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October 2012

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ffmpeg-devel.log.20121020
by burek 21 Oct '12

21 Oct '12

[00:08] <cone-191> ffmpeg.git 03Hendrik Leppkes 07b87ff3449662: vc1: implement vc1 field interlaced dxva2 decoding * 03http://tinyurl.com/8wlpu6r03 [00:08] <cone-191> ffmpeg.git 03Hendrik Leppkes 0733f2a4942380: vc1: only disable interlaced b-frames for software decoding * 03http://tinyurl.com/8lgvq8703 [00:33] <ubitux> saste: i have a strange issue with lavfi device and ffprobe, which is not reproducible with ffmpeg & ffplay [00:33] <ubitux> interested? :) [00:34] <saste> ubitux, what? [00:34] <saste> and don't forget to file a bug report, if you don't plan to work on it [00:34] <ubitux> i kind of try to understand it at the moment [00:34] <ubitux> i'm doing ./ffprobe -f lavfi -i 'amovie=input.mp3,ebur128=video=1 [out0][out1]' -show_frames [00:35] <ubitux> it stops immediately without error [00:35] <ubitux> av_read_frame() is returning an "invalid argument" err code [00:35] <ubitux> (coming from the request frame thing) [00:36] <saste> what if you remove ebur? [00:36] <ubitux> works fine [00:37] <ubitux> actually even a split & showwaves works [00:37] <ubitux> so maybe i miss something in the filter [00:37] <ubitux> but since it works fine with ffplay & ffmpeg.. [00:37] <saste> ubitux: gdb... [00:37] Action: ubitux prefers random av_log debugs [00:38] <saste> may be related to internal caching [00:38] <ubitux> i was just asking in case you may have an idea [00:48] <ubitux> saste: yes, the fifo looks empty [00:49] <saste> uhm... [00:52] <ubitux> it looks like the main difference with ffplay is that ffplay just doesn't care and continue to read frame anyway [00:55] <saste> ubitux: and ffmpeg? [00:55] <ubitux> ffmpeg has no issue either [00:55] <saste> and yes i remember that i saw a similar error [00:55] <saste> why not? [00:56] <ubitux> mmh maybe it has.. [00:56] <ubitux> heh it has an explicit error [00:58] <ubitux> well, i'll see later© [00:58] <ubitux> 'night [00:58] <ubitux> saste: tomorrow is your last chance to comment on the lavfi metadata btw ;) [00:58] Action: ubitux & [00:59] <saste> ubitux, are you waiting for reviews? [01:00] <saste> what about the issue spotted by nicolas? [01:00] <saste> anyway give some time, i'll try to review tomorrow [01:00] <saste> if not feel free to push it [01:37] <durandal_1707> seeking in he-aac-v2 have problems [02:02] <cone-191> ffmpeg.git 03Hendrik Leppkes 07953a3dcc4e28: Mark data symbols shared between libraries with av_export * 03http://tinyurl.com/8qq2auf03 [02:02] <cone-191> ffmpeg.git 03Hendrik Leppkes 0704bf2e7f0e9c: swresample: include ff_log2_tab for shared builds * 03http://tinyurl.com/9x793yo03 [02:03] <ramiro> hi [02:08] <durandal_1707> hi [03:06] <michaelni> hi ramiro [03:06] <michaelni> do you want a coverity account ? [11:31] <cone-120> ffmpeg.git 03Stefano Sabatini 076d6ccbae4cee: examples/decoding_encoding: add missing checks on avcodec_alloc_context3() * 03http://tinyurl.com/8sfr9n703 [11:31] <cone-120> ffmpeg.git 03Stefano Sabatini 078c4753f7f5f1: examples/decoding_encoding: remove misplaced and confusing comment * 03http://tinyurl.com/8l6ardw03 [11:31] <cone-120> ffmpeg.git 03Stefano Sabatini 077b116a94af99: examples/decoding_encoding: fix misc typos in the usage text * 03http://tinyurl.com/9ju6pws03 [12:12] <cone-120> ffmpeg.git 03Stefano Sabatini 071cd9c81ddb31: lavc/utils: extend feedback provided by avcodec_open2() * 03http://tinyurl.com/8sq24yg03 [12:12] <cone-120> ffmpeg.git 03Stefano Sabatini 077bc533c41b40: lavc/utils: fix a few case/punctuation inconsistencies in avcodec_open2() * 03http://tinyurl.com/8z2cs5c03 [12:12] <cone-120> ffmpeg.git 03Stefano Sabatini 07935ecfb00238: examples/scaling_video: remove unnecessary intermediary variable in fill_yuv_frame() * 03http://tinyurl.com/9aq6tmf03 [12:22] <cone-120> ffmpeg.git 03Stefano Sabatini 07cdea54b4c8cd: lavu/parseutils: rework rational reduction logic in av_parse_ratio() * 03http://tinyurl.com/9bs83ga03 [12:43] <cone-120> ffmpeg.git 03Michael Niedermayer 076bcdfe48d0d0: mpeg4videodec: Disable frame multithreading for GMC, its not implemented at all * 03http://tinyurl.com/9tgkg7k03 [12:43] <cone-120> ffmpeg.git 03Diego Biurrun 07c896aa984e88: build: Drop OBJS declaration for non-existing PCM_DVD encoder * 03http://tinyurl.com/9quocpn03 [12:43] <cone-120> ffmpeg.git 03Diego Biurrun 078fb1e2640505: lzo: Drop obsolete fast_memcpy reference * 03http://tinyurl.com/8nucuew03 [12:43] <cone-120> ffmpeg.git 03Diego Biurrun 074b587848ce03: configure: Disable Snow decoder and encoder by default * 03http://tinyurl.com/96m9mnq03 [12:43] <cone-120> ffmpeg.git 03Martin Storsjö 0712549db6535c: fate: Add proper dependencies for the tests in video.mak * 03http://tinyurl.com/9qdyr6303 [12:43] <cone-120> ffmpeg.git 03Hendrik Leppkes 07d2d08d706b8f: gitignore: ignore files created by msvc * 03http://tinyurl.com/8ncnbsc03 [12:43] <cone-120> ffmpeg.git 03Martin Storsjö 072f41eaa9c6b5: rtsp: Make sure the ret variable is initialized in ff_rtsp_fetch_packet * 03http://tinyurl.com/9hcvvju03 [12:43] <cone-120> ffmpeg.git 03Mans Rullgard 071846ddf0a787: ARM: fix overreads in neon h264 chroma mc * 03http://tinyurl.com/8ft4fht03 [12:43] <cone-120> ffmpeg.git 03Mans Rullgard 07c9ef43215c7d: fate-vc1: add dependencies * 03http://tinyurl.com/9zy67nl03 [12:43] <cone-120> ffmpeg.git 03Michael Niedermayer 0704c6ecb7da67: Merge commit 'c9ef43215c7d68c2cdcdbe02287aa114f27a32ed' * 03http://tinyurl.com/98b3o8m03 [13:30] <cone-120> ffmpeg.git 03Diego Biurrun 07c08536979be1: avutil/lzo: K&R formatting cosmetics * 03http://tinyurl.com/8pyva9y03 [13:30] <cone-120> ffmpeg.git 03Diego Biurrun 075532cf317838: avutil/mem: K&R formatting cosmetics * 03http://tinyurl.com/9audojc03 [13:30] <cone-120> ffmpeg.git 03Diego Biurrun 0779042ab37619: configure: Group math functions into a separate variable * 03http://tinyurl.com/966qcpy03 [13:30] <cone-120> ffmpeg.git 03Michael Niedermayer 076912e7a008ac: Merge remote-tracking branch 'qatar/master' * 03http://tinyurl.com/9cs2zuz03 [14:05] <cone-120> ffmpeg.git 03Michael Niedermayer 07bf52ad1e49d3: lavc: revert broken hunk from 1cd9c81ddb317ca00061d11d3562d3a34888d91b * 03http://tinyurl.com/9xw3wbu03 [14:41] <ubitux> <@saste> what about the issue spotted by nicolas? // i fixed it and sent a new version [17:03] <Compn> oh no! bsd fail! [17:03] <Compn> :P [19:39] <durandal_1707> ping for lvf [20:25] <durandal_1707> same_quant should be put back [20:28] Action: michaelni is wondering where cone went ... [20:29] <michaelni> j-b, is cone ok, she didnt report the last few commits ? [20:29] <JEEB> durandal_1707, even though it didn't work as it even was supposed to in the first place, and even if most damn users were misusing it? [20:29] <JEEB> it's like wtf [20:30] <JEEB> if it's put back [20:30] <JEEB> it was the most misused option ever [20:30] <JEEB> I think it even set quants to zero for a long time now instead of copying them [20:30] <durandal_1707> but there is guy, look above ffloger post, that is using it and it is useful to him [20:31] <JEEB> it's not useful for him, he just thinks it's the proper damn option [20:31] <JEEB> and I'm pretty sure he wasn't getting the same quants [20:31] <JEEB> also he could just use a low enough quant with the setting [20:35] <ubitux> didn't we add a dummy sameq/same_quant option displaying a warning? [20:35] <durandal_1707> yep [20:35] <JEEB> durandal_1707, not to mention that sameq IIRC would just give random results if the input format was different from the output one [20:36] <JEEB> I really don't see why sameq would have to be brought back to life [20:36] <durandal_1707> it just hack to be used with mpeg2? [20:36] <JEEB> ... [20:37] <JEEB> it was a hacked that used to be used for development of something or another to copy the quants of the same format input to the output. It has for all I know not worked for a relatively long amount of time and most of its users are thinking it's "same quality". [20:37] <JEEB> the first sentence already tells you why with MPEG-4 Part 2 -> MPEG-2 you shouldn't be using it [20:38] <JEEB> it was probably just setting a low enough quant there /by chance/ [20:38] <durandal_1707> omg [20:38] <JEEB> he would have better and more predictable results by just using a certain low enough quant instead [20:40] <ubitux> saste: well, i don't know how that buffering is supposed to work; i'll likely open an issue soon :( [20:40] <ubitux> (the lavfi device issue) [20:44] <ubitux> wtf some of my mails didn't reach ffmpeg-devel.. ? [20:45] <ubitux> arg again i replied in private.. [20:53] <JEEB> someone please copypasta what I said on the trac on that sameq issue or someone might actually end up putting it back >_> [20:53] <michaelni> one could fix the issues and then put it back [20:54] <JEEB> why? I understand it's a usable setting for development, but in this case it was different formats to begin with [20:54] <cbsrobot> JEEB: http://ffmpeg.org/trac/ffmpeg/wiki/Option%20%27-sameq%27%20does%20NOT%20mea… [20:54] <JEEB> he was depending on (most probably broken) utility [20:54] <JEEB> cbsrobot, and why am I getting that linked to me? [20:55] <cbsrobot> well it's already a subject on trac [20:55] <cbsrobot> we could also just point the ticket to the page [20:56] <JEEB> which doesn't mention what I noted [20:56] Action: cbsrobot reads it again [20:56] <JEEB> "...that used to be used for development of something or another to copy the quants of the same format input to the output...." [20:56] <JEEB> the guy on trac wants it for MPEG-4 Part 2 -> MPEG-2 [20:56] <JEEB> IMHO it'd be better to just set a low'ish quant or something [21:00] <cbsrobot> This option actually means "same quantizers" and should only be used to copy the quants from the input to the output of the same format (f.ex. mpeg2 -> mpeg2). [21:02] <durandal_1707> JEEB: what option should he use instead? [21:03] <JEEB> <JEEB> he would have better and more predictable results by just using a certain low enough quant instead [21:11] <saste> but we should at least warn the user [21:11] <saste> just dropping an option like this overnight is bad [21:12] <durandal_1707> saste: option is ignored [21:12] <saste> also, aren't there some proper use of the option? [21:13] <JEEB> yes, but I haven't seen it used in the last few years [21:13] <saste> copy quantization coefficients may make sense when one is transcoding+filtering and re-encoding to the same codec [21:14] <JEEB> personally I'm just happy a setting that has caused so much herp and derp is no longer there, and I don't think re-adding it for some development work would be much work if someone becomes to have a need for it during development [21:14] <JEEB> (although I've heard it didn't work in the end for a relatively long time, just setting some low'ish quants) [21:15] <durandal_1707> transcoders are evil [21:17] <saste> fb722a900fc5cc9e003b9fef25b27ed7fc5547a2 [21:17] <saste> i can agree on the rationale, but breaking API should be avoided [21:17] <saste> anyway maybe it should *fail* rather than print a warning [21:18] <durandal_1707> saste: than revert michaelni commit [21:20] <saste> ahah it was even mentioned in the FAQ [21:20] <saste> no wonder people were misusing it [21:20] <saste> i misused it as well [21:21] <durandal_1707> wait.... [21:21] <saste> that's because people looks for a magic option - keep same quality [21:21] <Compn> should just make ffmpeg -magic .... [21:22] <saste> having to set a quantization parameter is not that user-friendly (you have to get a clue about what the quantizer is in the first place) [21:22] <saste> Compn: ffmpeg -dwim [21:23] <Compn> ffmpeg -thatfilterfromavisynth-daemonkeepstalkingabout [21:23] <michaelni> ffmpeg `cat /dev/brain` [21:23] Action: Daemon404 has been hilighted [21:23] <Daemon404> michaelni, do you happen to know sort of input the mpeg2video decoder expects? [21:24] <Daemon404> im passing it a gop offset in a mpeg2 PS [21:24] <Daemon404> and its... -sorta- working [21:24] <Daemon404> [15:21] <@Daemon404> http://i.imgur.com/20jJm.png [21:24] <Daemon404> [15:21] <@Daemon404> ^ output luma [21:24] <Daemon404> [15:21] <@Daemon404> doesnt look quite right ;) [21:24] <Daemon404> e.g. i i know teh offset of a GOP in an mpeg2 ps file, and i read in from that offset, and pass to avcodec [21:24] <saste> but the thing is that -sameq sometimes seemed to work [21:25] <saste> at least it was better that no option at all [21:25] <saste> so the question is, why was it working? [21:25] <michaelni> Daemon404, the decoder expects MPEG-ES video frames [21:25] <Daemon404> michaelni, im not -that- fimiliar with mpeg-ps [21:25] <Daemon404> but should a gop offset point to mpeg-es frams? [21:25] <Daemon404> shoudlnt* [21:26] <michaelni> what is a "gop offset" ? [21:26] <Daemon404> sec [21:26] <Daemon404> http://neuron2.net/dgmpgdec/DGIndexManual.html#AppendixA [21:26] <Daemon404> the "position" marker [21:27] <Daemon404> "The position field is a decimal number that points to the byte position in the source file at which decoding of the indexed I picture should begin." [21:27] <saste> ugh i was wrong, there was no API break (no lavc option was removed) [21:28] <saste> michaelni, what about to fail rather than printing a warning in case of -sameq? [21:28] <michaelni> Daemon404, a I-Picture can be split in many lets call them chunks at the PS level [21:29] <michaelni> or more precissely, MPEG-ES frames get concatenated and then randomly choped in a series of bytes [21:29] <michaelni> then headers added and interleaved with audio then more crap added and you have MPEG-PS [21:29] <saste> and direct the user towards some option which may achieve the same effect [21:30] <Daemon404> i see [21:30] <michaelni> saste, theres no option that achives the same effect currently [21:32] <saste> michaelni, mention something about quality options? [21:32] <saste> there are so many of them [21:32] Action: saste feels like we need a proper manual for the lavc options [21:33] <michaelni> the most fitting would probably to mention that the user should select a bitrate similar to the input [21:33] <michaelni> but if people want this option so much we should think about fixing it and putting it back [21:33] <Daemon404> michaelni, is there anything tangible i can do with libavcodec given these "gop offsets" [21:33] <michaelni> convert the quality between formats and all that [21:34] <michaelni> (if i understand correctly what they are) you can use them as seed for our random number generator [21:34] <Daemon404> <.< [21:35] <saste> michaelni, I have no clue about how -sameq was working, but it was quite hackish [21:35] <Daemon404> it wasnt working [21:35] <saste> it was *somehow* working in some cases [21:35] Action: Daemon404 thinks it was a coincidence [21:36] <michaelni> the problem with sameq was that it was written 10 years ago and not maintained since then [21:36] <michaelni> or something like that :) [21:36] <JEEB> yes [21:36] <JEEB> I hear it had been broken for a /long/ time [21:36] <JEEB> and just used some zero/low quant [21:39] <Daemon404> michaelni, so given those offsets, is there a way to get it into an mpeg-es format easily? [21:39] <saste> JEEB: allright, it was copying the quality param of the input decoded frame, and putting it in the encoded frame [21:40] <saste> that info was possibly used by the encoder [21:40] <michaelni> Daemon404, if you have a MPEG-PS file you need to run it through a demuxer that produces ES frames [21:40] <michaelni> or a ES stream which you can feed through a AVParser [21:40] <saste> from avcodec.h: quality (between 1 (good) and FF_LAMBDA_MAX (bad)) [21:41] <Daemon404> michaelni, i may only access the ps at those offsets [21:41] <Daemon404> and start reading from there [21:41] <Daemon404> i dont know if avparser can handle that [21:41] <Daemon404> i.e. i am NOT using libavformat [21:41] <Daemon404> to open the ps [21:41] <nevcairiel> if there is PS headers in the middle of the coded frames, the parser wont remove those [21:41] <saste> FF_LAMBDA_MAX (256*128-1) defined in avutil.h [21:41] <nevcairiel> you may need to write some magic to do that [21:42] <Daemon404> nevcairiel, its certainly possible [21:42] <Daemon404> every d2v related filter in existence has done it [21:42] <saste> in general it seems like something related to some specific codec, so I agree it doesn't make much sense to rely on that option [21:42] <saste> i suppose it was working more or less when transcoding from MPEG-family-codec to MPEG-family-codec [21:42] <nevcairiel> its not like PS parsing is that complicated [21:43] <Daemon404> nevcairiel, yeah but i have no idea how? [21:43] <nevcairiel> i never looked at PS much. i could tell you how to deal with TS :) [21:44] <Daemon404> ;p [21:44] <Daemon404> im am readign dgdecode's source [21:44] <Daemon404> pity me [21:44] <michaelni> Daemon404, if it is PS you could try to feed it through libavformats PS demuxer [21:45] <Daemon404> mmm maybe [21:45] <Daemon404> this code is lol... [21:45] <Daemon404> // If the D2V filename has _P just before the extension, force [21:45] <Daemon404> // progressive upsampling. [21:45] <Daemon404> surely this cannot be a bad idea... [21:50] <Daemon404> heh.. can lavf even open a buffer [21:50] <Daemon404> or does it only take filenames... [21:50] <nevcairiel> you can feed it buffers, but need to give it a custom avio context [21:51] <Daemon404> what func? [22:04] <Daemon404> wtf is the ps demuxer called [22:04] <Daemon404> there isnt a mpegps [22:05] <michaelni> its called "mpeg" [22:06] <michaelni> ff_mpegps_demuxer ... name= "mpeg" [22:07] <Daemon404> i see this now... [22:11] <durandal_1707> is mkv streaming actually supported? [22:18] <ubitux> mpf. i need to introduce another side data thing [22:18] <ubitux> (codec srt -> codec subrip) [22:19] <michaelni> ubitux, btw make sure all the metadata stuff gets fate tests so it is all regularly tested [22:20] <ubitux> mmh [22:20] <michaelni> could easily break with noone noticing otherwise [22:20] <ubitux> maybe i should add a ffprobe test indeed [22:20] <Daemon404> aha.. works! [22:21] <ubitux> michaelni: do you have a video fate sample in mind with 2-3 scene changes? [22:22] <michaelni> no but one could run a scene detect filter over all the samples to find a good candidate [22:22] <michaelni> small file low resolution is probably preferable [22:22] <michaelni> so its faster [22:22] <ubitux> yep [22:23] <ubitux> i need to find a sample with silence too [22:28] <ubitux> michaelni: any other comment btw? [22:29] <michaelni> ubitux, didnt really look at the latest patch yet [22:29] <michaelni> just wanted to point that fate thing out ... [22:30] <ubitux> yes, good point :p [22:48] <ubitux> meh side data doesn't get transmitted [22:51] <ubitux> (with subtitles) [22:57] <ubitux> oh it's because ffmpeg isn't reusing the decoded subtitle& [23:03] <ubitux> rha the subtitles api really is completely out of time :D [23:15] <ubitux> seems related to AVFMT_FLAG_KEEP_SIDE_DATA [23:15] <ubitux> what's this side data merge thing? [23:20] <ubitux> michaelni: av_packet_merge_side_data() in lavf/utils.c:ff_read_packet is trashing the side data set in the subtitle demuxer [23:20] <ubitux> any idea why? [23:22] <michaelni> no [23:23] <ubitux> :( [23:30] <durandal_1707> j [23:30] <ubitux> oh ok the function is moving the side data at the end of the pkt data... huh. [23:31] Action: ubitux doesn't get the point [23:32] Action: ubitux doesn't get how that's supposed to work [23:33] <durandal_1707> is there nice way to read null terminated string from avio? [23:33] <michaelni> ubitux, the point is that older user apps do not preserve side data but just the main data between lavf and lavc [23:34] <michaelni> its easy to disable the merging if its unwanted [23:34] <ubitux> yes with keepside option [23:34] <ubitux> but& [23:34] <ubitux> how is the side data supposed to be restored? [23:34] <michaelni> magic :) [23:35] <ubitux> oh indeed the magic thing [23:35] <ubitux> FF_MERGE_MARKER? :p [23:35] <michaelni> av_packet_split_side_data() [23:36] <ubitux> yeah ok [23:36] <ubitux> so i'm supposed to get the data back [23:37] <ubitux> and that function is called only for audio & video, ok, get it. [23:37] <ubitux> thank you :) [23:37] <michaelni> ahh uhm, i forgot adding that for subtitles ? ^^;; [23:37] <ubitux> seems so :p [23:43] <durandal_1707> i forgot name of function that reads string up to NULL or X chars whatever comes first [23:44] <ubitux> something like bprint + a avio_r8() loop? [23:45] <ubitux> ff_get_line maybe [23:45] <ubitux> libavformat/aviobuf.c:int ff_get_line(AVIOContext *s, char *buf, int maxlen) [23:46] <ubitux> michaelni: could we simply disable it for subtitle btw? [23:47] <ubitux> since no subtitle use side data at the moment, the "old user app" should not need it, right? [23:47] <ubitux> (assuming the subtitles side data are always optional) [23:47] <ubitux> do you plan to keep the split side data thing indefinitely btw? [23:48] <durandal_1707> ubitux: actually i need it in lavc [23:48] <ubitux> durandal_1707: i'd be interested in a bprint version of ff_get_line btw [23:48] <ubitux> too bad :p [23:48] <ubitux> want a bytestream_get_line? :) [23:49] <durandal_1707> avio_get_str(utf8) variant for lavc [23:49] <j-b> good morning [23:50] <durandal_1707> j-b: good midnight [23:50] <michaelni> ubitux, we could but that would be inconsistent [23:54] <durandal_1707> i will use strnlen [00:00] --- Sun Oct 21 2012

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ffmpeg.log.20121020
by burek 21 Oct '12

