March 2014 - Ffmpeg-devel-irc

ffmpeg-devel.log.20140310
by burek 11 Mar '14

11 Mar '14

[02:00] <lindenle> Hi all. I am trying to decode an mp4 and then re-encode it to basically the same format. I am getting invalid argument when i do an interleaved write. i looked in the source and it looks like it gets thrown when AVPacket->dts is NOPT unless the format does not need timestamps. How do i set the packets dts? Shouldn't it get set on the decode of the packet when i read it in? [02:27] <michaelni> dts will often be set by the demuxer but not always, also if you use an encoder, then the dts from the demuxer have no meaning [02:28] <lindenle> michaelni: ok so my demuxer is setting dts in my pkt that i decode. how do i set the dts for the video encoder? [02:29] <lindenle> michaelni: also for context im basically doing decode -> Frame -> encode [02:30] <lindenle> using the demuxing.c and muxing.c examples as my starting point [02:32] <michaelni> the AVPacket output from the encoder should have dts set, if the input to the encoder had timestamps set [02:32] <lindenle> or i guess more appropriatly read -> pkt -> decode -> frame -> encode -> pkt -> write [02:32] <lindenle> so i need to set pts in the frame i pass to the encoder? [02:33] <lindenle> im seriously not grokking this by the way [02:34] <lindenle> michaelni: I guess i am asking how i set the dts before i do an interleaved write correctly as currently there is not pts in the fram i got from the encoder. [02:37] <lindenle> michaelni: ? [02:38] <michaelni> yes you set the AVFrame pts for whats input in the encoder, should also be correctly done in muxing.c [02:39] <lindenle> ok so i take my decoded frame and set the pts on it like in: http://ffmpeg.org/doxygen/2.1/doc_2examples_2muxing_8c-example.html? [02:39] <lindenle> see im reuing the frame that i decoded as my input frame to the decode (currently) id that crazy? [02:41] <lindenle> should i be using two frames and copying data between them? [02:41] <lindenle> michaelni: ^^ [02:42] <michaelni> there should be no need to copy frames unless you write into them [02:42] <michaelni> also that isnt the latest muxing example but that shouldnt matter [02:43] <michaelni> theres more recent example code in git master [02:43] <lindenle> ok [02:54] <lindenle> michaelni: im still getting non monotonically increasing invalid dts in encode [02:54] <lindenle> michaelni: i am setting pts in the frame i use to encode just as in the example [02:54] <michaelni> make sure you input valid pts to the encoder and makre sure all rescalings are done correctly [02:55] <michaelni> also keep in mind the timestamps in the input file might be faulty [02:55] <michaelni> print things out and you will maybe spot where things go wrong [02:55] <lindenle> here is the gist: https://gist.github.com/lindenle/72fab989fc053a89c6c0 [02:56] <lindenle> michaelni: does that look crazy? [02:57] <lindenle> michaelni: I guess im not understanding why i need anything from the decoded packet at all if i just calculate it with the output strem time base and codec time base. Or do i need to do two calculations? one from encoded to decoded then from decoded to encoded? [03:15] <cone-29> ffmpeg.git 03Timothy Gu 07master:53eb4e0799b7: Add fieldmatch test [05:23] <lindenle_> So im only getting green in my output video...anyone have a thought as to what I could be doing wrong? [05:33] <lindenle_> im merely trying to decode and encode mp4 h264 aac with the same settings. [09:12] <ubitux> BBB: so you might actually be right, and the '!' might be about post increment [09:13] <ubitux> though, i understand the official documentation as exactly the opposite [09:16] <ubitux> official