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May 2014
- 1 participants
- 62 discussions
[01:50] <cone-808> ffmpeg.git 03Michael Niedermayer 07master:7efe83996c02: tests/fate-run: add runecho command to run a test and display its output
[01:50] <cone-808> ffmpeg.git 03Michael Niedermayer 07master:565c321cd0d1: tests/fate/libavutil: run cpu test and display the cpus detected feature set
[02:51] <cone-808> ffmpeg.git 03Michael Niedermayer 07master:3af7e7b501d9: avfilter/vsrc_mandelbrot: use av_malloc_array()
[02:51] <cone-808> ffmpeg.git 03Michael Niedermayer 07master:f6d17d2aa98e: avformat/matroskaenc: use av_mallocz_array()
[02:51] <cone-808> ffmpeg.git 03Michael Niedermayer 07master:09cd22860fef: avformat/libnut: use av_mallocz_array()
[04:47] <cone-808> ffmpeg.git 03Alex Sukhanov 07master:8b96f31817be: libavformat/mov: Elimitate double reading of COVR metadata if MOV_EXPORT_ALL_METADATA is enabled
[11:41] <plepere> it's aliiiiive !
[11:42] <plepere> doing more testing, but hevc ASM dbf should be good to go very soon
[14:44] <nevcairiel> Daemon404: GCC 4.8.3 release candidate was cut today, 4.8.3 final is scheduled for the 23rd.. iirc you wondered about it before
[14:44] <Daemon404> oci
[14:44] <Daemon404> oic
[14:45] <Daemon404> 4.9.1 is more interesting, but also ages away
[14:45] <nevcairiel> yeah that'll be 2 month at least
[14:45] <nevcairiel> they dont change their release timing just because of a compiler bug
[14:45] <nevcairiel> :)
[14:46] <Daemon404> they dont have timing
[16:41] <plepere> nevcairiel, ASM dbf patch sent
[16:43] <J_Darnley> I'll download it an test it on win64 in just a moment
[16:50] <ePirat> Hello
[16:50] <ePirat> can someone tell me where is the sourcecode for the hls muxer? I can only find segment muxer and hls demuxer
[16:51] <nevcairiel> the segment muxer can mux in hls format
[16:51] <nevcairiel> and it has more features as the hls muxer, so why not use that?
[16:51] <nevcairiel> in any case, the hls muxer is in libavformat/hlsenc.c
[16:52] <ePirat> hmh the manual recommended using the hls muxer for apple hls streams&
[16:53] <nevcairiel> dunno, i use the segment muxer successfully for hls streaming, just set segment_list_type to hls
[16:53] <ePirat> there is no hlsenc file
[16:53] <nevcairiel> then your code is wrong or out of date, i'm looking at it right now
[16:54] <ePirat> oh I see
[16:55] <ePirat> I have a problem with latest ffmpeg
[16:55] <ePirat> it tells me hls_base_url is unknown: https://gist.github.com/ePirat/00710e09e36ef9fee85a
[16:55] <ePirat> but as I can see in the code the option is defined
[16:57] <nevcairiel> hls_base_url is new, its not in ffmpeg 2.2.1
[16:57] <ePirat> oh
[16:57] <nevcairiel> if you were to use the segment muxer, it had such a feature for a long time already :D
[16:57] <ePirat> would be nice if the docs would point that out
[16:57] <nevcairiel> anyway, need to go
[16:58] <ePirat> thank you very much
[16:58] <ePirat> :)
[17:22] <ubitux> ah, it seems diego wants to drop --build-suffix
[17:23] <ubitux> ofc in theory it could be used to differenciate the -ffmpeg and -libav flavor of the libav* libraries on some systems
[17:23] <ubitux> i guess he doesn't want that
[17:25] <nevcairiel> I use build suffix. Screw him
[17:27] <Daemon404> indeed i replied
[17:27] <Daemon404> and linked to nevcairiel
[17:28] <Daemon404> nevcairiel, did you forget to actually write something in your email
[17:34] <nevcairiel> Phone mailing is hard
[17:35] <nevcairiel> I wrote a proper one tho
[17:36] <nevcairiel> In before "that's FFmpeg"
[17:43] <ePirat> hls muxer seems a bit unusable in 2.2.1, paylist/segments generate with segment muxer cannot be played with quicktime or my iPhone. works in vlc. weird&
[17:44] <nevcairiel> Try segment. :D
[17:44] <ePirat> I did
[17:45] <nevcairiel> Oh right. May need to set a bunch of options. Format to mpegts, and that option that only generates headers for first and last
[17:45] <nevcairiel> That had success for me
[17:46] <nevcairiel> Don't have the code in front of me since I'm on the subway. :p
[17:46] <ePirat> I'll try, thanks
[18:24] <ePirat> wow
[18:24] <ePirat> discovered it after 2 hours
[18:32] <compn> ePirat : so what was problem ?
[18:33] <ePirat> mpegts seems not be playable
[18:33] <ePirat> it worked with h264/aac segments for which i had to use -vbsf h264_mp4toannexb
[18:34] <ePirat> and the -write_header_trailer option
[18:47] <cone-725> ffmpeg.git 03Anton Khirnov 07master:b70d7a4ac72d: lavc: add a native Opus decoder.
[18:47] <cone-725> ffmpeg.git 03Michael Niedermayer 07master:2c7d3ecfc962: Merge commit 'b70d7a4ac72d23f3448f3b08b770fdf5f57de222'
[18:47] <cone-725> ffmpeg.git 03Michael Niedermayer 07master:96cb4c871837: swresample: swr_close()
[18:47] <cone-725> ffmpeg.git 03Michael Niedermayer 07master:ffa05e0802fd: avcodec/opusdec: switch to swresample
[18:56] <ePirat> any idea how I can force ffmpeg to generate more acurate segments?
[18:57] <ePirat> specified length 5, videos total has 16 seconds
[18:57] <ePirat> first segment: 6, 2nd: 6, 3rd: 3, 4th: 1
[19:10] <cone-725> ffmpeg.git 03Carl Eugen Hoyos 07master:beeb7551d6ab: Fix make checkheaders if VDA is not available.
[19:21] <cone-725> ffmpeg.git 03Anton Khirnov 07master:0c1959b056f6: lavf: add AVFMT_FLAG_BITEXACT.
[19:21] <cone-725> ffmpeg.git 03Michael Niedermayer 07master:eacf7d650dfe: Merge commit '0c1959b056f6ccaa2eee2c824352ba93c8e36d52'
[19:31] <cone-725> ffmpeg.git 03Anton Khirnov 07master:c9281a01b78c: lavf: drop the zero-sized packets hack
[19:31] <cone-725> ffmpeg.git 03Michael Niedermayer 07master:f478e8500a3d: Merge commit 'c9281a01b78cc3f09e36300a0ca3f5824d1c74cf'
[19:37] <cone-725> ffmpeg.git 03Anton Khirnov 07master:efc7df6c1f11: lavc: preserve the original private data in avcodec_copy_context()
[19:37] <jamrial> michaelni: wouldn't it be better to let the opus decoder work with avr if ffmpeg is configured with it and not swr?
[19:37] <cone-725> ffmpeg.git 03Michael Niedermayer 07master:9b7cb02319b6: Merge commit 'efc7df6c1f11b20a48e60c3f743ce2331b661973'
[19:38] <jamrial> hardcoding it for swr is as bad as hardcoding it for avr
[19:38] <wm4> the APIs are slightly different
[19:38] <wm4> just enough to annoy users
[19:38] <wm4> the real solution would be to merge swr improvements into avr, and dropping swr
[19:41] <jamrial> in this case it's a matter of some ifdeffery. no need for drastic measures
[19:49] <cone-725> ffmpeg.git 03Anton Khirnov 07master:3b2fbe67bd63: lavc: properly handle subtitle_header in avcodec_copy_context()
[19:49] <cone-725> ffmpeg.git 03Michael Niedermayer 07master:098a699867e0: Merge commit '3b2fbe67bd63b00331db2a9b213f6d420418a312'
[19:49] <cone-725> ffmpeg.git 03Michael Niedermayer 07master:97f856a4c271: avcodec/options: avcodec_copy_context() Check subtitle_header_size instead of setting it
[19:52] <michaelni> jamrial, i know and maintain swr code, so its alot easier for me to use it and deal with it if something doesnt work, using avr sometimes and using swr sometimes means strictly more code that can cause bugs, so i think its not a good idea unless theres some gain from that, is there ?
[19:56] <jamrial> i was simply thinking that just like libav hardcoded the decoder to lavr, we hardcoding it to swr alienates half the user base that chooses one library over the other
[19:57] <michaelni> hmm, i see the argument, and i dont object if you want to maintain support for avr-opus
[19:58] <michaelni> but then i dont vounteer to help if it breaks somehow, i already wondered why opus-avr broke make examples today
[20:03] <jamrial> alright, i'll see how problematic it might end up being and think about it
[20:06] <kurosu_> michaelni, hi
[20:06] <kurosu_> there are people posting to libav about implementing h23's unquantize for neon
[20:07] <kurosu_> I'm wondering if it's actually worth it, considering one may unquantize on the fly while parsing
[20:07] <kurosu_> I'm not sure because of the AC prediction in h263+/mpeg4 part2
[20:09] <michaelni> wm4, the sugested solution seems hard given the existing hostilities. Also if anyone from avr wants to work together, everyone is welcome in ffmpeg or swr.
[20:09] <kurosu_> the alternative would be (albeit slowly) to convert to "unquantize on the fly" and remove dsp fucntions/set it to return functions
[20:29] <michaelni> kurosu_, an optimized dequant would be useful for a encoder which otherwise doesnt use many timeconsuming features
[20:30] <michaelni> unless someone wants to try to implement on the fly dequant for encoders
[20:31] <michaelni> also it would be nice if these discussions would be on ffmpeg-devel ...
[20:31] <kurosu_> unless there's some coeff optimization like r/d-optimized quantization, why wouldn't the encoder have the same info as a decoder would ?
[20:32] <kurosu_> michaelni, agreed, though I was asking to see if it was worth even investigating
[20:33] <kurosu_> oh, I see for the encoder, it might be difficult to remember where they are located
[20:33] <kurosu_> *where the coeffs are located (in the coeff buffer)
[20:36] <michaelni> the encoder probably could use the same on the fly dequant but it doesnt currently
[20:38] <ubitux> michaelni: how large are the samples?
[20:40] <michaelni> ubitux, opus? 122mb
[20:40] <ubitux> aw
[20:40] <ubitux> wtf?
[20:40] <ubitux> wasn't opus supposed to compress well? :))
[20:40] <michaelni> fate hevc-conformace is just 63mb
[20:41] <michaelni> its raw pcm float files
[20:41] <ubitux> can't be generated with aevalsrc or something?
[20:42] <ubitux> mmh wait why would you need raw pcm?
[20:46] <kurosu_> last question: so as to compute stddev and max diff; but you can indeed compress the files losslessly (to half the size?) and it may even reduce test runtime (less data to read)
[20:53] <jamrial> ubitux: same as dts tests, to compare the decoded data with the reference pcm file in question. md5/crc is out of the question for float
[20:58] <wm4> we should add support for NSA fuzzy md5 /troll
[20:59] <cone-725> ffmpeg.git 03Olivier Langlois 07master:af31d58a635b: doc: Add udp broadcast option description
[21:06] <cone-725> ffmpeg.git 03Janne Grunau 07master:5e2ba41d4b94: build: add avresample after avcodec to FFLIBS
[21:06] <cone-725> ffmpeg.git 03Michael Niedermayer 07master:fcb5849a62e5: Merge commit '5e2ba41d4b94de1fa5267081d6c4b6b262c8d86f'
[21:13] <cone-725> ffmpeg.git 03Tristan Matthews 07master:7c5ca546a074: configure: fix enable-libopus help string
[21:13] <cone-725> ffmpeg.git 03Michael Niedermayer 07master:91d7d790d176: Merge commit '7c5ca546a0747a20c7f7fb5550455c3042699ee9'
[21:16] <jamrial> i'm glad i wrote the float resample asm for swr seeing how opus uses it unconditionally in most cases
[21:20] <cone-725> ffmpeg.git 03Janne Grunau 07master:d3f5b94762fb: aarch64: opus NEON iMDCT and FFT
[21:20] <cone-725> ffmpeg.git 03Michael Niedermayer 07master:30cdf384d143: Merge commit 'd3f5b94762fb803c0f3b29f9ad6c5eaa813998ba'
[21:21] <mraulet> michaelni, what is the purpose of key_frame? I think the patch is not totally correct if the goal is to recover Radom Access Point
[21:22] <michaelni> purpose is to be a point from where decoding can start
[21:24] <mraulet> okay so a RAP
[21:24] <michaelni> yes
[21:31] <mraulet> michaelni so I am missing the purpose of the patch since key frame since to be already managed
[21:32] <mraulet> might be a bug somewhere else in the code
[21:32] <michaelni> should i revert it ?
[21:34] <mraulet> I would prefer yes
[21:34] <mraulet> I would like to get a bitstream
[21:34] <mraulet> I can check then
[21:40] <cone-725> ffmpeg.git 03Michael Niedermayer 07master:f8de1caa6e98: Revert "avcodec/hevc: fix outputted AVFrame.key_frame"
[21:41] <michaelni> mraulet, done, if you want to ask the patch author see "[FFmpeg-devel] HEVC decoder always outputs key_frame equal to 1"
[21:42] <mraulet> I was writing an email to him
[21:42] <mraulet> ccing you
[21:43] <michaelni> ok thx
[22:53] <cone-725> ffmpeg.git 03Michael Niedermayer 07master:e9602dcb4d7e: ffmpeg: bitstream filters require split out side data
[22:53] <cone-725> ffmpeg.git 03Michael Niedermayer 07master:7a4424e5ed3a: avcodec/opus: fix doxygen comments to be associated with the correct fields
[23:26] <cone-725> ffmpeg.git 03Christophe Gisquet 07master:d1310c591eb2: x86: sbrdsp: implement SSE qmf_deint_neg
[00:00] --- Fri May 16 2014
1
0
[00:16] <aarobc> So I have freetype2 installed, but it's complaining that it isn't...
[00:17] <sacarasc> Did you install the dev files?
[00:18] <aarobc> pkg-config --modversion freetype2 outputs 17.1.11 so it's definitely there
[00:18] <sacarasc> What does the config.log say?
[00:19] <aarobc> reading it right now...
[00:21] <aarobc> sacarasc: https://gist.github.com/aarobc/671a3e716710655944ef
[00:22] <aarobc> well that's the end of it.
[00:26] <aarobc> Okay, might be something with the configure arguments, because trying it with the arguements in the tutorial and it works fine
[00:28] <jedir0x> how long is ffmpeg.org going to be down
[00:31] <aarobc> well I guess the problem was it didn't like the idea of being static.
[00:31] <sacarasc> IIRC, to link statically, you have to have static libraries.
[01:56] <aarobc> ha ha! adding -j <cores> makes make go so much faster
[08:03] <kode54> wtf
[08:04] <kode54> I tried encoding with libvpx-vp9
[08:04] <kode54> and I got the error "Illegal instruction: 4"
[09:10] <mungustas> hi
[09:10] <mungustas> I want to make a gif from youtube video
[09:11] <mungustas> scratching my head how do I achieve these settings
[09:11] <mungustas> first 5 seconds, with total of 5 frames at final gif (for size)
[09:28] <ParkerR> mungustas, mplayer -vo gif89a -endpos 00:00:05 vid.mp4
[09:29] <ParkerR> http://stackoverflow.com/questions/6079150/how-to-generate-gif-from-avi-usi…
[09:30] <ParkerR> -ss 00:00:00 -to 00:00:05 if using ffmpeg
[09:36] <mungustas> hmm :)
[10:41] <mungustas> hm ParkerR mplayer produces horribly looking gif from .flv
[10:57] <pzich> try using a combination of ffmpeg and imagemagik http://superuser.com/a/730389
[12:16] <anton02> im using this to go from mkv to mp4. The audio track is DTS. What additional stuff should i throw in there to go from dts to a more widely supported format? ffmpeg -i 777.mkv -c: copy blah.mp4
[12:23] <relaxed> anton02: aac or ac3
[12:23] <anton02> relaxed: ac3 i guess
[12:24] <relaxed> No, I meant mp4 supports ac3 and aac
[12:25] <anton02> relaxed: how do i reencode the audio track to ac3?
[12:25] <anton02> ill append the necessary flag to ffmpeg -i 777.mkv -c: copy blah.mp4
[12:25] <relaxed> add -c:a ac3
[12:26] <anton02> so -c: copy -c:a ac3
[12:26] <anton02> to indicate the copying of the video track
[12:26] <relaxed> -c:v copy -c:a ac3
[12:26] <anton02> thanks
[13:01] <mungustas> ParkerR: pzich: what I came with is this
[13:02] <mungustas> ffmpeg -i chelsea.flv -vf scale=540:-1 -r 1 -t 20 frames/ffout%03d.png makes 20 png's
[13:02] <mungustas> convert -delay 80 -loop 0 frames/ffout001.png frames/ffout005.png frames/ffout009.png frames/ffout017.png output.gif
[13:03] <mungustas> stitches together some of those png's for result
[16:13] <Mathieu_Du> Hey folks !
[16:15] <Mathieu_Du> In gstreamer, the "lowres" option is exposed for h264 decoding, however the corresponding context attribute is deprecated and useless, and the recommandation is to use "decoder private options"
[16:15] <Mathieu_Du> I couldn't find out options allowing for downscaling in the H264Context, any ideas ?
[16:15] <Mathieu_Du> (not sure if that's doable either)
[16:16] <Mathieu_Du> I don't mind losing quality, the purpose is to provide a preview, and we don't need full HD samples to be rendered at their actual resolution
[16:23] <ePirat> Hello, I am trying to use ffmpeg's segment muxer, it works fine but I cannot specify a base url
[16:23] <ePirat> ffmpeg -i ~/Downloads/intro.mp4 -hls_time 5 -hls_list_size 0 -hls_base_url "http://epirat.rsync.at/Files/stream/" out.m3u8
[16:23] <ePirat> results in Unrecognized option 'hls_base_url'.
[16:28] <ePirat> Here full output https://gist.github.com/ePirat/00710e09e36ef9fee85a
[16:44] <t4nk100> hello
[16:45] <t4nk100> a simply dude, can I mix video&audio file and audio file in one concrete moment? for example have a 10 minutes audio& video file and need put other extra sound in minute 5
[17:26] <DeadSix27> anyone an iea what "Unable to parse option value "(null)" as sample format"
[17:26] <DeadSix27> means when converting flash videos?
[17:29] <DeadSix27> ye well
[17:29] <DeadSix27> sec
[17:30] <DeadSix27> c_14 http://pastie.org/private/dx9jexpiaxh3ofmtlyzpka
[17:30] <DeadSix27> i assume i just have to define one
[17:32] <c_14> ye, normally the sample format is taken from the input file, but ffmpeg couldn't find one in the input file so you'll probably have to define one.
[17:35] <DeadSix27> c_14: by using -sample_fmt i assume?
[17:35] <c_14> ye
[17:35] <DeadSix27> according to the message: Could not find codec parameters for stream 0 (Audio: pcm_u8, 5512 Hz, mono, 44 kb/s): unspecified sample format
[17:36] <DeadSix27> it says pcm_u8, isnt that the sample rate u8
[17:36] <DeadSix27> i mean what else could it be.
[17:36] <c_14> pcm_u8 is the codec
[17:36] <c_14> unsigned 8bit pcm
[17:37] <c_14> It is however, also the sample format (probably).