21 Oct '12

[00:07] <durandal_1707> needs_more_info [00:07] <durandal_1707> does mkv file with aac have same duration ? [00:07] <Endorgh> yes [00:08] <Endorgh> mkv containing aac_he_v2 has the same duration as the original input file [00:09] <durandal_1707> what you use to play mp4? [00:11] <Endorgh> deadbeef [00:12] <durandal_1707> does ffmpeg -i input.mp4 show correct duration? [00:17] <Endorgh> durandal_1707: yes, it shows correct duration [00:21] <durandal_1707> so it appears to be deadbeef problem and how it calculates duration [00:22] <durandal_1707> how are deadbeef libav* plugins build? [00:22] <durandal_1707> perhaps it uses older versions of library.... [00:24] <Endorgh> may be, I'll investigate about it... [00:25] <Endorgh> otherwise, when I convert flac to aac_he_v2 into mp4, deadbeef, aqualung and VLC are having problems playing it [00:26] <Endorgh> coding aac_he into mp4 works well on all of them, but not aac_he_v2 [00:26] <Endorgh> its confusing........ :S [00:29] <durandal_1707> does aac_he_v2 decodes correctly with ffmpeg? [00:29] <durandal_1707> (it may be that such profile is just broken) [00:32] <Endorgh> yes, perfectly. If I convert from mp4 to mp3, it works perfect... [00:32] <Endorgh> in any case, How can I check if 'aac_he_v2' profile is broken? [00:35] <durandal_1707> hmm, dunno perhaps deadbeef use broken aac decoder [00:35] <Endorgh> I understand [00:37] <Endorgh> in last place and in order to discard any possible error made by myself, this command is 100% correct? [00:38] <Endorgh> ffmpeg1 -i input.flac -acodec libfdk_aac -profile:a aac_he_v2 -ab 60k -ar 44100 -ac 2 output.mp4 [00:39] <Endorgh> it should give me an mp4 file containing high quality AAC-HE v2 audio? [00:43] <durandal_1707> if you play it with quicktime and it plays correctly then yes [00:46] <DelphiWorld> yo saste [00:46] <saste> hi DelphiWorld [00:46] <DelphiWorld> saste: do you know Sast-NG? [00:46] <saste> saste.-ng? [00:46] <Endorgh> durandal_1707: ok, I'll try it! [00:46] <Endorgh> thanks!! [00:46] <saste> an improved version of saste? [00:47] <DelphiWorld> saste: first of all what is sast ? [00:47] <saste> i have no idea [00:48] <DelphiWorld> saste: saste-ng is a softcam ;) [00:48] <DelphiWorld> saste: when i see you i remember it;-) [00:50] <saste> can't find it on the internet [00:50] <DelphiWorld> saste: please kill me:( [00:50] <DelphiWorld> saste: my assness [00:50] <DelphiWorld> saste: it's sasc:P [00:52] <DelphiWorld> nighty [12:14] <hendry> hi there, when I convert my OGV videos from recordmydesktop like so: ffmpeg -loglevel quiet -i out.ogv youtube-upload.avi [12:14] <hendry> the avi files become "artefacted" and not as clear as the original out.ogv [12:14] <hendry> it's like converting a PNG to JPG [12:15] <hendry> any tips to avoid this lossy conversion, yet still be able to upload to Youtube ? [12:15] <sacarasc> Capture your desktop using ffmpeg. [12:15] <Tjoppen> why? just upload the original file [12:15] <sacarasc> That's what recordmydesktop uses... [12:17] <hendry> sacarasc: is there a guide to doing that somewhere? [12:18] <hendry> Tjoppen: the orginal file that recordmydesktop produces: out.ogv, does not upload to Youtube. [12:19] <hendry> sacarasc: for e.g. `ffmpeg -f x11grab -s wxga -r 25 -i :0.0 -sameq /tmp/out.mpg` doesn't record audio [12:20] <Tjoppen> that's strange, because youtube uses ffmpge [12:20] <Tjoppen> also: don't use avi [12:20] <Tjoppen> ffmpeg -i out.ogv -vcodec libx264 -crf 20 -acodec copy out.mov or something like that [12:23] <hendry> Tjoppen: why not .avi ? [12:39] <hendry> Tjoppen: btw sound is not working in the out.mov [13:23] <RoyK> are there any support for opencl/cuda in ffmpeg for h.264 or web encoding? [13:27] <JEEB> RoyK, there is a patch that's under review currently, which moves lookahead me to opencl in x264 IIRC [13:27] <JEEB> but [13:27] <JEEB> it's the usual case of "do I really want to lower the quality and get a 10% speed boost with a high-end GPU instead of just investing on a new CPU" [13:29] <RoyK> it§s just that a colleague of mine told me he got a 4x speed gain by using the gpu on his machine with some proprietary software [13:29] <RoyK> s/it§s/it's/ [13:29] <JEEB> first of all, did he properly test or just see the speed with random settings? [13:30] <JEEB> because x264 can in general be very fast f.ex. [13:31] <JEEB> also I'm not even sure how fast those GPU-based encoders are nowadays, but they mostly suck because GPUs need you to be able to throw a lot of stuff at the same time, aka the workload really has to be well multithreadable. Unless you strip features out of H.264 to its barebones you're not going to get that simply on the GPU [13:32] <JEEB> when you have something using GPU for certain tasks, that /can/ be faster, but not always without a cost and the actual CPU algorithm might've not been fast to begin with [13:33] <hendry> anyone know how i get the playing length of an AVI file? [13:33] <JEEB> the integrated encoder chips in intel stuff f.ex. are really dedicated hardware for encoding and the results of those generally are quite a bit better than pure GPU encoders' [13:34] <JEEB> but those too are mostly currently riding on the side of "we're fast and we're not using the 'CPU'", and the speed might be attainable with similar/better quality from x264 with a fast preset [13:36] <JEEB> anyways, OpenCL/CUDA is not a silver bullet under any circuimstances. CUVID decoding can help on newer hardware stuff to take some stuff off of the CPU, but if you go fast enough on the encoder's side that might end up being your bottleneck since the hw decoder can only output as many frames per second [16:16] <tomreyn> hi, i'm trying to analyze the audio files we have in MegaGlest (an open source RTS game, http://megaglest.org ), to get an idea whether playing them back is causing elevated CPU load (memory matters, too, but less so), and to find out ways to reduce the overhead they cause without decreasing the audible quality more than a little bit. [16:17] <tomreyn> here's what i find, including high (?) bitrates of 1411 kbps: http://paste.ubuntu.com/1292282/ [16:17] <tomreyn> i'm looking into this because profiing the game turns out that a rather high percentage (up to 8%) of the processing power of the test system goes into audio processing. [16:18] <tomreyn> now any suggestions on what to look into would be welcome, i'm pretty helpless with audio and audio processing. [16:22] <ubitux> maybe ./ffmpeg -nostats -i input.wav -benchmark_all -f null - ? [19:57] <Element9> while trying to configure ffmpeg for compiling, I got "libtheora not found". I tried with ffmpeg 1.0 package and the latest code from git. is this a known problem could it be that I am doing something wrong? [19:58] <durandal_1707> Element9: paste relevant config.log somewhere [20:02] <hendry> i'm trying to grab to a mov, but [20:02] <hendry> ffmpeg -f x11grab -s 1366x768 -i :0.0 -vcodec libx264 -crf 20 -acodec libfaac -f alsa -i hw:0,0 out.mov # results in Unknown decoder 'libx264' [20:02] <hendry> any ideas? [20:03] <JEEB> before -i = settings for decoders/inputs [20:03] <JEEB> so in other words exactly what it told you [20:03] <JEEB> there is no such decoder as libx264 (and no libfaac decoder either) [20:03] <JEEB> probably because of the fact that you have two -is [20:04] <JEEB> also I would recommend fdk instead of faac if you want aac [20:04] <JEEB> https://github.com/mstorsjo/fdk-aac [20:06] <hendry> JEEB: `ffmpeg -f x11grab -s 1366x768 -i :0.0 -vcodec libx264 -crf 20 -acodec libfaac out.mov` works, so libx264,libfaac do exist [20:06] <hendry> or must exist [20:07] <hendry> it's just that cmdline has no sound [20:07] <hendry> so it's useless to me :) [20:08] <klaxa> well obviously since you don't specify an audio input [20:08] <JEEB> hendry, did you like ignore all of what I said? [20:09] <JEEB> re-read my lines [20:09] <JEEB> also encoders are not decoders [20:09] Action: JEEB sighs [20:09] <klaxa> kinda spoonfeeding here, but try: ffmpeg -f x11grab -s 1366x768 -i :0.0 -vcodec libx264 -crf 20 -f alsa -i hw:0,0 -acodec libfaac out.mov [20:10] <klaxa> and the fdk stuff... no idea about that though [20:12] <JEEB> faac is actually pretty bad, but I guess building yet another binary is not an alternative to this guy :P [20:12] <hendry> klaxa: that didn't work either [20:13] <hendry> you make spoonfeeding sound like a crime [20:13] <klaxa> opus looks good, i think it misses a lot of container implementations though... as a standalone audio-codec it has a great quality/bitrate ratio [20:13] <hendry> this stuff is really rather tricky ffs [20:13] <klaxa> did you google? did you look at the manpage? [20:13] <hendry> klaxa: yup ! [20:13] <JEEB> more like, put those -is before you start setting up the encoder/output side settings [20:13] <klaxa> what did you google? also, if you say it doesn't work, that doesn't help [20:14] <JEEB> otherwise ffmpeg will parse those things before an -i as input/decoder settings [20:14] <JEEB> as I said [20:14] <klaxa> complete ffmpeg log would help [20:14] <JEEB> <hendry> this stuff is really rather tricky ffs <- yes it is if you have never done it before, but you have gladly ignored what I told you. [20:15] <klaxa> hendry: sorry if we sound insulting, but it's actually kinda tiring to help if people ignore you [20:15] <hendry> https://gist.github.com/3924250 is the log [20:15] <hendry> JEEB: IIUC you want me to group -i together [20:16] <hendry> A) i tried that, I get hw:0,0: Protocol not found [20:16] <klaxa> are you running pulse by any chance? that would make things easier (less efficient though) [20:16] <klaxa> oh wait [20:16] <hendry> B) this is not a problem when I use huffyuv [20:16] <klaxa> you are missing -f alsa [20:16] <klaxa> i think? [20:16] <klaxa> -f also before -i hw:0,0 [20:16] <klaxa> *alsa [20:16] <JEEB> and after the video -i I would guess [20:16] <klaxa> you didn't run my command after all [20:17] <klaxa> oh no you did [20:17] <klaxa> however... >Unknown decoder 'libx264' [20:17] <JEEB> hendry, I would be surprised if it wasn't a problem in case of ffvhuff or huffyuv :) Because the inputs should be the same [20:17] Action: hendry wrote https://github.com/kaihendry/recordmydesktop2.0/blob/master/r2d2.sh#L24 today, trying to make ffmpeg more useful [20:18] <JEEB> ugh sameq [20:18] <JEEB> remove that [20:18] <JEEB> first of all it doesn't make sense with ffvhuff or huffyuv [20:18] <JEEB> because they are lossless [20:18] <JEEB> second of all I hear sameq has been broken for years and didn't mean what you think it does [20:18] <hendry> my experience of huffyuv is that they are lossy [20:19] <hendry> since the picture / colours are off [20:19] <JEEB> they are only lossy in the sense that there's a possible RGB->YCbCr conversion going on [20:19] <hendry> e.g. http://r2d2.webconverger.org/2012-10-21/ [20:19] <JEEB> if you need rgb, you take on a codec that can do RGB [20:19] <JEEB> I think ffvhuff might be able to do that, or maybe utvideo [20:19] <JEEB> pick your choice off the lossless codecs [20:19] <hendry> JEEB: well, I'm all ears. I don't want any loss ideally. I want it to look exactly like the screen capture [20:20] <hendry> and i want it to work across browsers :-) [20:20] Action: hendry tries ffvhuff [20:20] <JEEB> utvideo at least does support RGB [20:20] <JEEB> I coded that implementation ^^; [20:20] <Element9> durandal_1707: sorry, got carried away, trying to fix it myself. :) here's the log: http://pastebin.com/vjKs0W0p [20:20] <klaxa> JEEB: where would i get a list of video codecs with their respective color space? [20:21] <JEEB> <hendry> and i want it to work across browsers :-) <- you're not telling me that you want the in-the-middle capture to be playable with browsers, right? [20:21] <JEEB> klaxa, I'm not sure if it's listed anywhere easy :s [20:21] <klaxa> :S [20:21] <JEEB> also there could be defaults [20:21] <klaxa> utvideo is lossless? [20:21] <JEEB> f.ex. libx264 is defaulted to 4:2:0 [20:21] <JEEB> yes [20:21] <JEEB> it's similar to huffyuv/ffvhuff [20:21] <klaxa> ah [20:22] <durandal_1707> Element9: have you installed libtheora? [20:22] <JEEB> just has implementations on all major OS [20:22] <JEEB> VFW/DS for Windows [20:22] <JEEB> QT thingy for Macs [20:22] <JEEB> and now libavcodec dec/enc as cross-platform [20:22] <hendry> JEEB: I don't understand what you mean by "in-the-middle capture" [20:22] <JEEB> hendry, you don't usually throw the lossless file you capture onto the internet as-is [20:23] <JEEB> because if you do capture with one, it's usually quick with enc/dec and not exactly small [20:23] <Element9> durandal_1707: yes [20:23] <JEEB> you then usually convert those to whatever you need in the end [20:24] <JEEB> and in most cases you need to convert to 4:2:0 YCbCr (colloquially called "YUV") for web video [20:24] <JEEB> so if you don't want any change of color in the end result, that's a "tough luck" kind of thing, but the change should be minimal if you match RGB->YCbCr and YCbCr->RGB conversion matrices [20:25] <Element9> durandal_1707: I have there: codec.h theoradec.h theoraenc.h theora.h [20:25] <Element9> in /usr/include/theora [20:25] <Element9> s/there/these [20:25] <JEEB> of course 4:2:0 YCbCr means it will have chroma subsampling, but not like anything else is really supported in browsers (Except for Chrome or Flash) [20:25] <JEEB> Flash uses Mainconcept's decoder and thus can do up to 4:2:2 YCbCr (in both 8bit and 10bit), and unless Chrome removes features from their H.264 decoder they should be able to deal up to 4:4:4, or possibly RGB [20:26] <hendry> JEEB: just tried both utvideo & ffvhuff. I think ffvhuff.webm is better, but still the colours are off [20:26] <Element9> I tried configuring without theora and then it says libvorbis is missing. hmmm... [20:27] <JEEB> > ffvhuff.webm > WhatAmIReading [20:27] <JEEB> anyways, quality-wise they should be the same but the utvideo encoder is not threaded so prolly slower [20:27] <JEEB> I should really get to that one day :P [20:27] <JEEB> hendry, check your log btw [20:27] <JEEB> or gist it [20:28] <JEEB> that will tell you what possible colorspace conversions were made [20:28] <durandal_1707> Element9: you will probably need to add extra flags: --extra-cflags=-I/usr/include --extra-ldflags=-L/usr/lib [20:28] <JEEB> at least I would think that with utvideo there should be none, as I've supported all of the usual ones (RGB24, RGBA, 4:2:2/4:2:0 YCbCr) [20:29] <hendry> JEEB: https://gist.github.com/3924250 [20:30] <Element9> durandal_1707: lets try that [20:31] <klaxa> >-t is not an input option, keeping it for the next output; consider fixing your command line. [20:31] <klaxa> >Stream #0:0: Video: vp8, yuv420p, 1366x768, q=-1--1, 200 kb/s, 1k tbn, 29.97 tbc [20:31] <klaxa> looks like it gets converted to 4:2:0 YCbCr [20:32] <JEEB> yes [20:32] <JEEB> Stream #0:0 -> #0:0 (rawvideo -> libvpx) [20:32] <JEEB> and instead of ffvhuff you get vp8 [20:32] <JEEB> probably because it's webm [20:32] <JEEB> but this is weird [20:32] <klaxa> and vorbis instead of pcm_16se [20:32] <JEEB> it shouldn't just override like that [20:33] <hendry> http://r2d2.webconverger.org/2012-10-21/utvideo.webm.html versus http://r2d2.webconverger.org/2012-10-21/ffvhuff.webm.html [20:33] <klaxa> *pcm_s16le [20:33] <JEEB> it should instead kick you for using a container that couldn't take your selection [20:34] <JEEB> hendry, you could report this as a bug I guess? Because instead of herping a derp at you it auto-sets things to be compliant for webm. Instead you probably would have wanted to get an error that would tell you that those things don't go into webm in case you /explicitly/ set thme [20:34] <JEEB> *them [20:34] <Element9> durandal_1707: now I get "Symbol mangling check failed." [20:35] <durandal_1707> what OS is that? [20:36] <hendry> JEEB: where should I file a bug? [20:36] <Element9> durandal_1707: xubuntu and cross compiling for windows [20:36] <durandal_1707> Element9: not my experize [20:37] <Element9> durandal_1707: oh, I think I saw a flag for cross compiling. I'll try to set that one [20:37] <Element9> durandal_1707: I can always try to comment out that on check. maybe it goes well :) [20:37] <Element9> durandal_1707: thanks for the help [20:38] <JEEB> hendry, trac I guess [20:40] <JEEB> hendry, also in general I would really say that you shouldn't be encoding straight into something that you give out as-is on the web, because you won't be able to compress it as well when it also has to be realtime [20:40] <hendry> JEEB: tbh most of this stuff if over my head [20:41] <hendry> JEEB: so if you could spoon feed me what I should be doing, I'd be extremely grateful [20:41] <hendry> JEEB: I'll add it to https://github.com/kaihendry/recordmydesktop2.0/blob/master/r2d2.sh#L24 [20:43] <klaxa> hendry: record lossless to a temporary raw, then re-encode for publishing on the web [20:43] <hendry> klaxa: do you have a step-by-step example please? [20:44] <JEEB> hendry, I'm not going to be copying your long line, but look at this, more or less, ffmpeg <inputs> -vcodec ffvhuff -acodec pcm_s16le out.avi [20:44] <JEEB> <inputs> meaning that mumbo-jumbo on your live input devices etc. [20:44] <klaxa> hendry: i write something up for you [20:46] <hendry> JEEB: my mumbo jumbo is pretty standard? x11grab & alsa for sound [20:46] <hendry> klaxa: that would be much appreciated [20:46] <JEEB> hendry, I'm just not well versed in that but it looked more or less correct [20:46] <JEEB> but you should understand how my example would build around your inputs :P [20:47] <klaxa> JEEB: i'll assume utvideo is very well suited for lossless screencasts? [20:48] <hendry> added -report logs to http://r2d2.webconverger.org/2012-10-21/ if you are interested [20:48] <hendry> I can do this in one step with my script like so: ./r2d2.sh -d 5 -u r2d2.webconverger.org:/srv/www/r2d2.webconverger.org -c ffvhuff ffvhuff2.webm [20:48] <JEEB> as a format it's nice enough, supports RGB as well as various YCbCr formats, but it might not be fast enough, but I'm not sure if the native library encoder that Daemon404 wrote is any faster :) [20:49] <klaxa> libx264 ultrafast lossless it is? [20:49] <hendry> JEEB: I'm OK with slow encoding speed as long as results look like my desktop I'm trying to capture. [20:49] <hendry> http://r2d2.webconverger.org/2012-10-21/utvideo2.webm.html is failing the "look like my desktop I'm trying to capture" test [20:49] <hendry> and so is http://r2d2.webconverger.org/2012-10-21/ffvhuff2.webm.html [20:50] <JEEB> ... [20:50] <JEEB> did you actually read ANYTHING I wrote? [20:50] <Mavrik> O.o [20:50] <JEEB> and did you not read why I told you reporting that one thing might be a good idea AT ALL? [20:50] <JEEB> if you set your output file name to be something dot webm [20:50] <JEEB> it will auto-set stuff to be compatible with it [20:50] <JEEB> that is not ut video and that is not ffvhuff [20:51] <hendry> JEEB: yes, I do read but i might not understand [20:51] <JEEB> real ffvhuff or ut video DO NOT DECODE IN YOUR BROWSERS [20:51] <hendry> JEEB: I understand you want me to report a bug about something I understand very little [20:51] <durandal_1707> webm is vp8 + vorbis [20:51] <JEEB> there's a reason why I set the output file name as out.avi there [20:51] <JEEB> meh [20:52] <JEEB> durandal_1707, I think everyone by now knows that, but the funny thing is that instead of erroring out on incorrect video/audio codecs (that are set explicitly) it will just switch them over to compatible ones :P [20:53] <durandal_1707> gonna fix it asap [20:53] <hendry> ok, I understand `ffmpeg <inputs> -vcodec ffvhuff -acodec pcm_s16le out.avi`... and that's it? maybe i'll just shut up now and wait for klaxa [20:54] <durandal_1707> hendry: what ffmpeg version you are using? [20:55] <klaxa> JEEB: what audio codec would you suggest for lossy audio? [20:55] <JEEB> hendry, yes? [20:55] <hendry> durandal_1707: ffmpeg version 1.0 Copyright (c) 2000-2012 the FFmpeg developers built on Sep 29 2012 11:22:50 with gcc 4.7.1 (GCC) 20120721 (prerelease) [20:55] <hendry> durandal_1707: http://r2d2.webconverger.org/2012-10-21/ffmpeg-20121021-024453.log may help [20:56] <hendry> durandal_1707: i'm using Archlinux [20:56] <durandal_1707> hendry: latest ffmpeg disallows muxing other codecs for webm [20:58] <JEEB> klaxa, depends on what your intended use case is [20:58] <klaxa> hendry: what is your intended use case? [20:58] <hendry> durandal_1707: so I'm not using the latest IIUC [20:58] <hendry> klaxa: recording my desktop [20:58] <klaxa> will you record your soundcards audio or your voice or both? [20:59] <hendry> klaxa: ideally to container that works with HTML5 video across browsers. I don't mind 2/3 copies if it means getting it working on IOS6 safari etc [20:59] <klaxa> i see [20:59] <hendry> klaxa: both I guess [20:59] <durandal_1707> hendry: this is strange 1.0 should not allow it either [20:59] <hendry> klaxa: "-acodec pcm_s16le -f alsa -i hw:0,0" has been fine for me [21:00] <klaxa> well yeah that's raw pcm [21:00] <klaxa> you don't want that to be sent over the internet [21:00] <hendry> klaxa: ok, sure, whatever is the appropriate codec for voice really [21:00] <klaxa> i think with webm you're pretty much bound to vorbis and vp8 [21:01] <hendry> klaxa: fine, but Safari IOS6 I assume is a different beast altogether [21:02] <klaxa> anyone know the variables for libvx? i for one don't, i don't use libvx [21:02] <klaxa> i.e. profile, preset, good crf or qp values [21:03] <JEEB> libvpx? [21:03] <JEEB> unfortunately it has no such easy things like the three first ones you mentioned :P [21:03] <JEEB> it does have constant quant'ish thing [21:03] <JEEB> but nothing like crf [21:08] <klaxa> D: [21:08] <klaxa> do the default values suffice for publishing on the web? [21:08] <JEEB> it has some vpre files for certain stuff, but no idea how good those in reality are [21:08] <klaxa> well.. hendry will find out for us :) [21:08] <JEEB> you are most probably best off setting a constant quant or a bit rate with libvpx [21:10] <klaxa> https://gist.github.com/7dcccbd86fdcce3c4ced [21:10] <klaxa> hendry: ^ [21:11] <klaxa> ah fixed small thing, second step: raw.avi -> raw.mkv [21:12] <hendry> klaxa: https://gist.github.com/3924250 [21:12] <hendry> klaxa: first step worked, second didn't [21:12] <klaxa> <klaxa> ah fixed small thing, second step: raw.avi -> raw.mkv [21:13] <klaxa> reload the paste [21:13] <klaxa> wait what.. [21:13] <hendry> -strict -2 # didn't work either as the output suggests [21:14] <klaxa> there is no audio in the .mkv [21:14] <hendry> klaxa: when i `mplayer raw.mkv` i hear audio [21:15] <klaxa> wat... no there is audio in the mkv [21:15] <klaxa> but ffmpeg doesn't... someone... why doesn't vorbis work? [21:15] <klaxa> that's beyond my knowledge... try specifying bitrate with -ab 192k or something after -acodec vorbis [21:16] <durandal_1707> hendry: do not use internal experimental aac encoder (it have limited quality) [21:16] <durandal_1707> neither internal vorbis encoder (which is just not for users) [21:17] <durandal_1707> just do not use experimental things [21:17] <hendry> durandal_1707: can you suggest edits to https://gist.github.com/7dcccbd86fdcce3c4ced please? [21:18] <klaxa> yeah i'm actually a novice myself [21:18] <klaxa> i feel like i should know a lot more to be able to give advice at all [21:19] <hendry> i've got to go to bed since it's 3AM here, i'll be connected though and I'm desperately looking to improve https://github.com/kaihendry/recordmydesktop2.0 [21:21] <durandal_1707> hendry: use copy for last -acodec [21:23] <hendry> durandal_1707: https://gist.github.com/3924250 [21:23] <hendry> i guess thre is a problem with klaxa's https://gist.github.com/7dcccbd86fdcce3c4ced step 1 [21:24] <durandal_1707> hendry: what are you doing? [21:24] <klaxa> >[webm @ 0x1c2a6e0] Only VP8 video and Vorbis audio are supported for WebM. [21:24] <klaxa> i think vorbis is mandatory for webm, no? [21:25] <hendry> ffmpeg -f x11grab -s 1366x768 -r 30 -i :0.0 -f alsa -i hw:0,0 -acodec pcm_s16le -vcodec ffvhuff -acodec vorbis raw.mkv # doesn't work: Codec is experimental but experimental codecs are not enabled .... [21:25] <klaxa> wait... don't use -acodec twice [21:25] <hendry> durandal_1707: trying to record my desktop [21:25] <klaxa> remove the -acodec vorbis [21:25] <klaxa> that's for the reencode to webm [21:25] <klaxa> the raw is supposed to be lossless anyways [21:26] <hendry> klaxa: oops, no different if i remove 2nd -acodec vorbis [21:26] <hendry> so it looks like we are stuck here or is there some intermediate stage ? [21:27] <durandal_1707> klaxa: mkv can have anything for audio [21:27] <durandal_1707> i mean hendry [21:27] <klaxa> yeah the initial raw should have ffvhuff video and pcm_s16le audio [21:27] <klaxa> then re-encode with vp8 and vorbis for webm [21:28] <mark4o> should use -acodec libvorbis not -acodec vorbis [21:28] <durandal_1707> hendry: you cant use webm for this [21:28] <durandal_1707> pcm_s16le is overkill, it can take lot of space, use flac [21:29] <amendes365> good morning/afternoon/evening everyone, im dont know why ffmpeg shows 'Unknown AudioCodec: aac' even when [ffserver -formats | grep aac] recognizes it [21:30] <durandal_1707> amendes365: you want to stream aac? [21:31] <klaxa> ah right... libvorbis, will flac be encoded fast enough? (it's audio so i guess so?) [21:31] <amendes365> durandal_1707: im trying to stream H.264 video with AAC audio in MPEGTS [21:32] <durandal_1707> klaxa: if disk space is small and disk is slow but CPU is reasonaly faste use flac [21:32] <amendes365> following one of the tutorials in ffmpeg.org [21:32] <hendry> ffmpeg -f x11grab -s 1366x768 -r 30 -i :0.0 -f alsa -i hw:0,0 -acodec libvorbis -vcodec ffvhuff raw.mkv && ffmpeg -i raw.mkv -acodec copy -vcodec libvpx encode.webm # but my machine is too slow to play this. worried the audio is out of sync.... need slep [21:32] <durandal_1707> klaxa: if disk is big and very fast it is irrelevant [21:32] <klaxa> hendry, reload the gist, go to sleep, come back tomorrow with a clear mind [21:32] <klaxa> looks like you are a bit tired [21:33] <durandal_1707> amendes365: you need aac encoder there are many of them supported by ffmpeg [21:33] <hendry> klaxa: if we use flac then we get the Only VP8 video and Vorbis audio are supported for WebM. problem [21:33] <hendry> web video is crazy hard :( [21:33] <klaxa> you are supposed to use it the other way around [21:33] <klaxa> flac for the raw, vorbis for the encode [21:33] <klaxa> because flac is lossless [21:33] <klaxa> and vorbis isn't [21:34] <amendes365> durandal_1707: i know, ffserver -formats show raw ADTS AAC, Advanced Audio Coding, libfaac AAC but when starting ffserver shows Unknown audio codec: AAC [21:34] <hendry> klaxa: so we need an intermediate step to convert the flac audio to vorbis for the final webm ? [21:34] <durandal_1707> klaxa: but it is irrelevant in this case becuase it just copy it when muxing into webm [21:34] <klaxa> hmm true [21:34] <durandal_1707> hendry: try flac if libvorbis encoding is soo slow [21:35] <durandal_1707> hendry: basically you recored as fast as possible to keep sync and reencode and mux after recording is finished [21:36] <durandal_1707> amendes365: formats are not codecs, use ffserver -codecs [21:36] <hendry> durandal_1707: understand that [21:37] <hendry> durandal_1707: guess i need to figure out the intermediate step's args after some sleep [21:39] <amendes365> durandal_1707: thanks. ffserver -codecs showing unrecognized option '-codecs' [21:39] <amendes365> klaxa: thank you for the help yesterday, trying mpegts now [21:39] <klaxa> i can run ffserver -codecs [21:40] <klaxa> the only encodable line says: DEA.L. aac AAC (Advanced Audio Coding) [21:41] <amendes365> ill try reinstalling ffmpeg [21:41] <durandal_1707> internal encoder is experimental [21:41] <durandal_1707> use libfdk-aac [21:41] <klaxa> how would you install and set up an external encoder? [21:41] <klaxa> install encoder program and re-compile ffmpeg with the apropriate flags? [21:41] <durandal_1707> yes [21:43] <klaxa> so it probably won't work with pre-compiled binaries from <insert arbitrary package-manager/repository here> ? [21:44] <amendes365> i've visited different forums where usually is recommended to use the medibuntu repositories for this type of codec [21:44] <durandal_1707> klaxa: some external encoders are not redistributable(non-free) [21:44] <durandal_1707> which means, to use best available encoder you need to compile it [21:45] <amendes365> is there any other audio codec i can use instead of aac for streaming H.264 video in mpegts? [21:46] <durandal_1707> mp2? [21:46] <durandal_1707> or aac [21:46] <durandal_1707> *ac3 [21:47] <amendes365> durandal_1707: will try mp2. this is my second day sleeping in the lab :/ [21:47] <durandal_1707> mpegts for audio acepts: mp2,mp3,ac3 and aac [21:48] <durandal_1707> mp2 quality is not perfect [21:48] <durandal_1707> amendes365: what bitrate you are targeting? [21:49] <amendes365> durandal_1707: i want a very lightweight setup to see if i can get realtime audio/video streaming from a robot [21:49] <amendes365> robot is running on debian [21:49] <amendes365> as long as the audio is understandable is ok [21:50] <durandal_1707> so robot is doing encoding? [21:51] <amendes365> yes, but im testing a scenario between two laptops first [21:51] <amendes365> and using a webcam [21:52] <amendes365> i was using mjpeg which worked perfectly, but it has no audio [21:53] <amendes365> and i was considering ffmpeg to have both video and audio, otherwise i would have to do them separately [21:54] <durandal_1707> amendes365: so you can only use mpegts? [21:55] <amendes365> i can use anything as long as it is lightweight and can get me close to real time streaming [21:56] <durandal_1707> well mkv wiht mjpeg + any audio codec should work tooo [21:56] <amendes365> durandal_1707: quality is not important as long as the stream is understandable [21:56] <durandal_1707> if bitrate can be very high you can be completly lossless [21:57] <amendes365> ok... will try that [22:01] <klaxa> i don't think lossless video is important if a webcam is the source... [22:02] <klaxa> durandal_1707: do you have experience with live streaming matroska? i had quite the trouble and didn't succeed in the end [22:06] <durandal_1707> klaxa: what trouble? [22:06] <klaxa> mostly that ffserver wasn't able to mux the videos correctly (h264 video, aac audio, ass subtitles) [22:07] <klaxa> actually i don't remember what went wrong... [22:08] <klaxa> i'll try again and come back... maybe i was just too stupid the last time i tried [22:20] <amendes365> durandal_1707: got it working with the official theora/vorbis example, will make a conf for mkv + mjpeg, which will be the container and which one is the codec? [22:20] <amendes365> audio will be mp2 [22:21] <amendes365> test [22:35] <durandal_1707> amendes365: mkv is container [22:36] <amendes365> excellent [22:36] <amendes365> durandal1707: i made a test with output from my webcam to a .mkv video, mjpg is working [22:36] <amendes365> im working on the audio right now : ) [00:00] --- Sun Oct 21 2012