doc says this: http://i.imgur.com/hWfKDyv.png [09:18] <ubitux> and slide 22 from neon_tutorial.pdf says the opposite [10:09] <j-b> Morning [10:12] <plepere> morning. :) [12:03] <cone-589> ffmpeg.git 03Martin Storsjö 07master:8cafeb8bca5d: mxfdec: Validate parameters to strftime [12:03] <cone-589> ffmpeg.git 03Michael Niedermayer 07master:4c63beeefe9f: Merge commit '8cafeb8bca5d079041739dbd72ccec0ead138eaf' [12:23] <cone-589> ffmpeg.git 03Anton Khirnov 07master:e854b8f9f409: Work around broken floating point limits on some systems. [12:23] <cone-589> ffmpeg.git 03Michael Niedermayer 07master:27b4d154cad0: Merge remote-tracking branch 'qatar/master' [13:27] <cone-589> ffmpeg.git 03Timothy Gu 07release/1.2:0d819ed318de: configure: use pkg-config to detect libbluray [13:28] <cone-589> ffmpeg.git 03Michael Niedermayer 07release/1.2:6ee47bd008ff: Merge remote-tracking branch 'TimothyGu/release/1.2' into release/1.2 [14:08] <cone-589> ffmpeg.git 03Michael Niedermayer 07release/1.0:08f53e1b14ff: update for 1.0.9 [15:00] <cone-589> ffmpeg.git 03Michael Niedermayer 07fatal: ambiguous argument 'refs/tags/n1.0.9': unknown revision or path not in the working tree. [15:00] <cone-589> Use '--' to separate paths from revisions [15:00] <cone-589> refs/tags/n1.0.9:HEAD: Merge remote-tracking branch 'qatar/master' [15:02] <plepere> ok, I've got a working version of 8-tap filter. I've gotten under 16 seconds from a 26 seconds video without any optimizations [15:06] <ubitux> :) [15:10] <nevcairiel> you should target 13, then you can say its 100% faster [15:15] <plepere> haha [15:16] <plepere> well I still have the SAO, the iDCT, the weighting and all to do in ASM [15:19] <plepere> ubitux : IIRC you work on AVX2. Does FFmpeg really support AVX2 ? I didn't see the --disable-avx2 in the configure. only up to avx [15:20] <ubitux> configure: --disable-avx disable AVX optimizations [15:20] <ubitux> configure: --disable-avx2 disable AVX2 optimizations [15:20] <plepere> I must have an old configure then [15:20] <plepere> so there's nothing on not using AVX2 ? [15:20] <ubitux> but i didn't do any avx2 yet, i'm still waiting for the hw... which i wonder why it hasn't arrived yet btw [15:21] <ubitux> i don't think we have avx2 yet [15:21] <ubitux> but infra is available [15:21] <plepere> I might get my hands on some haswell mid april. I'll see if I can get some interesting numbers [15:23] <cone-589> ffmpeg.git 03Michael Niedermayer 07release/0.11:aa1fdf1a6457: update for 0.11.5 [15:53] <plepere> hey vp9 guys, is there a performance hit on using a C wrapper for filter_8tap_2d_fn instead of having a full ASM function for the 8tap 2d ?? [15:53] <plepere> oops, second ? :/ [16:32] <cone-589> ffmpeg.git 03Michael Niedermayer 07fatal: ambiguous argument 'refs/tags/n0.11.5': unknown revision or path not in the working tree. [16:32] <cone-589> Use '--' to separate paths from revisions [16:32] <cone-589> refs/tags/n0.11.5:HEAD: Merge remote-tracking branch 'qatar/master' [18:12] <cone-589> ffmpeg.git 03Luca Barbato 07release/0.10:8b24e17d0920: twinvq: Cope with gcc-4.8.2 miscompilation [18:12] <cone-589> ffmpeg.git 03Luca Barbato 07release/0.10:a89acaa0b0db: get_bits: change the failure condition in init_get_bits [18:12] <cone-589> ffmpeg.git 03Luca Barbato 07release/0.10:976a7b72a3f5: avi: directly resync on DV in AVI read failure [18:12] <cone-589> ffmpeg.git 03Michael Niedermayer 07release/0.10:d04194db4571: vqavideo: check chunk sizes before reading chunks [18:12] <cone-589> ffmpeg.git 03Michael Niedermayer 07release/0.10:ef6c90e102a3: dsputil/pngdsp: fix signed/unsigned type in end comparison [18:12] <cone-589> ffmpeg.git 03Michael Niedermayer 