[17:38] <DeadSix27> never used -sample_fmt, using -sample_fmt u8 doesnt fix it, so i assume i have to specify the stream? and according to the message its stream 0:0 so i tried -sample_fmt:0:0 u8
[17:38] <DeadSix27> doesnt work either
[17:39] <c_14> Same error? As an input or output option?
[17:40] <DeadSix27> same error
[17:40] <DeadSix27> oh wait
[17:40] <DeadSix27> right
[17:40] <DeadSix27> keep forgetting that
[17:41] <DeadSix27> well same error on input option and same on output
[17:42] <c_14> Can you play the swf with ffplay?
[17:43] <DeadSix27> sec
[17:43] <DeadSix27> nopeyes
[17:43] <DeadSix27> yes**
[17:43] <DeadSix27> nope was from another chat window, ignore it
[17:43] <DeadSix27> plays fine.
[17:44] <DeadSix27> even though it gives the same warning
[17:44] <c_14> Which audio stream do you want?
[17:44] <c_14> The pcm stream, the mp3 stream or both?
[17:44] <DeadSix27> both preferred muxed into mkv as seperate streams
[17:44] <DeadSix27> but mp3 has more priority
[17:45] <DeadSix27> in fact i would be happy if i get atleast the mp3 to work
[17:45] <c_14> If you want both, you'll have to add a -map 0:a to the output. But just to test, can you add -map 0:2 as an output option?
[17:45] <c_14> I just want to test if it works if we ignore the pcm stream for now.
[17:53] <DeadSix27> c_14, yep ill try it out
[17:53] <aleb> If I convert with "ffmpeg -i x.MTS -acodec copy -vcodec copy -sn x.MKV", do I loose anything besides the "s" channel?
[17:54] <c_14> That depends on how many streams the input has. ffmpeg by default only copies one audio and one video stream. Other than that, you shouldn't lose anything.
[17:55] <DeadSix27> c_14 how would i even handle non /2 width's ?
[17:56] <aleb> I see it only has one video and one audio stream - ok, thanks!
[17:56] <DeadSix27> (also using -map 0:2 worked (audio only)
[17:56] <c_14> sec, I did that yesterday. Let me search my history.
[17:57] <c_14> -vf scale="trunc(iw/2)*2:trunc(ih/2)*2"
[17:57] <DeadSix27> answer to me or aleb?
[17:57] <c_14> That'll round down to the nearest even width/hight.
[17:57] <c_14> to you DeadSix27
[17:57] <DeadSix27> ah thamks
[17:57] <DeadSix27> n*
[17:58] <DeadSix27> oh i have to mention, the video in the flash is looped to the length of the audio
[17:59] <DeadSix27> so in fact the resulting video has to be the length of the audio
[17:59] <DeadSix27> and keep the video looped
[17:59] <DeadSix27> (thats pure assumption)
[17:59] <DeadSix27> because the resulting video was only 1 frame (even though original is longer)
[18:04] <DeadSix27> c_14 using only the video stream works btw.
[18:06] <DeadSix27> c_14 is it even possible to loop a stream?
[18:06] <DeadSix27> e.g something like -loop -map 0:1
[18:06] <c_14> You can try reading the audio stream from the input file and then using the movie filter to read the video stream from the swf and use :loop=0 and -shortest to cut to the length of the audio stream, but iirc that currently just gets ffmpeg caught in an infinite loop: https://trac.ffmpeg.org/ticket/1799
[18:07] <DeadSix27> hm
[18:07] <c_14> If you know exactly how often you want to loop the video you can use :loop=x, and that _should_ work, but I've never tested it successfully.
[18:08] <c_14> There is a -loop option, but that works only as an input option for static images and as an output option for formats like gif where you can set a loop count.
[18:10] <DeadSix27> ill try out that movie filter combination first
[18:10] <DeadSix27> never used that before
[18:12] <DeadSix27> c_14 using the movie filter results in a corrupted video
[18:12] <DeadSix27> its deformed etc.
[18:13] <DeadSix27> hm lets try something else
[18:19] <c_14> If you know how often you want to loop the video, you can use the concat filter, I've done that before. It _is_ rather tedious, but I don't know a better way of doing it with ffmpeg.
[18:20] <DeadSix27> in fact i dont know how often i want to loop it
[18:20] <DeadSix27> theres prob a way to calculate it: e.g (length of audio)/(len of video)
[18:20] <pkug> hi there, i'm trying to figure out how to run ffmpeg to record rtsp stream for the last hour (only video of the last hour or other specified amount of time should be kept in the recorded file), can anyone help ?
[18:20] <DeadSix27> but i doubt you can do that in the same command line
[18:21] <pkug> ..and to do it efficiently with fair encoding so that resulting video is not gigs of raw data
[18:24] <c_14> DeadSix27: Yeah, I don't think you could do it in the same command line. You'd have to get the length of the audio stream, the length of the video stream and then calculate how often you want to loop the video.
[18:24] <DeadSix27> c_14 in fact using the movie filter i get just tons of dropped frames
[18:24] <c_14> Hmm, that shouldn't be happening.
[18:24] <DeadSix27> also c_14
[18:25] <DeadSix27> i have the resulting video as h264 stream, and the audio as aac
[18:25] <DeadSix27> from there on there must be a easier way to just loop the h264 stream
[18:25] <DeadSix27> (theY're both seperated files)
[18:25] <DeadSix27> thats theo nly way i yet got it working hm
[18:25] <DeadSix27> (which is now a 2 line command line :|)
[18:26] <DeadSix27> going from a 200char command line to a 2line++ python script is annoying step :p
[18:27] <c_14> pkug: Recording the rtsp stream and encoding it to a file is relatively easy, the problem is that you (if I'm reading you correctly) want only one hour's worth of video in the file and then want to delete any output older than an hour. You _could_ use the nice properties of ts streams to capture one hour of video and then remove x minutes of video from the beginning every x minutes, but that would get rather
[18:27] <c_14> complicated really fast. You might want to wait and see if someone else has a better idea.
[18:28] <c_14> DeadSix27: At this point you can just use the concat filter to concat the video stream n times and mux that with the audio stream.
[18:28] <DeadSix27> ye
[18:28] <DeadSix27> was about to say
[18:29] <DeadSix27> guess ill just write a concat file
[18:29] <c_14> Since loop doesn't do what you want/doesn't work that is probably the best option.
[18:29] <pkug> c_14, hmm, current solution i have now is to record files into separate segments by date and "rotate" them with another daemon/service (sort of like logrotate but for video files) but it requires two components :\
[18:32] <c_14> pkug: You're probably going to need 2 components in either case unless you write something yourself. Your solution has the same base as mine, just that you're using multiple files and I'm using a single ts file and then just cutting blocks out of it every now and then. Your solution is also easier to implement.
[18:35] <DeadSix27> c_14 thanks anyway, using concat i got it fully working.
[18:36] <DeadSix27> c_14 requires a python script, but the result is worth it
[18:36] <c_14> If at first you don't succeed, try scripting it.
[18:36] <DeadSix27> im also highly suprised how that 2xx*something tiny video
[18:37] <DeadSix27> can be upscaled to my 2k hd screen
[18:37] <DeadSix27> and look rather nice
[18:37] <DeadSix27> i mean going by the original size
[18:37] <DeadSix27> thats kinda awesome
[18:37] <DeadSix27> sure is blurry, but not thaat muc
[18:47] <pkug> c_14, thanks though. One lame question: what sort of video encoding would be the most efficient for CCTV records (good quality not needed) ?
[18:48] <c_14> I'd probably just go with h264 with a low crf and -preset faster
[18:49] <c_14> s/low/high
[18:49] <pkug> thanks, i'm currently using it, will try those additional options
[20:02] <AndrzejL> guys my laptop is powering down while using ffmpeg due to the critical temperature. Its a kernel / acpi bug. Can I use something to force cpu to not to use 100%? Would using nice help or something? I want to use ffmpeg just restrict how much cpu can it use...
[20:05] <AndrzejL> forgot to add - cpu is clean / fan is clean / fresh cooling grease arctic mx-4 applied less then 2 weeks ago...
[20:08] <AndrzejL> I would need to acpi=off it but then bluetooth isnt working and no fn keys and no touchpad... No biggie since I use mouse and rarely use bluetooth and the vol up keys but its an annoyance
[20:11] <aarobc> Why would an older version of ffmpeg compiled with libfaac transcode m4p audio faster than the newest version using the superior libfdk-aac?
[20:12] <pzich> quality takes time
[20:12] <Reventlov> still nobody to review that ? https://trac.ffmpeg.org/ticket/3630
[20:15] <dloo> hi
[20:16] <dloo> i'm tring to record from v4l2i, but the -vc 1 is not working :-/
[20:38] <fajung> how can I do to record my desktop with sound from the speakers (ubuntu 14.04?)
[21:02] <aarobc> stupid question: how do I send something from ffmpeg to sout?
[21:03] <pzich> aarobc: use a '-' instead of an output filename
[21:04] <aarobc> ffmpeg -i hot.m4a -ab 128k -v 0 -f mp3 - ?
[21:04] <sacarasc> You'll also need -f something
[21:04] <sacarasc> To force a format.
[21:05] <pzich> aarobc: that looks about right, having problems?
[21:05] <aarobc> pzich: well I'm testing this locally because I'm having an issue on my subsonic server on the new version of ffmpeg
[21:06] <aarobc> so I'm testing it by trying to pipe standard output into vlc just to make sure the basics are working
[21:06] <aarobc> ffmpeg -i hot.m4a -ab 128k -v 0 -f mp3 - | vlc doesn't work
[21:06] <pzich> is there a reason you need to pipe it into vlc instead of creating the file and then opening it?
[21:06] <aarobc> pzich: because I want transcoding to happen as fast as possible
[21:07] <aarobc> So I want the file to be able to start playing before waiting for the transcode to finish
[21:08] <pzich> I'm not sure if piping will change that, you may need other flags that optimize for streaming
[21:09] <aarobc> Indeed, that's what I'm wondering, but I have to at least get the basic concept working so I have somewhere to start from
[21:09] <pzich> can you paste a link with the output?
[21:10] <relaxed> aarobc: you need "| vlc -"
[21:10] <pzich> ah
[21:10] <pzich> makes sense
[21:10] Action: pzich has not piped things into vlc before
[21:11] <aarobc> https://gist.github.com/aarobc/7750d2ef12cbeaba427e
[21:11] <relaxed> well, I tested with mpv
[21:11] <relaxed> but the same logic aplies
[21:11] <relaxed> applies*
[21:11] <sacarasc> Haha, video players and logic.
[21:11] Action: relaxed pipes sacarasc to /dev/null
[21:12] <sacarasc> :(
[21:12] <aarobc> relaxed: no worky, see link
[21:12] <relaxed> try, ffmpeg -i hot.m4a -ab 128k -v 0 -f mp3 - 2>/dev/null| vlc -
[21:13] <aarobc> main stream error: cannot pre fill buffer
[21:13] <aarobc> relaxed: same err
[21:14] <relaxed> use mplayer, mplayer2, or mpv
[21:14] <relaxed> that's whait also works with ffplay
[21:14] <relaxed> er, it also works with ffplay
[21:15] <aarobc> relaxed: okay, mplayer actually makes sound
[21:16] <relaxed> mplayer is king
[21:16] <aarobc> kinda skips the first 20 seconds though
[21:16] <aarobc> but it's at least a place to start from
[21:17] <relaxed> why do you need this on the fly?
[21:18] <relaxed> fg sacarasc
[21:20] <aarobc> relaxed: using this software, trying to acomidate more codecs though and the precompiled version included with it doesn't handle opus http://www.subsonic.org/pages/transcoding.jsp
[21:25] <aarobc> okay
[21:25] <aarobc> confirmed
[21:26] <aarobc> I copied the ffmpeg binary to my local machine, ran the command to pipe it into mplayer and it started playing INSTANTLY
[21:27] <aarobc> the thing is I've compiled newer versions of ffmpeg using the exact same libraries and it's still buggy and laggy
[21:27] <aarobc> but the older version works much faster and I have no idea why
[21:28] <aarobc> relaxed: any idea why this would happen?
[21:37] <relaxed> No, I'm using the latest from git master and it works fine.
[21:44] <aarobc> ugh. I dread the idea of cloning that repo, so freaking huge
[21:46] <aarobc> relaxed: you're telling me you can run this command 'ffmpeg -i hot.m4a -ab 128k -v 0 -f mp3 - | mplayer -' and it plays instantly and correctly with an m4a file?
[22:08] <aarobc> ...
[22:09] <aarobc> relaxed: ping
[22:41] <aarobc> Okay, just compiled from the git repo and still having the same issue
[22:42] <aarobc> however when I'm running from an older version that I didn't compile it works instantly
[22:44] <aarobc> in fact the older version I can run this and it works perfectly: "./ffmpeg -i song.m4a -ab 128k -v 0 -f mp3 - | cvlc -"
[22:50] <aarobc> was ffmpeg forked?
[22:51] <sacarasc> Yes.
[22:52] Action: aarobc punches a baby
[22:52] Action: sacarasc waves at Rincewind.
[22:57] <aarobc> Okay, the problem comes down to not being able to send to standard output. getting error "main stream error: cannot pre fill buffer"
[22:57] <aarobc> version compiled in 2011 works, version compiled just now from git repo doesn't
[23:04] <aarobc> someone willing to test this on an m4a file (aac) "ffmpeg -i song.m4a -ab 128k -v 0 -f mp3 - | cvlc -"
[23:06] <ubitux> aarobc: if you find a regression please open a ticket
[23:06] <ubitux> https://trac.ffmpeg.org/
[23:09] <ubitux> ffmpeg -f lavfi -i 'aevalsrc=sin(440*2*PI*t)' -ab 128k -v 0 -f mp3 - | mpv -
[23:09] <ubitux> this works for me
[23:09] <ubitux> works with cvlc as well
[23:09] <ubitux> aarobc: can you confirm that it works for you with the above cmd line?
[23:10] <aarobc> ubitux: testing...
[23:11] <aarobc> don't have mpv installed, using mplayer...
[23:12] <ubitux> replace with cvlc
[23:12] <ubitux> it works as well
[23:13] <aarobc> ubitux: that does work with the version I just compiled
[23:13] <aarobc> assuming it's supposed to make a tone
[23:13] <ubitux> yes
[23:13] <ubitux> but it doesn't with your m4a?
[23:14] <ubitux> can you share that m4a?
[23:14] <aarobc> ubitux: sure. one sec
[23:15] <aarobc> ubitux: aarobc.com/hot.m4a
[23:16] <ubitux> seems to work here
[23:16] <ubitux> what's the issue?
[23:17] <aarobc> ubitux: so you run that string using cvlc and it plays?
[23:17] <ubitux> mmh nothing happens with vlc, but it works with mpv and mplayer
[23:18] <aarobc> ubitux: try just vlc then. should still work
[23:18] <ubitux> it doesn't
[23:18] <ubitux> nothing happens
[23:19] <aarobc> ubitux: you're getting the "cannot pre fill buffer" error?
[23:19] <ubitux> no
[23:19] <ubitux> i'm not getting anything
[23:19] <aarobc> ubitux: when did you compile your current ffmpeg?
[23:19] <ubitux> current HEAD
[23:20] <ubitux> i tried 2.2.2 as well
[23:21] <aarobc> ubitux: what did you use when compiling? Any chance you could post the binary somewhere?
[23:22] <aarobc> Or did you download a precompiled one?
[23:22] <ubitux> no option in particular
[23:22] <aarobc> ubitux: what do you see when you just run 'ffmpeg'?
[23:22] <ubitux> tell me what you want to know exactly
[23:23] <ubitux> because it's probable the mismatch is on vlc side here
[23:23] <ubitux> i'm using 2.1.4, built with ffmpeg
[23:23] <ubitux> (ffmpeg 2.2.2)
[23:24] <aarobc> but no additional libraries when you compiled it?
[23:26] <aarobc> ubitux: post the output when you run 'ffmpeg' somewhere
[23:27] <ubitux> well i have libmp3lame ofc
[23:27] <ubitux> why do you think this would affect the test?
[23:27] <aarobc> ubitux: because I really can't think of anything else why it wouldn't work
[23:27] <ubitux> well
[23:28] <ubitux> does it work for you with mplayer?
[23:28] <aarobc> ubitux: not with the current one that I just compiled
[23:29] <ubitux> ffmpeg 2.2.2, lame 3.99.5, vlc 2.1.4, mplayer 37051
[23:29] <ubitux> this is what i'm using
[23:30] <aarobc> maybe my sintax is off for mplayer. should I jjust be able to substitute 'mplayer' for 'cvlc' in the string?
[23:31] <ubitux> that's what i did yes
[23:32] <aarobc> okay. am going to try downloading latest binary, maybe that will work :-p
[23:33] <aarobc> same problem
[23:36] <aarobc> ubitux: what distro are you on?
[23:36] <ubitux> arch
[23:37] <aarobc> could that possibly be the issue?!
[23:37] <aarobc> I'm on ubuntu
[23:37] <ubitux> ubuntu uses the fork for every app
[23:37] <ubitux> every multimedia* app
[23:38] <ubitux> that's a completely different setup anyway
[23:44] <aarobc> but I compiled my own ffmpeg!
[23:45] <aarobc> why does the universe hate me
[23:49] <goulard> how does ffmpeg interleave audio and video when muxing ts?
[23:49] <goulard> does it doing it based on pts values?
[00:00] --- Fri May 16 2014
1
0
[00:03] <J_Darnley> Which country has a timezone of GMT-10224?
[00:03] <J_Darnley> (yes, I did have that calculated)
[00:03] <ubitux> maybe the moon
[00:04] <J_Darnley> :)
[00:17] <jamrial> well, dc set the bar so high everything else feels a bit disappointing
[00:18] <jamrial> still, all six are faster than their ssse3 counterparts
[00:18] <jamrial> i'll send a patchset to the ml later
[00:40] <tmm1> how can i run a subset of the fate tests locally?
[00:43] <tmm1> ah, make fate-list to get a list of targets and make target
[01:57] <cone-70> ffmpeg.git 03Michael Niedermayer 07master:c3417ed7fd92: swscale/utils: Add check that ensures that the hardcoded struct offsets are valid
[02:43] <michaelni> Skyler_, is the use of the per level reference frame num reducing code in "http://permalink.gmane.org/gmane.comp.video.ffmpeg.devel/176739" (its based on a few lines of af2a4ecd7bcefc97c8aa83913c9a2980206f9cd0 from you) under LGPL in libavcodec/libx264.c ok ?
[02:44] <Skyler_> zYes
[02:44] <Skyler_> Yes
[02:51] <michaelni> thx
[03:08] <cone-70> ffmpeg.git 03Michael Niedermayer 07master:0aac9b76bc77: avcodec/libx264: Implement reference frame count limiting based on level
[06:22] <cone-419> ffmpeg.git 03Michael Niedermayer 07master:14e2e40f3b5b: avfilter/vf_hqdn3: use av_malloc_array()
[06:22] <cone-419> ffmpeg.git 03Michael Niedermayer 07master:6ab37221c8ee: avfilter/vf_removelogo: use av_malloc_array()
[07:51] <cone-419> ffmpeg.git 03Clément BSsch 07master:bc2e39c4cc35: avfilter/lut3d: move the scale calc out of the inner loop.