1 0

ffmpeg-devel.log.20121019
by burek 20 Oct '12

20 Oct '12

[00:04] <durandal_1707> michaelni: why i get stream start is in unreadable format? 2286340.981000 [00:32] <durandal_1707> what to use to convert ascii to integer? [00:34] <pengvado> strtol [00:35] <durandal_1707> and how to check for failure? [00:36] <pengvado> *endptr==ptr implies no conversion happened [01:48] <cone-541> ffmpeg.git 03Michael Niedermayer 07df727d408c7e: mlp_parser: print error when ff_combine_frame() fails to add the current buffer * 03http://tinyurl.com/8zok49203 [01:48] <cone-541> ffmpeg.git 03Michael Niedermayer 0705b0337025f6: motionpixels/mp_decode_frame_helper: assert that the first pixel doesnt reuse the last. * 03http://tinyurl.com/9etjwdy03 [02:08] <cone-541> ffmpeg.git 03Paul B Mahol 07aadb7b3ac4fb: lavc/c93: use meaningful error codes * 03http://tinyurl.com/9ezxumo03 [02:08] <cone-541> ffmpeg.git 03Paul B Mahol 07527224830aad: idcinvideo: if decoding fails return error * 03http://tinyurl.com/9q3wtnr03 [02:08] <cone-541> ffmpeg.git 03Paul B Mahol 074c6c6a266e9c: jvdec: use more meaningful error code * 03http://tinyurl.com/8twxyjs03 [02:08] <cone-541> ffmpeg.git 03Paul B Mahol 070bf40e0ef421: lavc/yop: remove redudant YOP in av_log() messages * 03http://tinyurl.com/9nz3q8u03 [02:08] <cone-541> ffmpeg.git 03Paul B Mahol 07be536f084a98: xxan: return more meaningful error codes * 03http://tinyurl.com/9p8n6zs03 [02:08] <cone-541> ffmpeg.git 03Paul B Mahol 07445f36d7c83f: kmvc: use meaningful error codes * 03http://tinyurl.com/9yj2mu203 [02:08] <cone-541> ffmpeg.git 03Paul B Mahol 074ebf30595106: lavc/tta: do not overwrite bits_per_coded_sample * 03http://tinyurl.com/9r9toe503 [02:20] <durandal_1707> some guy on libav-user ml says that libswscale is too slow [02:21] <Compn> lets rm it and rewrite it hen [02:21] <Compn> :P [03:03] <cone-541> ffmpeg.git 03Michael Niedermayer 07c0b17ea106b9: roqaudioenc: Fix crash with very small roq files * 03http://tinyurl.com/975s86m03 [03:51] <cone-541> ffmpeg.git 03Michael Niedermayer 07a06f943f9d65: roqvideodec: replace dead code by assert * 03http://tinyurl.com/9kmsbqy03 [03:51] <cone-541> ffmpeg.git 03Michael Niedermayer 07d50aa006fb34: tiffenc: fix integer overflow * 03http://tinyurl.com/9u432en03 [09:32] <divVerent> ubitux: I have seen the commit [09:32] <divVerent> what is with all this tinyurl waste BTW? [09:32] <ubitux> i don't know [09:32] <ubitux> i don't like it :( [09:32] <divVerent> do we really need to allocate a new tinyurl for each commit? [09:32] <divVerent> it just worsens the tinyurl service for everyone else [09:33] <divVerent> why not http://ffmpeg.org/gd50aa00 [09:33] <divVerent> would be a job for mod_rewrite ;) [09:33] <ubitux> even the full url with a reduced hash is fine [09:33] <TimNich> +1 [09:33] <divVerent> well, I see the point of using short urls [09:33] <divVerent> but compare [09:33] <divVerent> http://tinyurl.com/9u432en [09:33] <divVerent> http://ffmpeg.org/gd50aa00 [09:33] <divVerent> equally long [09:33] <divVerent> and the latter doesn't eat up tinyurl namespace [09:34] <divVerent> but only means ffmpeg can no longer have URLs starting with g and having no dot inside ;) [09:34] <divVerent> could even do http://ffmpeg.org/d50aa00 with some careful regexing [09:34] <ubitux> ffmpeg.org/h/<hash> [09:34] <divVerent> or that [09:34] <divVerent> that's one more character than the tinyurl though ;) [09:34] <ubitux> i think we have similar redirections with sources.ffmpeg.org [09:35] <ubitux> source.ffmpeg.org sorry [09:35] <divVerent> o the tinyurl TOS even allow this? [09:35] <divVerent> yes, it is [09:35] <divVerent> well, then that's tinyurl's fault ;) [09:36] <ubitux> curl -I 'source.ffmpeg.org/?test' [...] Location: http://git.videolan.org/?p=ffmpeg.gittest&test [09:36] <ubitux> something looks fishy :D [09:36] <divVerent> hehe [09:36] <divVerent> that'd work too [09:36] <divVerent> http://ffmpeg.org/?hash [09:36] <ubitux> curl -I 'source.ffmpeg.org/?foo=bar' [...] Location: http://git.videolan.org/?p=ffmpeg.gitfoo=bar&foo=bar [09:37] <ubitux> the dup is weird [09:37] <divVerent> ubitux: it MAY still make sense [09:37] <divVerent> there may be a separator that isn't contained [09:37] <divVerent> but I can't find the one :P [09:37] <ubitux> curl -sI 'source.ffmpeg.org/?foo=bar' | cat -e [...] Location: http://git.videolan.org/?p=ffmpeg.gitfoo=bar&foo=bar^M$ [09:38] <ubitux> doesn't look like to [09:38] <divVerent> was just a guess, becuase such duping happens easily in shell scripts splitting strings [09:38] <divVerent> if s is foo/bar, then ${s%%/*} is foo and ${s#*/} is bar [09:38] <divVerent> but if s is just foo, both will be foo [09:39] <divVerent> so MAYBE source.ffmpeg.org uses a matching technique with similar issues [09:39] <divVerent> sure not a shell script though ;) [09:40] <ubitux> source.ffmpeg.org/?;a=commit;h=d50aa006 [09:41] <ubitux> i'm able to trick it with this [09:41] <ubitux> but it's ugly :p [09:41] <divVerent> but it still dupes, I suppose [09:41] <divVerent> just that this then at least works [09:41] <ubitux> yes [09:42] <ubitux> Location: http://git.videolan.org/?p=ffmpeg.git;a=commit;h=d50aa006&;a=commit;h=d50aa… [09:42] <ubitux> it's a big ugly :D [09:42] <divVerent> source.ffmpeg.org/?;h=d50aa006 [09:42] <divVerent> suffices BTW [09:42] <ubitux> indeed [09:42] <divVerent> because a=commit is default if h is given [09:42] <ubitux> i like a=commitdiff though [09:43] <ubitux> i'd go for source.ffmpeg.org/?;a=commitdiff;h=d50aa006 [09:43] <divVerent> I don't, for a very simple reason [09:43] <divVerent> because the subject line then is less visible :P [09:43] <ubitux> how so? oO [09:44] <ubitux> commitdiff has the commit description [09:44] <ubitux> in addition [09:44] <ubitux> ah no sorry it's also in commit as well [09:50] <divVerent> yes [09:50] <divVerent> it is there in commitdiff [09:50] <divVerent> but in commit, it is right above the commit message [09:51] <divVerent> and in committdiff it's way more above [10:49] <ubitux> hey, it would be nice to have a shout protocol [13:36] <cone-191> ffmpeg.git 03Martin Storsjö 07b760ffdd07ea: aviobuf: Remove a senseless ifdef in avio_seek * 03http://tinyurl.com/9ukqgnp03 [13:36] <cone-191> ffmpeg.git 03Martin Storsjö 0753e8cd68b722: configure: Split out msvc as a separate target OS * 03http://tinyurl.com/8sttjkg03 [13:36] <cone-191> ffmpeg.git 03Martin Storsjö 07fc085c5b33a9: gxf: Add a local copy of the relevant parts of the frame rate table * 03http://tinyurl.com/8rrhfe903 [13:36] <cone-191> ffmpeg.git 03Martin Storsjö 07eaa9b2e66c04: avcodec: Rename avpriv_frame_rate_tab to ff_mpeg12_frame_rate_tab * 03http://tinyurl.com/9g8tx6h03 [13:36] <cone-191> ffmpeg.git 03Martin Storsjö 07d66c52c2b369: Add support for building shared libraries with MSVC * 03http://tinyurl.com/8ehw33803 [13:36] <cone-191> ffmpeg.git 03Mans Rullgard 07c0329748b04e: fate: add a dependency helper macro * 03http://tinyurl.com/8bw2wlb03 [13:36] <cone-191> ffmpeg.git 03Michael Niedermayer 07b0554fec0470: Merge commit 'c0329748b04e1f175dad8c9c2ebf22a5e2dc5b72' * 03http://tinyurl.com/9kxlvbp03 [13:54] <nevcairiel> guess now i have to figure out what needs to be fixed for shared msvc ffmpeg builds [13:54] <cone-191> ffmpeg.git 03Mans Rullgard 07eccd0671cc89: fate: dependencies for demux tests * 03http://tinyurl.com/9u2k9wf03 [13:54] <cone-191> ffmpeg.git 03Mans Rullgard 07b55dda4a7205: fate-als: add dependencies * 03http://tinyurl.com/9ogn8x503 [13:54] <cone-191> ffmpeg.git 03Mans Rullgard 072e05143a8df9: fate-indeo: add dependencies * 03http://tinyurl.com/9f2fhbg03 [13:54] <cone-191> ffmpeg.git 03Mans Rullgard 0708c6a12a562d: fate-mpc: add dependencies * 03http://tinyurl.com/8pr5yor03 [13:54] <cone-191> ffmpeg.git 03Mans Rullgard 0722f5149ec461: fate-twinvq: add dependencies * 03http://tinyurl.com/97d4waa03 [13:54] <cone-191> ffmpeg.git 03Mans Rullgard 0755351de9f43d: fate: add dependencies for misc microsoft codecs * 03http://tinyurl.com/8hmbtak03 [13:54] <cone-191> ffmpeg.git 03Martin Storsjö 0722310eef9f78: changelog: Mention the MSVC DLL support * 03http://tinyurl.com/9h8hzyf03 [13:54] <cone-191> ffmpeg.git 03Martin Storsjö 070af1fe845a9d: avformat: Fix references to the removed function av_write_header in comments * 03http://tinyurl.com/9btmu3d03 [13:54] <cone-191> ffmpeg.git 03Michael Niedermayer 0793f244e3abc3: Merge commit '0af1fe845a9d7112da0a58d33a4fc81fe7c47e95' * 03http://tinyurl.com/95tlxyk03 [14:10] <cone-191> ffmpeg.git 03Diego Biurrun 077b2121e7e2e1: riff: Move functions around to be covered by appropriate #ifdefs * 03http://tinyurl.com/8jr27k203 [14:10] <cone-191> ffmpeg.git 03Diego Biurrun 07e8fe208be810: fate: dependencies for screen codec tests * 03http://tinyurl.com/9x3kjv303 [14:10] <cone-191> ffmpeg.git 03Mans Rullgard 077a12d97eb1aa: aac: fix build with hardcoded tables * 03http://tinyurl.com/8dcsffq03 [14:10] <cone-191> ffmpeg.git 03Martin Storsjö 070de9380be54e: rtp: Update the check for distinguishing between RTP and RTCP * 03http://tinyurl.com/8ey2ojh03 [14:10] <cone-191> ffmpeg.git 03Martin Storsjö 071c37744963ac: rtsp: Vertically align a constant definition * 03http://tinyurl.com/8bf6l4k03 [14:10] <cone-191> ffmpeg.git 03Martin Storsjö 073f055f8f5f71: rtsp: Allow setting the reordering buffer size via an AVOption * 03http://tinyurl.com/8qtg8s203 [14:10] <cone-191> ffmpeg.git 03Mans Rullgard 071cd432e167b1: configure: fix libcdio check * 03http://tinyurl.com/8jjwqzr03 [14:10] <cone-191> ffmpeg.git 03Michael Niedermayer 0781ff0c24ef05: Merge commit '1cd432e167b1a80853760c89a33606e2b5f229c2' * 03Error03 [14:20] <cone-191> ffmpeg.git 03Kostya Shishkov 078774d5835806: bmv: get a new frame on every decode_frame(), so we can use direct rendering * 03http://tinyurl.com/9hmms5e03 [14:20] <cone-191> ffmpeg.git 03Kostya Shishkov 07169514c440a7: indeo3: do not try to output more lines than we can fit * 03http://tinyurl.com/9vrhule03 [14:20] <cone-191> ffmpeg.git 03Martin Storsjö 0791485e744f8c: fate: Add proper dependencies in lossless-video.mak * 03http://tinyurl.com/9ntr68f03 [14:20] <cone-191> ffmpeg.git 03Martin Storsjö 0761cc99748c11: fate: Add proper dependencies in qt.mak * 03http://tinyurl.com/8htepdb03 [14:20] <cone-191> ffmpeg.git 03Michael Niedermayer 07c5fd9d3c35cb: Merge remote-tracking branch 'qatar/master' * 03http://tinyurl.com/9gkanw703 [15:02] <cone-191> ffmpeg.git 03jamal 079434ead2f34b: fate: Handle lavf-fate tests in a makefile * 03http://tinyurl.com/9vmkneu03 [15:13] <gnafu> Don't we really only need tiny URLs if we plan to tweet about a commit? [15:13] Action: gnafu scuttles. [15:36] Action: cbsrobot advises gnafu to fix his tinyurl before fixing others: http://tinyurl.com/gnafu [15:39] <gnafu> Huh, there's a Gnafu on Twitter. [15:54] <durandal_1707> troller too? [16:11] <cone-191> ffmpeg.git 03Petter Ericson 077abf394814d8: mov.c: Check for stsd + m1s tag indicating MOV-wrapped MPEG-PS, and force continued probing if found. * 03http://tinyurl.com/9erz4sh03 [16:26] <gnafu> durandal_1707: Huh, I guess I never quite considered myself a troller. I didn't think I was good enough. I do look up to kshishkov and av500, though ;D. [16:27] <gnafu> And I dunno about that Gnafu, as it seems most of their tweets are in French (and I didn't bother trying to translate). [16:27] <durandal_1707> gnafu: sorry, I was just trolling [16:27] <gnafu> :-D [17:01] <durandal_1707> Compn: ping [17:07] <j-b> http://technet.microsoft.com/en-us/security/msvr/msvr12-017 [17:10] <durandal_1707> they are still secret, so? [17:13] <durandal_1707> now that there is riff tags it is time add exif tags support [17:22] <ubitux> the hard part with supporting old weird subtitles format is to actually find some samples :P [17:22] <nevcairiel> how about a tool that can create them? :) [17:23] <ubitux> might do the trick [17:23] <ubitux> i have one sample for subviewer1, one for mpsub (though, i should be able to generate some more with mplayer) [17:23] <mateo`> hi, dumb question about the sony D10 spec, is 8 channels at 24bit allowed ? [17:23] <ubitux> now about vplayer, aqt or pjs.. [17:28] <ubitux> j-b: would vlc be interested in using lavf/lavc for subtitles at some point? [17:30] <durandal_1707> j-b: would vlc be interested in QDMC ? [17:31] <j-b> ubitux: subtitles being? [17:32] <j-b> durandal_1707: I think this is used in RTSP too right? [17:33] <ubitux> j-b: right now we have subrip, realtext, sami, subviewer2, jacosub, webvtt (basic markup for this one though), and a few others; basically the more complex ones [17:33] <ubitux> i plan to add the old ones supported by mplayer pretty soon [17:35] <ubitux> we're going to change the api anyway, but i was wondering if at some point the vlc project would be interested in moving to lavf/lavc for this [17:37] <j-b> ubitux: no, I mean: do you mean decoding or rendering? [17:37] <ubitux> demux and decode [17:37] <ubitux> right now you would get an ASS string which you could directly send to libass [17:38] <ubitux> it will change [17:38] <ubitux> here is how it would look like: http://lists.mplayerhq.hu/pipermail/ffmpeg-devel/2012-September/130474.html [17:39] <ubitux> basically you would be in the video player path in the diagram [17:39] <ubitux> (btw if you have any comment/suggestion on this) [17:39] <ubitux> (work is already started) [17:41] <j-b> decoding, maybe, re-rendering, no. [17:42] <ubitux> you would be responsible for the rendering [17:42] <ubitux> what do you mean by "re" rendering? [17:43] <ubitux> the "libass to render" is in your scope; if your player has libass you can decide to encode the avsubtitle into ass markup; otherwise you can encode it in "raw text" so it can be printf'ed or displayed with a basic osd [17:44] <ubitux> (that's what the rendering filter in lavfi would do) [17:44] <nevcairiel> i wish someone would finish the work on libass to make it free of fontconfig on windows [17:47] <j-b> zgreg did [17:49] <Plorkyeran> a patch to vlc is not exactly what I would call non-fontconfig support in libass [17:50] <j-b> He released 0.10.1 and updated his work on harfbuzz in order to go to this point [17:50] <nevcairiel> there is a branch in the google code git repository, but it has some open TODOs [18:11] <j-b> nevcairiel: but it is on the way [18:20] <cone-191> ffmpeg.git 03Ash Hughes 072470851f1228: lavc: enable recursively using avcodec_open2/close. * 03http://tinyurl.com/9w5p8t903 [18:54] <ubitux> - h->cbp= cbp; [18:54] <ubitux> + h->cbp = cbp; s->current_picture.f.mb_type[mb_xy] = mb_type; [18:54] <ubitux> am i the only one to find this ridiculous? [18:54] <ubitux> (\n after ';', irssi ate it) [18:56] <ubitux> same for the switch cases.. :/ [18:56] Action: ubitux really doesn't get it [18:57] <Compn> durandal_1707 : pong [18:57] <Compn> ubitux : you saw the sub samples in the repo, right ? [18:58] <durandal_1707> Compn: does playing QDMC works for you? [18:59] <Compn> in mplayer ? [18:59] <Compn> i dont have latest svn mplayer, a lot of things are broken due to planar output atm [18:59] <durandal_1707> but svn mplayer have fixed that [19:00] <Compn> http://samples.mplayerhq.hu/A-codecs/QDMC/rumcoke.mov works with lavf demuxer and binary quicktime codec [19:00] <durandal_1707> and it does not need to be latest, because QDMC works only via binary codec [19:00] <ubitux> Compn: the mpsub in a lost directory on mphq [19:01] <Compn> for me , using r34835 [19:01] <ubitux> (not the samples directory iirc) [19:01] <durandal_1707> Compn: good [19:02] <Compn> ubitux : whut? theres http://samples.ffmpeg.org/sub/ [19:02] <cone-191> ffmpeg.git 03Michael Niedermayer 07d86ef54476b3: avcodec_align_dimensions2: add missing breaks * 03http://tinyurl.com/999d77v03 [19:02] <cone-191> ffmpeg.git 03Michael Niedermayer 070fa26bd4703c: utvideoenc: fix theoretical integer overflow * 03http://tinyurl.com/8l96hz603 [19:02] <Compn> ubitux : http://samples.ffmpeg.org/sub/manyfmts/ contains the mpsub samples generated by some random subtitle program [19:05] <ubitux> wow. :) [19:05] <ubitux> thx [19:06] <Compn> theres also this [19:06] <Compn> http://www1.mplayerhq.hu/DOCS/tech/mpsub.sub [19:12] <Compn> durandal_1707 : anything else you want me to test with quicktime stuff ? [19:13] <durandal_1707> no, i just need brainz to start REing [19:16] <Compn> ah [19:16] <Compn> wonder if kostya did any work on it [19:17] <Compn> qdm2 has lots of authors :) [19:17] <Compn> decoder i mean [19:20] <Compn> durandal_1707 : you've been busy RE'ing lots of stuff :) [19:20] <durandal_1707> Compn: that was old times... [19:20] <Compn> ben larsson wrote this up : http://multimedia.cx/mirror/qdmc2.pdf [19:21] <Compn> has some info on qdmc i think [19:22] <Compn> oh , no it doesnt :D [19:22] Action: Compn says stuff before reading stuff [19:34] <Compn> qdm2.c is hard to follow :) [19:56] <rrivera> Hey guys, oss colleague of mine is sitting down with the Visual C++ guys re: why oss devs might not be using Visual Studio/VC++ compiler. Looking to collect your feedback and pass it on as I've hacked on ffmpeg/win32 in the past. [19:56] <rrivera> I understand C99 is important, anything else? [19:56] <JEEB> C99 features are the biggest reason unless the project is C++ [19:57] <JEEB> ffmpeg now has MSVC support because there's a clang-based converter now [19:57] <JEEB> which converts /some/ C99 features into C89 [19:57] <rrivera> Ah yeah, saw that. That's kinda lame tho isn't it, lol [19:57] <JEEB> and makes ffmpeg compile'able in the first place [19:57] <JEEB> rrivera, well how else would you support it then o_O Because it's pretty sure that MSVC ain't going to support C99/11 any time soon [19:58] <JEEB> you take the most realistic path that's possible [19:58] <rrivera> I hear you, yup. [19:59] <Compn> rrivera : patches are welcome to make ffmpeg work easier in vc++ compilers and apps [19:59] <JEEB> and then there's plenty of windows hate in various OSS circles, certain people not being ready to make limitations or extra changes regarding windows or MSVC specifically unless pushed a lot [19:59] <Compn> as long as they dont clutter the code in huge ways... [20:00] <JEEB> those are the two biggest reasons I know why OSS projects don't use the MS compilers [20:00] <Compn> actually we support a lot of mingw/msvc now JEEB [20:00] <JEEB> yes I know that [20:00] <JEEB> I just noted [20:00] <Compn> especially because chrome uses msvc to compile ffmpeg ... [20:00] <Compn> ah [20:00] <cone-191> ffmpeg.git 03Michael Niedermayer 0731fdf3065dac: vf_idet: reorder operations to avoid division by 0 * 03http://tinyurl.com/9f6zo4903 [20:00] <cone-191> ffmpeg.git 03Michael Niedermayer 07042a738b4599: vf_mp: check list in querry_format() * 03http://tinyurl.com/8jtsj5903 [20:00] <cone-191> ffmpeg.git 03Michael Niedermayer 07aaf78e4d14b4: vf_mp: fix null ptr deref in case of ENOMEM * 03http://tinyurl.com/9mg26dh03 [20:00] <JEEB> I'm just talking about overall (F)OSS traits that lead to MSVC not being used [20:00] <Compn> ah [20:01] <rrivera> Yep gotcha. I'll put another tick in the C99 list haha [20:01] <Compn> rrivera : most of the devels use linux. so they stick to linux tools. [20:01] <Compn> some use msvc and its debugging tho [20:02] <JEEB> the first being the fact that stuff uses C99/11 features (and possibly GNU extensions), second being linux/*nix developers and possible wish to not "filth" the project with MSVC-specifics (given the thing can even be built with MSVC to begin with) [20:02] <Compn> and one of ffmpeg's core goals is to be cross-os-compatable [20:03] <Compn> so vc++ stuff probably doesnt work well on mac for instance... [20:03] <Compn> or on ppc, arm, and other arch [20:04] <JEEB> they did re-add ARM again lately, but yeah [20:04] <JEEB> it's very specific, and if you're cross-platform then... [20:04] <Compn> yeah i dont think visualstudio was cross-plat back in the year 2000/2001 when ffmpeg was started [20:05] <rrivera> I would think Windows is a big consumer of ffmpeg [20:05] <Compn> it is [20:05] <rrivera> Interesting that it's a back burner OS for ffmpeg-dev [20:05] <Compn> vlc distributes millions of copies every year for windows [20:06] <kierank> 19:00:17 <Compn> especially because chrome uses msvc to compile ffmpeg ... --> iirc not yet [20:06] <kierank> not sure about that though [20:06] <Compn> there was some talk of a script that converts c99 to what vc++ stuff needs [20:06] <Compn> a pre-compiler script [20:07] <Daemon404> you mustnt be very observant >.> [20:07] <Daemon404> chrome does use msvc now (or will extremely soon-- theres stuff sitting in gerrit) [20:07] <Daemon404> and ffmpeg has built with msvc for a while now [20:07] <JEEB> <rrivera> Interesting that it's a back burner OS for ffmpeg-dev <- people've used it with mingw toolchains for years, then linking to their MSVC-built software [20:08] <Compn> yeah, people have been using ffmpeg on windows successfully [20:08] <Compn> there is a ffmpeg-msvc port , why arent more vc++ people helping to develop that ? [20:09] <Daemon404> Compn, ffmpeg git master builds with msvc [20:09] <Daemon404> stop spreading misinfortmation [20:09] <Compn> Daemon404 : tell it to kierank [20:09] <kierank> i mean i don't think chrome use ffmpeg with msvc yet [20:09] <kierank> iirc [20:10] <Compn> rrivera : what problems are there wtih ffmpeg ? submitting some bug reports is the quickest way to get ffmpeg to fix things .... [20:10] <Compn> http://trac.ffmpeg.org/trac/ffmpeg [20:10] <rrivera> Compn: Hmm, what? [20:11] <Compn> Daemon404 : http://trac.ffmpeg.org/trac/ffmpeg/wiki/KnownCompilationIssues and http://trac.ffmpeg.org/trac/ffmpeg/wiki/CompilationGuide both mention msvc being uncompilable [20:11] <Daemon404> i read it on teh internet [20:11] <Compn> Daemon404 : if the wiki needs to be updated, yell at it for spreading misinfo [20:11] <Daemon404> on a wiki even! [20:11] <Daemon404> must be true! [20:11] <Compn> our wiki! [20:11] <rrivera> lol [20:11] <Daemon404> never mind http://ffmpeg.org/platform.html [20:11] <Daemon404> that OFFICIAL docs [20:12] <Compn> oh yeah [20:12] <Compn> rrivera : so what problems are there with ffmpeg and visual studio ? [20:12] <Compn> i ask because i dont know them [20:12] <rrivera> Compn: Not sure, I typically cross-compile. I'm just asking for feedback to pass onto Visual C++ team. :) [20:13] <Compn> heh [20:13] <Compn> which team / company are you talking to ? :) [20:13] Action: Compn updates wiki [20:13] <rrivera> I'm just passing it onto my colleague Garrett (http://twitter.com/fearthecowboy/status/259340782337671168) [20:14] <Compn> well the answer is ---- ffmpeg supports vc++ :) [20:17] <Compn> there. [20:18] <Compn> rrivera : send him that info on the c99 to c89 script. thats probably useful for a lot of projects... [20:21] Action: durandal_1707 compiling mplayer, result: libavcodec/x86/cabac.h:170:9: error: ran out of registers during register allocation [20:22] <Compn> durandal_1707 : ehe, michael vs gcc bug :) [20:22] <Compn> it should have been fixed [20:22] <Compn> re run the last command with -fomit-frame-pointer [20:22] <durandal_1707> i'm using clang [20:22] <Compn> ah [20:22] <Compn> well -fomit-frame-pointers then (i cant remember pointers / pointer) [20:22] <Compn> if thats possible with clang... [20:23] <durandal_1707> why mplayer recompiles ffmpeg - how to forece it to use shared lib? [20:23] <Compn> ./configure --help | grep -i ffmpeg |grep -i external [20:23] <Compn> or something [20:23] <Compn> or rm -rf mplayer/ffmpeg [20:23] <Compn> maybe [20:23] <Compn> no idea [20:24] <Daemon404> the better question is [20:24] <Daemon404> why does mplayer use a shitty fucked up custom build system for ffmpeg [20:24] <Daemon404> at least mplayer2 got rid of that iirc [20:25] <durandal_1707> Daemon404: last time i checked it still use some internal stuff [20:25] <Compn> have to ask dondiego [20:26] <Compn> mplayer uses internal stuff, for some filters and other things [20:26] <Daemon404> durandal_1707, i thought mplayer2 supported system ffmpeg now [20:26] <Compn> mplayer2 got rid of those filters i think :D [20:26] <Compn> or maybe just for mplayer2 shared ffmpeg [20:26] <Daemon404> i bet theyre nto even useful filters [20:26] <durandal_1707> Daemon404: mplayer2 and mpv support system libav* [20:27] <Daemon404> yeah [20:27] <Daemon404> i havent used 'normal' mplayer in a long time. [20:28] <durandal_1707> wtf, libpostproc can not compile with latest lavu [20:28] <durandal_1707> according to mplayer output [20:29] <beastd> durandal_1707: huh [20:29] <durandal_1707> Checking for libpostproc ... Package 'libpostproc' requires 'libavutil = 51.41.100' but version of libavutil is 51.65.100 [20:30] Action: Compn runs away from version hell [20:31] <beastd> durandal_1707: Some pkg_config problem it seems [20:31] <durandal_1707> yea [20:31] <durandal_1707> i forgot to remove it when removing lib [20:39] <durandal_1707> Compn: looks like with latest mplayer i cant play QDMC [20:41] <durandal_1707> forget that it works [20:41] <durandal_1707> something was corrupted in binary blobs [20:53] <Compn> durandal_1707 : yeah , theres some versions of the binary which dont work (qt 6.5 i think does not, but qt 6.3 does) [21:58] <ubitux> <@Compn> http://www1.mplayerhq.hu/DOCS/tech/mpsub.sub // i was talking about this one :) [22:55] <cone-191> ffmpeg.git 03Michael Niedermayer 073d48dd01fd3d: avidec: remove unneeded null check * 03http://tinyurl.com/9nyphoz03 [22:56] <cone-191> ffmpeg.git 03Michael Niedermayer 07d30351363f6f: url_alloc_for_protocol: fix use of uninitialized variable * 03http://tinyurl.com/9cb38se03 [22:56] <cone-191> ffmpeg.git 03Michael Niedermayer 07a96577df3846: avio: fix sizeof argument * 03http://tinyurl.com/9hfsfp803 [23:36] Action: durandal_1707 Yes!! [23:37] <ubitux> @_@ [23:40] <cone-191> ffmpeg.git 03Michael Niedermayer 07395caf3de84a: hls: fix integer overflow * 03http://tinyurl.com/8snqvdk03 [23:40] <cone-191> ffmpeg.git 03Michael Niedermayer 07ba39303050c1: gxfenc: fix null ptr dereference * 03http://tinyurl.com/9osqqhj03 [23:40] <cone-191> ffmpeg.git 03Michael Niedermayer 078fb8d539a4c5: hlsproto: fix integer overflow * 03http://tinyurl.com/96eg79o03 [00:00] --- Sat Oct 20 2012