07release/0.10:cb5d0ea0bec1: flashsv: Check diff_start diff_height values [18:12] <cone-589> ffmpeg.git 03Michael Niedermayer 07release/0.10:b7a750f67f68: Merge commit 'cb5d0ea0bec119ecbe327bd7d3834987ab42ec1a' into release/0.10 [18:24] <cone-589> ffmpeg.git 03Justin Ruggles 07release/0.10:b68e5b119588: avutil: use align == 0 for default alignment in audio sample buffer functions [18:24] <cone-589> ffmpeg.git 03Luca Barbato 07release/0.10:2c0bfce4cb2d: avi: DV in AVI must be considered single stream [18:24] <cone-589> ffmpeg.git 03Luca Barbato 07release/0.10:e964207e6c17: cavs: Check for negative cbp [18:24] <cone-589> ffmpeg.git 03Anton Khirnov 07release/0.10:23144c5f060d: h264_cavlc: check the size of the intra PCM data. [18:24] <cone-589> ffmpeg.git 03Anton Khirnov 07release/0.10:6a56d16dc136: segafilm: fix leaks if reading the header fails [18:24] <cone-589> ffmpeg.git 03Martin Storsjö 07release/0.10:2f4e066d66b3: mov: Free an earlier allocated array if allocating a new one [18:24] <cone-589> ffmpeg.git 03Anton Khirnov 07release/0.10:979f77b0dc40: h264: check that an IDR NAL only contains I slices [18:24] <cone-589> ffmpeg.git 03Michael Niedermayer 07release/0.10:0d82c3a0ca6c: Merge commit '979f77b0dc40571761999633a38d97be9a1670c8' into release/0.10 [18:36] <cone-589> ffmpeg.git 03Anton Khirnov 07release/0.10:716ee73c991c: h264: reset num_reorder_frames if it is invalid [18:36] <cone-589> ffmpeg.git 03Michael Niedermayer 07release/0.10:30c8a5e4f6c0: vc1: Always reset numref when parsing a new frame header. [18:36] <cone-589> ffmpeg.git 03Anton Khirnov 07release/0.10:e03b875c0b57: mathematics: remove asserts from av_rescale_rnd() [18:36] <cone-589> ffmpeg.git 03Anton Khirnov 07release/0.10:b0db7a523df5: oggparseogm: check timing variables [18:36] <cone-589> ffmpeg.git 03Reinhard Tartler 07release/0.10:5522c564d48e: Updated Changelog for 0.8.10 [18:36] <cone-589> ffmpeg.git 03Anton Khirnov 07release/0.10:8cade1352bde: lavf: make av_probe_input_buffer more robust [18:36] <cone-589> ffmpeg.git 03Reinhard Tartler 07release/0.10:36017d49e2f7: Prepare for 0.8.11 Release [18:37] <cone-589> ffmpeg.git 03Michael Niedermayer 07release/0.10:15efd9a7c0a6: Merge commit '36017d49e2f797f7371dc24848a2285ca63e39ab' into release/0.10 [18:47] <cone-589> ffmpeg.git 03Anton Khirnov 07release/0.10:cf676c159b81: rpza: limit the number of blocks to the total remaining blocks in the frame [18:47] <cone-589> ffmpeg.git 03Vittorio Giovara 07release/0.10:a6003760bd07: h264: Lower bound check for slice offsets [18:47] <cone-589> ffmpeg.git 03Luca Barbato 07release/0.10:4279e0e8d09a: h264: Fix a typo from the previous commit [18:47] <cone-589> ffmpeg.git 03Justin Ruggles 07release/0.10:9786c24bb756: samplefmt: avoid integer overflow in av_samples_get_buffer_size() [18:47] <cone-589> ffmpeg.git 03Anton Khirnov 07release/0.10:43aa7eb38efc: shorten: pad the internal bitstream buffer [18:47] <cone-589> ffmpeg.git 03Anton Khirnov 07release/0.10:de0e442e9d67: truemotion1: check the header size [18:47] <cone-589> ffmpeg.git 03Ronald S. Bultje 07release/0.10:2c1d84499bfe: lagarith: pad RGB buffer by 1 byte. [18:47] <cone-589> ffmpeg.git 03Michael Niedermayer 07release/0.10:5640ea43d720: Merge commit '2c1d84499bfe06d75e9160b824eeffd9f5587337' into release/0.10 [18:52] <cone-589> ffmpeg.git 03Anton Khirnov 07release/0.10:2fb0a52e70fa: lagarith: avoid infinite loop in lag_rac_refill() [18:52] <cone-589> ffmpeg.git 03Anton Khirnov 07release/0.10:3da4fdd5acdd: lagarith: reallocate rgb_planes when needed [18:52] <cone-589> ffmpeg.git 03Janne Grunau 07release/0.10:fd2fc130b24c: arm: hpeldsp: prevent overreads in armv6 asm [18:52] <cone-589> ffmpeg.git 03Janne Grunau 