[09:10] <ubitux> note: it seems audacity is considering dropping ffmpeg support because 1) no one to update their code 2) api is changing too much 3) gst has ff/av support
[09:11] <ubitux> see http://bugzilla.audacityteam.org/show_bug.cgi?id=606#c2 and https://ffmpeg.org/pipermail/libav-user/2014-May/006791.html
[09:12] <wm4> oh yeah that's sad
[09:13] <wm4> they're considering gst because ffmpeg is moving too much
[09:13] <wm4> can't they use ffms2? afaik that has audio support
[09:13] <wm4> oh, they need encoding
[09:14] <wm4> would be so great if ffmpeg just had a simple and stable high-level API
[09:15] <thardin> hmm, I do use audacity
[09:15] <wm4> maybe we should even consider turning ffmpeg.c/ffplay.c into a library
[09:15] <wm4> (not that that'd be a great idea, but it could lead to quick success without too much work)
[09:16] <wm4> ubitux: also this was 2013
[09:16] <ubitux> my distro is packaging a "compat ffmpeg" (aka ffmpeg 0.10) just (almost) for audacity :p
[09:16] <wm4> more than 6 months ago
[09:16] <ubitux> the mail is recent though
[09:16] <wm4> yeah
[09:17] <ubitux> there are exactly 4 packages that needs the old ffmpeg version here
[09:17] <ubitux> sorry, 5
[09:17] <wm4> audacity, and which others?
[09:17] <ubitux> audacity, motion, synfig, openscenegraph and xjadeo
[09:17] <ubitux> i think openscenegraph is the only relevant one with audacity
[09:18] <ubitux> but maybe i'm wrong
[13:38] <BBB> plepere: if you need help with using these macros, feel free to ask me (or any of the x264 people in here, or ubitux would know also, probably michaelni also)
[14:01] <cone-419> ffmpeg.git 03Anton Khirnov 07master:a738540366c9: lavf: properly document the distinction between flags and ctx_flags
[14:01] <cone-419> ffmpeg.git 03Michael Niedermayer 07master:6a72260e6baf: Merge commit 'a738540366c9b114949b7914c0d08e2c28982cfb'
[14:02] <ubitux> did i miss a question?
[14:04] <compn> [08:38] <plepere_> BBB : I'm trying to use the x86util.asm transpose code instead of the one defined in hevc_deblock.asm
[14:04] <compn> was the last message i saw
[14:04] <compn> yesterday?
[14:04] <BBB> yes
[14:05] <BBB> basically "if you have questions about x86util.asm/x86inc.asm, ask any of these people"
[14:14] <cone-419> ffmpeg.git 03Janne Grunau 07master:7e90133f6420: build: do not append $(FFLIBS-) to $(FFLIBS) if $(NAME) is not defined
[14:14] <cone-419> ffmpeg.git 03Michael Niedermayer 07master:dbd1fdd21400: Merge commit '7e90133f6420b1c53652f972b9561600822881ee'
[15:07] <plepere> BBB : thanks. I'll ping you if I need anything, but I think that there are some important problems in the code that need resolving.
[15:07] <plepere> for instance, I can replace TRANSPOSE8x8W as it is in store, but not in the load.
[15:29] <Daemon404> [14:17] < tsieg> RiCON: btw CyanogenMod merged FFmpeg into Android: http://review.cyanogenmod.org/#/c/63444/
[15:29] <Daemon404> lul
[15:30] <wm4> what is that even about
[15:31] <wm4> I see something about libstagefright
[15:31] <Daemon404> the message is a mess
[15:33] <nevcairiel> sounds like they have a a module to use ffmpeg through libstagefright to decode all sorts of media otherwise not supported
[15:33] <Daemon404> ffmpeg has libstagefright support
[15:33] <wm4> I thought libstagefright was for the other way around
[15:33] <Daemon404> maybe libstagefright can call ffmpeg to call libstagefright
[15:33] <Daemon404> wm4, its both.
[15:34] <nevcairiel> the ffmpeg module is mostly to get to the hardware decoder, isnt it
[15:35] <compnn> i thought libstagefrieght called omx hw ?
[15:35] <compnn> ;P
[15:41] <Daemon404> we need to go deeper
[15:58] <cone-419> ffmpeg.git 03Michael Niedermayer 07master:0fdc3cd86fdd: avutil/version: add FF_CONST_AVUTIL53
[15:58] <cone-419> ffmpeg.git 03Michael Niedermayer 07master:acb6f3af4fce: avutil/fifo: delay addition of const from 78d3453c4a2efef9bc079e8f3458653beafcf990 until next major ABI bump
[15:58] <cone-419> ffmpeg.git 03Michael Niedermayer 07master:3690393f681d: avutil/dict: delay addition of const from e12a73246d8ce7d0fc4036522688934e26de4bb1 until next major ABI bump
[16:00] <ubitux> huh, how can this cause build errors?
[16:03] <michaelni> assigning function pointers and them misatching is one way
[16:03] <ubitux> mmh ok
[19:04] <Voicu> hello, does anyone know how to make a callback for an avio context in java?
[19:05] <Voicu> I'm trying to extend avformat.Read_packet_Pointer_byte_int but I get a "invalid indirect reference"
[19:05] <Voicu> when the callback gets called
[19:08] <cone-419> ffmpeg.git 03batguano999 07master:434ba17b22c6: configure: add encoding to --enable-libopus doc.
[20:34] <cone-419> ffmpeg.git 03Peter KováY 07master:973de9ebf879: avcodec/hevc: fix outputted AVFrame.key_frame
[20:35] <nevcairiel> this can't be right, certainly there can be I frames which are not key frames
[20:39] <Daemon404> yes
[20:39] <Daemon404> many times actually.
[20:39] <compnn> depends on codec ?
[20:39] <Daemon404> types*
[20:42] <muken> wtf
[20:42] <nevcairiel> compnn: it clearly says which codec we're talking about, pay attention!
[20:45] <compnn> bah
[20:46] <compnn> like anyone has actually read the entire hevc spec :P
[20:49] <Daemon404> even h264 had nonkeyframe I frames
[20:51] <Paranoialmaniac> because of multiple references...
[21:41] <cone-419> ffmpeg.git 03Michael Niedermayer 07master:56193d33be93: avformat/http: remove never twice executable loop
[21:43] <wm4> so, an I frame which is not key frame means a frame following the I frame could have references to frames before that I frame?
[21:47] <michaelni> yes
[21:47] <michaelni> i should have fixed the commit message better
[21:47] <michaelni> no question its not always correct after the patch
[21:48] <michaelni> but before every was marked as keyframe
[21:49] <wm4> but in lavf, "keyframe" should mean "you can start decoding from here"
[21:51] <michaelni> yes
[21:52] <Paranoialmaniac> that is a bit wrong since we mark frame with recovery point sei as a keyframe
[21:52] <Daemon404> yeah, and open-gop-style h264 doesnt seek properly
[21:53] <Daemon404> its been a bug for many years
[21:53] <Daemon404> not that there's a "good" way to do that unless you keep an index of what references what
[21:53] <Daemon404> like a d2v file does
[21:54] <michaelni> "you can start decoding from here" doesnt mean that this very packet contains a fully decodeable and artifact free frame
[21:55] <michaelni> so in the recovery point case theres a bit to feed into the decoder first before the first "clear" image can come out
[21:56] <michaelni> also about h264, if anyone has testcases that show artifacts when seeking, iam interrested
[21:56] <michaelni> ive fixed all that have been reported IIRC
[21:58] <Paranoialmaniac> leading frames after a keyframe in output order are always decoded incorrectly when seeking :P
[21:58] <michaelni> about hevc, just to make sure i dont misunderstand anyone, do you agree that patch was ok to apply and just not a complete solution to the keyframe value or is there some issue iam missing and i shuld revert ?
[21:58] <Plorkyeran> for recovery points it'd be very helpful if lavf actually reported the recovery duration
[21:59] <Plorkyeran> there's a field on the context with a comment claiming it's the recovery duration
[21:59] <Plorkyeran> but it's never set
[22:00] <Daemon404> is it possible to set that for PIR too?
[22:00] <Paranoialmaniac> we need parse frames to get the recovery duration unless the container has it
[22:00] <Daemon404> which would be all P, no?
[22:00] <Plorkyeran> the recovery duration is a SEI field
[22:00] <Plorkyeran> which is already parsed
[22:00] <Plorkyeran> and then discarded
[22:00] <Paranoialmaniac> no. the duration is not given by the sei
[22:01] <Plorkyeran> sei_recovery_frame_cnt
[22:02] <Plorkyeran> is the recovery duration in frames
[22:02] <Paranoialmaniac> recovery point sei tells us frame_num (avc) or poc (hevc) which completes the recovery
[22:02] <Paranoialmaniac> not duration
[22:02] <Plorkyeran> those are equivalent if you know where you are in the stream
[22:03] <Paranoialmaniac> but how do you know that without parsing
[22:04] <Plorkyeran> what?
[22:04] <Plorkyeran> if you have values from the sei fields then you're parsing stuff
[22:05] <Paranoialmaniac> you can't know the duration until you get corresponding frame_num or poc
[22:05] <michaelni> Daemon404, do you have a Hevc PIR sample ?
[22:05] <Plorkyeran> that's fine
[22:06] <Plorkyeran> I am okay with having to parse the entire file to be able to seek accurately in it (because that's pretty unavoidable)
[22:10] <Daemon404> michaelni, i meant h264 PIR
[22:14] <michaelni> hmm, yes its possible to extract the recovery frame count for h264 PIR, not the duration though without reading ahead
[22:14] <Daemon404> michaelni, we (and Plorkyeran) always preindex the file
[22:14] <Daemon404> just fyi
[22:15] <Plorkyeran> I'm not terribly concerned about the form the data comes in
[22:15] <Plorkyeran> anything vaguely sane would be useful
[00:00] --- Thu May 15 2014
1
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[02:21] <nitero> hi, are there any gui's for ffmpeg, for linux? i need to convert .mkv to .avi with a certain resolution. i'm not adverse to learning CLI but i'm not very experienced with it ... i'm reading the manpage for ffmpeg right now
[02:25] <c_14> There are, but I don't know any good ones. I'd just go with the cli if I were you. It's pretty easy to learn and allows you to accomplish rather complex things once you're used to it.
[02:46] <c_14> If you're just trying to convert to something specific, there's usually a wiki page with examples that you can look at.
[02:47] <nitero> okay, that sounds good, i'll look that up, thank you =)
[08:45] <maksimkaaa> Hi, I want to type a timestamp on every image of a set of 1000 frames I have, I am planning to convert the images into mp4 but I need a guide about putting a TimeStamp on every images.. is there any good doc about that?
[08:45] <maksimkaaa> **The timestamp is different for every image.
[08:54] <blippyp> maksimkaaa: do you have a table of the filenames and timestamps?
[09:13] <maksimkaaa> Hi, I want to type a timestamp on every image of a set of 1000 frames I have, I am planning to convert the images into mp4 but I need a guide about putting a different TimeStamp on every image.. is there any good doc about that?
[09:22] <blippyp> maksimkaaa: um yeah -> have you made a table with the filenames and timestamps yet?
[11:17] <Wu> c_14: thanks for the info and the link yesterday, I managed to replace ffmpeg2theora with ffmpeg, same resulting ogv
[11:40] <maksimkaaa> Hi, I have 300 frames ready to generate video using ffmpeg, I know the timestamp of every image and want to print the timestamp of every image over that image while generating the video. how is that possible with ffmpeg?
[14:09] <mrskman> Hi! Is there any way how convert videos to OGV with ffmpeg and using multiple threads? I guess libtheora has no multi-thread support but is there any good alternative?
[14:10] <klaxa|work> if you have multiple videos you can run multiple ffmpeg instances
[14:13] <mrskman> Sure, but I need to convert only single video file
[15:16] <coreb1te> mrskman: ffmpeg have a "-threads" parameter
[15:17] <coreb1te> for example, ffmpeg -i video.avi -threads 4 video.mp4
[15:18] <coreb1te> in this case use 4 threads to convert the video and is faster
[15:19] <relaxed> libtheora isn't multithreaded
[15:21] <coreb1te> upps.. sorry, xD
[15:22] <relaxed> I think libx264 now uses all cores by default
[15:24] <coreb1te> relaxed: yes, but I think that is necesary ffmpeg compiled with option "--enable-runtime-cpudetect"
[15:26] <relaxed> No, I just tested it.
[15:26] <Mavrik> coreb1te, no it isn't
[15:26] <Mavrik> x264 always does runtime cpudetect
[15:27] <coreb1te> ok, perfect
[15:27] <Mavrik> it's a separate library that does not rely on ffmpeg and thus ffmpeg configuration parameters have no effect on it.
[15:27] <relaxed> libx264 did not used to be multithreaded by default when using ffmpeg.
[15:28] <relaxed> you had to pass -threads 0
[15:28] <Mavrik> yes, ffmpeg now passes threads 0 by default.
[15:29] <relaxed> that contradicts your last statement
[15:31] <Mavrik> no.
[15:31] <coreb1te> I'm using now ffmpeg 2.2.2 and when convert a video to .mp4 using libx264 codecs by default use all cores
[15:31] <Mavrik> you just didn't understand that "configuration parameters" in those context are parameters passed to "./configure" like the "--enable-runtime-cpudetect" mentioned by coreb1te
[15:31] <Mavrik> not parameters passed when running ffmpeg.
[15:31] <relaxed> right
[15:32] <relaxed> coreb1te: it's been the default for some time
[16:35] <kriegerod> is there a way to look at ffplay-ish audio visualizer without having audio output to any audio devices?
[16:35] <kriegerod> i.e. i need either to show visualizer in ffmpeg or to mute ffplay
[16:43] <Sembiance> Guides online say to use 'x264 --help' to see available presets, but there is no 'x264' command on my system.
[16:49] <c_14> kriegerod: ffplay -an ?
[16:49] <Sembiance> figured it out, I was missing 'x264-encoder' package ;)
[17:13] <Sembiance> So, using ffmpeg with x11grab, I am NOT getting a cursor, but I want one :)
[17:13] <Sembiance> Using command `ffmpeg -f x11grab -r 30 -s 1280x720 -i :0.0+1721,262 -vcodec libx264 -preset ultrafast -threads 0 -y test.mp4` with ffmpeg 2.2.1
[17:14] <sacarasc> -draw_mouse 1
[17:14] <sacarasc> I think.
[17:14] <Sembiance> tried that :)
[17:14] <Sembiance> just tried using a 0,0 window position, worked then
[17:15] <Sembiance> so it appears to have some weird bug capturing the cursor when on my second monitor
[17:21] <Sembiance> hrm, err hrm...
[17:24] <Sembiance> oh wait, so I guess no, I didn't see a cursor after all even with 0x0 window position
[17:24] <Sembiance> hrm....
[17:31] <Sembiance> no, no cursor, no matter what I try. hrm...
[17:36] <Sembiance> well, recordmydesktop won't draw one either and trying --dummy-cursor with it seg faults it.
[17:36] <Sembiance> so maybe my xmonad config somehow isn't playing nicely with cursor recording ;)
[19:29] <mrskman> https://trac.ffmpeg.org/wiki/Seeking%20with%20FFmpeg#Fastandaccurateseeking Is it true that -ss output param is no longer needed when using version >=2.1?
[19:30] <c_14> Where did you hear that?
[19:31] <mrskman> I'm guessing from changelog: - when transcoding with ffmpeg (i.e. not streamcopying), -ss is now accurate
[19:31] <mrskman> even when used as an input option. Previous behavior can be restored with
[19:31] <mrskman> the -noaccurate_seek option.
[19:36] <c_14> Hmm, if it's in the changelog it should be true. I also just tested it and it seems to be working.
[19:37] <tyler1> is there a way to make an inputted file be divisble by 2 both height and width?
[19:42] <c_14> If you have a given height: scale="trunc(oh*a/2)*2:720" for a given width scale="1280:trunc(ow/a/2)*2"
[19:42] <tyler1> @c_14 the problem is that i don't know the height or width ahead of time
[19:43] <tyler1> if I have 147x89 source, I need to output it as 148x90
[19:43] <tyler1> (or 148x88) or whatever
[19:44] <c_14> you can use iw and ih as variables which contain the input width and the input height respectively.
[19:45] <tyler1> could you give me an example with that?
[19:50] <c_14> scale="trunc(iw/2)*2:trunc(ih/2)*2" should do it
[19:50] <c_14> If my understanding of trunc is correct, that should round down every time.
[19:55] <c_14> Ye, it'll round every non-even number down to the nearest even number.
[19:58] <tyler1> awesome let me try that, thanks c_14
[20:10] <tyler1> c_14: i'm getting this error: Unable to find a suitable output format for 'scale="trunc(iw/2)*2:trunc(ih/2)*2"'
[20:10] <tyler1> here's my command: ffmpeg scale="trunc(iw/2)*2:trunc(ih/2)*2" -i temp/monq%05d.png -vcodec libx264 -an -pix_fmt yuv420p -preset fast -y out/monq.gif.mp4
[20:10] <c_14> ffmpeg -vf scale[..]
[20:10] <tyler1> ah
[20:10] <c_14> And after the -i $file
[20:11] <tyler1> after -i but is it required to be before something else? or just after -i
[20:11] <c_14> After -i and before the output file name
[20:13] <jspencer> ohai...I have a multi-channel audio question
[20:14] <klaxa> go ahead
[20:14] <jspencer> when I create a 6 channel (5.1) audio file (using amerge) channel 6 is always distorted (sounds like a low pass) but when I create a 7 channel (6.1) all channels are clean
[20:15] <jspencer> I assumed the distortion in the 6 channel is because it's the LFE channel
[20:15] <jspencer> but as none of the channels in the 7 channel file have the same distortion I'm not so sure
[20:16] <jspencer> so...what I'd really like is a 6 channel file without any distortion on that 6th channel...if possible
[20:35] <jspencer> my workaround will probably be to use 6.1 with a silent channel, I just don't understand why 5.1 has one distorted channel but 6.1 is clean throughout...
[21:52] <tyler1> hey c_14 still having issues
[21:52] <c_14> What seems to be the problem?
[21:53] <tyler1> I ran : ffmpeg -i monq%05d.png -vf "crop=((in_w/2)*2):((in_h/2)*2)" -vcodec libx264 -an -pix_fmt yuv420p -preset fast -y ~/www/gifvid/output.mp4
[21:53] <tyler1> and still getting: height not divisible by 2 (500x495)
[21:53] <c_14> you need trunc(in_w/2)*2
[21:54] <c_14> Or in this specific case trunc(in_h/2)*2
[21:54] <tyler1> ah i did crop instead of scale
[21:54] <tyler1> I think its working now
[21:54] <c_14> If you're not truncating, you won't get an even number.
[21:55] <tyler1> this worked: -vf scale="trunc(iw/2)*2:trunc(ih/2)*2"
[21:55] <tyler1> thanks again for your help :)
[21:56] <c_14> np
[21:57] <tyler1> it is working from ffmpeg command line but not with this node package : https://www.npmjs.org/package/fluent-ffmpeg
[21:57] <tyler1> :\
[21:58] <tyler1> [Parsed_scale_0 @ 0x7fddbc800000] [Eval @ 0x7fff565d9440] Unknown function in '"trunc(iw/2)*2'
[22:00] <c_14> I don't know anything about that, and that isn't supported here; but are you using .addOption('-vf','scale="trunc(iw/2)*2:trunc(ih/2)*2"') ?
[22:01] <tyler1> I think so, I have this: .addOptions(['-vf scale="trunc(iw/2)*2:trunc(ih/2)*2"', '-pix_fmt yuv420p', '-preset fast'])
[22:02] <tyler1> oh maybe I need to seperate the -vf, let me try that
[22:04] <tyler1> cf_14 this was the full error:
[22:04] <tyler1> [Parsed_scale_0 @ 0x7fea93000000] [Eval @ 0x7fff5f14b440] Unknown function in '"trunc(iw/2)*2'
[22:04] <tyler1> [Parsed_scale_0 @ 0x7fea93000000] [Eval @ 0x7fff5f14b440] Invalid chars '"' at the end of expression 'trunc(ih/2)*2"'
[22:04] <tyler1> Error when evaluating the expression 'trunc(ih/2)*2"'.