1 0

ffmpeg.log.20121019
by burek 20 Oct '12

20 Oct '12

[00:47] <whoom> hi [00:48] <whoom> anyone in here [00:51] <relaxed> we're all waiting for a valid question. [01:21] <arakn0> is it possible to use a bit stream filter in ffserver? any hint? Thanks [01:56] <creep> both vlc and mplayer should be useful, but the coders surely suck in user interaction [02:41] <t4nk157> hello! anyone know how i can convert a byte array of 2 channel 16 bit audio into a planar format that i can encode with? [02:55] <durandal_1707> t4nk157: that should be pretty trivial and there are and was numerous examples in FFmpeg source code and on the web [02:56] <t4nk157> yeah i've spent 4 days on it so far - any ideas where to look? I've spent hours on google looking for it [02:56] <durandal_1707> have you wrote some code at all? [02:57] <t4nk157> yes massive amounts, it's working with single channel, as soon as i add another channel it gets distorted - i think i was chatting to you yesterday about it [02:57] <durandal_1707> t4nk157: i mean code that does interleaving [02:59] <t4nk157> i'm not sure where to start - the code i had worked with the 53 version. It doesn't work with the new version. I get a byte array of AV_SAMPLE_FMT_S16 which I can't encode with unless I use AV_SAMPLE_FMT_S16P - but I have no idea how to get that byte array into a planar format [03:01] <durandal_1707> for planar and 2 channels it is 2 planes or 2 byte arrays [03:01] <durandal_1707> how do you call encode_audio2 ? [03:02] <t4nk157> avcodec_encode_audio2(c, &pkt, frame, &got_packet); [03:02] <durandal_1707> and where you put your pointer to array? [03:03] <t4nk157> i get a load of byte data at a time that i have to split into frames [03:04] <durandal_1707> you put it into AVFrame->data[] ? [03:04] <t4nk157> ret = libffmpeg::avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt, pSoundBuffer, size, 1); [03:07] <durandal_1707> in that case your pSoundBufer need to be interleaved [03:07] <durandal_1707> *deinterleaved [03:07] <durandal_1707> so lets assume it have 100 samples in each frame - each sample takes 2 bytes [03:08] <t4nk157> yeah that's what i'm after - how to do that [03:09] <durandal_1707> size & 1 must be 0 for stereo [03:09] <durandal_1707> so you divide size by 2 (because there are 2 channels in stereo) [03:10] <durandal_1707> there are many ways to do actual deinterleaving, one of it is using libswresample [03:10] <t4nk157> do you know of any example code that does that? [03:11] <durandal_1707> ffplay.c use it [03:12] <durandal_1707> and some filters libavfilter [03:12] <durandal_1707> do you use any of them? [03:13] <t4nk157> i've been through an enormous amount of source code trying to find an example of deinterleaving without result :( [03:13] <durandal_1707> i'm mentioning that lib because it have simd for conversion [03:15] <durandal_1707> naive approach: you take first 2 bytes and put them in X buffer. then take next 2 bytes and put them into Y buffer, you repeat this until you run out of samples [03:15] <t4nk157> yeah i haven't gotten anywhere, my old code does deinterleaving but i can't see how it relates to the new api - everything it's calling has either been deprecated or removed [03:15] <durandal_1707> last step you concatenate X and Y buffer [03:16] <t4nk157> ok so each packet has to be left or right channel? [03:16] <durandal_1707> packet is what encoder gives you [03:17] <durandal_1707> aka output [03:17] <t4nk157> yeah but i need to feed into the encoder arrays of left and then right channel? [03:17] <durandal_1707> the pSoundBUffer you mentioned in avcodec_fill_audio_frame is X+Y [03:19] <t4nk157> someone must have done this before right? [03:20] <durandal_1707> what? [03:21] <t4nk157> written some code that deinterleaves data for avcodec_encode_audio2 [03:22] <t4nk157> https://www.google.com/search?q=deinterleave+avcodec_encode_audio2 [03:22] <durandal_1707> no, you deinterleave pSoundBuffer you use in fill_audio_frame [03:23] <durandal_1707> it must be deinterleaved before calling fill_audio_frmae [03:26] <t4nk157> yes, therein lies my problem - you said there are loads of examples of doing this - could you give me a link? [03:29] <t4nk157> I thought all this was automatic - which is why you specify channel_layout and channels etc [03:30] <durandal_1707> t4nk157: i could be but it is not, you can not specify your sample format and expect fill_array to do the job [03:31] <durandal_1707> that is job for swr_convert [03:31] <t4nk157> ok is there an example of using swr_convert? [03:31] <durandal_1707> but this could be implemented .... [03:31] <durandal_1707> t4nk157: only if you have libswresample actually instealled. do you? [03:31] <t4nk157> yes [03:35] <durandal_1707> t4nk157: http://ffmpeg.org/doxygen/trunk/ffplay_8c-source.html [03:37] <t4nk157> ok thanks for that [07:34] <lake> i would like to use jack as an input format with ffmpeg. i'm on archlinux but i cannot seem to get the "-f jack" flag working [07:39] <lake> looks like i have to compile it with --enable-indev=jack [11:59] <gazzwi86> hi all [12:00] <gazzwi86> I'm trying to get a raspberry pi to stream a webcam to my computer [12:00] <gazzwi86> but when i go to ip/webcan.mjpeg it starts to download a file [12:10] <Yulth> hi! is fdk_aac be able to convert to AAC-HE? or is libaacplus better for this task? [12:12] <durandal_1707> yes. no. [12:16] <Yulth> extremely precise and concise :D [12:19] <Yulth> and, How I can be sure that I'm achieving AAC-HE quality typing this command:?? [12:19] <Yulth> ffmpeg -i input_file.flac -acodec libfdk_aac -ab 60k output_file.mp4 [12:24] <durandal_1707> Yulth: doing blind ABX test with reference that have AAC-HE quality [12:27] <durandal_1707> libfdk-aacenc supports several profiles: LC, HE-AAC, HE-AACv2, LD and ELD [12:28] <Yulth> I understand, but in the command line I can't specify what type of AAC I want to use for encoding. I only can type "mp4" or "aac". [12:29] <durandal_1707> there is -profile [12:29] <Yulth> mmm, interesting, I'm going to try it [12:29] <durandal_1707> ffmpeg -h full [12:39] <Yulth> this command: ffmpeg -i file_in.flac -profile:aac_he_v2 -ab 60k file_out.mp4 [12:39] <Yulth> is giving me this error: [12:39] <Yulth> Unable to find a suitable output format for '60k': invalid argument [12:39] <Yulth> :S [12:39] <Yulth> I think I've followed all steps shown by man page... [12:44] <durandal_1707> -profile HE-AAC [12:51] <Yulth> mmm the same: [12:51] <Yulth> "Please use -profile:a or -profile:v, -profile is ambiguous" [12:51] <Yulth> and I've followed the same sintax shown by man and even given by you... [12:56] <Yulth> ok solved: [12:56] <Yulth> -profile:a aac_he_v2 [13:13] <Element9> why doesn't "ffmpeg -formats" list the wmv1 and wmv2? [13:14] <durandal_1707> Element9: because they are not formats [13:14] <Element9> durandal_1707: formats == containers? [13:15] <durandal_1707> something like that [13:16] <Element9> durandal_1707: can I read somewhere about the difference? [13:17] <durandal_1707> it is just wording [13:18] <durandal_1707> in this specific case formats are containers [13:18] <Element9> can I choose which codecs to include and which to exclude when I compile ffmpeg? [13:19] <durandal_1707> yes [13:19] <Element9> durandal_1707: great. thanks! [15:16] <cykl> Hi. I currently have the following workflow: {SWFs} --swfrender--> {PNGs} --ffmpeg(image2)--> avi. I'm wondering if I could remove the dependency on swfrender and do all the stages with ffmpeg. [15:17] <cykl> Basically, each SWF file is an image, and I want to encode all the image into a movie. [15:18] <durandal_1707> can ffmpeg recognize swf files? [15:19] <cykl> According to google, uncompressed swf could be supported. [15:20] <durandal_1707> but does it work with your files? [15:21] <cykl> nope. I get this error message [swf @ 0x8e61a60]Could not find codec parameters (Video: mjpeg) [15:26] <durandal_1707> have you compiled ffmpeg with zlib support - paste full ffmpeg output [15:27] <cykl> http://pastebin.com/NBerPDUH [15:28] <durandal_1707> that is 3 years old [15:30] <cykl> Running a Debian stable. I need to embedded this process into an application, so user will have "outdated" versions anyway [15:31] <durandal_1707> cykl: sorry, i cant give you such support [15:32] <cykl> I will try on with an up to date version in few hours and see if it works or not. [15:32] <cykl> durandal_1707: thanks for your time. BTW: If somebody already did this kind of workflow I'm interested by a testimonial ;) [16:21] <dericed> :( [16:45] <meekohi> Hey how can I figure out which codec corresponds to AVCodec with codec id: 142? [16:45] <meekohi> avcodec_get_context_defaults3 [16:46] <meekohi> ? [16:46] <meekohi> No that doesn't make any sense... [16:47] <durandal_1707> most likely VP8 [16:50] <meekohi> durandal_1707: Thanks [16:51] <meekohi> I just realized XCode switched to clang and now none of my stuff works, what a pain :( [16:53] <teratorn> meekohi: well I guess you are fucked [16:53] <teratorn> meekohi: thanks Apple [16:53] <teratorn> can you even have more than one xcode installed at once? [16:53] <meekohi> lol [16:53] <teratorn> is that permitted? [16:54] <meekohi> Even if it is I'm sure you wouldn't want to. [16:54] <teratorn> hahaha [16:55] <meekohi> avcodec_find_encoder keeps returning null when I look for VP8, despite linking in lvpx, and this all worked in the past. [16:55] <meekohi> So I'm a bit stumped what I'm missing. [16:56] <Plorkyeran> you can have multiple versions of 4.x installed [16:56] <Plorkyeran> there's a cli tool to choose which one is active for command line stuff [16:57] <Plorkyeran> probably need a developer account to actually get anything other than 4.5 though [16:58] <meekohi> Any recommendations for debugging missing codec problems? [18:11] <sedulous> hi. i'm trying to remux a live stream with avconv: |avconv -re -i pipe:0 -acodec copy -f mp4 -movflags empty_moov [18:11] <sedulous> however i get: Undefined constant or missing '(' in 'empty_moov' [18:12] <sedulous> how do I add an empty moov atom? [19:54] <amendes365> #Hello everyone. I'm new to ffmpeg and would like to know if its possible to configure a webcam stream with both audio and video, and no flv? [19:55] <klaxa> what container do you want to use instead of flv? [19:56] <amendes365> I've tried mjpeg, which is almost real time, but it has no audio [19:56] <amendes365> @klaxa : anything but asf or flv [19:57] <klaxa> what is your setup so far? [19:58] <amendes365> @klaxa: pastebin.com/nqqE8bdq following this tutorial http://www.area536.com/projects/streaming-video/ [20:01] <klaxa> you could try simply replacing asf by mp4 i think [20:01] <klaxa> just a guess, i'm no expert [20:02] <amendes365> @klaxa: thanks, i'll try it right now [20:15] <amendes365> @klaxa: do you happen to know a good tutorial for streaming a webcam using mp4? [20:15] <amendes365> i'm lost here [20:15] <klaxa> actually no, i rarely use ffserver [20:15] <amendes365> i've found tutorials to stream mp4 videos only [20:15] <amendes365> it's ok... thanks anyway [20:16] <klaxa> using mp4 didn't work properly? [20:16] <klaxa> did it give any errors? [20:33] <klaxa> amendes365: i tried using the mpegts container, mpeg2video and vorbis as codecs, i get segmentation faults with the ffmpeg command though [21:29] <arakn0> hey guys... is it possible to use a bit stream filter in ffserver? any hint? Thanks [21:34] <amendes365> thank you @klaxa [21:36] <klaxa> did you try it and it worked? [21:42] <amendes365> no, but I appreciate the help [21:42] <amendes365> I will reinstall and start again from scratch [21:56] <rainmaker1> Hi, can we perform multipass encoding with live stream? It sounds impossible to me, but maybe... [22:11] <amendes365> @klaxa got the webcam working following http://bit.ly/PfJyd5 but getting 'file mystream.ffm not found' in browser [22:13] <klaxa> amendes365: did you copy the config? because that config isn't complete (lacks xml tags) [22:17] <amendes365> @klaxa here it is: pastebin.com/zTJz4KLm [22:19] <klaxa> swf container and flv codec? i thought you didn't want flash? [22:19] <klaxa> and you wanted audio [22:35] <amendes365> @klaxa thats the goal, but I was focusing on a working solution to stream the webcam and then change the format [22:36] <klaxa> try setting the contaner as mpegts, the videocodec as mpeg2video and vorbis as audio codec [22:36] <klaxa> then try again and see if anything crashes [22:36] <amendes365> ok [22:47] <ghostrookie> hey guys. i've joined 2 videos (X = A+B) with mp4box and when i open X in a player it only shows B and it says that duration of X is the same as for B. but the file size for X is s(X) ~ s(A)+s(B). ffprobe for X says [22:47] <ghostrookie> Duration: 00:00:29.44, start: -12.027937 [22:47] <ghostrookie> that 12.027937 secs is A. how can i reset that "start" to be 0 ? [23:03] <amendes365> @klaxa gives me an error 'Unknown AudioCodec ogg' in ffserver [23:03] <klaxa> replace ogg by vorbis [23:03] <amendes365> ok [23:03] <klaxa> ogg is a container [23:05] <amendes365> yes. ffmpeg shows 'Could not find input stream matching output stream #0.0 ioctl(VIDIOC_QBUF)' [23:06] <klaxa> what command did you run? [23:06] <amendes365> http://pastebin.com/Xws1LUzu [23:07] <amendes365> i ran ffmpeg -r 15 -s 640x480 -f video4linux2 -i /dev/video0 http://127.0.0.1:8090/feed1.ffm [23:09] <klaxa> not sure if this is the error, but ffmpeg isn't providing an audio stream [23:09] <klaxa> so there is nothing for the audio to be encoded [23:09] <amendes365> that's weird [23:09] <klaxa> if you run pulse you can use -f alsa -i pulse and change the recoding device with the pulseaudio volume control [23:11] <amendes365> ok. pulseaudio shows 'No application is currently recording audio' [23:12] <klaxa> it says so while running ffmpeg? [23:15] <amendes365> no, i checked it out in the pulseaudio volume control [23:16] <amendes365> and ffserver shows [ffm @ 0x9a14cf0]Could not find codec parameters (Audio: vorbis, 44100 Hz, mono, s16, 32 kb/s) [23:16] <Endorgh> Hi folks! I don't understand why the output file (aac format, -profile aac_he_v2) generated has a length of only 2 minutes plus 20 seconds, when the input file has 5minutes length..... I'm converting to aac-he v2 into mp4, using version 1.0 under FreeBSD. Any ideas? :S [23:16] <Endorgh> the command is: [23:17] <Endorgh> ffmpeg1 -i input_file.flac -profile:a aac_he_v2 -ab 60k -ar 44100 -ac 2 output_file.mp4 [23:17] <Endorgh> I'm typing wrong any option? [23:20] <durandal_1707> Endorgh: is output really that long? [23:24] <Endorgh> durandal_1707: interesting observation!! The MP4 output file has the same SIZE as a raw aac output file, but when I play both files, the raw aac file doesn't show any metadata although it appears as correct lenght. Otherwise, the MP4 output shows full metadata, but player only can play first 2 min + 20 secs... [23:25] <Endorgh> any incompatibility with mp4 - m4a containers? [23:28] <durandal_1707> Endorgh: does same happens with mkv container? [23:29] <Endorgh> I don't know, let me try it! [23:35] <Endorgh> durandal_1707: with mkv or mka containers it seems to work... [23:36] <Endorgh> what could be the cause? or..., when I can find more information about containers, etc.... [23:37] <durandal_1707> Endorgh: you sure it works? and it is not just number set in container header? [23:38] <Endorgh> mmm [23:39] <durandal_1707> do: ffmpeg -i input -f null - [23:39] <Endorgh> I don't know if this could give any extra info, but ffmpeg says this when encoding into mp4: [23:39] <Endorgh> [flac @ 0x2a4e3110] max_analyze_duration 5000000 reached at 5015510 [23:39] <Endorgh> ok [23:39] <durandal_1707> that is just spam (completly irrelevant) [23:40] <Endorgh> ok [23:44] <Endorgh> this is the output: [23:44] <Endorgh> http://pastebin.com/q2PzAqTL [23:48] <durandal_1707> Endorgh: that is expected, flac support is not broken, but do same with aac in mkv [23:49] <DelphiWorld> Hey [23:49] <DelphiWorld> i'm transcoding a mpeg2 stream into mpeg4 [23:49] <DelphiWorld> how to output it to RAW UDP? [23:51] <Endorgh> durandal_1707: flac support is not broken? what it means? [23:55] <durandal_1707> Endorgh: that duration of flac file is correct [23:58] <Endorgh> durandal_1707: ok, and what is the conclusion? [00:00] --- Sat Oct 20 2012

1 0

ffmpeg-devel.log.20121018
by burek 19 Oct '12

19 Oct '12

[00:27] <durandal_1707> does anyone have wav files with INFO chunks aka tags? [00:29] <durandal_1707> foobar2k can write such files, question are they spec compliant [00:29] <llogan> durandal_1707: doesn't audacity allow you to add those too? [00:31] <durandal_1707> i do not have it installed [00:33] <llogan> oh, I forgot you use FreeBSD [00:35] <durandal_1707> there is package but i do not want to download and install all that stuff just to test possible non-compliant software [00:35] <durandal_1707> and reading forums WMP can not read audactity wav tags [00:36] <durandal_1707> so nobody cares about compatibility [00:37] <llogan> i was just being a douche [00:37] <llogan> as the kids say around here [00:40] <durandal_1707> there is even 'id3 ' chunk in wav [00:41] <llogan> i didn't really look into what audacity does, but here is a file: http://greasedscotsman.com/tone.wav [00:44] <llogan> apparently it ignored album title, track number, and genre although those fields are available in audacity's "metadata editor" [00:46] <durandal_1707> looks like audacity picks safe side, it puts even chunks size [00:51] <durandal_1707> well if invalid INFO chunks are detected whole LIST should be just ignored, instead of failure to decode anything [01:14] <cone-343> ffmpeg.git 03Michael Niedermayer 07ce739e66f4f9: cmdutils: add missing check for ftell() return * 03http://tinyurl.com/clttcht03 [01:14] <cone-343> ffmpeg.git 03Michael Niedermayer 07eb19d89d8eb5: cache: check lseek() return * 03http://tinyurl.com/cctqgvn03 [01:17] <durandal_1707> huh, err_detect flags does not work in lavf [01:20] <durandal_1707> hah -f_err_detect does the job but it is deprecated ..... [01:21] <durandal_1707> michaelni: why is avconv mentioned in ffmpeg? [01:25] <Compn> to explain that ffmpeg supports everything avconv does ? [01:26] <durandal_1707> no, err_detect (same for lavf and lavc) does not work for ffmpeg in lavf context [01:26] <durandal_1707> so there is f_err_detect [01:27] <durandal_1707> if option is named same in lavf and lavc, there is no way to set it to be for lavf(or both) [01:38] <michaelni> durandal_1707, hum, it should work [01:38] <michaelni> i mean same option name in both [01:40] <durandal_1707> if i have only one option it does not work for lavf (obviously) [03:02] <burek> did anyone here compile static ffmpeg for raspberry pi [03:02] <burek> cross-compile [03:02] <bogie> hahah seems someone else got his pi today :D [03:02] <burek> :) [03:03] <burek> very cool thingy :) [03:03] <bogie> i need to think of how i will stream video from my NAS to the PI :) [03:08] <burek> i was following this guide http://lumux.co.uk/2012/07/03/cross-compiling-ffmpeg-for-the-raspberry-pi/ [03:08] <burek> but all the tools from that git are 32bit [03:08] <burek> no 64bit compilers there [03:08] <burek> something is wrong i guess :) [03:14] <burek> I can't believe that many people in deed build ffmpeg on the raspberry itself.. :S [03:14] <burek> how painful is that :S [06:18] <burek> heh :) it works :) [06:18] <burek> let me see now how strong is raspberry pi :) [08:07] <nevcairiel> not very [11:05] <cone-541> ffmpeg.git 03Justin Ruggles 07abd8b9e7e05b: libmp3lame: resize the output buffer if needed * 03http://tinyurl.com/c7qnnvr03 [11:05] <cone-541> ffmpeg.git 03Mans Rullgard 0721fed588cba7: fate: add macros useful for conditionally enabling things * 03http://tinyurl.com/dycrfkt03 [11:05] <cone-541> ffmpeg.git 03Mans Rullgard 07b262e4559710: fate: dependencies for vsynth tests * 03http://tinyurl.com/cpqtf4603 [11:05] <cone-541> ffmpeg.git 03Mans Rullgard 07292d1e787438: fate: dependencies for acodec tests * 03http://tinyurl.com/cfzmql303 [11:05] <cone-541> ffmpeg.git 03Michael Niedermayer 0717e4b0644b5a: Merge commit '292d1e78743855404c7d07e3e7cb3f9c9ae6275b' * 03http://tinyurl.com/c3zzwhh03 [11:05] <cone-541> ffmpeg.git 03Michael Niedermayer 07aadaa8112a19: fate: dependencies for ffmpeg acodec tests * 03http://tinyurl.com/c4dsmmg03 [11:05] <cone-541> ffmpeg.git 03Michael Niedermayer 075806cd93651f: fate: dependencies for ffmpeg vsynth tests * 03http://tinyurl.com/dx4nmnw03 [11:46] <cone-541> ffmpeg.git 03Mans Rullgard 07fa26335003c7: fate: handle lavf test dependencies entirely in make * 03http://tinyurl.com/d2kyb6w03 [11:46] <cone-541> ffmpeg.git 03Mans Rullgard 0736ac9a16a19a: fate: dependencies for seek tests * 03http://tinyurl.com/ccxvz2a03 [11:46] <cone-541> ffmpeg.git 03Michael Niedermayer 079317b73f11c3: Merge commit '36ac9a16a19a365ce58cc871484c20cffe9b6401' * 03http://tinyurl.com/chbxr8a03 [12:18] <durandal_1707> wtf, i did not write single bit of ffv1.3 code [12:19] <cone-541> ffmpeg.git 03Mans Rullgard 07206a070dce5a: fate: list lavfi tests in a makefile * 03http://tinyurl.com/d85qroe03 [12:19] <cone-541> ffmpeg.git 03Mans Rullgard 07b93e934aeea4: mips64: mark hi/lo registers clobbered in MAC64/MLS64 macros * 03http://tinyurl.com/c8ov89p03 [12:19] <cone-541> ffmpeg.git 03Michael Niedermayer 073777e6b3bf65: Merge remote-tracking branch 'qatar/master' * 03http://tinyurl.com/d4tv3do03 [12:49] <merbanan> durandal_1707: well it is written on the internet, must be true [12:50] <durandal_1707> but i did not write 1.3 at all i just cleaned up some code and added more trivial code like extra colorspace stuff for ffv1 [12:57] <merbanan> durandal_1707: you can not argue with the internetz [16:13] <cone-541> ffmpeg.git 03Michael Niedermayer 071350dffdc6ad: riff: dont discard truncated metadata * 03http://tinyurl.com/99x9usw03 [16:13] <cone-541> ffmpeg.git 03Michael Niedermayer 07d0c27e88d2bb: riff: retry reading metadata without padding if it fails with * 03http://tinyurl.com/9d9pekr03 [20:37] <cone-541> ffmpeg.git 03Michael Niedermayer 072b1a2466c7f0: dv: change assert(a2 < 4) to av_assert() * 03http://tinyurl.com/9ml5hdu03 [20:45] <cone-541> ffmpeg.git 03Michael Niedermayer 0723b203014f5d: indeo4: prevent printing uninitialized variable * 03http://tinyurl.com/8v8v54u03 [21:13] <cone-541> ffmpeg.git 03Michael Niedermayer 0797d1cb5cd48c: bmv: remove unreachable default case * 03http://tinyurl.com/8lt5a7503 [22:11] <cone-541> ffmpeg.git 03Michael Niedermayer 072472f3facbb4: lzwenc: change assert to av_assert * 03http://tinyurl.com/9q6952s03 [22:11] <cone-541> ffmpeg.git 03Michael Niedermayer 075537c92f84db: mpegvideoenc: check return value of ff_MPV_frame_start() * 03http://tinyurl.com/93tku5l03 [22:11] <cone-541> ffmpeg.git 03Michael Niedermayer 07c753b56b4d56: ff_convert_matrix: fix integer overflow * 03http://tinyurl.com/8rfd6gw03 [22:46] <ubitux> divVerent: you there? [22:49] <cone-541> ffmpeg.git 03Clément BSsch 07e807a2b64899: lavc: add raw text subtitles decoder. * 03http://tinyurl.com/8w34gf703 [22:50] <ubitux> divVerent: anyway, see above ^ [23:05] <durandal_1707> saste: ping [23:05] <saste> durandal_1707, pong [23:06] <durandal_1707> saste: you ok with idcinvideo explanation about fate ref chage? [23:06] <durandal_1707> *change [23:06] <saste> durandal_1707, yes, please add the explanation to the commit log [23:06] <saste> the other remarks are nits that you can safely ignore [23:07] <durandal_1707> i'm writing yet another demuxer [23:09] <saste> durandal_1707, BTW what happened of the XPM encoder/decoder? [23:09] <durandal_1707> no motivation [23:10] <ubitux> :( [23:10] <saste> well maybe i'm going to "adopt" it if you don't plan to work on it [23:11] <durandal_1707> saste: i have hnm4 (intra only working) if you want to work on that too [23:11] <saste> so i can use the original XPM glyphs of the effectv matrix filter [23:12] <saste> ham4? [23:12] <durandal_1707> hnm4 - game codec [23:13] <saste> ok i was confusing with IFF HAM images [00:00] --- Fri Oct 19 2012

