] <JEEB> either an error or just ignored [18:39] <JEEB> -x264opts are after all, specific to the x264 encoder [18:40] <DannyZB> ugh better not do that .. a bad habit [18:40] <DannyZB> "/usr/bin/ffmpeg -i $source_path -strict experimental -x264opts bitrate=756:vbv-maxrate=756:vbv-bufsize=378 -vcodec $vcodec -c:a libfdk_aac -movflags +faststart -preset faster /var/www/media/video/{$hash}.mp4 -y >/var/www/ffmpeg_logs/{$hash}.log 2>&1 &" [18:41] <DannyZB> I don't think I actually need -strict experimental .. am I right ? [18:41] <JEEB> also wtf @ setting bitrate and all in x264opts :D , and yes -- I don't think you're using anything that needs the "experimental" features enabled [18:42] <JEEB> -b:v , -maxrate and -bufsize [18:42] <JEEB> ^ same things as those x264opts [18:42] <DannyZB> for some reason that used to crash the encoder and give me weird errors [18:42] <DannyZB> I know this way is messed up but it doesn't crash haha [18:58] <DannyZB> http://pastebin.com/pa7P4bYm [18:58] <DannyZB> to those of you using php .. enjoy :P [19:01] <DannyZB> ok to you video enthusiasts out here .. how do you avoid blur in scene changes during conversion when using superfast preset ? [19:17] <trose> does anyone have experience writing frames to ffmpeg from C? [19:17] <trose> Here's my piping code [19:17] <trose> http://pastebin.com/2EV91cPh [19:17] <trose> I get this error in ffmpeg... [19:18] <trose> http://pastebin.com/JD7U8G49 [19:19] <trose> pipe:: Input/output error isn't very descriptive... [19:26] <teratorn> trose: doesn't a PPM have ascii integers, not raw byte values? (unrelated to your pipe error) [19:28] <teratorn> well I *guess* its unrelated.. [19:28] <trose> teratorn: there are two versions, distinguished by either P6 or P3 on the first line [19:29] <trose> P6 is the raw byte "standard" form [19:29] <teratorn> ok :) [19:29] <trose> maybe ffmpeg expects the ascii form? [19:29] <teratorn> I wouldn't know [19:29] <trose> teratorn: haha [19:30] <trose> I'm pretty noob with C coding so I'm not sure if i'm doing something wrong there or not [19:32] <trose> this might be worth noting... [19:32] <trose> if i write the output of osdemo to a file and then pipe the file into ffmpeg using image2, I get a working video [19:36] <DannyZB> if I want to get the average bitrate of a video .. what do I do? [19:36] <DannyZB> ffprobe doesn't seem to retrieve the bitrate for AVI/XVID videos [19:51] <Mavrik> DannyZB, calculate it from filesize. [19:51] <DannyZB> I found another way [19:51] <DannyZB> even when ffprobe doesn't output video bitrate [19:51] <DannyZB> it -always- outputs total bitrate [19:52] <DannyZB> total - audio = video [19:52] <Mavrik> hmm, does it always output total bitrate? I don't think so [19:52] <Mavrik> but if it does for your formats, it's ok as well. [20:03] <DannyZB> avi/mp4/mkv it does [20:03] <DannyZB> haven't tested others .. but we'll see [20:04] <Mavrik> those are fine, streaming formats are the ones where these things usually fail :) [20:13] <DannyZB> streaming formats? [20:13] <DannyZB> which ones for example ? [20:13] <DannyZB> mp4 is streamable [20:13] <Mavrik> mp4 is about as crappy for streaming as it gets [20:14] <Mavrik> formats like MPEG2-TS and similar [20:14] <LithosLaptop> TS is the best [20:14] <DannyZB> mp4 is what you got with flash [20:14] <DannyZB> nobody cares whats better when it's either that or oh-so-terrible flv [20:15] <DannyZB> *flash/html5 [20:16] <DannyZB> i don't want to convert the same video twice [20:18] <Fjorgynn> :D [20:18] <DannyZB> lets say I convert a video to the same format .. is direct transcode the only way to downsample ? [20:18] <Fjorgynn> lol flash [20:18] <DannyZB> no matter what you say .. flash video quality is great [20:18] <Mavrik> DannyZB, what do you mean by "downsample"? [20:19] <DannyZB> turn Full HD to HD [20:19] <Mavrik> yes, you have to transcode it. [20:19] <Mavrik> "flash video quality"? what? :D [20:19] <downwater> hello [20:20] <Mavrik> hey there. [20:21] <downwater> is "wmv3 encoding" currently supported by ffmpeg ? i'm asking because a friend seems to have produced a wmv3 video with it (Debian) or it looks like it's not supported [20:21] <Mavrik> uh [20:22] <Mavrik> downwater, nope, only WMV8 is supported, not WMV9 [20:22] <Mavrik> D.V.L. wmv3 Windows Media Video 9 [20:22] <Mavrik> only the decoder is present [20:22] <JEEB> years ago there was talk about a VC-1 encoder patch, but that never really materialized IIRC [20:23] <JEEB> if you need to encode WMV3/VC-1, I recommend you just use MS's expression encoder 4+ [20:23] <Mavrik> mhm, I think people just stopped caring about the format after H.264 really got adopted [20:27] <downwater> MPlayer output : selected video codec: [ffwmv3] vfm: ffmpeg (FFmpeg WMV3/WMV9) [20:27] <downwater> so it's not wmv3 ? [20:28] <Mavrik> what makes you think that? :) [20:28] <Mavrik> it's just telling you it has chosen ffmpeg's WMV decoder :) [20:28] <downwater> aha... [20:28] <downwater> indeed [20:28] <downwater> how can i check the format so ? [20:29] <downwater> wait [20:29] <Mavrik> ffprobe the file? [20:29] <file> keep your hands away from me [20:29] <downwater> thanks Mavrik [20:29] <downwater> Audio: wmav2 [20:30] <downwater> Video: wmv3 [20:30] <downwater> Oo [20:31] <LithosLaptop> lol @ probing file [20:32] <downwater> Mavrik, does Debian patch ffmpeg ? [20:32] <Mavrik> uh [20:32] <Mavrik> last I checked, debian shipped libav instead of ffmpeg [20:32] <Mavrik> symlinked "ffmpeg" to avconv and kept telilng people ffmpeg is deprecated [20:32] <downwater> raah [20:33] <Mavrik> downwater, use static builds from ffmpeg download page :) [20:33] <downwater> the funniest : wmv2 video from ffmpeg are not playable by windows media player [20:34] <JEEB> :) [20:34] <downwater> my options : -vcodec wmv2 -acodec wmav2 -s 720x576 -b:v 1600k -r 25 [20:37] <LithosLaptop> eek [20:40] <downwater> thanks anyway [20:40] <downwater> it helped me [20:40] <downwater> bye ! [20:40] <LithosLaptop> downwater: my advice is to not use wmv/wma with ffmpeg [20:40] <LithosLaptop> use MS's expression encoder [20:40] <downwater> yes of course [20:40] <downwater> MS's expression encoder ? [20:41] <LithosLaptop> the wma audio encoder in ffmpeg is nearly as bad as vo-aac/// [20:41] <JEEB> downwater, the same I mentioned [20:41] <downwater> JEEB, sorry didn't see it [20:41] <JEEB> the free version should support avisynth input and wmv3/vc-1 input [20:41] <JEEB> uhh [20:41] <JEEB> output [20:41] <LithosLaptop> http://en.wikipedia.org/wiki/Microsoft_Expression_Encoder [20:42] <downwater> if it works on wine... [20:42] <downwater> maybe [20:42] <JEEB> downwater, http://www.microsoft.com/en-us/download/details.aspx?id=18974 [20:42] <JEEB> you can try [20:43] <downwater> thanks ! [20:43] <downwater> i'll try [21:47] <pdbogen> Hi, guys. I'm trying to use ffmpeg to assemble JPEGs into a timelapse video, except that the JPEGs are continuously arriving, so I actually want to append one JPEG to the end of the video. So the simplest case of this is that I have two JPEGs. I want to encode the first one to video (ffmpeg -i img1.jpg -vcodec mpeg4 result.avi), then concatenate the second one (ffmpeg -i result.avi -i img2.jpg -filter_complex concat -vcodec mpeg4 tmp.avi; [21:47] <pdbogen> Running with -v debug has '*** drop!' at the end, and then drop=1 at the summary, but no indication as to /why/ there was a frame dropped. [23:02] <brontosaurusrex> pdbogen, isn't concat limited to same codec? [23:37] <pdbogen> brontosaurusrex: Since filters handle "uncompressed data," I think the codec(s) don't matter? [00:00] --- Fri Jul 12 2013

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