[22:04] <tyler1> Maybe the expression for out_w:'"trunc(iw/2)*2' or for out_h:'trunc(ih/2)*2"' is self-referencing.
[22:04] <tyler1> [Parsed_scale_0 @ 0x7fea93000000] Failed to configure output pad on Parsed_scale_0
[22:04] <tyler1> Error opening filters!
[22:05] <c_14> If you have an error that's more than one line long, please use a pastebin so you don't spam the channel.
[22:05] <tyler1> ah okay
[22:05] <klaxa> remove the quotes though
[22:05] <relaxed> it looks like you ned .withVideoFilter('scale=\'trunc(iw/2)*2:trunc(ih/2)*2\'')
[22:05] <klaxa> >Invalid chars '"'
[22:05] <relaxed> from a quick glance
[22:07] <relaxed> that's my 2 seconds, 2 cents :)
[22:07] <tyler1> thanks all, ill try the back slashes
[22:08] <tyler1> worked!
[22:08] <tyler1> thanks c_14 klaxa relaxed
[22:18] <aarobc> So I have a subsonic server, and I'm trying to get it to use a new version of ffmpeg. Problem is, the newer version doesn't allow streaming while it's transcoding, and it seems slower too. Any ideas?
[22:38] <tyler1> one other question - how can I set the output fps (using -r) as a multiple of whatever the input is
[22:41] <c_14> setpts=x*PTS
[22:41] <c_14> Wait, no.
[22:43] <c_14> Can I ask why you would want to do that?
[22:46] <tyler1> sure, I'm converting gifs to video, but the outputted video ends up being significantly faster than the gif, and since I can't really read the "fps" of a gif (doesn't exist), I need to estimate it
[22:47] <c_14> In that case, setpts might be what you're looking for.
[22:47] <c_14> https://trac.ffmpeg.org/wiki/How%20to%20speed%20up%20/%20slow%20down%20a%20…
[22:48] <tyler1> ah, perfect
[22:50] <tyler1> so making the output twice as slow is eesentially doubling the frames and thus the file size?
[22:52] <c_14> Usually, yes.
[23:53] <jedir0x> so i'm generating a video with ffmpeg and i'm finding that the fps as described by ffprobe -i (and players) is half of what i expect.
[23:53] <jedir0x> vCodecCtx->time_base.num = 1 vCodecCtx->time_base.den = 30;
[23:53] <jedir0x> that's how i'm setting it - is there anything else that could be causing it to do this?
[00:00] --- Thu May 15 2014
1
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[00:22] <BBB> plepere_: you made the change on both sides of that macro right, i.e. in both "mov [%1+4], r8" as well as "mov r8,[%1+4]"
[01:06] <cone-717> ffmpeg.git 03Aaron Graham 07master:72dcd48c19a6: avformat/asfenc: honor the maximum of 63 payloads
[03:57] <compn> is it possible to have binary identical frame multithreading ? i think people would be happy to have that available even if it causes a small performance hit. as an option anyways
[04:11] <BBB> Compn: what does that even mean?
[04:11] <BBB> Compn: decoding of most media formats is by definition binary identical
[04:11] <BBB> Compn: and yes, all ffmpeg frame-mt is binary identical to single-threaded decoding
[04:14] <BBB> Compn: or do you mean encoding?
[04:15] <compn> encoding
[04:17] <BBB> ah, yes, that can be done for academic purposes, I wouldn't be surprised if x264 already does it
[04:17] <Skyler_> that's probably not possible, at least not without some severe constraints
[04:17] <BBB> oh
[04:17] <BBB> well ignore me for that then
[04:17] <Skyler_> most of which would consist of "apply the multithreaded constraints to singlethreaded mode"
[04:17] <Skyler_> along with I guess "no ratecontrol"
[04:18] <Skyler_> (you can always cheat by making singlethreaded do the multithreaded thing, then they're the same ;-) )
[04:24] <compn> Skyler_ : what about doing chunks of video in paralell until hit a keyframe ?
[04:25] <compn> i guess it would require a larger readahead / buffer or precise seeking
[04:25] <compn> wouldnt work on streamed content either
[04:26] <Skyler_> gop threading is possible, but that -definitely- couildn't be bit identical if you had any form of scenecut detection (unless that ran separately ahead of everything) or ratecontrol
[04:28] <compn> sure no ratecontrol then, cbr only ?
[04:34] <Skyler_> CBR is ratecontrol
[04:34] <Skyler_> so, CRF/QP only
[04:35] <Skyler_> well, really, CQP only, since macroblock tree would break too unless you did it all in a single lookahead.
[09:07] <plepere_> BBB : of course. :p
[11:41] <Voicu> can I make the read functions used by avio contexts blocking?
[11:42] <Voicu> or what should I do if the read callback is requested a larger data piece than currently available?
[11:44] <nevcairiel> there is no reason it can't block
[11:44] <nevcairiel> or you can return a smaller piece
[11:44] <nevcairiel> whatever fits your design better
[11:54] <Voicu> nevcairiel, ok thank you
[12:38] <BBB> plepere_: ok - and still the same functions that break?
[12:42] <BBB> plepere_: I also see other lines like "add r3, 4" and same for r2
[12:43] <BBB> plepere_: so I guess maybe the 4 is intentional
[12:43] <BBB> plepere_: if that's the case, just do mov dword [%1], r7 and mov sword [%1+4], r8 in SHIFT_LEFT_PARAM
[12:43] <BBB> r7d also then
[12:44] <BBB> or change it to %if gprsize == 8 mov r7, qword [%1] shl r7, 2 mov qword [%1], r7 %else current code %endif
[12:45] <BBB> plepere_: I think the fundamental problem is that the second mov at the end of SHIFT_LEFT_PARAM overwrites stack that is owned by the calling function
[12:46] <BBB> plepere_: and am just trying to figure out what the intention of that code is, I don't have the c code in front of me so I don't really know
[12:47] <BBB> plepere_: HTH ;)
[12:56] <BBB> plepere_: (I guess I'm starting to see the comments that r2 is betas and r3 is tcs, which are arrays of type uint8_t and size [8], not pointer-to-pointer which is what I thought it was)
[13:54] <Daemon404> i wonder if all these ER and mpegdec patches lately from michaelni have anything to do with spacex
[13:54] <Daemon404> /conspiracytheory
[14:01] <nevcairiel> they do
[14:01] <nevcairiel> iirc, someone asked
[14:01] <Daemon404> nevcairiel, is tehre anywhere i can peek at progress?
[14:01] <Daemon404> or rather, how will michael publish his results
[14:01] <Daemon404> ive never seen him on reddit :P
[14:07] <ubitux> Daemon404: i had to add support for \r in subtitles recently generated
[14:07] <Daemon404> yeah ive seen that in the wild
[14:07] <Daemon404> iirc some formats even mandate it
[14:07] <Daemon404> (ancient formats)
[14:09] <michaelni> Daemon404, http://spacexlanding.wikispaces.com/, http://forum.nasaspaceflight.com/index.php?topic=34597.0
[14:10] <Daemon404> are you in that thread/
[14:10] Action: Daemon404 sees 40 pages
[14:11] <Daemon404> ah i see some references to you
[14:11] <Daemon404> nice mini introduction to video coding lol.
[14:23] <BBB> someone is working on ER?
[14:23] <BBB> omg that is like ancient
[14:24] <BBB> good thing I guess
[14:24] <BBB> how about hevc-er and vp8/9-er?
[14:24] <smarter> that would be nice
[14:33] <mraulet> BBB: we have one implementation in hevc decoder that replaces the current frame with the previous one (it is already in ffmpeg)
[14:33] <Daemon404> thats "ER"
[14:35] <mraulet> if ER means what I think :)
[14:35] <cone-70> ffmpeg.git 03Mickaël Raulet 07master:053fdacde76a: hevc: a much faster implementation of intra prediction (up to 1s on haswell 2.6GHz on basket ball drive all intra qp 27)(cherry picked from commit 5de9739176f0eb4c205e80a91628a0196c9924b2)
[14:35] <cone-70> ffmpeg.git 03Mickaël Raulet 07master:38df49a068c8: hevc/intrapred: fix indent(cherry picked from commit ab167f3158cf37bd679bda28566170e2c3691686)
[14:35] <cone-70> ffmpeg.git 03Mickaël Raulet 07master:bb9e5116fe5b: hevc: fixing code indenting in hevcpred(cherry picked from commit 28a5916845928de2951013b5398465573b181925)
[14:36] <iive> michaelni: one quick question. mpeg ts packets have a bit in the ts header that says if the packet is damaged. is that bit set for any of these packets?
[14:37] <plepere_> BBB : thanks for your insight
[14:37] <michaelni> iive every bit is probably set in one of the xx thousand packets
[14:38] <iive> michaelni: well, if there have been FEC correction, then broken packets should have been marked.
[14:38] <plepere_> BBB : I'm trying to use the x86util.asm transpose code instead of the one defined in hevc_deblock.asm
[14:38] <iive> if the data have been written in flash memory, then it wouldn't matter.
[14:59] <ubitux> nevcairiel: so MS is not gonna fix the ebur128 vectorization bug in 2012?
[14:59] <nevcairiel> 2012 is dead to them
[15:00] <Daemon404> next april 1st maybe ill try and add visual studio 6 support
[15:01] <ubitux> i'd love to see you submitting a VB6 binding instead
[15:01] <Daemon404> maybe i can compile ffmpeg as an ocx
[15:02] <ubitux> can you try to make a build < 64k so it sticks in a .com?
[15:02] <Daemon404> 16bit ffmpeg?
[15:59] <cone-70> ffmpeg.git 03Olivier Langlois 07master:f1c167496e41: Support broadcast destination for udp protocol
[16:09] <cone-70> ffmpeg.git 03Matt Oliver 07master:cef99e12bc49: Use rac_get_prob branchy version when used within a conditional branch.
[20:30] <ubitux> i can do some testing on mine if necessary btw
[20:30] <ubitux> ah and well i guess i could let it up somewhere and provide ssh accesses if someone wants one
[20:34] <jamrial> that'd be great, yes
[20:34] <ubitux> ok, let me think how i'm gonna do that
[20:36] <nevcairiel> i could also provide ssh access to avx2
[20:36] <nevcairiel> or even rdp access to a windows box if someone ever needs to :p
[20:37] <Skyler_> the intel SDE can at least be used for testing/FATE
[20:40] <jamrial> i've been using that to check asm i write is valid, but it can't be used to benchmark it
[20:40] <jamrial> also, it's too slow. fate runs could take hours with it
[20:40] <Skyler_> yeah
[20:40] <Skyler_> true
[20:40] <cone-70> ffmpeg.git 03Janne Grunau 07master:9aa4592076d4: aarch64: assembler in clang-3.4 ignores the division by two
[20:41] <cone-70> ffmpeg.git 03Michael Niedermayer 07master:76581ab83387: Merge commit '9aa4592076d4dbb29d1198b0e258f9f85c0c00b5'
[20:45] <kierank> I have a lot of AVX2 machines on my desk
[20:45] <kierank> but no permanent internet connection
[20:49] <ubitux> jamrial: do you have a pub key somewhere?
[20:50] <jamrial> One sec
[20:54] <cone-70> ffmpeg.git 03Reimar Döffinger 07master:d90ba411aba2: dpx: use intfloat.h instead of deprecated intfloat_readwrite.h.
[22:40] <jamrial> 439 decicycles in dc, 131070 runs, 2 skips
[22:40] <jamrial> 1219 decicycles in dc, 131070 runs, 2 skips
[22:40] <jamrial> This being dc_32x32
[22:41] <jamrial> first is avx2, of course
[22:42] <ubitux> :)
[22:44] <Skyler_> nice!
[22:52] <cone-70> ffmpeg.git 03Janne Grunau 07master:449511740f06: build: handle library dependencies in configure
[22:52] <cone-70> ffmpeg.git 03Michael Niedermayer 07master:3d7218d93292: Merge commit '449511740f06a4675b0066730fa45cdb764ffafc'
[22:52] <cone-70> ffmpeg.git 03Michael Niedermayer 07master:efa959683174: common.mak: fix (not)building disabled libs
[22:52] <cone-70> ffmpeg.git 03Michael Niedermayer 07master:f0ae467e6c80: configure: swresample depends on avutil
[22:54] <michaelni> ubitux, can you move EBUR128LIBS dependancy thing be moved to configure ?
[22:55] <michaelni> move THE .... s/ be moved//
[22:57] <ubitux> something like "enabled ebur128_filter && enabled swresample && prepend avfilter_deps swresample"?
[22:59] <ubitux> i don't remember how i'm supposed to test that
[23:00] <jamrial> https://gcc.gnu.org/bugzilla/show_bug.cgi?id=60902 finally backported to gcc-4.9 branch
[23:01] <ubitux> jamrial: the gcc 4.9 you have access to should have that patch anyway
[23:02] <michaelni> ubitux, yes i think so, should be quite trivial
[23:02] <jamrial> cool
[23:03] <ubitux> michaelni: i've the patch locally but dunno how to test
[23:05] <michaelni> does en/disabling the filter & swresample work or is that not possibly ?
[23:12] <ubitux> ok seems good
[23:15] <cone-70> ffmpeg.git 03Clément BSsch 07master:c683e6aa2c69: build: move ebur128 swr dep to configure.
[23:31] <michaelni> ubitux, thx
[23:42] <Zeranoe> Ignore the caps, but this seems odd: http://ffmpeg.zeranoe.com/forum/viewtopic.php?f=5&t=1971 How are dates determined
[23:45] <J_Darnley> Zeranoe: it uses the __DATE__ and __TIME__
[23:49] <Zeranoe> J_Darnley: What defines that
[23:49] <iive> gcc
[23:50] <J_Darnley> (or any other compile you mat use)
[23:50] <Zeranoe> So any ideas whats happening here
[23:50] <J_Darnley> (or any other compiler you may use)
[23:51] <J_Darnley> 1 - it looks like you are not pulling new code since the copyright date is from last year and the revision number is ~10k less
[23:52] <J_Darnley> 2 - The clock on the computer is wrong
[23:52] <J_Darnley> 3 - Tie travelling VMs
[23:52] <J_Darnley> *time
[23:53] <J_Darnley> For the first I would suggest the fix-all solution: pulling from git into a clean directory
[23:54] <Zeranoe> date reports: Tue May 13 17:54:23 EDT 2014
[23:54] <Zeranoe> cloning now
[23:57] <iive> time traveling might be related to timezone
[00:00] --- Wed May 14 2014
1
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[01:06] <maksimkaaa> If I had 2 sets of frames that are parts of one clip (1000 frames taken at 5 FPS and 1000 taken as 12 FPS, how can I generate one video of the two sets using ffmpeg without losing quality ? it seems like ffmeg takes only fixed frame rate variable while I have 2 frame rates ..
[01:07] <c_14> I'd look at the concat filter in a complex filtergraph with at least one fps filter.
[01:07] <c_14> https://trac.ffmpeg.org/wiki/How%20to%20concatenate%20(join,%20merge)%20med…
[01:08] <maksimkaaa> c_14: thank you very much sir! just what I was looking for!
[16:18] <xreal> Nice! I've build ffmpeg for Windows in Ubuntu AND cygwin! cygwin makes more problems...
[17:54] <Wu> hi everbody
[17:54] <Wu> s/ever/every
[17:55] <Wu> maybe someone in here is familiar with this error...
[17:55] <Wu> Assertion v>0 && v<=(1 ? 32 : 16) failed at libavutil/mem.c:224
[17:55] <Wu> ?
[17:55] <Wu> running ffmpeg2theora (latest release, but also happened with a binary compiled from the latest sources)
[17:56] <Wu> ffmpeg installed on that box is the latest available for freebsd 10 (2.1.1)
[17:57] <Wu> ffmpeg2theora starts the conversion just fine and after some minutes, it segfaults
[18:01] <c_14> This channel doesn't give support for ffmpeg2theora, could you try encoding the file using ffmpeg to see if that segfaults as well?
[18:10] <Wu> c_14: mmm, can the same conversion be done with ffmpeg?
[18:12] <c_14> Pretty sure. Depends on what kind of complicated stuff you're doing in your ffmpeg2theora commandline since I don't know much about the tool. You might want to look at: https://trac.ffmpeg.org/wiki/TheoraVorbisEncodingGuide
[19:34] <jedir0x> i'm linking against ffmpeg to generate videos on android devices and i'm finding that it's pretty slow. What are some tips for speeding this up?
[19:34] <jedir0x> this is how i configure ffmpeg: https://github.com/Cheers-Dev/android-ffmpeg/blob/master/configure_ffmpeg.sh
[19:34] <jedir0x> and libx264: https://github.com/Cheers-Dev/android-ffmpeg/blob/master/configure_x264.sh
[19:44] <ChocolateArmpits> Hello, can anyone help compiling ismindex.c under windows with mingw ?
[20:18] <xreal> Is it good to decode AC3 with 5-7 channels to Dolby ProLogic2 or better to normal stereo?
[20:33] <edrx> hi - I am trying to clean up some mp3 files that have lots of errors like "Too many audio packets in the buffer: (4096 in 2850733 bytes)", and that seem to point to an image...
[20:34] <edrx> I'd like to use ffmpeg to read the audio in them and create another audio file withou the image. I am trying to find in the ffmpeg manpage how to do that
[20:35] <edrx> question: what is the terminology for those images in mp3s? they're usually used for album covers...
[20:37] <rsevero> Hi. I've been recording continously TV output to mp4 format with the "segment" format. Each file had 15 min as I asked and when played on Firefox and Chrome the video duration shown was 15 min also. Now I just changed the container format to webm and the files have a increasing duration with a delay that makes the final duration to be the expected 15 min. How can I make ffmpeg create sequencial webm files with 15 min duration as the mp4 ones?
[20:40] <rsevero> edrx: Googling showed me http://goo.gl/hYjtVF
[20:49] <edrx> rsevero: excellent, thanks! =)
[21:03] <rsevero> Answering my own question, I finally found the solution: set the "-reset_timestamps" flag to 1.
[22:18] <cepheus> Anyone around? I'm having issues running ./configure with VS2013/Win32-x64 with git rev cef99e1 when using the --enable-libopus flag
[22:18] <cepheus> cl -Werror=missing-prototypes -D_ISOC99_SOURCE -D_FILE_OFFSET_BITS=64 -D_LARGEFILE_SOURCE -D_WIN32_WINNT=0x0502 -Dstrtod=avpriv_strtod -Dsnprintf=avpriv_snprintf -D_snprintf=avpriv_snprintf -Dvsnprintf=avpriv_vsnprintf -nologo -D_USE_MATH_DEFINES -D_CRT_SECURE_NO_WARNINGS -Dinline=__inline -FIstdlib.h -Dstrtoll=_strtoi64 -c -Fo./ffconf.peDPdrQW.o ./ffconf.DNHgcyrz.m
[22:18] <cepheus> cl : Command line error D8021 : invalid numeric argument '/Werror=missing-prototypes'
[22:18] <cepheus> looks like it's generating flags for the wrong compiler
[22:19] <cepheus> nvm, looks like it might be just because i don't have pkg-config installed. rtfm i guess!
[22:32] <nehaljwani> On my 32 bit Ubuntu, I am getting the error: http://fpaste.org/101507/40001315/
[22:32] <nehaljwani> Can somebody please help?