1 0

ffmpeg.log.20121018
by burek 19 Oct '12

19 Oct '12

[00:43] <arakn0> I'm trying to do point to point streaming using rtsp but not working yet. The command im using and its output: http://pastebin.com/2TUyMRq6 Any suggestion/example that can iluminate me? Thanks! [00:46] <burek> arakn0, why not just use udp output directly if you need p2p [00:47] <burek> also, your ffmpeg might be old [00:47] <burek> try 1.0 [00:49] <arakn0> burek: it's one of my requirements. In my scenario, I have an embeded device that acts a rtsp client. [00:50] <burek> req for what? [00:50] <burek> oh rtsp [00:50] <burek> ok [00:50] <burek> can you update your ffmpeg [00:51] <arakn0> sure [00:51] <arakn0> I built ffmpeg on the 3rd of nov [00:51] <arakn0> but i wonder if I'm in the right track [00:51] <burek> it doesn't matter when did you build it, but what source code did you use [00:55] <arakn0> ffmpeg version N-45010-g5e6439a [00:55] <llogan> arakn0: you said you built ffmpeg but your pastes look like the repo version [00:55] <burek> that one should be ok [01:01] <arakn0> burek: weird. I just pulled master and rebuilding. [01:03] <arakn0> just in case, but I'm not sure that my ffmpeg version is the problem [01:04] <arakn0> more concern about the options [01:04] <burek> we'll see :) [01:04] <arakn0> that I'm using [01:05] <arakn0> burek: just fyi, I've tried rtp over udp and tcp and both work with no issues [01:19] <addisonj> hey, so I am having some issues trying to do some processing of multicast udp stream (ffprobe gives me this: mpeg2video (Main), yuv420p, 1920x1080 [PAR 1:1 DAR 16:9], 65000 kb/s, 30.22 fps, 29.97 tbr, 90k tbn, 59.94 tbc), but I am getting lots of artifacts and corruption, when I use vlc however the stream looks great [01:33] <addisonj> burek: thanks for the tip, here is what I am trying to do: http://pastebin.com/tn1sH7Dj [01:33] <arakn0> burek: same problem [01:33] <burek> addisonj, you are not using ffmpeg [01:34] <burek> join #ubuntu and ask them to explain to you what are you using actually [01:34] <burek> or read this [01:36] <addisonj> burek: oh sorry, just did this on a new machine, forgot about the fork, will switch and try again [01:36] <burek> addisonj, try using ffmpeg's git [01:41] <arakn0> the command and its output to start an rstp server: http://pastebin.com/zKuF8Kzq [01:43] <burek> you are still using old ffmpeg :) FFmpeg version 0.6-4:0.6-2ubuntu6.3 [01:43] <burek> latest is like 1.0 [02:07] <arakn0> burek: i don't know what happened. [02:07] <arakn0> here it's the right version: http://pastebin.com/b6LbnGkk [02:08] <burek> [tcp @ 0x2664c40] TCP connection to 0.0.0.0:554 failed: Connection refused [02:09] <arakn0> I've seen that. But I don't wan to connect to that IP. I want to bind to that IP and open that port [02:09] <burek> ffmpeg is not an rtsp server [02:09] <arakn0> So, I can stream point to point with rtp...but not rtsp... [02:10] <burek> you can feed the rtsp server [02:10] <burek> but if you are going to use p2p [02:10] <burek> what's the point of rtsp then [02:10] <burek> why not just send udp/rtp directly [02:11] <arakn0> the requirements [02:11] <arakn0> I know, I know... [02:11] <burek> well ok [02:11] <burek> then don't use ffmpeg [02:11] <burek> use something else, vlc for example [02:12] <burek> use something that can act as an rtsp server [02:12] <burek> i think ffserver also can [02:12] <burek> let me check [02:12] <arakn0> yes, I was going to mention that [02:12] <arakn0> ffserver [02:12] <burek> http://ffmpeg.org/sample.html [02:12] <burek> ctrl+f rtsp [02:14] <arakn0> yes, I see that. I was going in that direction too. It wasn't 100% if I could stream using just ffmpeg [02:14] <burek> http://ffmpeg.org/trac/ffmpeg/wiki/Streaming%20media%20with%20ffserver [02:14] <burek> that might help too [02:15] <burek> the only problem is I'm not sure how to specify -vcodec copy inside ffserver's config file [02:16] <burek> but since you want to stream a file [02:16] <burek> there is no need for that and you can safely ignore me :) [02:16] <burek> just see the sample and rtsp examples there :) [02:23] <arakn0> burek: thanks so much. I'm going to study it and I'll let you know! [02:25] <burek> :beer: :) [02:36] <addisonj> burek: our issues disappear with ffmpeg, avconv-- [02:36] <addisonj> thanks for the push in the right direction :) [02:36] <burek> +1 :) [03:53] <dericed> Hi all, for an unknown audio source is there a way to transcode it to an uncompressed output without possible loss in bit depth? For instance with video I can use -c:v rawvideo which will produce uncompressed video in native chroma subsampling and bit depth. There doesn't seem to be a rawaudio equivalent. [05:03] <jivetalkingturke> hello i have a question, i need to input 3 videos crop and position into one ouput.. can this be done? [05:03] <jivetalkingturke> like a tile effect [05:04] <llogan> jivetalkingturke: sup cracka. [05:04] <jivetalkingturke> i am capturing 3 webcams at 720p and would like to join them into one video output [05:05] <llogan> it is possible. let me find an example. [05:05] <jivetalkingturke> ok thanks [05:06] <dericed> the side-by-side example in the overlay filter documentation is close https://ffmpeg.org/ffmpeg.html#overlay-1 [05:06] <jivetalkingturke> ok.. thanks i'll have a look [05:06] <jivetalkingturke> overlay is the word i needed.... [05:08] <jivetalkingturke> thank you.. i'll have a look at overlaying and see what i can get... [05:09] <llogan> that should be a good starting point. my example is ancient anyway [05:10] <llogan> this binder has too many examples [05:10] <jivetalkingturke> no worries thanks lllogan... [05:10] <jivetalkingturke> and thank you dericed [05:11] <dericed> ffmpeg -f lavfi -i testsrc -f lavfi -i testsrc -f lavfi -i testsrc -filter_complex "[0:0]pad=iw*2:ih*2[a];[1:0]negate[b];[2:0]vflip[c];[a][b]overlay=w[x];[x][c]overlay=w:h" -y -t 1 out.avi [05:11] <dericed> aw, he's leave the room once I finish an example [05:12] Action: llogan gives dericed +10 FFmpeg Bucks anyway [05:12] <dericed> sweet FFmpeg Bucks!! can i apply to a trac bounty? [05:16] <llogan> dericed: consider adding your example here: https://ffmpeg.org/trac/ffmpeg/wiki/FilteringGuide [05:17] <llogan> i see similar requests every once in a while [05:19] <dericed> llogan: working on it [05:21] <dericed> llogan: where do i upload thumbnails for wiki? [05:22] <llogan> dericed: click on "Attach file" button on bottom of page, IIRC. [05:23] <dericed> thx [05:23] <llogan> ah...duh. i forgot there already is an example at FancyFilteringGuide [05:23] <llogan> s/Guide/Examples [05:24] <llogan> but a three or 4 input version would still be interesting [05:24] <llogan> dericed: sorry about that [05:25] <llogan> and that example still only have one input. [05:31] <dericed> llogan: my image doesn't work but it is added to here: https://ffmpeg.org/trac/ffmpeg/wiki/FilteringGuide [05:32] <llogan> dericed: thanks [05:39] <llogan> dericed: i fixed the image. apparently you have to attach it to the specific page you are editing. i didn't know that until now, and if i did i forgot because my memory is => lesser ape. [12:00] <zamba> hi there.. i want to create a timelapse of some web cam images [12:01] <zamba> the images are sorted like: cam/<year>/<month>/<day>/img-<year><month><day>-<hour><minute>.jpg [12:01] <zamba> and i want to pick one image each day [12:02] <zamba> any smart way ffmpeg can do that? [12:02] <Tjoppen> write a shell script to rename them to 1.jpg 2.jpg etc. [12:02] <Tjoppen> that's about it [12:03] <zamba> *shrug* [12:03] <zamba> ffmpeg can't take its input from a text file? [12:03] <zamba> so i can just put the absolute paths in there? [12:04] <Tjoppen> ffmpeg can't take explicit names in image sequences, AFAIK [12:04] <zamba> maybe look to mplayer, then [12:05] <Tjoppen> no, just write a shell script to rename them [12:05] <zamba> there's LOTS of images [12:05] <Tjoppen> or, copy [12:05] <Tjoppen> so? just throw disk and time at the problem. that or use an NLE like kdenlive [12:05] <zamba> and i don't want to rename any images, that's irreversible, and i don't want to copy them either, because that's a huge overhead [12:05] <zamba> i use mplayer instead [12:05] <Tjoppen> ln -s then [12:06] <zamba> ok, that's more interesting [12:06] <zamba> ln is a more interesting approach, yeah [12:06] <zamba> but the find job takes forever..... :p [12:07] <zamba> 1440 images a day for 6 months :) [12:08] <Tjoppen> somethin glike I=0; for f in `find .|grep jpg` ; do ln -s $f $I.jpg && I=`expr $I + 1` ; done [12:08] <Tjoppen> then ffmpeg -i %d.jpg yadda yadda [12:08] <zamba> yes :) [12:09] <zamba> but they need to be sorted [12:09] <Tjoppen> I hacked support for pipe separated image sequence names at work, but it's somewhat hackish [12:09] <Tjoppen> just pipe the find output to sort then [12:10] <Tjoppen> lunch [12:18] <tapas> hi, i'm fiddling around with avconv and ffserver.. [12:19] <tapas> running the server works fine.. just avconv doesn't like to produce the feed: http://pastesite.com/82142 [12:20] <tapas> i also tried just plain avconv -f video4linux2 -i /dev/video0 http://localhost:8090/feed1.ffm [12:20] <tapas> same error [12:20] <tapas> i wonder what avconv tries to tell me.. what codecs did it try? what paremeters did it try? [12:20] <tapas> which ones failed? :D [12:21] <tapas> works fine in vlc btw.. [just to note that the hardware does work] [12:22] <tapas> and in mplayer.. [12:26] <tapas> hmm, i guess it's a broken package in ubuntu.. [12:44] <zamba> i tried building my time lapse now, but i'm only getting 9 seconds worth of video [12:45] <zamba> my files are named 0001.jpg to 0584.jpg [12:45] <zamba> and i'm running: ffmpeg -i %04d.jpg -r 30 -s 640x480 -vcodec libx264 -b 2000k cam-01.mp4 [12:46] <zamba> Input #0, image2, from '%04d.jpg': [12:46] <zamba> Duration: 00:00:23.36, start: 0.000000, bitrate: N/A [12:46] <zamba> Stream #0.0: Video: mjpeg, yuvj420p, 2560x1920 [PAR 1:1 DAR 4:3], 25 fps, 25 tbr, 25 tbn, 25 tbc [12:46] <zamba> why does it say duration 00:00:23.36? how can it know that? [12:48] <durandal_1707> from fps and number of images [12:48] <zamba> but why is only a 9 second long video created? [12:49] <zamba> ah, error in input [12:49] <zamba> three broken images [12:49] <zamba> let's try that again [12:50] <zamba> argh.. the images have to be in order [12:51] <zamba> and there's three jpegs that are broken [12:51] <zamba> ffmpeg has no option to "skip" broken input? [12:56] <zamba> just removed those images.. size 0 anyway [12:56] <zamba> that was the problem [13:51] <jivetalkingturke> hello... was wondering if someone could help me with my terminal command for ffmpeg.. [13:52] <jivetalkingturke> i am trying to run multiple jobs at the same time [13:53] <jivetalkingturke> i don't want to use a bash script.. would rather do it all from terminal [13:53] <jivetalkingturke> i am using 3 webcams 1 audio interface and 1 microphone [13:54] <jivetalkingturke> would like to capture them at the same time [14:23] <jivetalkingturke> anyone know about multiple simultaneous input? [15:07] <t4nk124> hey guys, I'm updating my project to use the latest version of ffmpeg - it's working for single channel streams but anything stereo is coming out the other side very distorted - the playback rate is fine but the sound is like it's dropped 2 octaves - example: http://hosting.ispyconnect.com/example.mp3 [15:09] <durandal_1707> t4nk124: you have not support for planar sample formats [15:09] <durandal_1707> so only mono will work [15:09] <durandal_1707> almost every decoder is switching to that [15:10] <durandal_1707> some, like flac and alac supports both [15:11] <t4nk124> when i try and use AV_SAMPLE_FMT_S16 I get an error: Specified sample_fmt is not supported. [15:11] <durandal_1707> as i said only some decoders support selecting sample fmt [15:12] <t4nk124> ... but it works if it's single channel [15:12] <t4nk124> and this used to work with an old version of ffmpeg [15:12] <durandal_1707> t4nk124: as i said decoders switched to planar sample format [15:13] <t4nk124> ah ok so basically what are my options? [15:13] <durandal_1707> and interleaving is now handled in libswresample [15:13] <durandal_1707> interleave output manually or use libswresample [15:14] <durandal_1707> you will need to take care of channel layout - that it match with output device [15:14] <durandal_1707> and libswresample AFAIK do not support that [15:14] <durandal_1707> ^custom remapping [15:15] <durandal_1707> so you are force to do it manually until this get cleaned up either in libswr or in lavc itself [15:15] <durandal_1707> *forced [15:16] <t4nk124> do you know of any examples? I'm way out of my depth :( [15:16] <durandal_1707> but if you are interested in stereo this should not matter at all [15:16] <t4nk124> ideally it would spit out a stereo mp3 [15:17] <durandal_1707> you meen encoding? [15:18] <durandal_1707> *mean [15:18] <t4nk124> yes [15:19] <durandal_1707> you just need to deinterleave samples [15:20] <t4nk124> ok sounds promising - how do i do that? [15:20] <durandal_1707> but this is overkill considering that libmp3lame support interleaved samples, so fork just removed that code.... [15:21] <durandal_1707> actually not, it was never done that way, deinterleaving was done in encoder istself :roll [15:22] <t4nk124> yeah this was all working fine in the previous version - now i can't encode anything stereo into mp3 using AV_SAMPLE_FMT_S16 [15:23] <durandal_1707> you need to give encoder each channel in separate plane [15:24] <t4nk124> so i should be using AV_SAMPLE_FMT_S16P ? [15:24] <durandal_1707> previously: Chan.1_sample.1|Chan.2_sample.1|Chan.1_sample.2|... [15:26] <durandal_1707> now: Chan.1_sample.1|Chan.1_sample.2|Chan.1_sample.3 | ... | Chan.1_sample.N | Chan.2_sample.1 | ... | Chan.2_sample.N [15:27] <durandal_1707> t4nk124: yes, but make sure you give deinterleaved samples [15:30] <t4nk124> hmmm... all iv'e got coming into the method is a pointer to a byte array [19:40] <Amnesia> hi question, I'm trying to decode my webcam's output to something compatible in multiple oses using a pipe [19:40] <Amnesia> what type of codecs/contains would be advisable? [19:40] <Amnesia> (it has to become a stream) [19:47] <Endorgh> hi everyone! After 3 hours searching on internet, I haven't found out how I can preserve metadata after a conversion. I'm using version 0.7.13 under FreeBSD 9.0. Any suggestions? [20:13] <creep> hi [20:14] <creep> do somebody know how to set the headphone's left to audio track 1 right to track 2 converted to mono ? [20:14] <creep> or maybe usgin gmplayer? [20:45] <divVerent> creep: ffmpeg sure can do it using -filter_complex [20:45] <divVerent> but I don't know exactly how [20:45] <divVerent> look for "amerge" in the manpage, it does something like that [20:55] <alezakos> Good morning/evening. Can I make ffmpeg produce TIFF images in the RGB colourspace? [20:57] <durandal_1707> alezakos: several rgb colourpspaces are supported (except planar)) [21:01] <arakn0> I'm trying to stream an mp4 file over RTSP using just ffserver, but for some reason ffserver crashes. http://pastebin.com/pjGDzRH8 Any help/hint ??? [21:02] <alezakos> Good. As my experience on this field is zero, could you please provide me with help on how to split a video to tiffs readable by imagemagick? [21:05] <durandal_1707> alezakos: ffmpeg -i input out%06d.tiff [21:09] <alezakos> Unfortunately, that command produces tiffs in the YCbCr colourspace, and imagemagick doesn't support it (the colours are weird) [21:10] <klaxa> arakn0: not sure if this is still up to date, but the documentation says streaming from files is broken: http://ffmpeg.org/ffserver.html#What-can-this-do_003f [21:11] <klaxa> alternatively you could try using ffmpeg as a feed [21:11] <klaxa> i think? that kinda worked for me [21:12] <durandal_1707> alezakos: your input is than not in rgb colorspace which means you need to specify it via -pix_fmt rgba [21:14] <Endorgh> Hi folks! I've ffmpeg compiled with 'libaacplus' on FreeBSD, and it doesn't keep into output file the metadata of input file. I've tried all -map options but nothing works. Any ideas? [21:14] <alezakos> durandal_1707: That works perfectly! Thank you very much! [21:14] <arakn0> klaxa: damn.... but it works for mpg :( [21:15] <durandal_1707> Endorgh: what container? [21:15] <klaxa> arakn0: like i said, no idea if that's still up to date, i have given up on setting up ffserver myself, because it lacks some functionality i want (streaming matroska with subtitles) [21:17] <Endorgh> durandal_1707: flac to aac [21:20] <Endorgh> durandal_1707: exactly this command: ffmpeg -i file_in.flac -ab 60k file_out.aac [21:21] <Endorgh> durandal_1707: if I type ffmpeg -i file_in.flac -ab 60k file_out.mp4 the file_out.mp4 has the medatada, but its length is only 2 minutes, not the same length of input file... Is very strange [21:21] <JEEB> uhh, raw AAC doesn't have anything really to put metadata into [21:21] <JEEB> a container like "MP4" is needed to accomodate that [21:24] <Endorgh> JEEB: hence, is needed to specify ".mp4" extension [21:24] <Endorgh> ? [21:24] <durandal_1707> Endorgh: flac to mp4 with libaacplus works fine here [21:25] <JEEB> Endorgh, or anything else that has possible spots for the tags you need, yes [21:25] <Endorgh> durandal_1707: do you recommend libaacplus in place of FAAC? [21:25] <Endorgh> ok I understand... [21:25] <JEEB> no, you should use fdk [21:25] <JEEB> instead of both faac and libaacplus [21:25] <JEEB> handles both HE-AAC and LC-AAC [21:25] <JEEB> https://github.com/mstorsjo/fdk-aac [21:26] <Endorgh> ohh, great info! [21:26] <JEEB> you can't give out binaries with it, just like faac and libaacplus, but it's the least bad aac encoder atm :P [21:27] <durandal_1707> JEEB: you did blind test? [21:28] <JEEB> not me, but in general that has been the concensus. It's the fraunhofer encoder and it definitely is better than faac and vo-aacenc with LC [21:28] <Endorgh> JEEB: hehe, the least bad aac encoder. Then, what is the best one? [21:28] <divVerent> durandal_1707: really need no blidn test to compare with faac [21:29] <divVerent> given I DO hear difference between 128kbit/s faac and original [21:29] <divVerent> faac really IS that bad [21:29] <JEEB> and it doesn't sound bad with HE-AAC so it's at least on par with libaacplus [21:29] <JEEB> yeah, faac is pretty bad [21:29] <JEEB> and vo-aacenc is at times even worse than the ffaac [21:29] <divVerent> blind test may be needed for 192kbit/s faac :P [21:29] <JEEB> (internal aac encoder) [21:29] <cbsrobot> JEEB: sorry for the late replay, but about the libaac_fdk delay issue [21:29] <divVerent> and the ffmpeg builtin "aac" encoder is buggy [21:29] <divVerent> it does clipping ;) [21:29] <divVerent> or a noise similar to that [21:29] <JEEB> well, every encoder does that [21:29] <cbsrobot> I was not able to get it to work [21:29] <JEEB> oh [21:30] <cbsrobot> I thought it is a samplerate issue [21:30] <divVerent> I mean, apparently the "aac" encoder in ffmpeg has some overflow issue somewhere [21:30] <divVerent> it really is only good if "there is nothing else available" [21:30] <cbsrobot> but it does not seem so [21:30] <divVerent> and if you really NEED aac [21:30] <JEEB> divVerent, more like the ffaac encoder was left to be after some development :/ [21:30] <divVerent> the aac codec needs -strict -2 for a reason :P [21:30] <JEEB> at least it's better than the WMA encoder [21:30] <JEEB> lol [21:31] <divVerent> jeeb:didn't test THAT one [21:31] <JEEB> Endorgh, not sure -- Fraunhofer or Apple's (Dolby's?) [21:31] <divVerent> but I actually did use the aac encoder for a while [21:31] <divVerent> and so did I use faac [21:31] <divVerent> for libfdk_aac I still have to look for the artifacts ;) [21:31] <JEEB> well, the internal aac encoder is the best that you can still distro [21:31] <divVerent> even though I changed bitrate from 128 to 96 when changing from faac to libfdk_aac [21:31] <JEEB> vo_aacenc according to a blind test is not really better [21:32] <divVerent> jeeb: that I don't believe [21:32] <divVerent> unless vo_aacenc has similar issues :P [21:32] <divVerent> but well [21:32] <divVerent> the average case of the "aac" codec is fine [21:32] <JEEB> it's just the reference code >_> [21:32] <divVerent> it just has a bad worst case [21:32] <divVerent> if you want, I can switch to using it again [21:32] <JEEB> (and yes, I know that fraunhofer and friends base on the reference code, too) [21:32] <divVerent> and wait till I get a good sample to demonstrate the "clipping" issue [21:32] <JEEB> but at least fraunhofer generally has been optimized [21:32] <divVerent> when I did encounter one in the past, I verified the weird noise goes away when reducing volume before encoding [21:33] <cbsrobot> JEEB: Do you know how to prevent the .5 sec delay in libfdk_aac ? [21:33] <JEEB> cbsrobot, if it's the priming samples you need to signal them in the container and make sure the thing that's playing it back actually understands the way of setting priming samples [21:34] <JEEB> I think ffmpeg should support that (maybe even by default?), and L-SMASH's "MP4" muxer does it too [21:34] <cbsrobot> aac in mp4, played back with quicktime [21:34] <JEEB> basically you should check the first PTS of the first audio frame/packet, you should have the amount of priming needed there [21:34] <JEEB> (there's a negative pts IIRC) [21:35] <cbsrobot> how do I check that ? [21:35] <cbsrobot> hexedit ? [21:35] <JEEB> no idea, people usually use the API to get that figure :P [21:35] <cbsrobot> or ffprobe ? [21:36] <JEEB> also check if the audio length is the same in ffmpeg before and after [21:36] <JEEB> when you do f.ex. ffmpeg -i derp.mp4 and have it show stuff [21:36] <JEEB> (ffprobe probably shows too) [21:37] <cbsrobot> Ok thanks [21:38] <cbsrobot> I'll try and ping you later (maybe) [21:38] <JEEB> basically if there's a length difference then priming samples aren't for whatever reason pushed there :P [21:38] <JEEB> (output should be somewhat longer) [21:39] <JEEB> if that is the case, you'd have to get the amount of priming needed and re-mux it with L-SMASH's muxer or something from raw AAC and setting the amount of priming samples [21:41] <cbsrobot> basicly it a mov with prores and pcm sound convertet to h264 and aac in mp4 [21:48] <Endorgh> ffmpeg -i input_file.flac -ab 60k output_file.mp4, generates an output file with less time length than the original input file. Why?? O_O [21:49] <Endorgh> it results incomprehensible for me... [21:53] <durandal_1707> Endorgh: i said i can not reproduce it, you are using buggy version [21:55] <creep> nah i set software mixer on gmplayer and opened up same video 2 times, synced them, and set one language with balance left on one, and the other language on the other gmplayer on the right, i can watch CSI in both language now [21:56] <Endorgh> durandal_1707: ok, understood! [21:58] <durandal_1707> Endorgh: what version you are using? [22:00] <Endorgh> ffmpeg-0.7.13_6,1 [22:01] <JEEB> uhh, does that even have libaacplus >_> [22:01] <JEEB> also, that is /quite/ old [22:02] <Endorgh> its the default version on FreeBSD port collection [22:02] <JEEB> you should build it yourself then, current git if possible [22:03] <JEEB> you are not going to get much support or help with as old version as that. [22:03] <JEEB> as sad as it is [22:03] <durandal_1707> Endorgh: there is ffmpeg-011 and ffmpeg-devel [22:03] <durandal_1707> and ffmpeg1 (1.0) [22:06] <Endorgh> durandal_1707: mmm d'oh! [22:37] <dominikd> is there a "just work" setting for ffmpeg? [22:38] <dominikd> as in eat as many stream encodings as possible [22:38] <dominikd> without throwing errors [23:08] <Amnesia> question, what are the required arguments for webm? [23:15] <wlan2> required? use extension .webm (?) [23:15] <wlan2> And then it automatically chooses vp8 and vorbis [23:15] <Amnesia> hm don't know what parameters to give with webm [23:16] <Amnesia> Encoder (codec none) not found for output stream #0:0 [23:16] <Amnesia> and [23:16] <Amnesia> Could not write header for output file #0 (incorrect codec parameters ?) [23:18] <Amnesia> ffmpeg -f video4linux2 -i /dev/video0 -vcodec mpeg4 -f webm -y /dev/stdout | nc -lp 5555 [23:21] <dirty_d> Stream #0:2(eng): Subtitle: subrip [23:21] <dirty_d> is this the same as SRT fromat? [23:25] <iive> yes, srt is subrip [23:25] <Amnesia> iive: how can I determine what arguments are required for a specific container/codec ? [23:26] <iive> no idea. sorry. [23:26] <iive> btw, if you try to stream it through a pipe, nut may be the container. [23:27] <creep> ok mplayer hung again, shit "MPlayer interrupted by signal 2 in module: unknown" would someone tell mplayer devs that module unknown has bugs? [23:27] <Amnesia> hm ok thx [23:27] <Amnesia> I seriously suck at ffmpeg-.- [23:27] <iive> flv may be a little slow, there is v3 in development, no idea how to select it. [23:27] <iive> creep: LoL! [23:28] <iive> what was signal2? bus error or ctrl+c? [23:29] <Amnesia> iive: looks like asf is my best bet so far:) [23:29] <Amnesia> thanks [23:30] <creep> iive<< controlc did the job now [23:31] <creep> there are really lame errors like double pressing play, or pressing controls while playlist is open resulting infinite loops, or setting window to always on top and hitting some buttons on controls [23:56] <iive> creep: interesting, i assume these are gmplayer related. could you try to fill some of them in the bugzilla, in case they are not already there. [00:00] --- Fri Oct 19 2012