[22:37] <c_14> Do you have build-essentials installed?
[22:47] <nehaljwani> c_14, the problem was the option -fstack-protector-strong
[22:59] <nehaljwani> Now I get ffmpeg: relocation error: /usr/lib/libswresample.so.0: symbol av_calloc, version LIBAVUTIL_52 not defined in file libavutil.so.52 with link time referenc
[22:59] <nehaljwani> :-/
[22:59] <nehaljwani> Whats the solution to this?
[23:01] <nehaljwani> c_14, ^
[23:02] <c_14> When compiling or when running?
[23:03] <nehaljwani> c_14, after compilation ,during runnning
[23:05] <nehaljwani> c_14, config options used: http://fpaste.org/101519/15105140/
[23:05] <c_14> Try removing --enable-shared from the configure
[23:22] <nehaljwani> c_14, http://fpaste.org/101526/40001613/
[23:24] <c_14> Can you paste the output of ldd ffmpeg
[23:24] <c_14> ?
[23:28] <nehaljwani> c_14, I rebooted the vm, starting afresh now
[23:28] <nehaljwani> with clean image
[23:28] <c_14> Have you looked at: https://trac.ffmpeg.org/wiki/CompilationGuide/Ubuntu ?
[23:28] <c_14> Might give some pointers.
[23:29] <nehaljwani> c_14, on 64 bit, http://fpaste.org/101528/00165561/ works perfectly
[23:29] <nehaljwani> but not on 32 bit
[23:29] <nehaljwani> :'(
[23:33] <c_14> The main thing I can see is that you're using a lot of custom cflags, you might want to try it with only the basic flags and then if that works add the others incrementally so you know where it's going wrong.
[23:34] <nehaljwani> c_14, I am trying to start with the ones with which the ffmpeg was built in fedora20 i386 atrpm repo
[23:35] <c_14> Those might be specific to fedora and may not work with ubuntu. You might want to start with the ubuntu compilation guide and then expand on that.
[23:43] <nehaljwani> c_14, \m/ \woot/ Finally! http://fpaste.org/101529/14000173/
[00:00] --- Wed May 14 2014
1
0
[00:01] <smarter> _aca_: you need a patch
[00:02] <smarter> _aca_: http://sprunge.us/VTWO
[00:03] <smarter> (that patch comes from the libav package in Ubuntu originally)
[00:06] <smarter> _aca_: if you want a packaged version you can get mine from https://launchpad.net/~smarter/+archive/ffmpeg but I'm not gonna maintain it
[00:09] <_aca_> michaelni: It builds, but then the symbol ff_resample_int16_rounder is missing from libswresample. Was this intended?
[00:10] <_aca_> smarter: I'm going to try your patch, thanks for the link.
[00:13] <michaelni> no, wasnt intended
[00:14] <michaelni> if smarters solution doesnt work, maybe check what is the difference to existing and working (i assume) DECLARE_ASM_CONST + MANGLE()
[00:15] <cone-266> ffmpeg.git 03Luca Barbato 07master:ebc29519d163: hwaccel: Support specific frame allocators
[00:15] <cone-266> ffmpeg.git 03Michael Niedermayer 07master:c4fe50b954a0: Merge commit 'ebc29519d1634bfeb386c20a5d8a52837aae2436'
[00:29] <cone-266> ffmpeg.git 03Anton Khirnov 07master:dd2d3b766b20: lavc: Add hwaccel private data and init/uninit callbacks
[00:29] <cone-266> ffmpeg.git 03Michael Niedermayer 07master:2ec4586d77c1: Merge commit 'dd2d3b766b20196d0b65a82e3d897ccecbf7adb8'
[00:29] <_aca_> smarter: Thanks! Your patch works great. May I credit you in the changelog?
[00:30] <smarter> _aca_: sure
[00:30] <smarter> maybe say the patch originally comes from the libav packaging too
[00:32] <_aca_> OK.
[00:39] <siretart> _aca_: i guess that's the original patch: http://anonscm.debian.org/gitweb/?p=pkg-multimedia/libav.git;a=commitdiff;h…
[00:42] <_aca_> siretart: Yes, thanks.
[00:43] <smarter> of course, it'd be better if that patch wasn't needed at all
[00:58] <cone-266> ffmpeg.git 03Anton Khirnov 07master:66e6c8a7b4e3: lavu: Add a pixel format for new VDA hwaccel
[00:58] <cone-266> ffmpeg.git 03Michael Niedermayer 07master:14aef38a6982: Merge commit '66e6c8a7b4e3ac926302529606bd3e323c28b20f'
[01:05] <michaelni> smarter, yes, i also would prefer if we could minimize distro specific changes
[01:06] <smarter> then someone needs to find a way to make ffmpeg build with "LDFLAGS=-Wl,-Bsymbolic-functions ./configure --enable-pic --enable-shared --toolchain=hardened" :]
[01:47] <cone-266> ffmpeg.git 03Anton Khirnov 07master:31a46750c7ea: vda: use hwaccel private data for internal bitstream buffer
[01:47] <cone-266> ffmpeg.git 03Michael Niedermayer 07master:c6a683bf2291: Merge commit '31a46750c7eafe6bce6a3e4f09853cd0a585292b'
[02:32] <cone-266> ffmpeg.git 03Anton Khirnov 07master:67afcefb3593: lavc: Add new VDA hwaccel
[02:32] <cone-266> ffmpeg.git 03Michael Niedermayer 07master:5449239a031e: Merge commit '67afcefb35932b420998f6f3fda46c7c85848a3f'
[02:32] <cone-266> ffmpeg.git 03Michael Niedermayer 07master:2856332719d8: avcodec/vda_h264: fix null pointer dereference
[03:06] <cone-266> ffmpeg.git 03Anton Khirnov 07master:1839fafa9513: avconv: Support VDA hwaccel
[03:06] <cone-266> ffmpeg.git 03Michael Niedermayer 07master:5460ab73d2e8: Merge commit '1839fafa951308237c4672ffd9a94f927b26eac7'
[04:11] <cone-266> ffmpeg.git 03Michael Niedermayer 07master:96abdf2ffd7d: avcodec/vda_h264: fix bistream typo
[04:11] <cone-266> ffmpeg.git 03Matt Oliver 07master:3b2ab8a78306: configure: Allow log2 with msvc12 onwards.
[09:02] <olderz> Ciao a tutti :D
[09:08] <olderz> - sorry is an automatic login message :|
[10:13] <cone-717> ffmpeg.git 03Matt Oliver 07master:4d93e758d732: configure: Ensure log2 is disable based on msvcrt version.
[12:32] <cone-717> ffmpeg.git 03Michael Niedermayer 07master:e951e04a4a69: avcodec/buffersrc: use av_mallocz_array()
[12:32] <cone-717> ffmpeg.git 03Michael Niedermayer 07master:b9cce9ee5c7b: avfilter/buffer: Use av_malloc_array()
[13:43] <plepere> hmmm. I've got a weird problem : I've got a function in ASM. fate gives me OK when compiled with gcc, but a fail when compiled with clang
[13:55] <BBB> plepere: inline asm?
[13:55] <BBB> plepere: show it :)
[13:56] <nevcairiel> is someone fixing windows builds btw? :p
[14:00] <BBB> there's a problem with the windows build?
[14:00] <nevcairiel> well not with the build
[14:00] <BBB> hm, right
[14:00] <nevcairiel> but when you run it and decode hevc, it goes all kaboom
[14:00] <BBB> so mark all functions as taking 16 xmm registers
[14:00] <BBB> not ideal but that's essentially what it does
[14:01] <plepere_> ok, wait a minute BBB
[14:02] <BBB> you can also re-arrange registers
[14:02] <plepere_> it might be a little complicated : it's a big ASM file : http://pastebin.com/wjs4AiAV
[14:03] <plepere_> and hevc_v_loop_filter_luma_10 and hevc_h_loop_filter_luma_10 are the problematic ones
[14:03] <plepere_> but they work in gcc (which is strange because their compiler should be yasm in either way)
[14:06] <BBB> what's the function prototype?
[14:07] <BBB> all your macros already exist in x86util.asm btw, please don't duplicate all these transpose macros
[14:07] <plepere_> ok (it's not my code, BTW)
[14:08] <plepere_> the arguments are (uint8_t *_pix, ptrdiff_t _stride, int *_tc, uint8_t *_no_p, uint8_t *_no_q)
[14:08] <BBB> so typically when stuff crashes in a compiler-dependent way, you're either slightly overrunning stack, or slightly overrunning allocated memory
[14:08] <BBB> you could also be clobbering GPRs
[14:08] <plepere_> ok
[14:09] <plepere_> is there an easy way of debugging ?
[14:10] <BBB> you could try valgrind or address sanitizer
[14:10] <BBB> are you using stack?
[14:10] <BBB> seems like you don't?
[14:11] <plepere_> I don't think so, no
[14:12] <BBB> let's first do a bisect
[14:13] <BBB> disable all asm functions in this file except one, and then see if it crashes
[14:13] <BBB> if it does, that's probably your issue
[14:13] <BBB> if not, enable another one
[14:13] <BBB> once you know which entry point causes the crash, let's look at that one closer
[14:14] <plepere_> I've done that : clang gives fate errors when enabling luma h or v on 10bit
[14:14] <BBB> ok
[14:16] <plepere_> I'll try using valgrind. I've used the tool before
[14:17] <BBB> there's a lot of movdqus in there that should possibly be movdqas
[14:19] <BBB> what is shift_left_param
[14:19] <BBB> does that operate on pointers?
[14:19] <BBB> it seems to
[14:19] <BBB> but it hardcodes their size as 4
[14:19] <BBB> that should be gprsize, not 4
[14:20] <BBB> that would cause memory corruption
[14:20] <BBB> that looks like your bug
[14:20] <BBB> mov r7,[%1]
[14:20] <BBB> mov r8,[%1+4]
[14:20] <BBB> should be
[14:20] <BBB> mov r7,[%1]
[14:20] <BBB> mov r8,[%1+gprsize]
[14:20] <BBB> (since gprsize is 8 on x86-64)
[14:20] <BBB> and then same for the store 4 lines down
[14:25] <nevcairiel> that was easy, i found the windows problem
[14:25] <nevcairiel> it punched a gpr in the face!
[14:36] <BBB> that doesn't help indeed :-p
[14:36] <nevcairiel> silly windows, always exposing the developers mistakes
[14:36] <BBB> nevcairiel: patch ok, just commit already
[14:36] <nevcairiel> I cannot
[14:37] <BBB> o
[14:37] <BBB> michaelni: ^^
[14:37] <BBB> need auto-commit robot or so
[14:38] <BBB> if message == 'OK' and person.flags has FLAG.Maintainer: <indent> auto-commit
[14:38] <BBB> was that proper python?
[14:38] <nevcairiel> python needs more whitespace
[14:38] <smarter> "in" instead of "has" too probably
[14:43] <ubitux> BBB: you can remove the indent in a one-liner
[14:44] <ubitux> more like if message == 'OK' and FLAG_MAINTAINER in person.flags: auto_commit() should do it
[14:48] <DHE> Last week I asked for comments on a patch I was making - libx264 to copy closed captions from upstream sources. I've updated it and am seeking new comments: http://www.dehacked.net/ffmpeg-a53-to-libx264.txt
[14:49] <kierank> DHE: ga94 is pretty standard
[14:50] <kierank> DHE: but some of the bits are wrong
[14:50] <kierank> oh you write market bits below
[14:51] <kierank> I would make some of the random numbers #defines
[14:52] <DHE> kierank: in my research (and in the MPEG1/2 decoder) multiple standards are supported. GA94 is just the one I have access to (north american ATSC feeds)
[14:52] <kierank> GA94 is the only standardised one
[14:52] <kierank> the rest are vendor specific
[14:52] <kierank> yes, I have documents about echostar
[14:52] <kierank> and other thingas
[14:52] <kierank> but those in theory are not documents I can share
[14:53] <DHE> I have nothing beyond things I can google, though I have a real IPTV set top box with pretty good access to it
[14:54] <kierank> https://github.com/kierank/broadcastencoder/blob/master/filters/video/cc.c
[14:54] <kierank> that's my implementation
[14:59] <DHE> mpeg1/2 is just reading in the main ccdata payload and saving it, so I'm just writing it back out making up the missing headers as I go. I admit it could be better, but it Works For Me and other people suggested I avoid modifying what the mpeg1/2 decoder collects
[15:07] <kierank> the headers are all documented in atsc and in dvb
[15:07] <kierank> there's no need to make them up
[15:28] <plepere_> BBB : thanks for the help. I'll try that now
[15:28] <plepere_> nevcairiel, thanks for the debugging !
[15:30] <plepere_> BBB : nah, it's not enough. :( I'll keep looking, thanks for the help !
[15:55] <Voicu> I'm sending a stream to a rtmp server and the video packets don't seem to arrive
[15:56] <Voicu> i.e. in VLC I get sound but no video and on the codec information screen there are packets for the audio stream but not for the video one
[15:56] <Voicu> I'm sending video packets with the right stream_index set and the right timestamps
[15:56] <Voicu> (I'm 99% sure about the timestamps)
[15:57] <Voicu> actually nevermind
[17:09] <cone-717> ffmpeg.git 03Hendrik Leppkes 07master:87f2d8079ac6: hevcdsp: correctly indicate that hevc_put_hevc_bi_epel_h uses 9 GPRs
[18:10] <arpu> saste, hello can you help us to port this patches to ffmpeg 2.2 or master http://paste.ubuntu.com/7453051/ (against ffmpeg 1.1) first patchset was from michaelni http://ffmpeg.org/pipermail/ffmpeg-devel/2012-July/127773.html
[18:42] <cone-717> ffmpeg.git 03Carl Eugen Hoyos 07master:a01fedd9b87f: Fix compilation with --enable-lto --enable-shared with icc.
[18:42] <cone-717> ffmpeg.git 03Carl Eugen Hoyos 07master:ab21acecc72a: Use Matroska document version 2 for WebM files if possible.
[18:49] <nevcairiel> ... i fix one issue in hevc, and now some hevc file causes a ICE in msvc 2013u2
[18:50] <Daemon404> laff
[18:51] <nevcairiel> hevcpred_template.c(236) : fatal error C1001: An internal error has occurred in the compiler.
[18:51] <nevcairiel> those are always fun
[18:51] <Daemon404> time to wait N more momnths
[18:51] <Plorkyeran> that's been happening for a while for me
[18:51] <nevcairiel> didnt happen with 2013u1
[18:51] <nevcairiel> i d idnt install any of the pre-releases
[18:52] <Plorkyeran> yeah, ever since updating to 2013u2
[18:52] <nevcairiel> well, the final was just released today
[18:53] <Plorkyeran> ah, I forgot it was a RC
[18:53] <Plorkyeran> they improved linking performance a lot so I upgraded asap
[18:54] <nevcairiel> x64 build seems to work though
[18:54] <Plorkyeran> deleting the 9 and 10 bit versions makes it stop crashing for me
[18:54] <Plorkyeran> I didn't bother investigating further
[18:56] <nevcairiel> maybe the file size hits some magic limit with the inclusion of all these c templates into one compile unit
[18:56] <nevcairiel> lets try using the 64-bit compiler that outputs 32-bit code!
[18:57] <nevcairiel> too bad my fate box doesn't have that one, though
[19:10] <nevcairiel> the code causing the issue is so extremely simple... can'
[19:10] <nevcairiel> can't even change it much
[19:12] <nevcairiel> its a for loop that say "for (i = 0; i < size; i++)" ... if i change that to say "i <= size-1", it passes :P
[19:16] <michaelni> saste, is "0510 10:14 Anshul ( 10K)
[19:22] <saste> michaelni, i'll reply on ML in a 20 minutes
[19:40] <Kanerix> Hello
[19:40] <Kanerix> I am having trouble cross-compiling ffmpeg for a specific x86 machine from my x64 machine using a specific toolchain
[19:41] <Kanerix> I think I'm not getting the cross compile paths correct
[19:42] <Kanerix> What argument is --toolchain looking for?
[19:42] <Kanerix> I keep getting unknown toolchain
[19:43] <Kanerix> Is it looking for the binary prefix or the gcc version?
[19:43] <nevcairiel> toolchain is only for a few pre-defined setups
[19:43] <nevcairiel> you can manually pass --cc --ld --as and such
[19:44] <Kanerix> Am I using the full path or relative path to those?
[19:44] <Kanerix> What is --cross-prefix supposed to point to?
[19:45] <nevcairiel> its the name prefix of your cross chain
[19:45] <Kanerix> ah
[19:45] <nevcairiel> ie. i use "x86_64-w64-mingw32-" and gcc is then called x86_64-w64-mingw32-gcc
[19:46] <Kanerix> --sysroot points to which directory?
[19:52] <Kanerix> Trying to shotgun this thing
[20:54] <cone-717> ffmpeg.git 03Michael Niedermayer 07master:d05588e21e47: avformat/utils: limit probe score in the id3 case only if there is a chance that data after the id3 tag will become available
[20:54] <cone-717> ffmpeg.git 03Michael Niedermayer 07master:b3b505c73bf5: avformat/utils: if the id3 tag is too large for data after it to become available, use the full AVPROBE_SCORE_EXTENSION if extension matches
[21:30] <cone-717> ffmpeg.git 03Michael Niedermayer 07master:a8bc175dd9ce: avformat/mp3dec: check the number of frames per buffer size
[22:20] <a141982112> stefano, I have done some changes as you requested, though i am not sure of nit:refactor, can u please check following pastebin
[22:20] <a141982112> http://pastebin.com/CZXVQehf
[22:29] <a141982112> stefano, I am mailing new patch, hope this is what you meant. :)
[23:22] <cone-717> ffmpeg.git 03Carl Eugen Hoyos 07release/2.1:c5a2c26c55eb: Use Matroska document version 2 for WebM files if possible.
[23:22] <cone-717> ffmpeg.git 03Carl Eugen Hoyos 07release/2.2:bb01956d67f9: Use Matroska document version 2 for WebM files if possible.
[23:25] <cone-717> ffmpeg.git 03Michael Niedermayer 07master:81d1fcf37d4b: avcodec: add option to make is_intra_more_likely() from error concealment return "no"
[00:00] --- Tue May 13 2014
1
0
[00:17] <sim590> I'm trying to convert mp3 files to mp3 files with id3v2.3 tags for changing it with id3 program which only does up to id3v2.3 tags.
[00:18] <sim590> I use cmus (ncurse music player) and mpv to confirm if the file has tags set
[00:19] <sim590> at first I just try to set the tags with id3, but when it fails, I do $ ffmpeg -i $file -id3v2_version 3 out.mp3; mv out.mp3 $file
[00:19] <sim590> and then I set the tags and it "works". mpv shows the tags, but cmus doesn't find them.. I'm doing something wrong somewhere
[00:33] <sim590> the clip info displayed for a file that doesn't show tags in cmus is http://ix.io/cg9
[00:34] <sim590> and for a working one : http://ix.io/cga
[00:34] <sim590> I see there's additionnal information for the working one. May be it can give a clue about what's wrong concerning my ffmpeg line.
[01:38] <xreal> Nice, I've figured it out to compile ffmpeg from GIT using cygwin to "real" binaries without need of cygwin.
[09:47] <jkli> hi guys
[09:47] <jkli> is there a way to skip broken segments?
[09:48] <jkli> there are some videos who convert for infinity
[09:48] <jkli> ffmpeg just keeps encoding and so on
[10:51] <sammy007> Guys, how to convert 720x576 avi to mov in exactly same quality?
[10:51] <sammy007> I am tired to google one more time :(
[11:02] <maksimkaaa> Hi, if I had two sets of frames (1000 images with frame rate of 5fps and 1000 with frame rate 12 fps) how can i create an mp4 video out of the two? I could't find any direct answer online.. all of the examples use one frame rate and not multiple ones..
[11:05] <xreal> Building ffmpeg in cygwin is fun, it even works without cygwin dependencies.
[11:07] <xreal> Gizmoomzig: Guten Tag.
[11:15] <Gizmoomzig> Hi
[11:42] <jonkri> When I record using the following command, the video will be faster than how I recorded it. Why is this? (I have tried to remove the "ultrafast" preset but had the same problem.) Thanks!
[11:42] <jonkri> ffmpeg -f x11grab -r 30 -s 1280x720 -i :0.0+0,832 -vcodec libvpx -preset ultrafast -crf 0 -threads 0 -qmin 20 -qmax 20 output.webm
[11:58] <c_14> What is the fps of output.webm?
[12:35] <jonkri> c_14: Hmm... I can't seem to reproduce the problem now. Perhaps it doesn't happen for WebM. I've got to go now, but thanks for your help!
[14:06] <sruli1> Hi all, I am wondering if someone can help me use a few different filters in 1 command, I have many videos which i need to get the following result, trim, setpts, logo overlay, to achive this currently i run them in seperate commands thus creating a new file from each command and using the next fiter from the previous new file, the commands i use are;
[14:06] <sruli1> 1. trim -s xx:xx:xx -tt xx:xx:xx -an
[14:06] <sruli1> 2. trim -s xx:xx:xx -tt xx:xx:xx -vn
[14:06] <sruli1> 3. setpts=0.10*PTS
[14:06] <sruli1> 4. sox tempo 10.0 (as ffmpeg can only handle upto 2.0 speed)
[14:06] <sruli1> 5. -i vidfile -i audfile "movie=wm.png [wm];[in][wm] overlay=0:0 [out]"
[14:06] <sruli1> is there anyway i can make this into 1 command (for audio speedup i can call the tempo multiple times (atempo=2.0,atempo=2.0...)
[14:14] <c_14> https://trac.ffmpeg.org/wiki/FilteringGuide Look at filtergraphs and chains or use -filter_complex.
[14:17] <arpu> hello how can i see if a mpegts (dvb-s) stream is interlaced or not ?
[14:20] <c_14> look at the idet filter
[14:21] <c_14> ffmpeg -i file -vf idet -f rawvideo /dev/null
[14:22] <sruli1> c_14: i read that page, i mostly understood from it about using multiple sources, i could not figure out how to use a filter_complex in my case, this is one of the commands i tried <ffmpeg -i test.MOV -vf "movie=wm.png [wm];[in][wm] overlay=0:0, setpts=0.10*PTS;[out];atempo=2.0,atempo=2.0,atempo=2.0,atempo=2.0,atempo=2.0" -c:v libx264 -b:v 10000k test.mp4> i played around with this command i get non detailed errors so i cant figure out what to fix
[14:25] <c_14> That [out] looks wrong and the setpts/overlay filter doesn't have an output to go to. The atempo filterchain also doesn't have an input.
[14:27] <sruli1> c_14: i was able to get it to work with just <-vf "movie=wm.png [wm];[in][wm] overlay=0:0, setpts=0.10*PTS[out]"> but cant figure out where to add the atempo filter to this command
[14:28] <arpu> c_14, thx a lot
[14:30] <c_14> I think you could put the atempo stuff after PTS. ie: *PTS,$atempostuff[out]
[14:33] <arpu> c_14 if i use -vf yadif=0:-1:1 only interlaced frames are deinterlaced right?
[14:34] <c_14> Theoretically, yes.
[14:35] <c_14> sruli1: I might go with a complex filtergraph though and just define wm.png as another input source.
[14:36] <sruli1> for < -vf "movie=wm.png [wm];[in][wm] overlay=0:0, setpts=0.10*PTS, atempo=2.0,atempo=2.0,atempo=2.0,atempo=2.0,atempo=2.0[out]"> i get an error <[Parsed_setpts_2 @ 0x39e4360] Media type mismatch between the 'Parsed_setpts_2' filter output pad 0 (video) and the 'Parsed_atempo_3' filter input pad 0 (audio)
[14:36] <sruli1> [AVFilterGraph @ 0x39d5040] Cannot create the link setpts:0 -> atempo:0
[14:36] <sruli1> Error opening filters!
[14:36] <sruli1> >
[14:37] <sruli1> c_14: i am really an not very good at this, may i ask you to to give me an example for this in full?
[14:40] <c_14> ffmpeg -i in1 -i in2 -filter_complex "[0:v][1:v]overlay[tmpv];[tmpv]setpts=0.10*PTS[v];[0:a]atempo=2.0[a]" -map '[v]' -map '[a]' output
[14:40] <c_14> That should work if memory servers
[14:40] <c_14> -r
[14:42] <c_14> in2 would be your wm.png in this case and in1 would be test.MOV
[14:47] <sruli1> thanks, how would i add a trim to the chain? (-ss 00:01:30 -t 00:10:30) i cannot add it just like that as the setpts dosent work well with that, i get different timings when i try that
[14:53] <c_14> "[0:v]trim=90:630[tr];[tr][1:v]overlay[...]
[14:54] <c_14> you might need "[0:v]trim=90:630,setpts=PTS-STARTPTS[tr];[tr][1:v]overlay[...]
[14:59] <c_14> Ok, that's giving me rather b0rky results...
[15:10] <c_14> ffmpeg -i in1 -i in2.png -filter_complex "[0:v]trim=60:120,setpts=PTS-STARTPTS[tr];[0:a]atrim=60:120,asetpts=PTS-STARTPTS[atr];[tr][1:v]overlay[tmpv];[tmpv]setpts=0.10*PTS[v];[atr]atempo=2.0,atempo=2.0,atempo=2.0,atempo=2.0[a]" -map '[v]' -map '[a]' output.mkv
[15:10] <c_14> That looks like it's worknig for me.
[15:10] <c_14> The audio is a tad faster than the video though.
[15:11] <c_14> But I think you can work around that.
[15:11] <sruli1> works a treat for me
[15:11] <sruli1> thanks a mil
[15:11] <c_14> np
[15:11] <c_14> complex filtergraphs can be a bit confusing at first, but they're pretty awesome.
[15:13] <sruli1> c_14: i used ur previous one <[0:v]trim=98:1138,setpts=PTS-STARTPTS[tr];[tr][1:v]overlay[tmpv];[tmpv]setpts=0.10*PTS[v];[0:a]atempo=2.0,atempo=2.0,atempo=2.0,atempo=2.0,atempo=2.0[a]" -map '[v]' -map '[a]'> worked great, if i would have a breakdown for each chain it would be great for me to understand it better!
[15:14] <c_14> The only major difference between the one you used and my previous one is that I also trimmed the audio input in the second one.
[15:15] <sruli1> didnt notice, i better use that one
[15:16] <c_14> Ok, so the [0:v]->[tr] part cuts out part of the video from the 0th input stream and saves it to [tr]
[15:16] <c_14> The [0:a]->[atr] part cuts out part of the audio from the 0th input stream and saves it to [tr]
[15:17] <c_14> s/tr/atr
[15:17] <c_14> The [tr][1:v]->[tmpv] part takes the video stream from the first input stream and overlays it over [tr] and saves it to [tmpv]
[15:17] <c_14> The [tmpv]->[v] part speeds up the video and saves it to [v]
[15:18] <c_14> The [atr]->[a] part speeds up the audio from [atr] and saves it to [a]
[15:18] <c_14> The -map '[v]' -map '[a]' part takes the video in [v] and the audio in [a] and maps it to the output file
[15:19] <sruli1> i am gonna copy this and read in detail later...
[15:19] <sruli1> with regards to the last command, when i exc it it seems like its wanting something more.. i get a line with >
[15:20] <c_14> Then you're probably missing a quote somewhere or have an excess one.
[15:20] <sruli1> yep, found it
[15:20] <sruli1> thanks so much for ur help
[15:21] <c_14> No problem.
[15:31] <sruli1> c_14: the [0:a]atrim=60:120, does not trim the audio, i still have audio running after the video finished playing
[15:32] <c_14> Did you output the atrim to [atr] and use that as an input in the atempo chain?
[15:32] <c_14> Can you show me your current commandline?
[15:33] <sruli1> this is the full one <ffmpeg -i raw/2014-05-11.MOV -i raw/wm.png -filter_complex "[0:v]trim=98:1138,setpts=PTS-STARTPTS[tr];[0:a]atrim=98:1138,asetpts=PTS-STARTPTS[atr];[tr][1:v]overlay[tmpv];[tmpv]setpts=0.10*PTS[v];[atr]atempo=2.0,atempo=2.0,atempo=2.0,atempo=2.0[a]" -map '[v]' -map '[a]' -c:v libx264 -b:v 10000k -c:a mp3 -b:a 128k -preset slow split/tempo/test.mp4>
[15:34] <sruli1> oops, got atempo only 4 times
[15:34] <sruli1> i guess thats it, will fix and try again
[16:37] <Devrim> Do you guys know if it is possible to crop an MPEG-2 file without re-encoding it
[17:21] <sruli1> c_14: fyi, when calling atempo multiple times each works on the previous one, as my video was 10*pts i thought i have to call atempo=2.0 5 times, however the correct way to get atempo 10xspeed is atempo=1.25 followed by atempo=2.0 3 times (1.25 x 2 = 2.5, x 2 = 5, x 2= 10)
[17:23] <c_14> Ah, that would explain why the audio never synced up.
[17:24] <pk_vid> Does anyone know how to set a cue point for a WAV file? It is possible?
[17:30] <sruli1> yep, before when i told u the audio was longer then the video i called atemo 2.0 x4 (dont understand why, it should have been much shorter!) when I called it 5 times it was 1/3 of video length so played with it until i saw the problem
[17:35] <sruli1> c_14: if i want to do the same but skip the setpts and tempo, would this be correct <"[0:v]trim=98:1138,setpts=PTS-STARTPTS[tr];[0:a]atrim=98:1138,asetpts=PTS-STARTPTS[atr];[tr][1:v]overlay[v]" -map '[v]' -map '[a]'>?
[17:35] <c_14> either change atr to a or a to atr
[17:36] <sruli1> yep, saw that now
[17:36] <sruli1> thanks a mil, i really appriciate ur time
[17:37] <c_14> No problem.
[19:30] <kolizer> how overlay gif on video?
[19:30] <kolizer> i use -filter_complex bat gif no moves
[19:32] <kolizer> lol c_14
[19:33] <kolizer> help my?
[19:34] <c_14> If I see your filter_complex, I might be able to see what's going on.
[19:34] <c_14> Which is why I asked you to paste it.
[19:34] <c_14> s/I/I or someone else
[19:39] <kolizer> http://pastebin.com/fnCNwV95 c_14
[19:39] <sacarasc> kolizer: That's not the complete command line, or the complete output.
[19:43] <kolizer> http://pastebin.com/2ZrFXXzk c_14
[19:47] <kolizer> and c_14
[19:52] <kolizer> http://pastebin.com/rGGR7uU9 c_14
[19:52] <kolizer> this
[19:55] <kolizer> why you ignore c_14
[19:55] <c_14> I'm currently testing, also it's normal that it might take some time for someone to get back to you.
[20:12] <kolizer> why is this chat. then no help
[20:13] <Kanerix> I don't know about you, but I'm at work and doing a lot of things at once
[20:13] <Kanerix> So, I'll help if I can, but don't expect everyone to drop everything to answer your questions
[20:15] <kolizer> what ocerlay gif on video? http://pastebin.com/rGGR7uU9 Kanerix
[20:16] <kolizer> overlay*
[20:16] <kolizer> this gif no move
[20:18] <Kanerix> I have no idea
[20:18] <Kanerix> There's no source files to look at
[20:20] <kolizer> 244 people and no help. ok
[20:23] <Kanerix> ...
[20:24] <Kanerix> ...because we're ALL here to help everyone ALL the time the instant they want it...
[20:25] <c_14> What, you're not being paid vast sums of money to help anybody who has some sort of problem and you're not insanely well trained that you can identify and sort out their problem in under 3 minutes?
[20:26] <Kanerix> Hehe
[20:27] <Kanerix> if only
[20:27] <Kanerix> Then I might know how to fix my problem...
[20:27] <Kanerix> too busy to spell it out at the moment
[22:24] <rafael2k> people, even when I give --disable-programs and --disable-outdev=sdl, ffmpeg keeps placing "-lSDL" inside the .pc files
[22:27] <rafael2k> anyway to disable ffmpeg from linking to SDL, when it's present in the system?
[22:51] <ChocolateArmpits> How do I exactly write filters for ffmpeg ?
[22:52] <Plorkyeran> sed -i '' '/enable sdl/d' configure is the derpy solution I use
[22:52] <Plorkyeran> since --disable-sdl isn't a thing
[22:54] <Mavrik> IIRC you just disable ffplay
[22:55] <Plorkyeran> there's nothing in the sdl check that checks if ffplay is enabled
[23:01] <rafael2k> Plorkyeran: I edited configure to disable sdl, just commenting out all the sdl stuff
[23:01] <rafael2k> no way to just --disable-sdl
[23:01] <rafael2k> when I have time I'll write a patch for it
[23:02] <rafael2k> the software I'm writing badly breaks when SDL1.2 and SDL2 are linked together, and I'm using SDL2 in it...
[23:02] <rafael2k> btw, tks!
[23:02] <rafael2k> ; )
[23:16] <brontosaurusrex_> can i go from m3u to bunch of wav files?
[23:17] <brontosaurusrex_> where m3u is a textual playlist
[23:56] <JoeyMorani> Hey. Does anyone know if it's possible to output video segments via a pipe?
[23:56] <JoeyMorani> Getting this error: "Could not write header for output file #0 (incorrect codec parameters ?): Muxer not found"
[23:57] <JoeyMorani> Works fine when I output to files though.
[00:00] --- Tue May 13 2014
1
0
[00:27] <cone-56> ffmpeg.git 03Reynaldo H. Verdejo Pinochet 07master:e0877aa544b0: ffserver: don't leak poll_table if poll fails
[00:29] <Daemon404> [21:07] <+nevcairiel> here, have this .. guy http://shrani.si/f/P/sK/2lp1q32o/3.png
[00:29] <Daemon404> he won
[00:29] <Daemon404> apparently.
[00:56] <JEEB> why am I not surprised to the least
[02:18] <cone-56> ffmpeg.git 03michael 07master:04e140daa21c: libmpcodecs/vf_uspp: check avcodec_open() return code
[03:26] <cone-56> ffmpeg.git 03Aman Gupta 07master:72fed6d12299: avformat/mpegtsenc: fix minor typo in omit_video_pes_length description
[06:45] <cone-532> ffmpeg.git 03Lukasz Marek 07master:ab57cbb7b10f: lavc/on2avc: silent warnings about const params
[14:10] <cone-266> ffmpeg.git 03Matt Oliver 07master:efcea0b91184: configure: Fix floating point conformance ith icl.
[16:40] <cone-266> ffmpeg.git 03Tim Walker 07release/2.2:b3d8276d2dc9: dcadec: Do not decode the XCh extension when downmixing to stereo
[16:40] <cone-266> ffmpeg.git 03Tim Walker 07release/2.2:194d12345d86: dcadec: Use correct channel count in stereo downmix check
[16:40] <cone-266> ffmpeg.git 03Michael Niedermayer 07release/2.2:ed1ad2f5ebe9: Merge commit 'b3d8276d2dc913d708d187fb15f6e98869d00500' into release/2.2
[16:40] <cone-266> ffmpeg.git 03Michael Niedermayer 07release/2.2:e5294f407a81: Merge commit '194d12345d86c15b20b7e44a3679c9df3075d7f6' into release/2.2
[16:51] <cone-266> ffmpeg.git 03Luca Barbato 07release/2.2:428b629eb289: vp9: Read the frame size as unsigned
[16:51] <cone-266> ffmpeg.git 03Luca Barbato 07release/2.2:a56a9e65c6a1: swscale: Fix an undefined behaviour
[16:51] <cone-266> ffmpeg.git 03Michael Niedermayer 07release/2.2:3fb754712c8a: Merge commit '428b629eb28907738ae98331b9f6c25cc9a8f3c8' into release/2.2
[16:51] <cone-266> ffmpeg.git 03Michael Niedermayer 07release/2.2:3a67865963de: Merge commit 'a56a9e65c6a17dd8b6303dd45bdc3e3368093092' into release/2.2
[17:09] <cone-266> ffmpeg.git 03Martin Storsjö 07release/2.2:7d97cc8d87eb: rtmpproto: Check the buffer sizes when copying app/playpath strings
[17:09] <cone-266> ffmpeg.git 03Reinhard Tartler 07release/2.2:6c3985713b77: Add missing changelog entries for v10
[17:09] <cone-266> ffmpeg.git 03Michael Niedermayer 07release/2.2:fab9a7be76ee: Merge commit '7d97cc8d87ebf6ebe7ec1865d6f932c652dbce3a' into release/2.2
[17:09] <cone-266> ffmpeg.git 03Michael Niedermayer 07release/2.2:a14969253a6c: Merge commit '6c3985713b77c5c1155f31c3fbb8236f9b4cdfb5' into release/2.2
[17:31] <cone-266> ffmpeg.git 03Reinhard Tartler 07release/2.2:e2a83d72da86: Update Changelog for v10.1
[17:31] <cone-266> ffmpeg.git 03Luca Barbato 07release/2.2:ff79f6b35a94: avfilter: Add missing emms_c when needed
[17:31] <cone-266> ffmpeg.git 03Janne Grunau 07release/2.2:40ffa99dfa39: arm: hpeldsp: prevent overreads in armv6 asm
[17:31] <cone-266> ffmpeg.git 03Michael Niedermayer 07release/2.2:6452b3159982: Merge commit 'e2a83d72da866a29f5307a56f66c469966f02473' into release/2.2
[17:31] <cone-266> ffmpeg.git 03Michael Niedermayer 07release/2.2:ea28034f5d35: Merge commit 'ff79f6b35a94dde69b3d52bd2abdfdb793f525c3' into release/2.2
[17:31] <cone-266> ffmpeg.git 03Michael Niedermayer 07release/2.2:358d1f6e01e0: Merge commit '40ffa99dfa39c574d8784a3c4eaf6406198d675d' into release/2.2
[18:09] <cone-266> ffmpeg.git 03Janne Grunau 07release/2.2:1bd6372cd349: arm: hpeldsp: fix put_pixels8_y2_{,no_rnd_}armv6
[18:10] <cone-266> ffmpeg.git 03John Stebbins 07release/2.2:8e9e57ed0c7f: Update Changelog
[18:10] <cone-266> ffmpeg.git 03Michael Niedermayer 07release/2.2:7f954ca502a1: avi: Improve non-interleaved detection
[18:10] <cone-266> ffmpeg.git 03Michael Niedermayer 07release/2.2:9bdb254b980f: Merge commit '1bd6372cd34935815c0a061adbf3bc68ca68ff73' into release/2.2
[18:10] <cone-266> ffmpeg.git 03Michael Niedermayer 07release/2.2:a05f86ec101f: Merge commit '8e9e57ed0c7f40741da1ca83e52c851263e0c168' into release/2.2
[18:10] <cone-266> ffmpeg.git 03Michael Niedermayer 07release/2.2:2b14d980863d: avformat/avidec: Check required demuxing buffer sizes in guess_ni_flag()
[18:10] <cone-266> ffmpeg.git 03Michael Niedermayer 07release/2.2:2aa659233810: Merge commit '7f954ca502a16feeca44d4e9a7d450b404dabc94' into release/2.2
[18:23] <cone-266> ffmpeg.git 03nu774 07release/2.2:e780c3daafe0: pcm-dvd: Fix 20bit decoding
[18:23] <cone-266> ffmpeg.git 03Michael Niedermayer 07release/2.2:0008a87cb14b: Merge commit 'e780c3daafe0588e035e752c771ebfcd2201746a' into release/2.2
[18:23] <cone-266> ffmpeg.git 03Michael Niedermayer 07release/2.2:bc0c49b83e4b: avformat/avidec: remove unused variables
[18:39] <houbahop69> hi, I have just succesfully compiled ffmpeg from git sources with --enable-opencl option on a debian jessie + nvidia BUT...