1 0

ffmpeg-devel.log.20121017
by burek 18 Oct '12

18 Oct '12

[00:11] <cone-412> ffmpeg.git 03Michael Niedermayer 076d55a40b0080: mov: print warning if ff_get_wav_header() fails * 03http://tinyurl.com/9vq6jdd03 [00:11] <cone-412> ffmpeg.git 03Michael Niedermayer 07cb65b32c97b0: mp3enc: remove unneeded null ptr check * 03http://tinyurl.com/8lgblqe03 [00:11] <cone-412> ffmpeg.git 03Michael Niedermayer 07a30972609ca3: yuvPlanartoyuy2_c: fix sign extension * 03http://tinyurl.com/8mblbvd03 [00:11] <cone-412> ffmpeg.git 03Michael Niedermayer 07a07e9d72a1d9: yuvPlanartouyvy_c: fix sign extension * 03http://tinyurl.com/9zhvwcu03 [00:11] <cone-412> ffmpeg.git 03Michael Niedermayer 073e0b29ccd075: ffmpeg: Make video filter graph reinit user selectable * 03http://tinyurl.com/9grcj2w03 [00:11] <cone-412> ffmpeg.git 03Michael Niedermayer 076cbb8a450f16: libavfilter/buffersrc: Do not fail hard on changes of input parameters. * 03http://tinyurl.com/8qmelxq03 [00:11] <cone-412> ffmpeg.git 03Michael Niedermayer 075d2b8850746b: lavfi: limit matching w/h/fmt asserts to non scale filters * 03http://tinyurl.com/98mjdlp03 [00:32] <michaelni> saste, your patch breaks compex filters [00:32] <michaelni> example: ffmpeg -i matrixbench_mpeg2.mpg -i tests/lena.pnm -filter_complex 'overlay=10:main_h-overlay_h-10' out.avi [00:32] <saste> michaelni, which patch? [00:33] <michaelni> "lavfi/scale: accept named option, make parsing more robus" [00:33] <saste> uhm, how's that? [00:33] <ubitux> auto insert maybe? [00:34] <michaelni> dont ask me iam just testing as i had a bad feeling [00:34] <saste> No option name near '(null)' [00:34] <saste> ok got it [00:35] <saste> well that is a pre-existing bug, but sure it must be solved [00:35] <saste> we don't have FATE test for -filter_complex? [00:36] <saste> i run make fate with my patchset and didn't catch the issue [00:37] <saste> goodnight people [00:37] <michaelni> n8 saste [00:37] <Daemon404> [18:37] <@Daemon404> how do i make a build-only fate instance? [00:37] <Daemon404> same question here [00:39] <michaelni> Daemon404, try makeopts="FATE=" [00:39] <Daemon404> "try" ? [00:40] <michaelni> sorry wrong escaping, try 'makeopts="FATE="' [00:40] <michaelni> :) [00:41] <ubitux> still using that trick? [00:42] <ubitux> is that really the official way? [00:42] <ubitux> i mean, that's the way i workaround the problem but.. :D [00:42] <Daemon404> 'official [00:42] <Daemon404> well once the tilera patch is merged [00:42] <Daemon404> ill add my tilera build to ffmpeg's fate too [00:42] <ubitux> Daemon404: you have some config examples on lucy.pkh.me btw [00:42] <ubitux> http://lucy.pkh.me/fate-configs/ [00:45] <michaelni> i never tried to find another way because it works ... [00:45] <ubitux> :D [00:46] <ubitux> i think i saw a build_only patch somewhere on libav [00:46] <ubitux> so even if there is another solution, it's likely to be hacky as well [01:25] <ubitux> michaelni: examples make works for me: http://pastie.org/5070391 [01:28] <ubitux> michaelni: maybe a shared build? [01:45] <michaelni> #define CONFIG_SHARED 0 [01:47] <michaelni> theres a "-L/usr/local/lib" in there, if i remove it it links [01:48] <michaelni> cc muxing.o -pthread -Lpc-uninstalled/../../../libavdevice -L/usr/local/lib -Lpc-uninstalled/../../../libavfilter ... [01:51] <ubitux> is it better if you replace: [01:52] <ubitux> CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS) [01:52] <ubitux> LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS) [01:52] <ubitux> with: [01:52] <ubitux> CFLAGS += $(shell pkg-config --cflags $(FFMPEG_LIBS)) [01:52] <ubitux> LDLIBS += $(shell pkg-config --libs $(FFMPEG_LIBS)) [01:52] <ubitux> ? [01:53] <michaelni> no, same problem [01:53] <ubitux> ok :( [01:54] <ubitux> well i have to wake up ~5h so time for me to sleep.. [01:54] <ubitux> 'night [01:56] <michaelni> ubitux, good night [02:45] <cone-412> ffmpeg.git 03Michael Niedermayer 077b8fd298161d: examples/muxing: fix video pts * 03http://tinyurl.com/9lu7wzx03 [02:54] <cone-412> ffmpeg.git 03Michael Niedermayer 07adbb75dbd8ea: mov: fix time types related to mov_metadata_creation_time * 03http://tinyurl.com/8t2524303 [03:37] <cone-412> ffmpeg.git 03Michael Niedermayer 074e2e3d943ef0: ffv1: fix packed rgb with 1.3 * 03http://tinyurl.com/9h7xxos03 [07:46] <cone-343> ffmpeg.git 03Clément BSsch 07711ffb84df29: lavf/swfdec: support DefineBitsLossless{,2} tag. * 03http://tinyurl.com/93kes8303 [10:05] <ubitux> is the aac ssr profile in use? [10:05] <ubitux> i mean, is there some encoders making use of it, and at least a few decoders supporting it? [10:51] <nevcairiel> i dont think i ever saw a file [10:51] <nevcairiel> and usually people give me the weirdest sample files [10:53] <ubitux> ok :) [11:17] <durandal_1707> how long in minutes is 2gb wav file (44100hz, 16bps, 2channels) ? [11:19] <cptspiff> 2*1024*1024*1024/(44100*2*2) [11:19] <nevcairiel> 203 minutes or so [11:28] <merbanan> ubitux: I saw one ssr file [11:28] <ubitux> and then you woke up ? [11:28] <ubitux> according to a comment on top of the aacdec, there was a gsoc to support it [11:29] <ubitux> i guess it never reached the review process? [11:29] <merbanan> no diego wanted the ssr stuff from the gsoc resurrected [11:29] <merbanan> then I woke up [11:29] <ubitux> :D [11:30] <merbanan> no one uses it though [11:30] <merbanan> faad supports it iirc [11:30] <durandal_1707> i cant generate silence with 2 chan with aevalsrc [11:31] <ubitux> aevalsrc=0 ... -ac ? :P [11:32] <durandal_1707> i get mono [11:32] <durandal_1707> and if i specify layout i get error [11:32] <ubitux> aevalsrc=0:0 ? [11:33] <durandal_1707> thanks [12:02] <cone-343> ffmpeg.git 03Michael Niedermayer 07364c60bf64a2: sws-test: parse command line args before initing contexts * 03http://tinyurl.com/cdmbgcv03 [13:59] <cone-343> ffmpeg.git 03Paul B Mahol 07d6ea59b8608d: tta: datalen is unsigned integer per reference library * 03http://tinyurl.com/cvjzkyg03 [14:13] <cone-343> ffmpeg.git 03Paul B Mahol 071ade37ae9c2b: lavc/tta: use meaningful error codes * 03http://tinyurl.com/cwts6hp03 [14:21] <cone-343> ffmpeg.git 03Mans Rullgard 073b20eb25e7b9: avserver: move avserver-specific code from ffmdec.c to avserver.c * 03http://tinyurl.com/cgnopxh03 [14:21] <cone-343> ffmpeg.git 03Luca Barbato 0721de6ba5c12f: nut: export codec_tag provided by rawvideo * 03http://tinyurl.com/cd9zvdl03 [14:21] <cone-343> ffmpeg.git 03Mans Rullgard 071fbaabefc4bb: network: #include stdint.h in network.h * 03http://tinyurl.com/c6x2g4d03 [14:21] <cone-343> ffmpeg.git 03Diego Biurrun 07c0a6cac2920e: fate: Add rangecoder test * 03http://tinyurl.com/clec3rs03 [14:21] <cone-343> ffmpeg.git 03Luca Barbato 079a978b334b9b: ffv1: K&R formatting cosmetics * 03http://tinyurl.com/c54ppwe03 [14:21] <cone-343> ffmpeg.git 03Christian Schmidt 074a7429203a39: pcm-mpeg: correct bitrate calculation * 03http://tinyurl.com/c2y73uu03 [14:21] <cone-343> ffmpeg.git 03Rafaël Carré 07a25d912dca9c: avcodec_encode_audio(): fix invalid free * 03http://tinyurl.com/cak54ek03 [14:21] <cone-343> ffmpeg.git 03Michael Niedermayer 07d6e87190fd07: Merge commit 'a25d912dca9cd553440167e0476c47581359c0fc' * 03http://tinyurl.com/brazqj603 [14:34] <cone-343> ffmpeg.git 03Anton Khirnov 07a119c64e38bd: avconv: fix disabling auto mappings with -map_metadata * 03http://tinyurl.com/cavmbvw03 [14:34] <cone-343> ffmpeg.git 03Victor Vasiliev 0771e92414bfd7: lavf: move RIFF INFO tag writing from avienc to riff * 03http://tinyurl.com/c4vkeu603 [14:34] <cone-343> ffmpeg.git 03Michael Niedermayer 07fadfbb354b30: Merge commit '71e92414bfd79e56ea6fff174a665ff7b9b86e68' * 03http://tinyurl.com/bt5kxc803 [14:59] <cone-343> ffmpeg.git 03Victor Vasiliev 070bca0283ccde: riff: do not write empty INFO tags * 03http://tinyurl.com/cvholko03 [14:59] <cone-343> ffmpeg.git 03Michael Niedermayer 07c079da507316: Merge commit '0bca0283ccded5e32da143a462168ad1988a58fd' * 03http://tinyurl.com/bo7umz503 [15:33] <cone-343> ffmpeg.git 03Victor Vasiliev 0758b619c8a226: wav muxer: write metadata * 03http://tinyurl.com/cxf7dsa03 [15:33] <cone-343> ffmpeg.git 03Michael Niedermayer 07d8cfa9835804: Merge commit '58b619c8a226cc4564ad5af291bc99a04f89ee56' * 03http://tinyurl.com/bsok2xw03 [15:40] <burek> one stupid question.. what does -flags +global_header apply to? (streams or containers) [15:47] <cone-343> ffmpeg.git 03Anton Khirnov 0731c54711cc3f: lavf: split wav muxer and demuxer into separate files. * 03http://tinyurl.com/dyl6zz803 [15:47] <cone-343> ffmpeg.git 03Michael Niedermayer 07df5e089da95b: Merge commit '31c54711cc3f1484af101d629bbb805820d37ad1' * 03http://tinyurl.com/c3bffys03 [15:51] <Tjoppen> wee - I got an IFF file at work [15:51] <Tjoppen> it's 1985 again [15:52] <durandal_1707> why you get such files? [15:52] <cone-343> ffmpeg.git 03Anton Khirnov 0779922d7237ab: wav: do not fail on empty INFO tags * 03http://tinyurl.com/crto9gw03 [15:52] <cone-343> ffmpeg.git 03Michael Niedermayer 07940ee636301a: Merge commit '79922d7237aba2b8c6abbd2e06a0c08e4f498ad4' * 03http://tinyurl.com/cotnawj03 [15:52] <Tjoppen> I have no idea. but the client seems to want support for them [15:53] <Tjoppen> might be time to do my old "split wav into wavenc/dec, then share code with the avi demuxer" idea [15:53] <Tjoppen> in which case an iff demuxer could share code too [15:58] Action: durandal_1707 notices conflict list is becoming bigger and bigger [16:00] <cone-343> ffmpeg.git 03Anton Khirnov 07a43283b6f4cf: wavdec: check size before reading the data, not after. * 03http://tinyurl.com/d99cgce03 [16:00] <cone-343> ffmpeg.git 03Derek Buitenhuis 07c75848cd4c09: configure: Add support for Tilera processors * 03http://tinyurl.com/bn6h47h03 [16:00] <cone-343> ffmpeg.git 03Michael Niedermayer 07775d41b617f0: Merge remote-tracking branch 'qatar/master' * 03http://tinyurl.com/d4przjb03 [16:00] <ubitux> does anyone have any idea of what particular case Nicolas is talking about in the lavfi metadata support? [16:10] <cone-343> ffmpeg.git 03Paul B Mahol 079b762e2cba16: idcinvideo: remove redundant " id CIN Video: " from av_log() * 03http://tinyurl.com/cj7b9p203 [17:11] <ubitux> saste: http://fate.ffmpeg.org/log.cgi?time=20121017114713&log=configure&slot=x86_6… [17:11] <ubitux> didn't you send some patches for this? [17:12] <ubitux> mmh or maybe it was carl [17:13] <Daemon404> it was carl [17:13] <Daemon404> and it was pushed [17:13] <ubitux> "just"? [17:13] <Daemon404> no [17:13] <Daemon404> and the patch was to make it fail when it wasnt foudn [17:14] <ubitux> why did i read "just".... [17:14] <Daemon404> instead of silently contining [17:14] <ubitux> ok [17:14] <ubitux> i'll mail him, thanks :) [17:14] <Daemon404> maybe you should check why it fails first :P [17:14] <ubitux> yes [17:14] <ubitux> though, it's a syntax error :p [17:15] <Daemon404> lol [17:53] <durandal_1707> michaelni: that dca merge looks fishy [17:54] <cone-343> ffmpeg.git 03Michael Niedermayer 07a4fe661157b2: mov_probe: fix integer overflows * 03http://tinyurl.com/blobmm603 [18:00] <ubitux> eh all my typo are spotted [18:16] <Daemon404> tilegx and tilepro fate builds added [19:55] <durandal_1707> what is nice way to look code changes of actual merges? [19:55] <Daemon404> git cli or web? none [19:56] <Daemon404> there might be a decent way in one of git's many guis [19:56] <Daemon404> (probably mac based...) [19:56] <durandal_1707> cli only [19:56] <Daemon404> there's only the ugly way [19:56] <Daemon404> which is the obvious one [19:57] <durandal_1707> git is just for linux kernel maintainer [19:58] <durandal_1707> hmm, so looks like i need to setup this web gui thing [19:58] <ohsix> git instaweb ? [19:59] <ohsix> there's gitk too [19:59] <ohsix> but if you want to set something up, cgit [19:59] <ohsix> gitk isn't bad [19:59] <Daemon404> ohsix, none of those let you see merge stuff that easily [19:59] <Daemon404> gitweb's is horrible [20:00] <ohsix> wah, i look at that all the time on gitweb [20:00] <ohsix> you mean with extra context, not just the changes? [20:00] <durandal_1707> real changes to files no virtual crap [20:00] <Daemon404> merge conflicts, any extra content in the merge commits themselves [20:01] <Daemon404> and how it applies to THIS tree [20:01] <Daemon404> not the tree you merged it from [20:01] <ohsix> so you mean between branches [20:01] <Daemon404> branches is too specific [20:01] <Daemon404> just trees [20:02] <ohsix> trees have branches, one or more, practically speaking it's always between branches [20:02] <Daemon404> i suppose [20:02] <ohsix> a tree has all branches, comparing all branches to another set of all branches doesn't make much sense [20:03] <Daemon404> but thats not the definition of tree i use [20:03] <ohsix> http://git.alsa-project.org/?p=alsa-lib.git;a=shortlog;h=HEAD commitdiff is what i use all the time, and it's good enough; interested to hear where it fails [20:03] <Daemon404> ohsix, for example [20:03] <Daemon404> micael merges the latest batch of libav changes into ffmpeg [20:03] <Daemon404> via git merge [20:03] <Daemon404> merge conflicts etc [20:04] <Daemon404> or maybe it applies slightly differently to teh ffmpeg tree [20:04] <Daemon404> but ig you view teh diffs for the commits themselevs [20:04] <Daemon404> theyre always against libav's, nto ffmpeg's [20:04] <Daemon404> and the merge commit itself is very confusing to look at [20:04] <Daemon404> if at all even relevant [20:04] <ohsix> ah, can't help the confusion there, i think; because when you do a merge, they _are_ against libav's [20:05] <Daemon404> exactly [20:05] <Daemon404> but it's really annoying to track merged things [20:05] <ohsix> you would have to do all the patches individually for that to not be [20:05] <Daemon404> there's no good way to view teh dleta as it was applied to our tree [20:05] <Daemon404> and there really should be [20:05] <Daemon404> since thats what it ends up as [20:05] <ohsix> ok, i took it to mean the web/gui interfaces were bad, not that merging was confusing [20:06] <Daemon404> no webgui/gui displays it well [20:06] <Daemon404> :P [20:06] <Daemon404> afaik [20:06] <ohsix> you put them on a different branch and then compare them, merging only when needed [20:06] <ohsix> right, but the web ui isn't what's confusing you, the merge is [20:06] <Daemon404> no i mean its way of displaying the merge [20:06] <Daemon404> is poor [20:06] <Daemon404> double + [20:06] <Daemon404> like [20:06] <Daemon404> + [20:06] <Daemon404> ++ a [20:06] <Daemon404> -+ a [20:06] <Daemon404> its a shitty way to visualize it [20:07] <ohsix> maybe i've gotten used to it, or it never really mattered; i happen to agree with you tho i couldn't think of a better or less interesting way to do it [20:07] <Daemon404> ;p [20:08] <ohsix> interesting in the "may you live in interesting times" sense, not wow this is amazing sense [20:08] <durandal_1707> i use log -m but it forks only for the first merge.... [20:10] <ohsix> hm, i'm not sure i wouldn't just apply the merge to another branch and compare them if i was in that situation often, isn't the patch posted before merges are done? or is this a situation where you land in the middle of a merge and you need to figure out what happened [20:11] <ohsix> i'm presuming you want to review a merge, not like, doing a bisect then having to untangle a merge [20:14] <durandal_1707> log -m is doing job well except i need to manually pick each merge [20:15] <ohsix> hm, well, i might be able to think of some other useful tool if the use-case is different [20:17] <Daemon404> bleh [20:17] <Daemon404> *** drop! [20:17] <Daemon404> ^ what causes this [20:17] <ohsix> falling into the middle of a series of patches is usually confusing, merge or not; so i hadn't considered it as a source of extra confusion :p thus only thinking about reviews [20:18] <ohsix> git grep? probably a frame with a bad timestamp or something [20:18] <ohsix> (or an actual, genuine missed frame) [20:19] <Daemon404> yes im looking [20:20] <Daemon404> maybe ill send a patch for a less shitty message [20:20] <Daemon404> by default it just drops thousands of frames for this clip\ [20:20] <Daemon404> and the "verbose" long is "*** drop!" [20:20] <Daemon404> s/long/log/ [20:21] <ohsix> it eventually drops the "your computer may be too slow!" or maybe i'm thinking of mplayer [20:21] <Daemon404> no [20:21] <Daemon404> nb_frames = FFMIN(nb_frames, ost->max_frames - ost->frame_number); [20:21] <Daemon404> if this is 0 [20:21] <Daemon404> it drops the frame [20:21] <ohsix> ah something else then [20:21] <ohsix> ok, gotta run; bbl [20:21] <Daemon404> of course there are no comments. [20:25] <Daemon404> and there is no indication what max_frames is at all [20:25] <Daemon404> :| [20:26] Action: durandal_1707 yes : git lgs --min-parents=1 libavcodec/dcadec.c [20:26] <Daemon404> looks like INT64_MAX... [20:27] Action: Daemon404 guess he asks michaelni wtf is going on [20:35] <cone-343> ffmpeg.git 03Michael Niedermayer 07fd9e88fe6018: libavfilter/lavfutils: remove useless NULL check on codec context * 03http://tinyurl.com/cwlzux403 [20:36] <cone-343> ffmpeg.git 03Michael Niedermayer 07657998b5ee46: libavfilter/lavfutils: remove useless NULL check on format context * 03http://tinyurl.com/d4xmvd203 [20:36] <cone-343> ffmpeg.git 03Michael Niedermayer 077fd65104f489: ffm_seek: fix division by zero * 03http://tinyurl.com/c7ez86k03 [20:36] <cone-343> ffmpeg.git 03Michael Niedermayer 07378a5b9c5f1a: ffm_write_write_index: check lseek() return code * 03http://tinyurl.com/c8v27fa03 [20:36] <cone-343> ffmpeg.git 03Michael Niedermayer 0771bc8c95d7ca: ffm_read_write_index: check lseek return code * 03http://tinyurl.com/d9chjb403 [20:50] <ohsix> Daemon404: is it going to be reference frames or something? cuz there'd be no need to drop them unless something was exceeded, or they got too old [20:51] <Daemon404> its not clear whats going on [20:51] <ohsix> right, but there are only a handful of circumstances where frames would be dropped [20:51] <ohsix> small enough to investigate each one, anyways [20:51] <Daemon404> [14:33] <@elenril> in ffmpeg it's dropping frames when 1) two frames have the same timestamps or 2) you want cfr [20:51] <Daemon404> [14:44] <@Daemon404> elenril, that cant be accurate... [20:51] <Daemon404> [14:44] <@Daemon404> this is neitehr vfr input [20:51] <Daemon404> [14:44] <@Daemon404> nor are there same timestamps [20:51] <Daemon404> [14:45] <@elenril> Daemon404: my crystal ball is out of ord [20:51] <Daemon404> [...] [20:52] <Daemon404> [14:35] <@Daemon404> avconv's behavior is SO much better [20:52] <Daemon404> [14:36] <@Daemon404> Press ctrl-c to stop encoding [20:52] <Daemon404> [14:36] <@Daemon404> ^C^C^C^C^C 1 fps= 0 q=0.0 size= -0kB time=0.01 bitrate= -17.6kbits/s [20:52] <Daemon404> [14:36] <@Daemon404> and starts using ALL of my memory [20:52] <ohsix> ah, interesting [20:53] <ohsix> i'm not in a situation where i can read the code, so i won't distract you further [20:54] <durandal_1707> perhaps it keeps buffering stuff because timestamps are nonsense [20:54] <Daemon404> the timestamps are fine [20:54] <Daemon404> i checked them maually [20:54] <durandal_1707> what container? [20:54] <Daemon404> mp4 [20:57] <durandal_1707> get sample so i can reproduce and play little with it? [20:57] <durandal_1707> *got [21:00] <Daemon404> i dont think i can share this sample [21:00] <Daemon404> work-related [21:05] <durandal_1707> how many streams it have? [21:06] <cone-343> ffmpeg.git 03Michael Niedermayer 07a0e0e1e19254: ffmdec: fix hypothetical overflows * 03http://tinyurl.com/cw3cn8r03 [21:06] <cone-343> ffmpeg.git 03Michael Niedermayer 07f03c0f6afcb1: ffmdec: check av_new_packet() return value * 03http://tinyurl.com/bsltcsk03 [21:06] <cone-343> ffmpeg.git 03Michael Niedermayer 07d185c8a79bbd: tiff: run strlen() after setting the pointer * 03http://tinyurl.com/bqv9wzm03 [21:08] <ubitux> Daemon404: did you check them with -debug_ts? [21:13] <Daemon404> ubitux, with ffms2 [21:13] <Daemon404> but same shared lib [21:14] <Daemon404> -debug_ts output is ... dense... to say the least [21:14] <ubitux> :( [21:26] <durandal_1707> so riff info chunks are not 2byte aligned? [21:33] <ubitux> saste, michaelni (and others): i'm going to rename AV_PKT_DATA_METADATA to FF_PKT_DATA_METADATA in the next version of the lavfi metadata patch if you don't mind [21:33] <cone-343> ffmpeg.git 03Nicolas George 07709628aa71f2: lavfi/avf_concat: fix invalid exclusive test. * 03http://tinyurl.com/dywb5h603 [21:33] <ubitux> i'm actually pretty uncomfortable with what is libav going to do [21:33] <durandal_1707> name will not help [21:33] <ubitux> and i'm afraid they will introduce a AV_PKT_DATA_METADATA as well at some point [21:33] <Daemon404> i thought FF_ was for internal only [21:34] <ubitux> Daemon404: yes, but there is no point (at the moment) to make this side data available for users [21:34] <durandal_1707> ubitux: AV_PKT_DATA_METADATA=666 [21:34] <Daemon404> as long as FF_* isnt public [21:34] <ubitux> durandal_1707: that doesn't solve the issue, it's already the case [21:34] <ubitux> durandal_1707: the problem is if they use the same name [21:35] <durandal_1707> then put our initials in name [21:35] <ubitux> i was just proposing to use FF so it's private and it doesn't matter [21:36] <llogan> AV_PKT_DATA_METADATA_BETTEREST [21:36] <ubitux> and it can be made public without much effort [21:36] <ubitux> llogan: AV_PKT_DATA_METADATA_FFMPEGGOTITFIRST [21:36] <Daemon404> if you make something with a FF_ prefix public [21:36] <Daemon404> that is wrong [21:36] <Daemon404> and inconsistent [21:37] <durandal_1707> just FFMPEG [21:37] <durandal_1707> is enough [21:37] <ubitux> so you would prefer a public AV_PKT_DATA_METADATA_FFMPEG? [21:37] <Daemon404> and thats just inconsistent with the entirety of thepublic api [21:38] <durandal_1707> AV_PKT_DATA_METADATA_LOL [21:38] <ubitux> Daemon404: you should worry, as a user, about a FF_PKT_DATAxxx defined somewhere imo [21:38] <ubitux> durandal_1707: AV_PKT_DATA_METADATA_COME_ON_LIBAV_BREAK_API_IF_YOU_CAN [21:39] <ubitux> anyway, what are your suggestions? [21:39] <durandal_1707> add _EX [21:39] <ubitux> AV_PKT_DATA_STRINGS_METADATA maybe? [21:40] <durandal_1707> so libav introducen jet another regression into ffmpeg [21:41] <Daemon404> theyve done nothing yet [21:41] <Daemon404> ubitux is beign preemptive [21:42] <durandal_1707> Daemon404: talking about merges - recent one being riff info thing [21:43] <ubitux> Daemon404: yep, i'm just cautious on this one, especially after elenril told me today :D [21:43] <ubitux> +what [21:45] <saste> what's the problem with anonymous typedefs? [21:45] <saste> *anonimous typdedeffed structs? [21:45] <durandal_1707> personal preference [21:46] <ubitux> it seemed to have caused "issues" with some public ones in header [21:46] <ubitux> otherwise it just seems like a new diego-ing thing as durandal_1707 says [21:46] <saste> anonymous typedeffed structs considered harmful? [21:47] <saste> what about removing gotos then? [21:47] <burek> [20:25:33] <wlan3> And what are the differences, not counting need to modify my scripts? [21:47] <burek> [20:25:48] <relaxed> nothing really, they operate the same [21:47] <burek> ? [21:47] <saste> funny project for a boring winter afternoon [21:47] <burek> why tell something like this and cofuse people even more [21:47] <burek> confuse* [21:48] <llogan> burek: ask relaxed in -user. [21:48] <durandal_1707> burek: that is libav agent [21:48] <burek> yes but still [21:48] <burek> it's inappropriate to answer like that [21:48] <llogan> it's all about perspective. [21:49] <llogan> (not that i'm agreeing with that perspective). [21:49] <burek> well not quite [21:49] <burek> the reaction that has caused was [20:34:12] <wlan3> I'll have to change every reference of ffmpeg in the scripts. Easy but annoying. [21:49] <burek> and tomorrow he'll come again asking for help saying some option is not working in avconv [21:50] <cone-343> ffmpeg.git 03Stefano Sabatini 071f7962625cf3: examples/muxing: remove video_outbuf unused and useless code * 03http://tinyurl.com/d3598js03 [21:50] <cone-343> ffmpeg.git 03Stefano Sabatini 075ca298df2df8: examples/muxing: remove misleading comment about pending API change * 03http://tinyurl.com/bnnn2kf03 [21:50] <cone-343> ffmpeg.git 03Stefano Sabatini 07d6196d942191: examples/muxing: fix bogus setting of st->id * 03http://tinyurl.com/cq9tmsn03 [21:50] <cone-343> ffmpeg.git 03Stefano Sabatini 07eda0a52bf161: examples/muxing: check on frame * 03http://tinyurl.com/caxacx903 [21:50] <cone-343> ffmpeg.git 03Stefano Sabatini 07eebde404bc8c: examples/muxing: merge add_audio_stream() and add_video_stream() * 03http://tinyurl.com/d2o4dhc03 [21:50] <burek> if it was the same then there would be no fork at all [21:50] <burek> . [21:50] <llogan> burek: yes, it's the same old bullshit. [21:53] <ubitux> saste: i need something faster than scene detection for my tests; i'm going to add some metadata in one of the black detection filter, which one is appropriate? [21:53] <ubitux> (we have two of them iirc) [21:54] <ubitux> blackframe & blackdetect [21:54] <saste> ubitux, blackdetect is better [21:54] <saste> blackframe is GPL, and has an inflexible syntax [21:54] <ubitux> ok, thanks [21:54] <saste> we should try to merge the remaining features to blackdetect and drop it [21:56] Action: ubitux is stupidely laughing at the racist debug joke he added in his local tree in the blackdetect filter [21:56] <saste> yes that filter is racist, that is colorist [21:57] <saste> why a pink frame is better than a black one? [21:57] <Daemon404> 4chan has a game with the same name as that filter [21:57] <ubitux> haha [21:57] <saste> colordetect would be a better filter (and less colorist) [21:58] <Compn> you should just call it what its for [21:58] <Compn> e.g. commercialdetect [21:58] <saste> why such color discrimination? [21:58] <saste> but a colordetect would be slower [21:58] <Daemon404> Compn, thats a pretty shitty commercial detect [21:58] <Daemon404> you need a logo mask too [21:58] <ubitux> saste: i think it's overkill [21:59] <Daemon404> and perhaps check channel changes and bitrate reductio [21:59] <Daemon404> n [21:59] <ubitux> saste: with vf compare it won't be needed anymore [21:59] <ubitux> assuming we fix the code [21:59] <ubitux> (aka making it work with a still image) [21:59] <ubitux> so we can detect one frame in a stream [21:59] <ubitux> black, white, pink, lena, etc [22:00] <Daemon404> i assume this has a threshold? [22:00] <ubitux> ask luca [22:00] <Daemon404> right [22:00] <Daemon404> it soudns really trivial [22:00] <saste> i wonder how our psnr would compare to that [22:00] <ubitux> do we have a psnr filter? [22:00] <saste> no it was discussed on ML [22:01] <saste> there were some optimization issues to be fixed [22:01] <saste> it was never committed [22:01] Action: ubitux is waiting for the filter to go in, fix the braindead N alloc & free per frame, make it work with still images, and rule the world [22:02] <Daemon404> why dont you just tell luca [22:02] <Daemon404> to fix it [22:02] <ubitux> i did [22:02] <ubitux> he will likely won't [22:02] <Daemon404> grammar-sploded my head [22:02] <ubitux> :D [22:03] <ubitux> he will likely not do it [22:10] <ubitux> meh the metadata don't reach the end of the graph [22:10] <Daemon404> :D [22:13] <saste> return -1 never seem to end... [22:14] <Daemon404> -1 is very descriptive though [22:14] <Daemon404> it tells me not to touch this part of the code [22:14] <Daemon404> lest i endure pain [22:15] <ubitux> saste: how is the buffer ref transmitted to the next filter in blackdetect? [22:15] <ubitux> i mean, i add some meta to inlink->cur_buf, but that buffer is ignored afterward [22:15] <saste> grep -R "return -1" (find -name '*.c') | wc -l => 3412 [22:16] <ubitux> so any idea how we can trick this? [22:16] <saste> ubitux: can't say with no code [22:16] <saste> my memory is overrated [22:17] <saste> Daemon404, yeah permission denied to touch this code [22:17] <ubitux> i think i missed something :p [22:19] <ubitux> it seems to only work when i inject the metadata in the picref passed to start_frame() [22:31] <ubitux> just curious [22:31] <ubitux> do you guys feel like this metadata string in side data is a hack? [22:33] <durandal_1707> sort of [22:34] <ubitux> how so? [22:34] <ubitux> (soon: batch normalization possible with a few lines of code!) [22:35] <durandal_1707> the whole idea - it is not going to be used by any demuxer [22:35] <ubitux> nicolas mentioned some [22:35] <durandal_1707> ? [22:36] <ubitux> http://ffmpeg.org/pipermail/ffmpeg-devel/2012-October/132263.html [22:36] <ubitux> * The v4l2 device uses the priv field to manage its DMA buffers, it could need to inject metadata if some device were able to report additional information, such as autofocus info or GPS position. [22:37] <ubitux> also, i wonder if i couldn't use it for the timecode :) [22:37] <ubitux> (when at format level, like most of the cases) [22:37] <durandal_1707> what about waiting evils action? [22:38] <ubitux> do you think those suggestions are not reasonable? [22:46] <durandal_1707> i dunno what to think of it [23:30] <cone-343> ffmpeg.git 03Michael Niedermayer 0735daf3ca8173: cmdutils: remove unneeded null check * 03http://tinyurl.com/chybjur03 [23:30] <cone-343> ffmpeg.git 03Alexis Ballier 07916352f28285: configure: do not quote arguments passed to filter{,_out} in check_ld. * 03http://tinyurl.com/d3zyabt03 [23:39] <ubitux> nyuhu: seems like my lavfi fate howto will be soon "deprecated" [23:40] <ubitux> (the lavfi test listing at configure time is going away soon) [23:41] <burek> if you find a better solution, let me know :) [23:43] <ubitux> sounds legit [23:43] <ubitux> fine with me burek :) [00:00] --- Thu Oct 18 2012