[18:41] <houbahop69> Unfortunately when I try "ffmpeg -opencl_bench" I get "Could not get OpenCL platform ids: unknown error" and "Could not get device list from environment" while vdpauinfo works well. I have tried on both amachine without and with bumblebee/optimus. Same behavior ... Any suggestion?
[19:14] <wm4> houbahop69: just by the way, vdpauinfo has nothing to do with opencl
[19:15] <houbahop69> @wm4 OK that's good news because I was unable to find any link between them !
[19:50] <wm4> any news on the debian ffmpeg package?
[19:52] <ubitux> isn't it available in experimental?
[19:52] <wm4> I don't see anything
[20:12] <cone-266> ffmpeg.git 03Michael Niedermayer 07master:e2c7816ab0d1: avcodec/cinepakenc: use av_malloc_array()
[20:12] <cone-266> ffmpeg.git 03Michael Niedermayer 07master:5e9555119987: avcodec/dct: use av_malloc_array()
[21:58] <cone-266> ffmpeg.git 03Anton Khirnov 07master:632ad2248e2e: lavc: Add an internal wrapper around get_format()
[21:58] <cone-266> ffmpeg.git 03Michael Niedermayer 07master:d93cf093f840: Merge commit '632ad2248e2e5d8cd4b51e6c87c943a38c3da425'
[23:15] <_aca_> Compiling the Debian FFmpeg package on Ubuntu trusty fails:
[23:15] <_aca_> /usr/bin/ld: libswresample/resample.o: relocation R_X86_64_PC32 against symbol `ff_resample_int16_rounder' can not be used when making a shared object; recompile with -fPIC
[23:16] <_aca_> Any ideas how to fix this?
[23:17] <Meths> Is it already using the flag the error suggests to add?
[23:17] <_aca_> Yes, everything is compiled with -fPIC.
[23:17] <J_Darnley> Is it really ffmpeg though?
[23:18] <_aca_> I am sure that it is FFmpeg, as I build the package. ;)
[23:18] <_aca_> resample.o is created by:
[23:18] <_aca_> gcc -I. -I./ -D_FORTIFY_SOURCE=2 -D_ISOC99_SOURCE -D_FILE_OFFSET_BITS=64 -D_LARGEFILE_SOURCE -D_POSIX_C_SOURCE=200112 -D_XOPEN_SOURCE=600 -DPIC -DHAVE_AV_CONFIG_H -Wall -U_FORTIFY_SOURCE -D_FORTIFY_SOURCE=2 -fno-strict-overflow -fstack-protector-all -Wall -std=c99 -fomit-frame-pointer -fPIC -pthread -I/usr/include/freetype2 -I/usr/include/p11-kit-1 -I/usr/include/freetype2 -I/usr/include/fribidi -I/usr/include/freetype2 -I/usr/inclu
[23:18] <_aca_> de/opencv -I/usr/include/opus -D_REENTRANT -I/usr/include/p11-kit-1 -I/usr/include/schroedinger-1.0 -I/usr/include/orc-0.4 -D_GNU_SOURCE=1 -D_REENTRANT -I/usr/include/SDL -g -Wdeclaration-after-statement -Wall -Wdisabled-optimization -Wpointer-arith -Wredundant-decls -Wwrite-strings -Wtype-limits -Wundef -Wmissing-prototypes -Wno-pointer-to-int-cast -Wstrict-prototypes -Wno-parentheses -Wno-switch -Wno-format-zero-length -Wno-pointe
[23:18] <_aca_> r-sign -O3 -fno-math-errno -fno-signed-zeros -fno-tree-vectorize -Werror=implicit-function-declaration -Werror=missing-prototypes -Werror=return-type -Werror=vla -Wno-maybe-uninitialized -MMD -MF libswresample/resample.d -MT libswresample/resample.o -c -o libswresample/resample.o libswresample/resample.c
[23:25] <_aca_> The problem might be in the assembler code in libswresample/x86/resample_mmx.h, where`ff_resample_int16_rounder' is defined.
[23:30] <JEEB> did you configure with enable-pic and clean and re-compile?
[23:38] <_aca_> Unfortunately this doesn't help.
[23:39] <cone-266> ffmpeg.git 03Anton Khirnov 07master:5c1d7246cd65: lavc: set AVCodecContext.hwaccel in ff_get_format()
[23:39] <cone-266> ffmpeg.git 03Michael Niedermayer 07master:f2f99f07a5f1: Merge commit '5c1d7246cd65dc4db1b6dc36e29ce39fc1068f3f'
[23:39] <JEEB> yeah, 32bit and PIC generally in many projects isn't really supported. Not sure about FFmpeg, but x264 at least doesn't seem to support 32bit and PIC (at least under certain circuimstances)
[23:41] <_aca_> Just for clarity: This is build on amd64 and it works fine in Debian, but not in Ubuntu.
[23:42] <JEEB> argh, missed the X86_64
[23:42] <JEEB> read it as X86
[23:47] <michaelni> does it work if you replace DECLARE_ALIGNED by DECLARE_ASM_CONST in libswresample/x86/resample_mmx.h
[23:48] <cone-266> ffmpeg.git 03Anton Khirnov 07master:08bb25675865: lavc: document which parts of AVHWAccel are private.
[23:48] <cone-266> ffmpeg.git 03Michael Niedermayer 07master:8d19dd91a51c: Merge commit '08bb2567586599d62a282c4cd31567a0807f3d92'
[23:56] <cone-266> ffmpeg.git 03Anton Khirnov 07master:a871ef0cc936: hwaccel: Rename priv_data_size to frame_priv_data_size
[23:56] <cone-266> ffmpeg.git 03Michael Niedermayer 07master:fa0f7c024e90: Merge commit 'a871ef0cc936612bdb0e643196be72efcf0c8986'
[00:00] --- Mon May 12 2014
1
0
[00:32] <aolin> does anyone know about ffmpeg generated videos playing too slow on QuickTime, but VLC plays just fine
[00:33] <aolin> I'm transcoding a video shot with mobile phone h264 -> mpeg4
[00:33] <aolin> tried ffmpeg 1.2.6, 2.2.2 and same effect. does not happen with all videos, only 50% of them
[00:34] <aolin> also the audio plays normal speed, but video too slow
[00:35] <c_14> Does it happen deterministically with an input video?
[00:35] <aolin> yes, for certain videos this always happens
[00:36] <c_14> Does ffplay play them fine?
[00:36] <aolin> I would think so as VLC plays them just fine
[00:36] <c_14> vlc might do weird magic, if ffplay plays it fine, the fault is most likely with quicktime
[00:39] <aolin> problem is that I'm developing software that does transcoding, so I'm trying to find a workaround for the quicktime problem. The source video plays just fine on all players, but after transcoding with FFmpeg the problem appears for certain videos
[00:39] <aolin> https://www.dropbox.com/s/rj27edwgfutl0xk/output2.mp4 so here's an example of the problem
[00:45] <c_14> Could you paste the command line you use to encode?
[00:46] <aolin> ffmpeg -i ../videotest/Original-RC_20140502_1612.mp4 -vcodec mpeg4 -strict -2 ../videotest/output2.mp4
[00:48] <aolin> https://www.dropbox.com/s/t615jpoh4p52uei/Original-RC_20140502_1612.mp4 here's the original video
[00:49] <aolin> my software does much more complicated thing through filtering, but I've narrowed the problem down to that
[00:51] <aolin> for me it looks like quicktime is delaying the playback for a reason or another
[00:52] <c_14> Is it adding an offset to the video or actually playing it at a lower fps?
[00:52] <aolin> lower fps
[00:52] <aolin> it's slowmotion but audio plays ok
[00:53] <aolin> so half of the video is video too slow normal audio, and then audio runs out it's only the slowmotion video for rest
[00:54] <aolin> there's a degree of jerkiness in the video, but cannot describe it more
[00:54] <c_14> I can't find anything wrong with the files in question, so I'm probably going to lay the blame on quicktime. What you can try doing is downscaling the video and seeing if that will work, use a different player, or try switching codecs.
[00:57] <aolin> I was thinking using the h263p codec, but ffmpeg only allows mkv container for that, so cannot play that with quicktime anyways
[00:57] <aolin> I'm out of options as I need LGPL codec
[00:57] <aolin> that is widely supported
[00:59] <aolin> if I upload a video that works on my players, would you be able to see what's different with them ?
[00:59] <aolin> because it's rather peculiar that some devices produce video that works after ffmpeg and some videos that doesn't
[01:02] <c_14> I'm not sure I'll be able to see what's different, but I might be able to cross of things it can't be.
[01:04] <c_14> Question, are all the functional videos coming from one source and all the non-functional videos coming from a different source or does a source sometimes create working videos and sometimes create broken videos.
[01:04] <c_14> (In this context broken means that quicktime has issues playing them after encoding with ffmpeg)
[01:05] <iive> one blind guess. maybe interlace is involved?
[01:05] <aolin> looks like videos created on device x always work and device y never work.
[01:05] <c_14> Maybe give a source video from device y
[01:06] <c_14> Or was the one you uploaded from y?
[01:06] <aolin> https://www.dropbox.com/s/t615jpoh4p52uei/Original-RC_20140502_1612.mp4 this is the source video
[01:06] <aolin> I have another, completely different type of camera that has same problem (a Liquid Image action camera)
[01:08] <c_14> According to idet, the source looks progressive so I don't think it's that.
[01:11] <aolin> https://www.dropbox.com/s/6kqo0nw0u5m2dlv/output3.mp4 here's a clip that works on quicktime
[01:15] <iive> you may try playing with -vsync
[01:15] <c_14> The only major difference I can see is that output3.mp4 has 14.92fps while output2.mp4 has 29.83fps.
[01:17] <aolin> hmm, that's little bit weird
[01:17] <c_14> I'm mostly out of ideas, but you could try taking a video from source y and dropping the framerate to 14.92 in the output just to see what happens.
[01:22] <aolin> the videos seem to playback fine when the fps is dropped below 15
[01:22] <aolin> not sure what that means
[01:22] <c_14> top kek
[01:22] <c_14> Quicktime being awesome?
[01:22] <aolin> but I do have some clips at 30 that work just fine
[01:23] <c_14> With the same codec?
[01:23] <aolin> yep same method
[01:25] <aolin> Stream #0:0(eng): Video: h264 (avc1 / 0x31637661), yuv420p, 1280x720, 11995 kb/s, -3.68 fps, 30 tbr, 90k tbn, 180k tbc (default)
[01:25] <aolin> thats the input video, -3.68 fps ?
[01:26] <aolin> that works
[01:26] <aolin> Stream #0:0(eng): Video: h264 (avc1 / 0x31637661), yuv420p, 1920x1080, 16965 kb/s, SAR 65536:65536 DAR 16:9, 29.79 fps, 29.83 tbr, 90k tbn, 180k tbc (default)
[01:26] <aolin> this doesn't
[01:26] <iive> -3.68 fps ?!
[01:26] <c_14> Why is the framerate negative?
[01:26] <klaxa> fuck logic?
[01:27] <aolin> I'll try 2.2.2 if it says the same
[01:27] <iive> use 2.2.2 or latest git master.
[01:28] <aolin> yep it's the same
[01:28] <aolin> I was testing with 1.2.6 to see if this was a late bug and forgot to change back
[01:33] <iive> hum, -debug_ts is in the manual, but 2.2.2 complains about it?!
[01:34] <iive> it works with ffmpeg, but not ffplay. Ok.
[01:38] <aolin> Stream #0:0(eng): Video: h264 (avc1 / 0x31637661), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], 10965 kb/s, 5.30 fps, 59.94 tbr, 90k tbn, 119.88 tbc
[01:38] <aolin> ^input
[01:38] <aolin> Stream #0:0(eng): Video: mpeg4 ( [0][0][0] / 0x0020), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], q=2-31, 200 kb/s, 60k tbn, 59.94 tbc (default)
[01:38] <aolin> ^output
[01:39] <aolin> I'm wondering about those tbn/tbc
[01:43] <aolin> the videos that work, have integers as those values. the videos that do not work have floats
[01:44] <aolin> might be coincidense, but seems suspicious
[01:47] <iive> aolin: try different -vsync modes.
[01:48] <aolin> iive: ok
[01:48] <iive> 0 then 1 then 2
[01:52] <aolin> doesn't seem to affect at all
[01:55] <iive> try -copyts -vsync 0
[01:56] <iive> you probably need to talk with somebody that knows (de)muxers and mp4 is particular.
[01:57] <iive> but guessing why QT doesn't like something is tricky thing.
[01:58] <iive> the sample seems to skip a frame here and there. this probably makes the fps non-integer. afaik mp4 is vfr format, so it shouldn't be issue.
[02:02] <aolin> the first sample doesn't work with vsync 0 "Invalid pts (541) <= last (541)", but my 60fps video was not fixed with that
[02:02] <aolin> tried the copyts
[02:08] <aolin> iive & c_14, thanks for support, I really appreciate it! too bad there's no obvious workaround for this ...
[02:09] <aolin> I'm out of ideas and energy :)
[02:10] <iive> maybe michaelni might have an idea. But once again... it is probably more a question of finding bugs in QT.
[02:15] <aolin> ok, I'll hang around and see if anyone can guide me to new directions
[02:17] <aolin> too bad I don't know anyone even somewhat close apple, I'm sure it would be nearly impossible to have them try those samples without knowing anyone
[02:48] <iive> aolin: btw, you may try ffmbc .it is an older and less popular fork, but the author is expert in mp4 and broadcast stuff.
[02:48] <iive> just for the test.
[02:49] <iive> afaik it is actively developed, but it doesn't merge libav and new ffmpeg stuff.
[05:42] <ParkerR> Does something like "ffmpeg -i input.ts -vcodec copy -acodec copy output.mp4" actually convert to an mp4 codec or just changes container?
[05:43] <ParkerR> Ahh just tested it. It doesnt.
[05:56] <BtbN> there is no such thing as an "mp4 codec"
[06:08] <ParkerR> Well libx264
[06:08] <ParkerR> I understand mp4 is just another container
[06:09] <ParkerR> But generally it contains an h264 stream
[06:26] <HD|Laptop> hi all
[06:26] <HD|Laptop> omxplayer for raspberry pi uses ffmpeg as backend, and I have a problem with it...or more so, two
[06:27] <HD|Laptop> 1) ffmpeg doesnt support direct playing of a HTTP Live Stream/HLS stream
[06:27] <HD|Laptop> so I decided to write an "assembler" script which just pipes the individual segments into a FIFO and plays with omxplayer pipe:0 for stdin piping or /tmp/fifo with a real fifo
[06:27] <ParkerR> HD|Laptop, it doesnt use ffmpeg as a backend. It renders the video directly on the GPU
[06:28] <ParkerR> And omxplayer plays internat content just fine
[06:28] <HD|Laptop> problem is, once the buffer runs empty once, omxplayer quits
[06:28] <HD|Laptop> ParkerR: huh? for what does omxplayer depend on ffmpeg then? there
[06:28] <HD|Laptop> 's no streaming-related content anywhere in the omxplayer souce
[06:28] <ParkerR> I dudnt remember it depending on it. Lemme look
[06:28] <ParkerR> *didn't
[06:29] <HD|Laptop> if i read the docs and google correctly, it uses ffmpeg for fetching and decoding anything thats not h264
[06:30] <ParkerR> Hmm ok I see it does. Umm, I don't think it does any transcoding in that regard. It will only play whatever formats the Pi supports (h.264, mpeg2, VC1, and one more...)
[06:31] <ParkerR> HD|Laptop, I've been able to supply the raw URL from a youtube video and it plays just fine
[06:31] <HD|Laptop> ParkerR: yes this works fine, the problem is that HLS is playlist based
[06:32] <HD|Laptop> the "master" file is just a index file, ffmpeg must periodically reload it to get the newest available snippets and then enqueue them
[06:32] <HD|Laptop> ParkerR: also ffmpeg apparently doesnt support PLS/m3u-based web radios, I have to supply it with the direct stream url
[06:32] <ParkerR> Heh, I think thats where Im out. Good luck on finding a solution.
[06:33] <ParkerR> Bit beyond what I know
[06:34] <ParkerR> HD|Laptop, I've always thought of ffmpeg as just transcoding. Never even thought about it being a rednerer of sorts
[06:34] <ParkerR> *renderer
[06:35] <ParkerR> HD|Laptop, isn't that why other applications are created as wrappers? mplayer for playlist parsing for example
[06:35] <HD|Laptop> ParkerR: yeah the problem is that the coupling behind omxplayer and ffmpeg is too deep
[06:35] <ParkerR> Understandable
[06:36] <HD|Laptop> if one could use the omxplayer acceleration on vlc, mplayer or whatever i'd be totally fine with that
[06:36] <ParkerR> I think I saw some work of direct to GPU VLC on the Pi
[06:36] <ParkerR> http://intensecode.blogspot.com/2013/10/tutorial-vlc-with-hardware-accelera…
[06:37] <ParkerR> ./configure --enable-rpi-omxil --enable-dvbpsi
[06:37] <ParkerR> --enable-x264 <other_modules_you_might_need>
[06:37] <ParkerR> So apparently that does use omxplayer in a way
[06:37] <ParkerR> Well the omx libraries
[06:38] <HD|Laptop> hmm interesting but vlc is a fucking big app, i guess it would take days for a compile
[06:38] <ParkerR> HD|Laptop, another bit more complicated way would be a script to parse out the stream URL then pass to omxplayer with window arguments
[06:38] <ParkerR> It's able to render fullscreen or even just set X Y coordinates
[06:38] <HD|Laptop> ParkerR: that works for the radio stuff but not for HLS
[06:39] <HD|Laptop> you see, HLS is quite a complex thing actually. you have individual snippets, usually 10sec or so, packed in MPEG TS
[06:39] <ParkerR> --win "x1 y1 x2 y2" Set position of video window
[06:39] <HD|Laptop> a HLS renderer is supposed to reload the HLS index file every X seconds, X being the snippet duration
[06:39] <ParkerR> HD|Laptop, ogod mpeg ts. Im dealing with deinterlacing some recorded tv as we speak haha
[06:40] <HD|Laptop> and then enqueue the snippets it doesnt already have in a playlist...which is played sequentially
[06:40] <HD|Laptop> ParkerR: just for transport, the codecs used dont matter
[06:40] <ParkerR> Aye
[06:40] <[mbm]> thought it only reloaded the index if you marked the event as live
[06:41] <HD|Laptop> [mbm]: that would be a normal playlist, HLS is by definition live ;)
[06:41] <HD|Laptop> but well at least an option for ffmpeg not to exit if the read-buffer runs empty would be cool
[06:41] <ParkerR> Currently running the first pass on a TV episode. ffmpeg -i vlc-record.ts -pass 1 -vf yadif=1 -vcodec libx264 -acodec copy put.mp4
[06:43] Action: [mbm] played with hls for a few days before determining it didn't fit what I needed
[06:44] <ParkerR> It's like "Yay I have some HD OTA content recorded" "Crap it's 1080i *assembles some ffmpeg code*"
[06:44] <HD|Laptop> its quite cool actually, especially for distribution - i can supply ten "big" PCs with one HLS stream
[06:44] <[mbm]> technically HLS isn't "live" it's delayed by whatever your segment length is
[06:44] <HD|Laptop> no more running ten RTMP streams (with the needed bandwidth!), just a squid proxy for the individual segments :)
[06:45] <HD|Laptop> the problem is that a rpi isnt as powerful as a real pc ;)
[06:45] <HD|Laptop> [mbm]: yeah, thats right - but only a problem with soccer matches imho
[06:46] <ParkerR> Why would only one sport be the problem...
[06:46] <[mbm]> HD|Laptop: well, problem was I couldn't take a stream that was actually live and convert it to HLS
[06:47] <HD|Laptop> huh, iirc ffserver should be able to output a HLS stream
[06:49] <[mbm]> what I mean is that I had a device that you could go to http://server/url?duration=10 and it'd give you an actual live stream, duration specified as 10 seconds
[06:50] <[mbm]> in theory I could put that url into an HLS index/playlist
[06:50] <[mbm]> but in practice it doesn't work so well
[06:52] <[mbm]> being an actual live stream it took 10 seconds for the player to download the segment and then the player would disconnect and reconnect to request another
[06:52] <[mbm]> which resulted in either a gap between or an overlap
[06:53] <[mbm]> basically the only way HLS actually works is if you buffer to a disk, and then you're delayed by whatever your length is
[06:54] <[mbm]> </useless trivia>
[06:57] <HD|Laptop> [mbm]: yeah that is what i tried to do using a fifo (a real file would kill the SD card in days, also it needs 360MB/h of free space)
[06:57] <HD|Laptop> but the problem is, it may work fine, but if you hit the buffer end only *once* ffmpeg stops reading and exits :(
[06:58] <HD|Laptop> I tried the "http dump" option too, but both apache and nginx couldn't handle the php script :/
[06:58] <[mbm]> I then tried to cheat and created an m3u8 with only a single entry of infinite length, the player didn't like that much ;)
[06:59] <HD|Laptop> [mbm]: :D second problem is that omxplayer cant do playlists at all
[07:56] <HD|Laptop> what the actual fuck is ffmpeg doing when attempting to play a http provided video file?!
[07:56] <HD|Laptop> first, it opens four(!) simultaneous connections in GET mode and not HEAD
[07:56] <HD|Laptop> second, why doesn't it obey the (missing) content-length and fetch until the socket is closed?
[07:58] <[mbm]> heh
[07:59] <HD|Laptop> I could accept two HTTP connections - first a HEAD to see if the stream/file is available, but why *four*?!
[07:59] <[mbm]> what's your video source?
[08:00] <HD|Laptop> the video source is raw MPEG TS
[08:00] <[mbm]> from an hdhomerun?
[08:00] <HD|Laptop> nope, HTTP HLS from a tv station
[08:00] <HD|Laptop> 10-sec snippets
[08:01] <HD|Laptop> the HTTP requests are all equal... with User-Agent: Lavf/55.33.100
[08:02] <[mbm]> hmm, in theory you could write your own hls parser to fetch each snippet and then just cat them all together and pipe the output
[08:02] <[mbm]> all the sgements should line up prefectly to produce an mpeg-ts
[08:02] <BtbN> i guess it's the estimator and stuff accessing the file
[08:03] <[mbm]> curious what stations actuallh have a public hls stream though
[08:03] <BtbN> twitch does
[08:04] <BtbN> i have no idea what's so great about HLS compared to just a continous http or rtmp stream
[08:04] <BtbN> it's just more complicated and overhead
[08:05] <HD|Laptop> mmint: yes, exactly this is what I do
[08:05] <HD|Laptop> [mbm]: ^^
[08:05] <HD|Laptop> [mbm]: the problem is with pipes that once the buffer underruns just ONCE, ffmpeg *exits*
[08:06] <HD|Laptop> [mbm]: if you're German, you might try http://daserste_live-lh.akamaihd.net/i/daserste_de@91204/index_608_av-b.m3u…
[08:06] <HD|Laptop> dunno if this one is region locked
[08:06] <[mbm]> hmm, seems like something that would be easy to fix in the ffmpeg source
[08:07] <HD|Laptop> [mbm]: more interesting is the reason behind the three unneeded http requests
[08:07] <BtbN> VLC plays hls fine, would propably be easier to just use it
[08:08] <HD|Laptop> BtbN: nope, raspberry pi - and i dont have 10+ hours to do a compile :/
[08:08] <BtbN> then, don't compile it yourself?
[08:08] <HD|Laptop> also, I dont have the needed SD card space, just a 2gb sd card
[08:08] <BtbN> vlc is a little smaller than 2GB
[08:08] <HD|Laptop> BtbN: the prebuilt raspbian packages dont carry the configure flag which enables the rpi hardware acceleration, the OS occupies much of the sd card already^^
[08:09] <HD|Laptop> and the "media center" i've built is totally centered upon omxplayer, if possible i'd like to avoid switching the backend to vlc :/
[08:09] <BtbN> well, if you just want to watch hls on an rpi somehow, you can allways put xbmc on there
[08:09] <BtbN> it definitly also plays hls
[08:09] <[mbm]> you could run vlc as a server
[08:10] <HD|Laptop> BtbN: thats not an option, too, because the rpi is not just used as a TV - there's also a chromium webbrowser which is normally displayed on the monitor
[08:10] <BtbN> if you have python on there you can also use livestreamer to play the hls stream and make it available as local http stream
[08:11] <HD|Laptop> heh, if such a solution already exists i'm taking it :D
[08:11] <BtbN> you might have to write a small python script to add support for your live streaming "service", but that's all
[08:12] <BtbN> it's intended to watch stuff like twitch in vlc and other players
[08:26] <HD|Laptop> BtbN: hmm doesnt work
[08:26] <HD|Laptop> root@raspberry-pi:~# livestreamer "hls://http://daserste_live-lh.akamaihd.net/i/daserste_de@91204/index_608_av…" -p "/usr/bin/omxplayer" -l debug -a "--no-keys {filename}" -v --player-http --yes-run-as-root
[08:26] <HD|Laptop> [cli][info] Found matching plugin stream for URL hls://http://daserste_live-lh.akamaihd.net/i/daserste_de@91204/index_608_av…
[08:26] <HD|Laptop> Available streams: live (worst, best)
[08:26] <HD|Laptop> and then it exits -.-
[08:27] <BtbN> well, you forgot to tell it which stream (best) to use
[08:27] <BtbN> and i realy would not run stuff like that as root
[08:31] <HD|Laptop> heh, that is just awesome! thanks :)
[08:35] <Reventlov> Hello.
[08:36] <Reventlov> What ffmpeg options would you recommand to stream a .mp4 using ffserver and ffmpeg ?
[08:36] <Reventlov> currently use -i *.mp4 -override_ffserver -vcodec mpeg2video -b:v 3000 http://127.0.0.1:8090/rtsp.ffm
[08:36] <Reventlov> but the quality is shitty
[08:37] <HD|Laptop> BtbN: huh, if I omit the log level(!) option from livestreamer, omxplayer seems to bark o.O
[08:43] <HD|Laptop> anyway looks like it's working BtbN thank you very much for the help!
[08:50] <aolin> Reventlov: how about -v:q 2
[08:52] <Reventlov> aolin: replacing -b:v ?
[08:52] <Reventlov> let me man.
[08:54] <Reventlov> aolin: -loglevel [repeat+]loglevel | -v [repeat+]loglevel
[08:54] <Reventlov> are you sure it's what you meant ? Or for me ?
[08:54] <aolin> no, sorry
[08:56] <aolin> apparently it's just -q 2
[08:56] <aolin> it's more of dynamic quality setting than setting bitrate AFAIK
[08:57] <aolin> your bitrate must be too low though, 3000 is nothing
[08:57] <Reventlov> oh, it's wayyy better
[09:00] <blippyp> reventlov - you might want to consider using -crf instead of -q I think it does a better compression/quality ratio... but I'm not sure. My understanding is that -q is really only used when setting it to 0 for lossless quality. Also you video bitrate is probably really high - you can probably get away with 1-1.5 mb instead (if not lower) - but it really depends on your video size.
[09:07] <aolin> blippyp: hmmm -crf seems to be x264 specific
[09:07] <blippyp> oh sorry - I assumed you were using x264 since you were talking about mp4 - at least I thought you were - but you're probably right about that - which codec are you using?
[09:08] <blippyp> forget it - I see now
[09:08] <aolin> looks like he's transcoding .mp4 (h264?) to mpeg2video
[09:09] <blippyp> not sure then - I never use that codec - I think -q is a general for for most codecs, you're probably okay leaving it with that if crf doesn't do anything for you
[09:09] <blippyp> but I would still try lowering your video bitrate - what's the resolution of the video?
[09:12] <Reventlov> ID_VIDEO_BITRATE=4969920
[09:12] <blippyp> what's the resolution though?
[09:12] <Reventlov> 4969.9 kbps (606.7 kbyte/s)
[09:12] <Reventlov> 200~VIDEO: [H264] 1920x1080 24bpp 23.976 fps 4969.9 kbps (606.7 kbyte/s)
[09:12] <Reventlov> orry.
[09:12] <blippyp> ah
[09:13] <blippyp> yeah, you're probably lucky to get away with 3mb then... ;)
[09:13] <blippyp> you might want to try and tweak it a bit, but I'm guessing you're good to go if you liked what you saw....
[09:14] <Reventlov> well, I liked. Still, errors in the file seems to "break" the video stream, plus, i still have this bug opened on the bugtracker
[09:14] <aolin> if this is a transcoding server for PlayStation3 or such then the stream bitrate can be as high as your network can handle
[09:15] <Reventlov> Thank you
[09:15] <Reventlov> well, it's a server somewhere
[09:15] <aolin> ah ok, then throughput might be issue
[09:15] <Reventlov> Good connection, cpu limited
[09:16] <aolin> mpeg2 tends to be very large in good quality
[09:17] <blippyp> try raising your bitrate to at least 5mb
[09:18] <blippyp> if not higher
[09:18] <blippyp> I'd aim for between 5 and 8 to start
[09:21] <blippyp> like I said I'm not familiar with that codec, but x264 if probably better and I most videos need at least 8 with that codec to look good (i.e. the best that youtube will produce) a full blown blu-ray shoots out like 30 I believe - so.....
[14:25] <haxie> I'm trying to use ffmpeg to take a list of files of different types and frame sizes (for example, avi @ 720x480, mp4 @ 640x580, mp4 @ 1280x760, etc) and concatenate them into a single output mp4 with a maximum frame-size (so each video should be scaled so that it can fit entirely within the frame, even if borders are required).
[14:26] <haxie> Is such a thing possible?
[14:26] <haxie> I've been trying to set it up, but haven't been having much luck. Just wondering if 1. it's possible and 2. there are any shortcuts for something like this?
[14:31] <haxie> my initial approach was to take each file and convert them to the same encoding (say, mp4) and then use concat demuxer to combine them, but the missing piece is the automatic frame scaling
[15:16] Action: penguin42 is looking for suggestions for dumb IP cctv camera (jpeg output) to video with a web interface but I don't need motion detection; I can convert the IP cctv output using ffmpeg, but then how would I also be able to monitor it in real time?
[16:21] <sweb> for get screenshot from movie i use this command :
[16:21] <sweb> ffmpeg -i mymovie.mp4 -ss 100 -vframes 1 out.jpg
[16:21] <sweb> but it's to slow when i want to capture from long movie
[16:23] <c_14> https://trac.ffmpeg.org/wiki/Seeking%20with%20FFmpeg
[16:23] <c_14> -ss as an output option seeks frame by frame and therefore takes longer than -ss as an input option which seeks by keyframes.
[16:23] <c_14> Use a mix of both for fast and accurate seeking.
[18:26] <houbahop69> hi, I have just succesfully compiled ffmpeg from sources with --enable-opencl option on a debian jessie + nvidia
[18:28] <houbahop69> Unfortunately whn I try "ffmpeg -opencl_bench" I get "Could not get OpenCL platform ids: unknown error" and "Could not get device list from environment" I can't undestand why any help will be welcome ... Thanks
[18:49] <ParkerR> [mbm], lol input was 1920x1080i@24 mpeg2 ac3 and was 2.5GB - output was 1920x1080p@59 h264 ac3 was 1.1GB
[18:49] <ParkerR> Double the framerate but due to h264 less than half the size
[18:49] <ParkerR> LD
[18:49] <ParkerR> *:D
[18:51] Action: penguin42 assumes most of those extra frames become 'yeh the same, no change' and are small
[18:53] <Sevil> Hi, Can someone help me to transcode live input
[18:53] <ParkerR> penguin42, it's because deinterlacing
[18:54] <penguin42> oh I hadn't noticed the i/p
[18:55] <ParkerR> Sevil, live input as in?
[18:56] <houbahop69> Hi, could anybody help with opencl ?
[18:57] <Sevil> i have live input as UDP
[18:57] <Sevil> ffmpeg -i udp://@127.0.0.1:5032 -map 0:0 -map 0:15 this settings is a must and i want it in h264 aac with 1400kb bitrate
[18:58] <Sevil> size 1280x720
[18:58] <ParkerR> Sevil, -acodec aac (might have to check if thats the right name for the codec) -vcodec libx264 -b:v 1400k
[19:06] <Sevil> at this moment i have ffmpeg -i udp://@127.0.0.1:4009 -map 0:0 -map 0:15 -strict -2 -acodec aac -vcodec libx264 -b:v 1400k -f flv bla2.flv
[19:09] <Sevil> on ffserver it gives mush lower cpu usage, how can i optimize that line
[19:09] <ParkerR> Sevil, what is it not doing well? High CPU usage?
[19:10] <ParkerR> Also if you are using h264 there's really no need for flv
[19:11] <Sevil> i use that as example whihc one you suggest
[19:12] <ParkerR> Sevil, drop the -f as it's not really needed and just use a .mp4 extension
[19:12] <ParkerR> * -f flv
[19:13] <ParkerR> Forcing it to flv may be causing some of the overhead
[19:13] <ParkerR> Or may just be a redundant option. Not sure.
[19:14] <Sevil> but still it gives high cpu usage
[19:15] <ParkerR> Well video transcoding is by nature CPU intensive
[19:15] <ParkerR> For the most part
[19:15] <ParkerR> Are you just wanting to capture the stream?
[19:16] <Sevil> no i want to restream it, it is for a project of school
[19:16] <ParkerR> ffmpeg -i udp://@127.0.0.1:5032 -map 0:0 -map 0:15 -vcodec copy -acodec copy out.flv
[19:16] <ParkerR> Should be the lowest CPU wise
[19:17] <Sevil> i get unicast from another computer and i want to make it into hls, but the first step is to get it transcoded with lower bitrate and the mapping
[19:17] <ParkerR> Basically just stores the data it receives without transcoding
[19:17] <Sevil> because after this i will make it hls
[19:19] <Sevil> when i copy it will give high bitrate :(
[19:19] <ParkerR> Well then try just changing bitrate
[19:20] <ParkerR> ffmpeg -i udp://@127.0.0.1:5032 -map 0:0 -map 0:15 -vcodec copy -b:v 1400k -acodec copy out.flv
[19:22] <Sevil> now it will ignore it still high bitrate
[19:23] <Sevil> maybe i can show you if you want
[19:23] <c_14> If you want to convert the video to h264, you're going to use a lot of cpu time. You can try switching the preset to {very,super,ultra}fast if you want it to encode faster.
[19:24] <c_14> You can't set a bitrate with -codec copy
[19:25] <ParkerR> c_14, ahh ok good to know
[19:26] <Sevil> its better now thanks :) only need to look quality etc..
[19:26] <Sevil> -vprofile high -preset faster
[19:27] <c_14> Try messing with the crf
[19:27] <c_14> actually nvmd
[19:27] <c_14> You're setting a bitrate manually.
[19:31] <Sevil> i think it works fine i need to test this for a day to look if it not crash
[19:32] <Sevil> one more question if someone moves fast it will give lines or something how can i fix that
[19:32] <c_14> Is the input interlaced?
[19:32] <Sevil> is that deinterlace
[19:32] <c_14> Fast movement producing lines sounds like the source is interlaced, yes.
[19:32] <liquidmetal> I'm trying to setup ffmpegthumbnailer on fedora - I get this error when gnerating thumbnails - Failed to load gio libraries
[19:32] <ParkerR> -vf yadif=1
[19:33] <Sevil> yes its interlanced then
[19:33] <ParkerR> Sevil, although that'll add to CPU load
[19:33] <Sevil> okey, i will test if it gives lot more load
[19:34] Action: liquidmetal is afk
[19:35] <Sevil> pff that option will give alot more cpu load :(
[19:44] <liquidmetal> https://code.google.com/p/ffmpegthumbnailer/issues/detail?can=2&start=0&num…
[19:44] <liquidmetal> Can someone help me figure out the code chang for this ticket?
[19:46] <c_14> I'm guessing it's this: https://code.google.com/p/ffmpegthumbnailer/source/detail?r=250 But this channel doesn't deal with support for ffmpegthumbnailer
[19:47] <Lokie> is there a limit of how many threads ffmpeg can use? same command on a 4core machine had avg of ~ 4 on a 8core i get ~ 5.5
[19:48] <Lokie> main thread cpu usage on the 4core was ~ 400% on the 8core ~ 500-600%
[19:48] <ParkerR> Lokie, avg FPS?
[19:48] <Hello71> -threads
[19:48] <Lokie> 15
[19:48] <Hello71> you're best off running multiple ffmpegs though because some steps are un-parallelizable
[19:49] <Lokie> Hello71 i used -thread 4 and -thread 8 respectively
[19:49] <Hello71> xargs -P $(nproc)
[19:49] <Lokie> been running it the last 3 hours same usage so far
[19:49] <Lokie> it's a vps that u can resize it's cpu from 1 up to 20 cores
[19:51] <Lokie> Hello71 no response from that
[19:54] <ParkerR> Lokie, you have to use it in combo with ffmpeg
[19:54] <ParkerR> iirc
[19:54] <liquidmetal> c_14, got it. That was it, thanks!
[19:54] <Lokie> ParkerR u mean multiple ffmpegs?
[20:01] <ParkerR> Lokie, xargs is usually for parallel processing
[20:01] <ParkerR> You have to supply it a commond and args
[20:01] <ParkerR> Not sure how to use it with ffmpeg
[20:05] <Lokie> thx will try googling a bit
[22:43] <barhom> Can i buffer the output of ffmpeg somehow? I want it to not start writing to file for exmample until it has processed 10 seconds of data
[22:58] <aleb> When I try to run multiple ffmpeg commands like this: "ffmpeg -nostdin -i $i -acodec libvorbis -q:a 4 -vcodec libvpx -minrate 8M -maxrate 8M -b:v 8M -sn $i.webm 2>&1 >$i.log &" it does not work and this is printed: "[2] + 822 suspended (tty output) ffmpeg -nostdin -i $i -acodec libvorbis -q:a 4 -vcodec libvpx -minrate 8M 8M"
[22:58] <aleb> any idea how to make ffmpeg work in the background?
[23:00] <c_14> Try running ffmpeg with nohup
[23:01] <c_14> ie: nohup ffmpeg $options
[23:05] <aleb> uu, nice, thanks!
[00:00] --- Mon May 12 2014
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