1 0

ffmpeg.log.20121017
by burek 18 Oct '12

18 Oct '12

[00:00] <Mista_D> "qp 0" ? [00:00] <Mavrik_> huh? [00:00] <Mista_D> Mavrik_: I need a "near lossless" [00:01] <llogan> Mista_D: for libx264? just use the highest crf value that still gives you an acceptable quality. [00:02] <llogan> and sameq is gone. also it did not mean "same quality" [00:02] <llogan> purge it from your memory! [00:02] <Mista_D> llogan: ~10? [00:03] <llogan> see if you can tell a difference between the source and -crf 18 [00:04] <Mista_D> llogan: MSU VQMT will, and with enough zooming, anything is possible. [00:14] <llogan> people watch videos, not "quality measurement tools". [00:17] <Mista_D> llogan: you won't notice a change in a minute or 5, after an hour of watching a movie, you can tell crf 18 from original. [00:21] <burek> :) [00:22] <burek> Mista_D, crf assumes variable quality [00:22] <burek> over different scenes [00:22] <burek> so you won't accumulate any quality loss during time [00:23] <burek> crf is something like "quality factor" [00:23] <burek> it's not bandwidth limitation to accumulate loss if too much bitrate has been already used in past [00:24] <Mista_D> burek: I'm up for pepsi challenge of AppleProres vs x264 cfr 18 (: [00:25] <burek> I don't use prores :) I believe you :) [00:35] <relaxed> Why would anyone want to use an intermediate codec that's lossy? [00:37] <pablomurad> Someone alive? [00:38] <relaxed> pablomurad: I am, but burek is a zombie. And he bites people that ask useless questions. [00:39] <burek> (") [00:39] <pablomurad> I don't believe my question is so useless, lol. I used to use ffmpegx and I couldn't edit the size of the subtitle and fonts too. [00:39] <pablomurad> Many forums I read told me the problem was with the Ffmpeg binaries, is this true? [00:40] <burek> it might be, unless if it is not [00:40] <relaxed> We only support ffmpeg directly, not guis from other projects that interface with ffmpeg. [00:40] <llogan> does it use ffmpeg for subtitle stuff? [00:41] <burek> pablomurad, if you can add some custom options to be forwarded to ffmpeg, add -report [00:41] <burek> or what is its name [00:42] <pablomurad> I only knows that maybe the problem is with the last ffmpeg, I know you dont support another Apps, but it would be nice if you guys take a look on this. However, the actual binary is great [00:42] <pablomurad> Sure, I will do it :) [00:42] <burek> pablomurad, can you somehow dig up actual ffmpeg logs, if you can, we might help [00:42] <burek> so turn on all the verbosity there is in your gui [00:42] <relaxed> It would be helpful if you could replicate the problem using ffmpeg. [00:42] <pablomurad> I will get a log soon as possible!! thank you guys [02:24] <lake> practically, why would one choose to use -vf yadif=1 over -vf yadif=0 [02:24] <lake> i used yadif=1 on some home videos and it made it look so fluid that it didn't look right [02:30] <relaxed> the default is yadif=0 but with some content 1 might look better. [02:31] <relaxed> lake: This is old but you might learn something. http://www.100fps.com/ [02:36] <lake> relaxed: so it's more or less subjective? [02:36] <lake> i'll check that link out. thanks [02:42] <relaxed> lake: I would always use 1 [04:30] <JamJams> Hey I'm trying to use -i in.avs, however I keep having issues with it not loading my avs plugins. But they are sitting in the correct directorys and such. Is there something I'm missing? [05:41] <lake> relaxed: i'm interested to know if you agree with this article http://renomath.org/video/linux/interlace/ [13:06] <markuman> i've made a fast motion video like that. http://paste.osuv.de/index.php/QCErl/ but how to display the first image for 5 seconds [13:33] <markuman> or how to combine these two commands? http://paste.osuv.de/index.php/ru6CV/ the first image should take 5 seconds. and after that 11 images per second [14:12] <relaxed> markuman: you can't concatenate video with two different framerates. [14:13] <markuman> relaxed: no i mean, the first image from time laps should display 5 seconds long (kind of introduction image). and then start the timelaps video [14:13] <relaxed> I would use rawvideo output for both and cat them into ffmpeg from stdin. [14:15] <relaxed> For the first command: ffmpeg -loop 1 -r 11 -i stapelzeitrafferintro.png -t 5 -pix_fmt yuv420p -f rawvideo 1.yuv [14:16] <relaxed> Second command: ffmpeg -r 11 -i $other_images -pix_fmt yuv420p -f rawvideo 2.yuv [14:18] <relaxed> Third: cat 1.yuv 2.yuv | ffmpeg -f rawvideo -pix_fmt yuv420p -r 11 -s $freamesize -i - -vcodec libvpx -b:v 5000k output.webm [14:19] <relaxed> where $framesize is the WxH of the images [14:21] <relaxed> also, there are presets for libvpx that you might want to look into, the defaults may suck. [14:21] <markuman> k thx [14:23] <relaxed> here's the list: http://git.videolan.org/?p=ffmpeg.git;a=tree;f=presets;h=e64871fca371b05ebc… [14:25] <relaxed> you would think libvpx would take a page out of libx264's book and make internal presets available [14:27] <ubitux> relaxed: i see you're using a bitrate for libvpx; were you able to achieve a decent quality with a quality factor or anything? [14:28] <relaxed> ubitux: I was just using the bitrate he chose. I honestly hope I never have to use libvpx because other than being more "open" it fails in everything compared to libx264. [14:29] <relaxed> Including ease of use. [14:29] <ubitux> i wasn't able to get a non-low bitrate quality by playing with the different quality, cfr etc [14:29] <ubitux> i can only get a good output with -b:v :( [14:30] <relaxed> Why do you need it? For html5? [14:35] <ubitux> relaxed: yes, playable inside the browser [14:35] <ubitux> quite handy for common videos [14:35] <relaxed> True, I guess that's where the web is heading. [14:39] <ubitux> if i want to show a sample to someone, i just make a sample in webm, upload it and give a direct link [14:39] <ubitux> that's way handy than looking for the video on youtube [14:39] <ubitux> get a crappy quality, bad edits, etc [14:42] <relaxed> You could achieve the same thing with an h264 video. [14:42] <ubitux> except that wouldn't be directly playable in the browser :( [14:42] <relaxed> And serve it from your website, dropbox, or whatever. [14:42] <ubitux> i wonder if html5 allow seeking through url [14:43] <ubitux> like a video attribute with a start [14:44] <relaxed> ubitux: It would be if you had a template html setup with jwplayer. Then you would only have to upload the video. [14:53] <burek> <@ubitux> except that wouldn't be directly playable in the browser :( <- why? [14:54] <ubitux> cause i don't want to use flash? :p [14:54] <burek> h264 video, not a container.. [14:54] <ubitux> ? [14:54] <burek> webm(h264+aac) [14:56] <relaxed> webm is vp8+vorbis [14:56] <burek> only that? [14:56] <burek> no wonder it sucks :) [14:56] <relaxed> well, for html5 I think [15:00] <relaxed> you can do (h264/aac).mp4 through html5 but most browsers won't ship with decoders. [15:01] <burek> why is it so difficult to create an open source container in the 21st century? [15:01] <burek> do we lack knowledge, time or what? [15:02] <relaxed> webm is pretty much matroska [15:06] <relaxed> the push to vp8/vorbis is due to patents [15:13] <Jan-> is there some way I can turn an m4v file to an mp4 file without recompressing it [15:13] <relaxed> I think you can simply rename it. [15:13] <Jan-> hmm [15:13] <Jan-> adobe premiere doesn't like it when we do that [15:14] <Jan-> I have no idea where you'd even GET an m4v file [15:14] <relaxed> ffmpeg -i input -map 0 -c copy output.mp4 [15:14] <Jan-> I mean I've come across m2v before. [15:14] <relaxed> Apple has to rename everything. [15:14] <Jan-> unrecognized option "c" [15:15] <relaxed> ffmpeg -i input -acodec copy -vcodec copy output.mp4 [15:15] <Jan-> that's what I did [15:15] <relaxed> and...? [15:15] <Jan-> hmn [15:16] <Jan-> premiere still only sees it as 3 frames long [15:16] <Jan-> but vlc plays it fine [15:16] <Jan-> Recompress time I guess :) [15:16] Action: relaxed blames Adobe [15:16] Action: Jan- too [15:16] <Jan-> ok, what would you suggest to comprehensively reencode this thing so it's completely redone [15:16] <divVerent> 14:56:23 relaxed | webm is vp8+vorbis [15:17] <divVerent> 14:56:48 burek | only that? [15:17] <divVerent> 14:56:55 burek | no wonder it sucks :) [15:17] <divVerent> sort of [15:17] <divVerent> if you put anything else in, it is no longer webm [15:17] <divVerent> but matroska [15:17] <divVerent> webm is basically a subformat of matroska only supporting these codecs [15:17] <divVerent> so a browser can say "we support WebM", which then means "matroska but only few codecs and no extra crap like attachments" [15:18] <burek> Jan- ffmpeg -i yourinput.m4v ? [15:18] <burek> what's inside? [15:18] <burek> divVerent thanks for clarification :) [15:18] <burek> it makes me think it sucks even more :D [15:18] <divVerent> then just use matroska :P [15:19] <divVerent> lavf demuxer BTW doesn't care at all if it is webm or matroska [15:20] <divVerent> and the muxer has a webm and a matroska mode, and all the webm mode does is skip/disable some matroska features [15:20] <divVerent> this actually does make sense though [15:20] <divVerent> WebM is a whole spec including codecs [15:20] <divVerent> so if WebM allowed "any codec", you could have a "WebM supporting" browser, and a "WebM video", but the browser won't play it [15:21] <divVerent> for Matroska you'd have exactly this situation [15:21] <divVerent> you could have a player with Matroska support that just can't decode the H.264 inside [15:22] <Jan-> burek wouldn't I need to set bitrates and suchlike [15:22] <burek> Jan-, what I meant was: "Can you type: ffmpeg -i yourinput" [15:22] <burek> and pastebin the output [15:22] <burek> so we can see what's inside [15:23] <burek> divVerent yes I understand [15:23] <burek> it's just funny how many people think html5 will auto-magically solve all their problems [15:23] <burek> and they praise it so much [15:23] <divVerent> hehe [15:23] <burek> until they test it [15:23] <divVerent> it almost did [15:23] <burek> :) [15:23] <Jan-> http://pastebin.com/br7LqwTF [15:23] <divVerent> if only it hadn't been for the codec quarrel [15:23] <Jan-> I wonder if that came off a blu-ray or something [15:23] <divVerent> and this isn't even WebM's fault [15:23] <Jan-> I mean, an mp4 file with chapters in it? [15:24] <divVerent> the issue is, half the world supports H.264, the other half supports WebM [15:24] <divVerent> and thus nothing was gained by html5 [15:24] <burek> uh Jan-... you really want to update your ffmpeg [15:24] <divVerent> because html5 did not decide on which codec/format to support [15:24] <divVerent> mainly Apple and Microsoft blocked WebM from being required in HTML5 [15:24] <Jan-> burek: no, I don't, because if i do that everything will change [15:24] <burek> i mean, if you want to encode files properly [15:24] <Jan-> and I won't be able to do ANYTHING. [15:24] <burek> well, ok [15:24] <divVerent> and many others blocked H.264, for obvious patent reasons [15:24] <divVerent> so in the end, HTML5 for video became worthless [15:26] <Jan-> webm is going nowhere the same way flac for audio is going nowhere. [15:26] <burek> Jan-, try to re-encode with: ffmpeg -i input -map 0 -acodec copy -scodec copy -vcodec libx264 -crf <0-50> -preset <preset-name> output.mp4 [15:26] <Jan-> de facto standards are HARD to move. [15:26] <burek> oh, right.. you can't even use -crf... [15:26] <burek> well I don't know what to suggest [15:27] <divVerent> Jan-: flac going nowhere? [15:27] <burek> use a static ffmpeg? [15:27] <divVerent> flac already IS somewhere [15:27] <divVerent> while webm didn't really establish itself, and never will [15:27] <divVerent> it's replacable, if Google Chrome from today to tomorrow stopped supporting WebM and would only do H.264, nobody would notice [15:27] <Jan-> I'm pretty sure I don't have access to -presets and if I did I wouldn't know which one to use [15:28] <divVerent> WebM was dead the moment Microsoft and Apple decided to block it [15:28] <burek> Jan-, too bad [15:28] <burek> that's the consequence of using ancient ffmpeg, sorry.. [15:28] <burek> you can use static ffmpeg [15:28] <burek> re-encode your files [15:28] <burek> and delete it [15:28] <burek> (ffmpeg) [15:29] <divVerent> while FLAC actually managed to establish itself DESPITE better formats being around (although none of the lossless formats got an absolute majority) [15:30] <burek> wav rules :) [15:30] <divVerent> simply because FLAC was quick enough and is now one of the only two lossless compressed audio codecs you can assume to be decodable even in 10 years [15:30] <divVerent> the other is probably Apple Lossless (ALAC) [15:31] <durandal_1707> and APE [15:31] <divVerent> the reason is mainly that when FLAC came, it actually filled a need... because it was clearly superior to other lossless formats [15:31] <divVerent> even about APE i wouldn't be so sure [15:31] <divVerent> given it has quite poor support on mobile devices for example [15:31] <Jan-> Unfortunately as we all know it's not really about it being better. [15:31] <divVerent> you find a flac player for anything [15:31] <Jan-> It's about a lot more than that. [15:31] <divVerent> but ape... [15:31] <durandal_1707> flac is little better than shorten [15:31] <durandal_1707> *compression [15:31] <divVerent> what is really great about flac is not the compression, but the support [15:33] <divVerent> but thing is, it doesn't even matter... [15:33] <divVerent> in the test I see here, the span between the best and worst lossless compressed format goes from 51% to 58% [15:34] <divVerent> nobody is gonna switch format just for 14% total space savings (and that's what you'd get by changing from the worst to the best) [15:34] <divVerent> now, look at WebM... it did not fill ANY need [15:34] <divVerent> we already HAD video playing in the browswer [15:34] <Jan-> can someone suggest a sensible value for -presety [15:34] <divVerent> sure, it was using awful codecs, and involved Flash [15:35] <divVerent> but that's mainly why WebM was a failure. It did NOT become the "supported everywhere" format it was trying to become... [15:35] <divVerent> mainly because two big players stopped it from becoming that [15:35] <divVerent> and if you don't need "supported everywhere", then you can just keep using H.264 which already was there for ages, including good encoders [15:36] <divVerent> which is "supported everywhere except where software patents are in the way" ;) [15:38] <Jan-> it says "the system cannot find the path specified" which is what it usually says when you try to use presets on windows [15:38] <Jan-> no matter where you put the presets it always just says that [15:42] <relaxed> pastie.org the command and all output [15:43] <Jan-> it's OK [15:43] <Jan-> I think I fixed it [15:43] <Jan-> what does it mean when it starts a line of output with "[libx264 @0xDEADBEEF]" or whatever [15:43] <Jan-> is that the memory location where x264 was loaded? [15:44] <divVerent> that looks like memory corruption if you actually see 0xDEADBEEF [15:45] <divVerent> otherwise, it's the address of the context object [15:45] <Jan-> con...text... object? [15:45] <divVerent> "some stuff holding some data" [15:45] Action: Jan- suspects this conversation is headed in the direction of load maps and function pointers, and begins to feel the Fear [15:46] <divVerent> Jan-: just type LOAD "*",8,1 to fix this [15:49] <burek> can "-bsf h264_mp4toannexb" be used with -c:v copy ? [15:50] <burek> and what exactly does this message say: [mpegts @ 0x995f40] H.264 bitstream malformed, no startcode found, use the h264_mp4toannexb bitstream filter (-bsf h264_mp4toannexb) [15:51] <burek> im just trying: bla1 | ffmpeg -f h264 -i - -c:v copy -f mpegts - | bla2 [15:52] <Jan-> Is the function location stuff output by av_log or something [15:52] <Jan-> I've not seen it on other programs [15:53] <divVerent> Jan-: it is [15:56] <Jan-> yay, I made an API reference about ffmpeg and it wasn't wrong :D [15:56] <Jan-> I knew it would happen one day :D [16:15] <Jan-> is there some way I can specify an actual bitrate [16:15] <Jan-> this was always hard with ffmpeg as it always seemed to ignore -b [17:40] <Jan-> why would ffmpeg create a video with no audio track [17:44] <Jan-> ...aargh premiere doesn't like AAC audio [17:44] <sine_> hi guys [17:45] <Jan-> how do I just get it to do uncompressed audio!? [17:45] <sine_> learn me please. 29.97 is that actually 30fps ? [17:45] <sine_> im looking at iphone footage [17:45] <sine_> iphone 4 [17:45] <Jan-> sine strictly speaking no but it's often mislabeled [17:46] <sine_> how do i correctly dump the frames as png and then recombine them [17:46] <sine_> dumping is not a problem i suppse its the recombining [17:47] <sine_> ffmpeg -i input %04d.png? [17:47] <Jan-> Not sure. [17:47] <Jan-> I would really like to know how to make it do AAC or uncompressed audio. [18:19] <relaxed> Jan-: -acodec pcm_s16le [18:55] <dury> hi there channel :-) [18:55] <dury> anyone available [18:56] <dury> my ffmpeg doesn't work properly in ubuntu 12.04.1 [18:56] <dury> can anyone help me to solve it, please? [18:57] <JEEBsv> have you built ffmpeg yourself or are you using the ffmpeg binary that comes with Ubuntu? [18:59] <dury> JEEBsv: I've tried to extract the audio from .mp4 file using it.. it says to use avconv instead of ffmpeg [18:59] <dury> JEEBsv: I guess the ones that comes with ubuntu [18:59] <JEEBsv> so you are using the 'ffmpeg' binary that comes with ubuntu's default packaging. Debian/Ubuntu uses a fork project called libav and not ffmpeg. So yes, you should switch to using the 'avconv' command if you want to use that [19:01] <JEEBsv> Basically libav rewrote parts of the ffmpeg app and renamed it 'avconv', and then left the 'ffmpeg' binary to rot for a short while. The ffmpeg project merged those updates into the ffmpeg app. [19:02] <JEEBsv> if you have problems with avconv you can either hit #libav with them, or try the static builds available from the links on the topic etc. [19:02] <dury> JEEBsv: I'm going to try one and give you the error results ok? [19:02] <JEEBsv> -.- you should point that towards #libav then [19:05] <dury> JEEBsv: tell me an url the paste the results for you [19:08] <dury> JEEBsv: I've found one pastebin [19:08] <dury> JEEBsv: http://pastebin.com/RuP3bAM6 [19:09] <JEEBsv> I think that's pretty obvious and actually doesn't have anything to do with the ffmpeg binary being old [19:09] <JEEBsv> you don't have the version of libavcodec installed that has the "extra" encoders and so forth [19:09] <JEEBsv> libavcodec-extra I think? [19:17] <dury> JEEBsv: if I do sudo apt-get install libavcodec-extra-53.... it says the following extra packages will be installed libavutil-extra-51 and libopenjpeg2 Suggested packages: libfaad0 the following packages will be removed: libavcodec53, libavutil51 [19:18] <JEEBsv> yes, because it will remove the libraries without those extra things linked in :P [19:19] <dury> JEEBsv: should I confirm them? [19:19] <JEEBsv> yes [19:19] <dury> JEEBsv: all right [19:20] <dury> done it [19:21] <dury> JEEBsv: so suppose to do that ffmpeg will work fine, then? [19:21] <dury> JEEBsv: or avconv? [19:22] <JEEBsv> avconv is the newer one in the setup that debian/ubuntu has made, so if you are going to use their packages then yes, use avconv -- as that binary is more up-to-date than the ffmpeg that they give out [19:22] <JEEBsv> (ugh, my English, ended up saying something twice >_>) [19:23] <dury> JEEBsv: let me test it [19:24] <dury> JEEBsv: one sec. [19:24] <JEEBsv> also if you are going to be using that ubuntu package, I recommend you switch to #libav [19:24] <JEEBsv> because they use that project, while other distros use ffmpeg [20:20] <wlan3> Ok, this is confusing: [20:20] <wlan3> *** THIS PROGRAM IS DEPRECATED *** [20:20] <wlan3> This program is only provided for compatibility and will be removed in a future release. Please use avconv instead. [20:21] <wlan3> What does that mean? [20:21] <relaxed> it says to use avconv [20:22] <relaxed> which is the forked equivalent of ffmpeg [20:22] <wlan3> Oh, a fork. [20:24] <wlan3> And what are the differences, not counting need to modify my scripts? [20:24] <relaxed> nothing really, they operate the same [20:26] <wlan3> So just a great way to annoy users. [20:27] <relaxed> I think that was everyone's intended goal for the fork. [20:31] <wlan3> It seems so. [20:32] <wlan3> I'll have to change every reference of ffmpeg in the scripts. Easy but annoying. [20:34] <durandal_1707> avconv != ffmpeg [20:35] <wlan3> avconv: Like ffmpeg but changing some commandline options. [20:35] <wlan3> That would be a quite good description for whatis [20:36] <durandal_1707> it is not same, avconv use different lib [20:36] <relaxed> Nobody said it was the same. [20:36] <durandal_1707> you did [20:36] <wlan3> durandal_1707, Did not say same, said like [20:36] <relaxed> You misread. [20:37] <durandal_1707> >> relaxed: nothing really, they operate the same [20:38] <relaxed> THEY DO [20:38] <durandal_1707> putting caps will not make statement true [20:38] <relaxed> Then stop trolling me. [20:39] <relaxed> They do operate the same. I use both quite often. [20:39] <wlan3> durandal_1707, Then state why they do NOT operate the same. [20:40] <durandal_1707> they are different and use differen lib so can not operate same, but may operate same some time, but not all the time [20:42] <relaxed> wlan3: If you need support for the evil fork that does not operate 100% the same as ffmpeg, I would be happy to help you in #libav. [20:42] <wlan3> durandal_1707, what I see is that I have to change my scripts or manually build ffmpeg times and times for my scritps to work. [20:43] <wlan3> relaxed, It would be useful, but first I needed to complain a bit. [20:43] <durandal_1707> wlan3: only if you use ubuntu [20:44] <relaxed> or debian [20:44] <wlan3> durandal_1707, I use debian on most of my systems. [20:45] <durandal_1707> wlan3: your choice - i'm not forcing you to use anything, your pick [20:46] <wlan3> durandal_1707, but you said ONLY if you use ubuntu. [20:47] <durandal_1707> if you do not want to use non-system build or whatever it is called [20:49] <wlan3> durandal_1707, for unattended systems, Debian is the best working I've used so far, if you want to suggest me another option, i would investigate a bit. [20:49] <durandal_1707> what another option? [20:49] <relaxed> Debian is best. [20:50] <wlan3> durandal_1707, You should know more than me. [20:50] <durandal_1707> about linux distros? i do not know much, because i use FreeBSD [20:51] <wlan3> I prefer hurd [20:51] <wlan3> But, since it is still incomplete, I use Linux [21:02] <sine_> I have a load of png from a video which was 29.97 fps [21:03] <sine_> how can i encode them back again [21:03] <sine_> can i lump the pngs together or do they need to be rerendered/encoded [21:03] <tun0> sure you can [21:03] <sine_> wanna keep max qual if you know what i mean [21:03] <tun0> since everything is posible [21:04] <tun0> Then you should read man ffmpeg [21:04] <sine_> ffmpeg -i 0001.png -vcodec h264 /out/lolcakes.mp4 [21:05] <tun0> maybe [21:05] <tun0> ffmpeg [21:05] <tun0> sorry [21:05] <sine_> ffmpeg -f image2 -i %4d.png -r 25 -sameq myvideo.mpg [21:05] <JEEB> no, sameq is not what you think [21:06] <sine_> ffmpeg -f image2 -r 24000/1001 -i C:\BLENDERTEST\%04d.png -vcodec mpeg2video -qscale 3 C:\BLENDERTEST\output.mpg [21:07] <sine_> i want it to keep the quality though [21:07] <tun0> My eyeballs hurt [21:07] <sine_> i need it 29.97 as well [21:08] <sine_> jeeb can you hit me with the syntax please [21:08] <tun0> ffmpeg does that by default [21:08] <tun0> I think (that about keeping quality [21:08] <sine_> but i need a container [21:08] <sine_> x264 [21:08] <JEEB> <tun0> I think (that about keeping quality <- no [21:08] <sine_> mp4 [21:09] <JEEB> ffmpeg or anything else has absolutely no idea of how to "keep quality" [21:09] <JEEB> we don't have thousands of chinese children in basements [21:09] <JEEB> looking at various results [21:09] <sine_> i just want to watch my cgi [21:09] <JEEB> we don't yet have things that work like your eyes and brains :) [21:10] <JEEB> ffmpeg -r 30000/1001 -i %4d -c:v libx264 -crf 18 out.mp4 [21:10] <JEEB> for example [21:11] <JEEB> ugh, %4d.png I meant [21:11] <sine_> and the 3000/1001 is the 29.97 [21:11] <JEEB> -c:v is a newer way of saying "vcodec", libx264 is the H.264 encoder [21:11] <sine_> oh ok cool [21:11] <sine_> codec video [21:11] <JEEB> and crf is the rate control mode that is closest to "constant quality" [21:12] <JEEB> the default is 23, and lower is more bits thrown at the source, higher is less bits thrown at the source [21:12] <JEEB> 18 should be more or less nice-looking esp. with HD content, SD content depends on the actual content and your eyes [21:12] <tun0> JEEB, I meant, the output comes from the input, so if you do not alter options, the quality of the output is *similar* to that from the input. [21:12] <JEEB> tun0, nope [21:13] <JEEB> unless you mean that the defaults for various encoders being 200kbps mean that ffmpeg wants you to have the "similar quality" :P [21:13] <JEEB> only mostly used encoders like libx264 have sane defaults [21:13] <JEEB> like libx264 having crf 23 as the current default that looks pretty good for most people while compressing nicely [21:13] <JEEB> sine_, idea with crf is to find the highest crf value that still looks good for you [21:13] <sine_> cool its working [21:13] <sine_> im saving the code in my treepad lite. i have a section for ffmpeg [21:15] <tun0> JEEB, when I want nice quality, I just use out.webm as output. [21:15] <JEEB> no idea what the vp8 default for libvpx is [21:15] <JEEB> you might want to check it out [21:15] <sine_> thanks it worked. however my first cgi render is crap! [21:15] <JEEB> but yes, it's a newer encoder so there's a chance it's a nice default [21:15] <JEEB> sine_, huh [21:15] <JEEB> and how are you checking? [21:16] <sine_> you talking to me ? [21:16] <JEEB> the line beginning with sine_ and the next from it, yes [21:17] <sine_> im watching my cgi render ? [21:17] <sine_> ive just encoded it [21:18] <sine_> it was my first camera mapping experiment. i worked out mapping markers etc [21:18] <JEEB> I'm asking you to be somewhat less vague about various things if only possible [21:18] <JEEB> since it seems like the output wasn't good enough for you [21:18] <sine_> no, i mean my tracking and cgi modelling skills to the track data [21:18] <sine_> not the ffmpeg render [21:18] <JEEB> k [21:37] <DelphiWorld> hey [21:37] <DelphiWorld> xwma is wma ? [21:39] <llogan> never hoid of it. [21:40] <llogan> <- Mr. Usefull [21:45] <llogan> tun0: you also forgot to weigh the fact that repository packages of "ffaux-ffmpeg"/avconv or ffmpeg age relatively quickly compared to development [21:46] <DelphiWorld> hey llogan [21:49] <llogan> hi [21:52] <relaxed> burek: I choose to help users with the tool they have handy instead of wasting both of our time trying to install another tool that does the same thing. Also, I lurk in #ffmpeg-devel :P [21:53] <relaxed> If someone is in #libav and they're using FFmpeg's ffmpeg I will help them too. [21:54] <relaxed> Why? Because I don't give a shit about the politics between the two camps. [21:54] <burek> relaxed I know you do, just it might be better to tell them what that is all about [21:54] <burek> and redirect them where they need to go [21:54] <burek> no need to waste time in here [21:54] <relaxed> I don't look at it as a waste. [21:55] <burek> and also, if you dont give a shit, which you dont, then no need to tell people it's the same thing [21:55] <burek> because it is not [21:56] <burek> btw, there is no avconv's ffmpeg (at least it shouldn't be, due to copyright) [21:56] <relaxed> libav's avconv [21:57] <burek> if it's a fork, then there can't be avconv's ffmpeg [21:57] <burek> there can only be avconv [21:57] <relaxed> they should have named it ffmpeg2 [21:57] <burek> they shouldn't [21:57] <burek> because it's not [21:57] <burek> it's a 3rd party fork [21:57] <relaxed> I know the whole story. [22:03] <relaxed> I've been helping people in this channel for years. Longer than any of the familiar names I see here today. I will continue to do so the way I have regardless of anyone's political agenda. If an op doesn't like it they can man up and ban me. Until please stop the whining! [22:03] <relaxed> s/Until/Until then/ [22:04] <burek> I'm not whining, I'm just saying you should stop telling people something that's not true [22:04] <burek> I don't have anything against you, personally [22:04] <burek> it's just wrong to tell them something like that because that doesn't help them it just confuses them even more [22:05] <burek> regardless of the "politics" [22:05] <tun0> llogan, packages age? Did not understand [22:06] <llogan> i simply mean that compared to git-master the packages are generally old and smell like moldy shoes. [22:06] <cbdev> i just heard, ffmpeg was to be deprecated? [22:06] <relaxed> lol [22:06] Action: llogan sighs [22:07] <llogan> cbdev: here is a summary: http://stackoverflow.com/a/9477756/1109017 [22:07] <relaxed> I do agree that message was the ultimate troll. Keep in mind i didn't write it. [22:07] <relaxed> Or feel the n eed to explain it everytime someone comes in here asking about it. [22:08] <cbdev> ah i see :) [22:08] <cbdev> thanks [22:08] <llogan> the link refers to an older, crappier message, but it's still relavent [22:13] Action: relaxed deprecates burek [22:13] <burek> :) :beer: [22:13] <relaxed> :) [22:14] Action: wlan2 deprecates ppp0 [22:15] <relaxed> burek: the word "deprecated" should trigger the bot [22:16] <relaxed> it could be like the secret word on peewee's playhouse. [22:16] <wlan2> relaxed, should trigger what action? [22:17] <relaxed> a message explaining the fork, etc... [22:18] <relaxed> So that we will never have to have this discussion again. [22:19] <wlan2> Good idea. [22:19] <wlan2> Also explaining why the debian package maintainer sucks? [22:20] <wlan2> Or trolls, if you prefer [22:20] <relaxed> well, there's enought trolling already. [22:20] <relaxed> enough* [22:21] <wlan2> Also, I have a friend. [22:22] <wlan2> She usually uses windows for recording "tv" [22:22] <wlan2> And I was trying to figure which command she should use in ubuntu. [22:23] <saste> burek: !fork pointing to an *external* source explaining the forking status? [22:24] <wlan2> I told her to try "mplayer tv://" But even that does not work. [22:25] <burek> which external source saste? [22:25] <burek> do you have a link [22:25] <saste> stackoverflow may be one [22:26] <saste> we (and with "we" I mean "you") could add a link to more resources [22:26] <saste> internal resources may be biased [22:26] <burek> :) [22:26] <saste> or a collection of links from both ff* and av* and external resources [22:27] <burek> I like all that "biased" talk :) [22:27] <burek> they intentionally put a message saying ffmpeg is dead [22:27] <burek> and you are playing "fair" with them :) [22:27] <burek> cool :) [22:27] <burek> they even try to take over the domain [22:27] <burek> but that's ok too :) [22:27] <burek> let's not be biased :) [22:28] <saste> let the users decide for them [22:28] <saste> (even if they mostly don't care at all, provided the bloody thing works for them) [22:28] <burek> exactly :) [22:28] <burek> they made some steps to lead the users "in the right direction" [22:29] <burek> so they are more than biased if you ask me [22:29] <burek> and now we bear the burden of that kind of behavior [22:29] <burek> but, as i said, let's not be biased after all :) [22:48] <llogan> wlan2: "does not work" is a common description of a multitude of problems. [22:57] <wlan2> llogan, she is descriptive no more than that [22:59] <wlan2> the capturer is one of this freacky things that encode the video themselves and send it throught usb [23:03] <wlan2> https://pad.riseup.net/p/nGsg5q9DrLAI [23:05] <wlan2> That was all she gave me when I asked a big bit of times [23:17] Action: llogan kicks this router [23:40] <arakn0> I'm trying to do point to point streaming using rtsp but not working yet. The command im using and its output: http://pastebin.com/2TUyMRq6 Any suggestion/example that can iluminate me? Thanks! [23:43] <burek> rtsp/rtp/udp ? [23:51] <arakn0> burek: rtsp using udp [00:00] --- Thu Oct 18 2012

1 0

ffmpeg-devel.log.20121016
by burek 17 Oct '12

17 Oct '12

[00:14] <ubitux> michaelni: any neat trick to check for the overflow in swfdec? [00:14] <ubitux> given w and h being int (at most 16 bits), an >= 0 check added [00:14] <ubitux> and* [00:15] <ubitux> (and linesize with max w * 4) [00:16] <ubitux> mmh maybe av_check_image_size().. [00:18] <ubitux> av_image_check_size* [00:25] <ubitux> my brain doesn't work anymore [00:25] <ubitux> 'night ppl :) [01:16] <cone-200> ffmpeg.git 03Michael Niedermayer 07db3a0aae9fdc: msvideo1enc: fix initial keyframe value * 03http://tinyurl.com/8qubk8e03 [01:16] <cone-200> ffmpeg.git 03Michael Niedermayer 07f5581266f767: ra288: assert order to be withinn supported range in do_hybrid_window() * 03http://tinyurl.com/8vkvw2p03 [01:16] <cone-200> ffmpeg.git 03Michael Niedermayer 07901f4bb6fc70: utils: consider mpeg4 in mp4/mov to have a unreliable timebase * 03http://tinyurl.com/8t22sks03 [01:16] <cone-200> ffmpeg.git 03Michael Niedermayer 07ba31e59e83ca: ffmpeg: Override r_frame_rate by corrected timebase only for interlaced streams * 03http://tinyurl.com/8ah7zst03 [02:25] <cone-200> ffmpeg.git 03Michael Niedermayer 071fe45903b81f: msvideo1enc: fix interframe encoding * 03http://tinyurl.com/c4y7de203 [03:15] <cone-200> ffmpeg.git 03Michael Niedermayer 073896cd11a107: mxfenc: explicitly truncate continuity count. * 03http://tinyurl.com/brad5pq03 [03:15] <cone-200> ffmpeg.git 03Michael Niedermayer 079c669672c7fd: x86/motion_est: widen before multiply. * 03http://tinyurl.com/cy5opwz03 [04:06] <cone-200> ffmpeg.git 03Michael Niedermayer 074cc4ca584736: mandelbrot: fix inner=period coloring routine * 03http://tinyurl.com/bszbv6c03 [04:17] <cone-200> ffmpeg.git 03Andrew Euell 0736b3b2376dd9: gitignore: add *.dylib for those of us on Darwin/OS X systems. * 03http://tinyurl.com/8sgr8ux03 [10:50] <ubitux> michaelni: question about a very old commit :) [10:50] <ubitux> c45388b1 [10:50] <ubitux> "if(s->streams[i]->codec->codec_tag == 5){" [10:50] <ubitux> is this a special case for nellymoser? [10:51] <ubitux> ah, i missed that this chunk was already present.. [10:53] <ubitux> 11a8e4257 then :) [10:55] <ubitux> anyway, looks like a nellymoser 8khz special case to me [10:56] <ubitux> mmh or maybe vp6... [10:56] <ubitux> yes vp6a actually :p [11:18] <ubitux> nyuhu: there is no virtualdub repository? [11:18] <ubitux> only tarballs? [11:50] <nyuhu> ubitux : theres the source code on sourceforge& [11:53] <ubitux> sourcecode right [11:53] <ubitux> but what about the history? [11:57] <nyuhu> theres a changelog with a brief of whats new since previous version :/ [11:58] <nyuhu> +the [12:51] <ubitux> nyuhu: ok :( [12:51] <ubitux> it's too bad because it could have given some hints about the constants [13:03] <nyuhu> yes& [13:21] <ubitux> nyuhu: it looks like random to me [13:21] <ubitux> or at least "designed" for random [13:22] <ubitux> and "jran" sounds like j random [13:22] <ubitux> http://sprott.physics.wisc.edu/phys505/lect07.htm [13:22] <ubitux> Example: A = 1366, B = 150889, C = 714025 [13:22] <ubitux> looks similar to your constants [13:24] <ubitux> nyuhu: and they looks totally constant [13:24] <cone-412> ffmpeg.git 03Diego Biurrun 077e68c91e247b: rmdec: Move SIPR code shared with Matroska demuxer to a separate file * 03http://tinyurl.com/8hv6tec03 [13:24] <cone-412> ffmpeg.git 03Luca Barbato 070fbb62a8e6c6: build: support asan and tsan toolchain shortcuts * 03http://tinyurl.com/9hhcz5703 [13:24] <cone-412> ffmpeg.git 03Mans Rullgard 074c995fafd861: configure: simplify get_version() function * 03http://tinyurl.com/9p8xsgv03 [13:24] <cone-412> ffmpeg.git 03Michael Niedermayer 079270a2b374ea: Merge commit '4c995fafd861f537360b3717901cdbed6a6844e7' * 03http://tinyurl.com/9cn7fqp03 [13:24] <ubitux> nyuhu: so you could remove them from the context [13:30] <cone-412> ffmpeg.git 03Mans Rullgard 0766a1ccd7467a: configure: simplify argument handling in check_ld * 03http://tinyurl.com/crpncb703 [13:30] <cone-412> ffmpeg.git 03Michael Niedermayer 07d15be9108e6a: Merge commit '66a1ccd7467ab1913cd8877114c6d4c2588bb12f' * 03http://tinyurl.com/c5yu9bh03 [13:35] <cone-412> ffmpeg.git 03Stefano Sabatini 077ca102a7d7bd: examples: add README file with simple compilation instructions * 03http://tinyurl.com/ch82q5f03 [13:52] <cone-412> ffmpeg.git 03Stefano Sabatini 0725f3827e1daf: tools/ffeval: do not use UNIX-specific /dev/std{in,out} files * 03http://tinyurl.com/8w6c56g03 [13:56] Action: ubitux just founds something pretty funny with vp6 and flv [14:01] <cone-412> ffmpeg.git 03Mans Rullgard 0784e65c763d8f: build: simplify enabling of compat objects * 03http://tinyurl.com/9ogafhq03 [14:01] <cone-412> ffmpeg.git 03Diego Biurrun 07c1fcfdec7546: rangecoder-test: Return in case of an error * 03http://tinyurl.com/9nnmaxu03 [14:01] <cone-412> ffmpeg.git 03Michael Niedermayer 075717562c789f: Merge commit 'c1fcfdec75468009dc7de29a5d1c6adf3b2ef77d' * 03http://tinyurl.com/9jxyv6j03 [14:02] <ubitux> ./ffmpeg -i ~/fate-samples/ea-vp6/g36.vp6 -c copy -y out.flv output is rotated %) [14:04] <JEEB> yeah, I think VP62F or whatever was actually upside down or whatever [14:05] <ubitux> it works fine when muxed in mkv [14:07] <cone-412> ffmpeg.git 03Diego Biurrun 07f1be514540dc: rangecoder-test: Drop timer output that clutters stderr * 03http://tinyurl.com/ccl3zmt03 [14:07] <cone-412> ffmpeg.git 03Mans Rullgard 07099294577c8c: fate: improve md5sum utility selection * 03http://tinyurl.com/dy2f3f403 [14:07] <cone-412> ffmpeg.git 03Mans Rullgard 073dc06b6972cf: tiny_psnr: check for specified sample size less than 1 * 03http://tinyurl.com/cj9cosd03 [14:07] <cone-412> ffmpeg.git 03Michael Niedermayer 0783962004f727: Merge commit '3dc06b6972cf389269e9c36ff0a4373f80f7149b' * 03http://tinyurl.com/bt4eqq703 [14:09] <ubitux> and same in mov [14:13] <JEEB> well, there was some weird thing with muxing vp6X into flv, that's all I remember [14:13] <JEEB> which I remember including upside-downess [14:18] <ubitux> :( [14:19] <cone-412> ffmpeg.git 03Mans Rullgard 07bf868c4a9bc6: tiny_psnr: fix range calculation for sample size of 32 bits * 03http://tinyurl.com/bn8839n03 [14:19] <cone-412> ffmpeg.git 03Mans Rullgard 0707b3790d3626: build: simplify linking tools with cmdutils.o * 03http://tinyurl.com/ck8hlzl03 [14:19] <cone-412> ffmpeg.git 03Michael Niedermayer 078227d36bd356: Merge remote-tracking branch 'qatar/master' * 03http://tinyurl.com/d9z5tfs03 [14:29] <burek> I'm sorry to bother you guys again, but is there any way I could modify the flv muxer to be _really_ verbose with its err output log, so that I can see what doesn't work in vlc, when it is being used through the libav linkage [14:30] <burek> it seems it doesn't mux well the raw h264 stream (coming from the webcam, encoded already, no need to re-encode) with alsa input (encoded at the same time with vlc) [14:31] <burek> described in ffmpeg's vocabulary, it would be like: ffmpeg -f h264 -i /dev/video0 -f alsa -i hw:0 -c:v copy -c:a libaacplus output.flv [14:32] <burek> that's basically what vlc is supposed to be doing prior to http streaming [16:36] <cone-412> ffmpeg.git 03Anton Khirnov 071846f3b5b15f: avconv: fix -force_key_frames * 03Error03 [16:36] <cone-412> ffmpeg.git 03Reimar Döffinger 078efae4cbbf51: avconv: fix parsing of -force_key_frames option. * 03http://tinyurl.com/co7nb5303 [16:36] <cone-412> ffmpeg.git 03Kostya Shishkov 0702b723946279: vc1dec: add flush function for WMV9 and VC-1 decoders * 03http://tinyurl.com/c67hjwo03 [16:36] <cone-412> ffmpeg.git 03Reimar Döffinger 07e9ac06160f45: sipr: fall back to setting mode based on bit_rate. * 03http://tinyurl.com/bllgnts03 [16:36] <cone-412> ffmpeg.git 03Anton Khirnov 07bed5847563d1: bmpdec: only initialize palette for pal8. * 03http://tinyurl.com/dx7q6o903 [16:36] <cone-412> ffmpeg.git 03Janne Grunau 07fdb708078174: Revert "nuv: check per-frame header for validity." * 03http://tinyurl.com/bmgqeyr03 [16:36] <cone-412> ffmpeg.git 03Janne Grunau 076704522ca9dd: nuv: check RTjpeg header for validity * 03http://tinyurl.com/d6t2ctq03 [16:36] <cone-412> ffmpeg.git 03Max Lazarov 0725a1a5b1b38b: eval: fix swapping of lt() and lte() * 03http://tinyurl.com/bt52k7b03 [16:36] <cone-412> ffmpeg.git 03Janne Grunau 077a7229b52d19: imgconvert: avoid undefined left shift in avcodec_find_best_pix_fmt * 03http://tinyurl.com/d6xpmor03 [16:36] <cone-412> ffmpeg.git 03Michael Niedermayer 07da0c45766347: mpegvideo: Don't use ff_mspel_motion() for vc1 * 03http://tinyurl.com/cfjwvtb03 [16:36] <cone-412> ffmpeg.git 03Anton Khirnov 077124fa5d3640: lavf: don't segfault when a NULL filename is passed to avformat_open_input() * 03http://tinyurl.com/cwoswto03 [16:36] <cone-412> ffmpeg.git 03Justin Ruggles 07d9ffa2aca1e4: golomb: check remaining bits during unary decoding in get_ur_golomb_jpegls() * 03http://tinyurl.com/czcvgcg03 [16:36] <cone-412> ffmpeg.git 03Michael Niedermayer 07a1b127515bb7: alsdec: check opt_order. * 03http://tinyurl.com/d4mzppa03 [16:36] <cone-412> ffmpeg.git 03Martin Storsjö 076d1b91324c30: h263: Add ff_ prefix to nonstatic symbols * 03http://tinyurl.com/cne5wp603 [16:36] <cone-412> ffmpeg.git 03Martin Storsjö 078c0bbe51561b: vlc/rl: Add ff_ prefix to the nonstatic symbols * 03http://tinyurl.com/ctscotf03 [16:36] <cone-412> ffmpeg.git 03Janne Grunau 07f695bd601640: rv34: use AVERROR return values in ff_rv34_decode_frame() * 03http://tinyurl.com/d4jy28603 [16:36] <cone-412> ffmpeg.git 03Janne Grunau 0790575bd7dd1e: rv34: Handle only complete frames in frame-mt. * 03http://tinyurl.com/dyq2hpy03 [16:36] <cone-412> ffmpeg.git 03Janne Grunau 07b1ad5a21da7a: rv34: error out on size changes with frame threading * 03http://tinyurl.com/bwlxpj903 [16:36] <cone-412> ffmpeg.git 03Mina Nagy Zaki 07e5f4e2494228: lavfi: avfilter_merge_formats: handle case where inputs are same * 03http://tinyurl.com/dxgkwhj03 [16:36] <cone-412> ffmpeg.git 03Kostya Shishkov 07d4f3abca6a76: indeo3: validate new frame size before resetting decoder * 03http://tinyurl.com/clgh5ex03 [16:36] <cone-412> ffmpeg.git 03Justin Ruggles 07e46cf805b100: vorbisenc: check all allocations for failure * 03http://tinyurl.com/c8np5qz03 [16:36] <cone-412> ffmpeg.git 03Alex Converse 079aaaeba45c41: vorbis: Validate that the floor 1 X values contain no duplicates. * 03http://tinyurl.com/ccj2rfu03 [16:36] <cone-412> ffmpeg.git 03Martin Storsjö 0731bc3fb563b1: snow: Check mallocs at init * 03http://tinyurl.com/cf4vuur03 [16:36] <cone-412> ffmpeg.git 03Michael Niedermayer 07be2dd2559f54: Merge remote-tracking branch 'qatar/release/0.8' into release/0.10 * 03http://tinyurl.com/cq2nks603 [17:12] <burek> michaelni, what is the usual way of sending patches for ffmpeg-web [17:12] <burek> ml/pastebin/..? [17:22] <ubitux> burek: ml [17:22] <ubitux> classic ffmpeg-devel [17:22] <burek> ok thx :) [17:27] <burek> btw, should I attach patch or just paste it into the mail body? [17:28] <ubitux> ideally git send-email [17:28] <burek> oh :$ [17:28] <ubitux> and no, never paste it [17:28] <ubitux> as an attachment is fine, but it's nice if it's inlined [17:30] <burek> hm, i dont have git-send-email ? [17:30] <burek> is that normal? [17:30] <nevcairiel> git send-email, no - after git [17:30] <burek> git: 'send-email' is not a git command. See 'git --help'. [17:31] <nevcairiel> how old is your git? =P [17:31] <burek> there is an entry in man git about git-send-email(1) [17:31] <burek> 1.7.10 [17:32] <burek> solved [17:32] <burek> debian provides it in separate package it seems [17:35] <nevcairiel> how i hate debian and the practice of breaking everything up into meaningless packages =p [17:35] <burek> it is easier for them to maintain it, i guess [17:35] <nevcairiel> how so, all these git tools get updated as one package in upstream [17:35] <burek> don't know, im no debian maintainer :) [17:36] <burek> just guessing [17:36] <burek> nice :) still no git-send-email :D [17:36] <burek> how more stupid this can be [17:36] <Compn> nevcairiel : you mean splitting up -dev headers from libs ? [17:37] <nevcairiel> no, apparently splitting git in multiple packages [17:37] <Compn> 'why wont xxx compile?' - 'because debian' [17:37] <nevcairiel> i can understand splitting off dev headers [17:37] <nevcairiel> normal users dont need them [17:37] <burek> i had to do apt-get install git-email to be able to use git-send-email, but through the use of: git send-email :) [17:37] <burek> shortly :D [17:38] <burek> so, git send-email --to=... --subject=... 0001-blabla.patch [17:38] <burek> right? [17:38] <nevcairiel> you give it the revisions to send, not a patch [17:39] <burek> wait, I don't need git format-patch? [17:39] <nevcairiel> no [17:45] <burek> f it.. i dont have time to setup everything including smtp [17:46] <burek> i added it as attachment and sent it manually [17:49] <ubitux> < nevcairiel> you give it the revisions to send, not a patch // you can as well [17:58] <cone-412> ffmpeg.git 03Mans Rullgard 07b102d5d97dae: h264: allow cropping to AVCodecContext.width/height * 03http://tinyurl.com/cdxflrt03 [17:58] <cone-412> ffmpeg.git 03Mans Rullgard 070054d70f23ed: mov: set AVCodecContext.width/height for h264 * 03http://tinyurl.com/cwa2v7n03 [17:58] <cone-412> ffmpeg.git 03Anton Khirnov 07a60eb6ef12df: ffmpeg: fix -force_key_frames * 03http://tinyurl.com/bqgpvdr03 [17:58] <cone-412> ffmpeg.git 03Kostya Shishkov 070173a7966b33: vc1dec: add flush function for WMV9 and VC-1 decoders * 03http://tinyurl.com/c9c7e6p03 [17:58] <cone-412> ffmpeg.git 03Janne Grunau 07f31170d4e7f9: nuv: check RTjpeg header for validity * 03http://tinyurl.com/buc3k3603 [17:58] <cone-412> ffmpeg.git 03Janne Grunau 078812b5f16410: imgconvert: avoid undefined left shift in avcodec_find_best_pix_fmt * 03http://tinyurl.com/bomlrjp03 [17:58] <cone-412> ffmpeg.git 03Michael Niedermayer 07899d95efe12f: mpegvideo: Don't use ff_mspel_motion() for vc1 * 03http://tinyurl.com/crnx9ug03 [17:58] <cone-412> ffmpeg.git 03Anton Khirnov 0777d43bf42d76: lavf: don't segfault when a NULL filename is passed to avformat_open_input() * 03http://tinyurl.com/c68lp5803 [17:58] <cone-412> ffmpeg.git 03Michael Niedermayer 07b6ba39f931a8: alsdec: check opt_order. * 03http://tinyurl.com/d8rscdg03 [17:58] <cone-412> ffmpeg.git 03Mina Nagy Zaki 07b6c5848a1f8f: lavfi: avfilter_merge_formats: handle case where inputs are same * 03http://tinyurl.com/cr8sesx03 [17:58] <cone-412> ffmpeg.git 03Justin Ruggles 0761ece4137298: vorbisenc: check all allocations for failure * 03http://tinyurl.com/brmmq4503 [17:58] <cone-412> ffmpeg.git 03Alex Converse 07d6e250abfc36: vorbis: Validate that the floor 1 X values contain no duplicates. * 03http://tinyurl.com/d6u6z2a03 [17:58] <cone-412> ffmpeg.git 03Michael Niedermayer 07e28814e0e16c: Merge remote-tracking branch 'qatar/release/0.7' into release/0.8 * 03http://tinyurl.com/ce6cgz703 [18:03] <cone-412> ffmpeg.git 03Mans Rullgard 074eea330a2afa: h264: allow cropping to AVCodecContext.width/height * 03http://tinyurl.com/c5s7kdw03 [18:03] <cone-412> ffmpeg.git 03Mans Rullgard 0744e6cf3f75f0: mov: set AVCodecContext.width/height for h264 * 03http://tinyurl.com/cnsdbjb03 [18:03] <cone-412> ffmpeg.git 03Kostya Shishkov 07aa4121276777: vc1dec: add flush function for WMV9 and VC-1 decoders * 03http://tinyurl.com/cpzwot903 [18:03] <cone-412> ffmpeg.git 03Janne Grunau 07459feb7cce03: nuv: check RTjpeg header for validity * 03http://tinyurl.com/c5l56co03 [18:03] <cone-412> ffmpeg.git 03Janne Grunau 07fd7426ed8985: imgconvert: avoid undefined left shift in avcodec_find_best_pix_fmt * 03http://tinyurl.com/c7bojmm03 [18:03] <cone-412> ffmpeg.git 03Michael Niedermayer 07c82ae85a8a78: mpegvideo: Don't use ff_mspel_motion() for vc1 * 03http://tinyurl.com/c9ukn4j03 [18:03] <cone-412> ffmpeg.git 03Mina Nagy Zaki 07a4e277312cac: lavfi: avfilter_merge_formats: handle case where inputs are same * 03http://tinyurl.com/dxjcxub03 [18:03] <cone-412> ffmpeg.git 03Alex Converse 070e2f415adf5d: vorbis: Validate that the floor 1 X values contain no duplicates. * 03http://tinyurl.com/d29ve7903 [18:03] <cone-412> ffmpeg.git 03Michael Niedermayer 07f48d1fb16720: Merge remote-tracking branch 'qatar/release/0.6' into release/0.6 * 03http://tinyurl.com/coj95b503 [18:17] <cone-412> ffmpeg.git 03Kostya Shishkov 079125aa9218c3: vc1dec: add flush function for WMV9 and VC-1 decoders * 03http://tinyurl.com/dya8op403 [18:17] <cone-412> ffmpeg.git 03Janne Grunau 07f695be22d89a: nuv: check RTjpeg header for validity * 03http://tinyurl.com/cafb5r503 [18:17] <cone-412> ffmpeg.git 03Janne Grunau 077296a6b5e9c2: imgconvert: avoid undefined left shift in avcodec_find_best_pix_fmt * 03http://tinyurl.com/c95tkjn03 [18:17] <cone-412> ffmpeg.git 03Michael Niedermayer 076d6373dc6441: mpegvideo: Don't use ff_mspel_motion() for vc1 * 03http://tinyurl.com/csycyuy03 [18:17] <cone-412> ffmpeg.git 03Mina Nagy Zaki 070dfcbe5285f0: lavfi: avfilter_merge_formats: handle case where inputs are same * 03http://tinyurl.com/bwoltvy03 [18:17] <cone-412> ffmpeg.git 03Michael Niedermayer 07776fb2e10d4a: Merge remote-tracking branch 'qatar/release/0.5' into release/0.5 * 03http://tinyurl.com/d6665sl03 [19:04] <creep> fried potatoes or beans? [19:46] <cone-412> ffmpeg.git 03Michael Niedermayer 07dd3e5baa5996: swr-test: check pointers before use * 03http://tinyurl.com/8saxjkw03 [19:46] <cone-412> ffmpeg.git 03Michael Niedermayer 079de5b11d58ed: swr-test: fix division by 0 * 03http://tinyurl.com/8doxm2g03 [20:12] <cone-412> ffmpeg.git 03Michael Niedermayer 07340305646a82: riff: fix division by zero for G726 * 03http://tinyurl.com/9jh7fzw03 [20:12] <cone-412> ffmpeg.git 03Michael Niedermayer 07de6c15044415: nut: fix int32 overflow * 03http://tinyurl.com/8ggtx8803 [20:12] <cone-412> ffmpeg.git 03Michael Niedermayer 0724d6af0f7e5e: sierravmd: flip reduce arg order. * 03http://tinyurl.com/9dvadjm03 [20:31] <cbsrobot> burek: you could also use the gmail smtp server [21:47] <saste> michaelni: ping on "lavfi/scale: accept named option, make parsing more robust" [21:57] <michaelni> saste, does it still work with changing input resolution after the patch ? [21:57] <saste> I think so [21:58] <saste> how can i test it? [22:00] <michaelni> that is a good question [22:03] <michaelni> you can try to concatenate 2 mpegs with different size and chec per avlog that the scale filter gets differenz input sizes while the whole filter graphs is NOT rebuild [22:04] <saste> I believed we implemented normalization at some point, going to test it now [22:05] <ubitux> yaaargl [22:05] Action: ubitux realized something is wrong with the lavfi metadata patch :( [22:06] <ubitux> we dup the dict in the AVFrame in the copy props [22:06] <ubitux> but by default apps won't manually destroy the dict [22:06] <ubitux> and actually, nothing will [22:06] <ubitux> (except in case of the lavfi device with what we did) [22:07] <ubitux> any suggestion? :( [22:10] <saste> ubitux: why do you need to destroy the dict? [22:11] <ubitux> because there is one allocated per frame? [22:11] <ubitux> which will leak [22:11] <michaelni> ubitux, "decode_frame" free last, dup current, "codec close" free last [22:14] <ubitux> av_dict_free(&avctx->coded_frame->metadata); so we would move this from raw decoders up to the general decode frame? [22:15] <ubitux> ok let's try to activate my brain for once. [22:25] <saste> michaelni, same behavior as current (that is broken) [22:25] <saste> but somehow works [22:25] <ubitux> michaelni: codec close can only access to "coded frame", which is likely no pointing on the last decoded frame most of the time afaict [22:26] <saste> it is broken because is not doing the expected thing [22:26] <michaelni> saste, how can i reproduce the "broken" ? [22:26] <saste> when the size changes, it assumes the new size [22:27] <michaelni> ubitux, you can store a pointer to the dict in the codec context [22:27] <saste> for example if you have scale=2*iw:2*ih it will just keep the new "changed" size [22:27] <ubitux> michaelni: right sure, ok [22:27] <saste> i'd rather expect scale to resize the video accordingly [22:27] <saste> but => unrelated to the patch [22:30] <michaelni> saste, but its working if you use a fixed size like 400:300 ? [22:35] <saste> michaelni, my patch is correct in the sense that it doesn't change the current behavior [22:36] <saste> which is: when the size changes, assume the new input size as output, whatever expression the user set [22:39] <saste> michaelni, BTW can you have a look at the examples/muxing issue? [22:39] <saste> it is disturbing that we ship a broken example, and I don't know which is the best way to fix it [22:43] <michaelni> about the scale patch, id like to look at things before its applied, if theres a bug it should be fixed before changing the code [22:46] <michaelni> about the muxing issue, how can i reproduce it ? [22:47] <saste> michaelni, muxing out.mov [22:50] <saste> michaelni, i don't want to end up fixing the world before committing a simple change (which I have in my local repo since a few weeks) [22:50] <saste> so if the patches doesn't change the current behavior, i'm for applying it [23:16] <ubitux> puting a metadata in the codec context was actually so simple.. [23:16] <saste> life is a mess [23:17] <saste> (^^ in case you need some useful and constructive feedback) [23:18] <ubitux> :D [23:35] <ubitux> yepee! [23:35] <ubitux> it works, and no leak [23:36] <ubitux> thx a lot michaelni, it was actually the simplest thing to do.. [00:00] --- Wed Oct 17 2012

1 0

ffmpeg.log.20121016
by burek 17 Oct '12

17 Oct '12

[01:04] <TheWarden> saste_: thanks so much, that did the job. [02:28] <pablomurad> Alo [02:31] <RunawayDevil> Someone here can provide me some support? [02:43] <RunawayDevil> Someone alive here? [02:49] <funyun> hi. is there a way to rotate a video without re-encoding it? [02:57] <RunawayDevil> im here sometime already and no one until now [02:58] <RunawayDevil> answered me :/ [03:10] <sacarasc> funyun: No, I don't think so. [03:10] <sacarasc> RunawayDevil: You've not asked a question... [04:05] <Santiago> Hello.. need some help with my command under windows =) [04:45] <llogan> funyun: you can rotate it upon playback [04:55] <Eryn_1983_FL> hey peeps how do i install yuv420p [04:58] <Eryn_1983_FL> its not letting me convert from it to mpeg 4 [04:59] <sacarasc> YUV450p is a pixel format, not a video codec. [04:59] <sacarasc> *420 [04:59] <Eryn_1983_FL> ? [04:59] <Eryn_1983_FL> er ok i pb what i got on the cli [05:01] <Eryn_1983_FL> http://sprunge.us/dJaR [05:01] <Eryn_1983_FL> tghere ya go [05:02] <sacarasc> Your ffmpeg is rather old, maybe you could update it? And if you do, something like this should work for Android non-HD: ffmpeg -i input -vcodec libx264 -preset:v slow -b:v 500k -profile baseline -acodec aac -b:a 128k output.mp4 [05:29] <Eryn_1983_FL> :( [05:29] <Eryn_1983_FL> its on my server where i encode stuff [07:01] <lake> i would like to try to pass these options to ffmpeg: x264 -q0 -m1 [07:01] <lake> -q0 works, but -m1 doesn't [07:30] <lake> what container would hold flac/mjpeg [07:31] <llogan> lake: -qp 0 -subq 1, or use x264opts [07:31] <llogan> mkv should hold that combination [07:31] <lake> llogan: thank you so much for the quick replies [07:31] <llogan> but why mess with those settings anyway? that's what the presets are for [07:34] <lake> llogan: i could not find any preset that works like -qp 0 -subq 1 [07:35] <llogan> -qp 0 is lossless [07:35] <lake> llogan: if you know what preset they fall under i would be happy to use it [07:35] <lake> i am streaming from a v4l device and i -subq 1 seems to be better for streaming live [07:35] <llogan> anypreset will work with lossles, but only two are really useful: ultrafast and veryslow depending on what you are doing [07:35] <lake> no xruns [07:37] <lake> i didn't try ultrafast, actually, that might work. [07:37] <llogan> you should be using lossless for streaming [07:37] <llogan> *shouldn't [07:37] <lake> but i have these home videos that i want to archive. [07:37] <lake> should i just use -crf 18 [07:38] <llogan> i don't understand what you are doing. first you mention live streaming, and then you mention archiving [07:40] <lake> llogan: sorry for being unclear. I am "streaming" analog (red/white/yellow cable) to a usb analog to digital converter (red/white/yellow in, usb out) to my pc. 4vl2 is my input device. [07:40] <lake> the source is a handycam from the 90s. [07:41] <lake> i see now that streaming implies over the network [07:41] <lake> that is not what i mean [07:42] <llogan> ive never used anything like that before. [07:44] <llogan> if you have unlimited hard drive space you can use lossless if you like. [07:44] <lake> mmm 120 minutes of film into lossless is going to massive [07:45] <lake> i have 45 films * 2 hours each [07:45] <lake> yikes [07:45] <llogan> of you're limited then -crf 18 should probably be indistinguishable [07:46] <llogan> what does ffmpeg show for your input streams? [07:46] <lake> and the only reason i would reconvert the output would be for dvd. [07:46] <lake> thanks llogan. i think that solves my problems! [07:47] <llogan> https://ffmpeg.org/trac/ffmpeg/wiki/x264EncodingGuide [07:47] <llogan> CRF section [07:48] <lake> 18 looks solid from here [11:31] <Exatrive> I'm capturing in windows.. the audio is off about -1000ms on capture.. its there a way to compisate for this during the realtime recording to disk? [11:32] <Exatrive> ffmpeg -f dshow -r 30 -aspect 16:9 -s %resolution% -i video="SCFH DSF":audio="Stereo Mix (Realtek High Defini" -rtbufsize 9835m -pix_fmt yuv444p -vcodec libx264 -preset ultrafast -tune zerolatency -qp 0 -acodec pcm_s16le Z:\%MM-DD-YYYY%.mkv [11:38] <Exatrive> btw this is being called by a bat file so hence the %.......% stuff [12:41] <gst-kaps> How to make sure ffplay outputs single channels output audio for single channel input audio ? [14:18] <burek> does anyone know of any tool that can tell me what is wrong with this flv video (h264+aac): http://teststream765.chickenkiller.com:8091/live.flv [14:19] <burek> something is wrong with it, but I just can't figure out what [14:19] <burek> it doesn't play in jw player but it plays somehow in vlc and ffplay... [14:24] <relaxed> burek: did you try remuxing it? [14:25] <burek> er.. no, why? [14:26] <relaxed> try it [14:26] <burek> well it breaks the whole point kinda.. [14:26] <burek> the webcam is encoding the h264 video [14:26] <burek> i just add the audio in vlc and mux it into flv and stream it [14:27] <burek> wait, what do you mean remux it, i am muxing it :) [14:27] <burek> i thought you said re-encode it :) [14:28] <burek> webcam gives raw h264 which I'm muxing together with aac in flv (using vlc) [14:34] <relaxed> Hmmm, maybe flash player can't decode the h264 stream. [14:35] <burek> let me copy a part from #ffmpeg-devel [14:35] <burek> described in ffmpeg's vocabulary, vlc is doing this: ffmpeg -f h264 -i /dev/video0 -f alsa -i hw:0 -c:v copy -c:a libaacplus output.flv [14:36] <burek> when i grab the raw video (not h264) and encode both a/v in flv, then it works [14:36] <burek> but if i let the webcam encode it and just pass the video + encode the audio, again in flv, then the video doesn't work, but audio does [14:37] <relaxed> right, flash player doesn't seem to like the h264 stream. [14:37] <burek> how can I tell what's wrong with it [14:38] <relaxed> That is a good question. [14:38] <relaxed> Maybe nothing is wrong with it and the porblem is with flash player. [14:39] <relaxed> Since ffplay and vlc can play it back that would be my guess. [14:39] <burek> :/ i updated everything regarding flash.. [14:39] <burek> i would just like to have some kind of a tool [14:39] <burek> that can compare 2 flv files [14:39] <burek> so that i can tell what is different [14:39] <burek> and try to fix it [14:40] <relaxed> That sounds magical :) [14:41] <burek> :) [14:44] <relaxed> Did you try using fmpeg instead of vlc to capture? [14:44] <relaxed> ffmpeg* [14:44] <burek> it can't capture -f h264 [14:44] <burek> i don't know why [14:44] <burek> vlc can [14:45] <burek> http://pastie.org/private/ajwaid5gea27orjimqmmmg [14:47] <relaxed> Wouldn't you use -f v4l2 ? [14:47] <burek> oh yes, my bad [14:48] <burek> ffmpeg -f v4l2 -vcodec h264 -i /dev/video0 ? [14:49] <relaxed> you may not have to specify the vcodec [14:49] <burek> ok, what cmd should i type [15:22] <relaxed> What is your vlc command? [15:24] <burek> (to avoid spamming the channel) [15:30] <burek> if only I was able to grab it only using ffmpeg [15:31] <burek> I could debug it a low easier [15:31] <burek> but this way I need to bug people from videolan too [15:31] <burek> not knowing where the bug actually is [15:31] <burek> s/low/lot [15:33] <burek> doesn't work: http://pastie.org/private/7dlsd9wcflbrl5gld3z1rg [15:33] <burek> works: http://pastie.org/private/fnfbnee7wthe14d8zkexlq [15:49] <burek> and duration is certainly not 11 minutes.. it is more like a minute or less [15:49] <burek> also "368 pixels" should be 360.. man.. it is so wrong.. [15:50] <relaxed> burek: does this work? ffmpeg -f v4l2 -s 640x360 -r 25 -i /dev/video0 -map 0:v -c copy output.mkv [15:51] <burek> input Stream #0:0: Video: rawvideo [15:51] <burek> output Stream #0:0: Video: rawvideo [15:51] <burek> :) [15:51] <relaxed> ffmpeg -f v4l2 -s 640x360 -r 25 -vcodec h264 -i /dev/video0 -map 0:v -c copy output.mkv [15:51] <burek> [video4linux2,v4l2 @ 0xf79520] Cannot find a proper format for codec_id 28, pix_fmt -1. [15:52] <burek> just that end it quits [15:52] <burek> and* [15:52] <relaxed> ffmpeg -f v4l2 -s 640x360 -r 25 -vcodec h264-pix_fmt yuv420p -i /dev/video0 -map 0:v -c copy output.mkv [15:52] <burek> [video4linux2,v4l2 @ 0x1a23620] Cannot find a proper format for codec_id 28, pix_fmt 0. [15:53] <relaxed> I guess you should file a bug report. [15:54] <burek> well that is a point :) what should I report? :) [15:55] <relaxed> ffmpeg -f v4l2 -s 640x360 -r 25 -vcodec h264 -pix_fmt yuvj420p -i /dev/video0 -map 0:v -c copy output.mkv [15:55] <burek> [video4linux2,v4l2 @ 0x25f4620] Cannot find a proper format for codec_id 28, pix_fmt 12. [15:58] <relaxed> The reason the stream doesn't work with flash player is because it is yuvj420p and not yuv420p. [16:01] <burek> fmt.fmt.pix.pixelformat = V4L2_PIX_FMT_H264; //replace [16:01] <burek> does this have to do anything with that? [16:02] <burek> #define V4L2_PIX_FMT_H264 v4l2_fourcc('H', '2', '6', '4') /* H264 with start codes */ [16:02] <burek> no, i guess it's not that sorry [16:30] <Exatrive> well I fixed my sync issues with directshow filters.. hand to flip the order, Audio=:video= and have -itsoffset 00:00:00.90 right after the video filter.. [16:31] <Exatrive> that is case anyone else comes looking for it. [16:31] <Exatrive> if I had -itsoffset 00:00:00.90 anywhere else it didn't work but it didn't give any errors ether. [16:35] <burek> yes, you basically put audio a little bit before the video that way [16:36] <andyhuzhill> join #libav [16:36] <ubitux> ? [16:37] <burek> he probably misspelled it :) [16:37] <andyhuzhill> :P [16:37] <ubitux> he's probably provocating [16:37] <burek> probably just missing / [16:37] <burek> :) [16:41] <andyhuzhill> I recently want to write a program to make video monitoring, but I never used ffmpeg, where should I start? [16:42] <andyhuzhill> I am also interested about the libav, what's the matter? [16:43] <andyhuzhill> excuse me? [16:44] <JEEB> you should just check the headers related to what you'd be doing [16:44] <JEEB> ffmpeg consists of multiple libraries that it presents [16:45] <JEEB> libavcodec, libavformat, libavutil, libswscale, libswresample, libavresample [16:47] <JEEB> libavformat handles parsing of "containers" into streams that can be thrown into decoders or "repackaged", libavcodec handles the decoding and encoding of things, libavutil has various utlilities that other libraries might as well use, libswscale handles conversions between colorspaces, libswresample and libavresample are two libraries that handle audio resampling/mixing [16:49] <andyhuzhill> Soulhunter, I should go and read some docs about libavformat & libavcodec [16:49] <andyhuzhill> s/Soulhunter/so/g [16:50] <JEEB> yeah [16:50] <JEEB> there are examples available as well [16:50] <JEEB> otherwise the doxygen and the source code are your friend [16:51] <andyhuzhill> JEEB, OK, thank you [17:14] <ubitux> andyhuzhill: help for command line or api usage? [17:14] <andyhuzhill> ubitux:api [17:14] <andyhuzhill> ubitux, I want to compiled it into my program [17:15] <ubitux> andyhuzhill: ok then, https://ffmpeg.org/about.html for an overview of the project, and http://git.videolan.org/?p=ffmpeg.git;a=tree;f=doc/examples;hb=HEAD is a good start for API usage [17:15] <ubitux> build examples etc [17:17] <burek> can anyone check what is wrong with this stream.. http://teststream765.chickenkiller.com:8091/live.flv [17:18] <burek> :S [17:18] <burek> does anything look odd/wrong or anything [17:18] <burek> in terms of video frame size, pixel format, encoding, etc [17:19] <burek> because it won't play in flash players [17:20] <ubitux> i can't atm, i'm behind a proxy [17:20] <burek> ok, thanks anyway [17:20] <ubitux> burek: do you have the issue if you wget it? [17:21] <burek> you mean, can flash player play it as a file? [17:21] <ubitux> i don't know what's your issue [17:21] <burek> flash players play only audio not video [17:21] <burek> so I don't know how to start debugging [17:21] <ubitux> you sure the h264 profile is supported by the player? [17:22] <ubitux> i know some apple devices are limited to a very limited level [17:22] <ubitux> you might need to check in that direction [17:23] <ubitux> https://en.wikipedia.org/wiki/H.264/MPEG-4_AVC#Levels [17:23] <burek> hmh, how to set the profile level via v4l2.. [17:23] <burek> is it even possible [17:24] <ubitux> the device is outputing h264? [17:24] <burek> yes c920 [17:24] <ubitux> huhu fun [17:24] <burek> you couldn't even imagine... [17:25] <ubitux> i'm not sure how you could influence the level [17:25] <ubitux> maybe just by requesting a different resolution [17:25] <ubitux> (see the -video_size input v4l2 option) [17:26] <ubitux> also look at -list_formats [17:26] <ubitux> to check if there are "different" h264 formats [17:27] <burek> yes, good idea, didn't try different video sizes [17:47] <perrth1> Is there any plan to support Freeframe API beside frei0r? [18:20] <arpu_> hello when i use http auth how should i escape the password if there is a ! [18:20] <arpu_> %21 does not work [18:33] <relaxed> arpu_: try \! [19:27] <arpu_> thx relaxed works fine! [20:35] <sine_> can someone explain to me how video gets changed from 29 fps to 25 fps correctly. do they just skip out hte od frame here and there ? [20:51] <lake> what is the benefit of 60fps vs 30fps [20:55] <JEEB> sine_, you can't really do that "correctly" :< [20:55] <JEEB> some people merge frames, others cut frames off [20:55] <JEEB> some make fields from frames and encode interlaced [20:55] <JEEB> etc. [20:55] <JEEB> lake, unless you have actually a 60fps source, absolutely nothing [20:57] <sine_> i have 60 fps mode on my camera and you can make smoother slow motion [20:57] <sine_> you have more source frames to use [20:58] <JEEB> yeah, sure [20:58] <JEEB> you can slow down 60fps to, say, 24fps [20:58] <JEEB> and it'll be still relatively fluid [20:59] <lake> so if i'm capturing from v4l2, how can i know the source [20:59] <lake> source fps [20:59] <JEEB> uhh, the API usually tells you the options >_> [20:59] <JEEB> some cameras over v4l2 give for example 15fps 1920x1080 but 30fps 640x480 [20:59] <JEEB> etc. [21:44] <lake> if i convert a .mkv (x264/flac) => ntsc-dvd, i get a ton of buffer underflow messages from ffmpeg [21:45] <relaxed> join the club [21:45] <lake> i feel so welcomed [21:48] <lake> relaxed: is this a known issue [21:50] <_Adolf_Hit-ler_> lmfao [22:31] <JEEB> lake, you did do it with multiple passes, right? [22:31] <JEEB> single pass rate control is completely derp, 2pass can be somewhat better [23:01] <sine_> Does anyone know a program that i can use to batch process .png or .jpg images. i want to change colour depth and contrast and invert the images for a better black and white camera tracking [23:07] <llogan> sine_: you can probably use a frei0r filter in ffmpeg to change contrast, and negate filter to invert [23:09] <llogan> or of course inverstigate imagemagick or graphicsmagick [23:10] <sine_> ok i have saved what you have said into a text file and will investigate tomorrow after work thanks and good night. [23:12] <llogan> sine_: the frei0r filter in particular is "contrast0r" [23:13] <llogan> and graphicsmagick is generally faster, AFAICT. [23:13] <llogan> (than imagemagick) [23:14] <lake> JEEB: thanks for the tip. i will try multiple pass [23:15] <JEEBsv> also underflows just mean that the encoder couldn't keep the stream within the maxrate/bufsize you set [23:16] <lake> JEEBsv: okay, when i play back the video it's choppy but it sounds great [23:16] <lake> i'll try multipass now and see how it goes [23:17] <lake> this has all been one crazy learning experience. i didn't realize how complex ffmpeg is. [23:17] <lake> very badass tool [23:59] <Mista_D> what is a value between "sameq" and [00:00] --- Wed Oct 17 2012

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