07release/0.10:0120e480bf0a: arm: hpeldsp: fix put_pixels8_y2_{,no_rnd_}armv6 [18:52] <cone-589> ffmpeg.git 03Michael Niedermayer 07release/0.10:ad4963a94464: Merge remote-tracking branch 'qatar/release/0.8' into release/0.10 [19:00] <cone-589> ffmpeg.git 03Michael Niedermayer 07release/0.10:98f44b24b56b: dnxhdenc: fix mb_rc size [19:01] <cone-589> ffmpeg.git 03Michael Niedermayer 07release/0.10:5cab56dc9e40: avcodec/vmnc: Check that rectangles are within the picture [19:01] <cone-589> ffmpeg.git 03Michael Niedermayer 07release/0.10:4e47ae4e7162: avcodec/vc1: reset fcm/field_mode in non advanced header parsing [19:01] <cone-589> ffmpeg.git 03Michael Niedermayer 07release/0.10:ca8c3ec11b8c: wmalosslessdec: make mclms arrays big enough for whats written into them. [19:01] <cone-589> ffmpeg.git 03Michael Niedermayer 07release/0.10:ae51d9398353: avcodec/wmalosslessdec: fix mclms_coeffs* array size [19:01] <cone-589> ffmpeg.git 03Michael Niedermayer 07release/0.10:68b14c044a4a: avformat/mpegtsenc: Check data array size in mpegts_write_pmt() [19:01] <cone-589> ffmpeg.git 03Michael Niedermayer 07release/0.10:f157f18b3473: avcodec/msrle: use av_image_get_linesize() to calculate the linesize [19:01] <cone-589> ffmpeg.git 03Michael Niedermayer 07release/0.10:43b1762ab869: avcodec/snow: split block clipping checks [20:10] <cone-589> ffmpeg.git 03Michael Niedermayer 07release/0.10:6f9e6ac6aa3c: update for 0.10.12 [20:28] <cone-589> ffmpeg.git 03Michael Niedermayer 07fatal: ambiguous argument 'refs/tags/n0.10.12': unknown revision or path not in the working tree. [20:28] <cone-589> Use '--' to separate paths from revisions [20:28] <cone-589> refs/tags/n0.10.12:HEAD: Merge remote-tracking branch 'qatar/master' [21:06] <lindenle> Requested output format 'mp4' is not a suitable output format <- does that mean i dont have the muxer enabled? [21:12] <Mavrik> mhm [21:19] <J_Darnley> lindenle: yes it does sound like that [21:19] <lindenle> J_Darnley: yeah, i had to look up mhm... ha ha. I had disable-demuxers...duh [21:21] <J_Darnley> :) [22:12] <ubitux> BBB: the fpel put functions are just memcpys? [22:51] <cone-589> ffmpeg.git 03Luca Barbato 07master:5eacbb532895: golomb: Add a get_se_golomb_long [22:51] <cone-589> ffmpeg.git 03Michael Niedermayer 07master:d79d5aed6d43: Merge commit '5eacbb53289570834f9a1acb15fd406ea224eef6' [23:00] <cone-589> ffmpeg.git 03Luca Barbato 07master:ee17be3fdd37: hevc: Use get_se_golomb_long [23:00] <cone-589> ffmpeg.git 03Michael Niedermayer 07master:b73cf4eb1561: Merge commit 'ee17be3fdd37f63f4b77676820e387858908b0f4' [23:20] <lindenle> hi all. using avcodec and it tells me that aac is experimental. do i need to change something in my configure flags? [23:21] <lindenle> https://gist.github.com/lindenle/61febecff20d9777e7c1 [23:21] <lindenle> --enable-libfaac for instance? [23:22] <JEEB> you can use the ffaac encoder (internal aac) with -strict experimental, or you can build ffmpeg with a third party AAC encoder, of which fdk-aac is currently the best [23:23] <JEEB> do note that all of the usable level external AAC encoders for one reason or another create a nondistributable binary [23:23] <lindenle> JEEB: im not using ffmpeg i wrote my own demuxing and muxing code. Can i somehow set th strict programatically? [23:23] <JEEB> yes [23:23] <lindenle> JEEB: How? [23:23] <JEEB> you'll have to read the doxy/headers for that :P [23:23] <lindenle> avcodec i assume [23:24] <JEEB> yup [23:24] <JEEB> strict_std_compliance should have a macro that sets it to experimental [23:24] <lindenle> ok thanks [23:24] <JEEB> -2 is the internal value, but there should be a named macro which is better than a magic number :P [23:25] <lindenle> yeah i see the macro but not the function [23:25] <JEEB> in the context you set the variable I would think [23:25] <lindenle> ah ok [23:26] <lindenle> yep i see it!thanks! [23:27] <lindenle> https://gist.github.com/lindenle/4957abd9671c485626b7 [23:27] <lindenle> so something like that JEEB ? [23:28] <JEEB> something like that I would guess, yes [23:28] <lindenle> :P [23:36] <cone-589> ffmpeg.git 03Luca Barbato 07master:4a8562394b68: configure: Use the right pkgconf file for openjpeg [23:36] <cone-589> ffmpeg.git 03Michael Niedermayer 07master:182c674f6813: Merge commit '4a8562394b685e83ae4a38a93eef43625755a231' [23:44] <lindenle> How do I know what sample formats are available for the internal aac codec? [23:48] <JEEB> one way would be to look at the definition of the encoder in the encoder's c file [23:48] <JEEB> I have no idea if you can enumerate the stuff that an encoder supports [23:48] <JEEB> you probably can [23:54] <cone-589> ffmpeg.git 03Tim Walker 07master:34bbc81de8a4: lavf: simplify ff_hevc_annexb2mp4_buf [23:54] <cone-589> ffmpeg.git 03Michael Niedermayer 07master:7f9005547060: Merge commit '34bbc81de8a49fbddb92b76dc733f40890480b2b' [00:00] --- Tue Mar 11 2014

1 0

ffmpeg.log.20140310
by burek 11 Mar '14

11 Mar '14

[01:29] <lindenle> HI all i am getting invalid argument from av_interleaved_write_frame. What does that mean? [01:43] <lindenle> Hello71: im using libavcodec i figured it out by looking at the source code [02:36] <Jack64> what's the best way to create a video from images ? like image1 from 0:00:00.000 to 0:01:00.000, image2 from 0:01:00.000 to 0:01:30.000, etc losslessly from png images? [02:36] <Jack64> image2 would actually start at 0:01:00.001 but who cares [02:47] <braincracker> hello my friends [02:47] <braincracker> can ffmpleg play .mid file? [02:50] <Hello71> I https://encrypted.google.com/search?q=ffmpeg+midi [02:50] <Hello71> https://encrypted.google.com/search?q=ffmpeg+midi [02:50] <Hello71> ffmpeg is not a midi synthesizer. [03:16] <klaxa> braincracker: you might want to try something like timidity or rosegarden [03:21] <Jack64> klaxa: hey there [03:21] <klaxa> hi [03:23] <Jack64> so I tried using the script on other videos... still getting those damn duplicate frames where the stitching goes wrong :S I figured I'd do it right and convert the images to video and then overlay the video but I can't seem to find the command that tels me how to do it like image1 from 0:00:00.000 to 0:01:00.000, image2 from 0:01:00.001 to 0:01:30.000, etc losslessly from png images [03:23] <Jack64> you got any ideas ? [03:54] <klaxa> good point, no clue [03:54] <klaxa> hmm... [03:54] <klaxa> you could hack around it a little [04:02] <braincracker> thanks, almost installed timidy [06:57] <adyzx> guys running ffmpeg 0.8.6-4 in *nix terminal, deprecated version, but just wanting to hardcode subs onto .mp4 file. i'm I still safe in using this version for this particular task? [08:03] <adyzx> how can i add libass to ffmpeg, even if its a deprecate version? [08:03] <adyzx> *deprecated [08:07] <JEEB> the binary you have at hand doesn't have the libass-based subtitling stuff (although even if you had a newer binary, the Libav project doesn't have that video filter), so you would have to build your own current ffmpeg from the FFmpeg's source code [08:08] <JEEB> the libass-based video filter is kind of messy so I recommend that if you want to do simple hardcoding, it's better to build a video player called mpv that uses libav* on the background [08:08] <JEEB> it would let you read the subtitles and fonts straight off a matroska file, for example [08:08] <JEEB> instead of extracting it all first [08:11] <adyzx> JEEB: ^ take it as with matroska, this applies to mp4 too [08:11] <JEEB> pardon me? [08:11] <JEEB> *I beg your pardon? [08:13] <adyzx> JEEB: in your last response, you used matroska file as an example, what i meant to say this would also apply to an mp4 file (replace matroska file in your comment with mp4) [08:14] <JEEB> that doesn't exactly work because mp4 files don't usually contain fonts or stylized subtitles :P [08:14] <JEEB> the output can be mp4 just as fine as with ffmpeg [08:14] <JEEB> it's just that in many cases people want to hardcode subtitles from a source that already has them, and the libass-based filter sure doesn't make that simple [08:15] <adyzx> JEEB: problem is i'm working for with a beginning mp4 file and .srt subs file [08:15] <JEEB> well that (in theory) should be as hard with either, I _think_ the current workflow of that filter will convert it to ASS internally and then render it [08:16] <JEEB> I don't really use that filter because it's a mess, unfortunately :s (esp. since I see files with fonts and stylized subtitles multiplexed into matroska almost daily) [08:18] <adyzx> JEEB: given your expertise, btw, newb in this video realm, your advice, given what i mentioned starting with .mp4 and .srt file, what's my best option to handle this? [11:57] <wiretap1> Hi there, I have a xeon server (4cpu with 6 cores each= total 24 cores) and I want to use ffmpeg to catpure a TS stream from http link (local) and transcode it to libx264 with low cpu usage and good quality (not more than 1000kbits). input is 720x576i and i want to keep the same resolution. my target is to encode as many streams as possible by utilising all the cpu [11:58] <wiretap1> any suggestions what to look out for? [11:58] <wiretap1> now i am using this format [11:59] <wiretap1> ffmpeg -i "http://127.0.0.1/stream.ts -acodec libmp3lame -ac 2 -ar 44100 -ab 128k -vcodec libx264 -crf 28.5 -r 25 -g 25 -vprofile High444 -level 3.1 -preset veryfast -f flv "rtmp://192.168.1.1/stream1" [12:00] <wiretap1> also i would like to know if deinterlacing is suggested and if it kills the cpu [13:50] <jnvsor> Anyone know an ffmpeg 2.1 PPA for ubuntu users? The official PPA only goes to 0.7 :/ [13:51] <sacarasc> Just grab one of the static downloads. [13:53] <jnvsor> Ooh, didn't know ffmpeg came in anything but source :P [14:03] <Keshl> It also comes in a balanced breakfast. oÉo. [15:49] <justMaku> hello [15:50] <justMaku> Any idea how can I output ffmpeg to a stdout on linux? [15:50] <justMaku> ffmpeg -i http://media.dev/TV/WeebTV/1246/stream -c:v libx264 -crf 23 -c:a libfaac -q:a 100 -f mp4 - [15:50] <justMaku> unfortunatelly doesn't work for me [15:50] <sacarasc> MP4 requires having the whole file to be able to be indexed. Use a different container. [15:50] <justMaku> sacarasc: any idea which container should I use for html5 live streaming? [15:50] <sacarasc> Nope. [15:51] <sacarasc> There might be a streaming MP4 thing available in ffmpeg, but I don't know the name for the option. [15:51] <sacarasc> (Sorry, llogan.) [15:51] <sacarasc> From a little bit of googling, this might work: -f segment -segment_format mp4 -segment_time 10 - [15:52] <sacarasc> But, I've never used them options, so they might not. [16:08] <justMaku> sacarasc: thanks for help, i think movflags option should work too [16:08] <justMaku> unfortunately osx binaries from homebrew project doesn't have that one [16:08] <justMaku> need to compile ffmpeg by my own then [16:10] <sacarasc> movflags would just wait until you have completed the file, then move the moov atom to the start. [16:11] <sacarasc> It wouldn't output anything until the entire file had been finished. [16:11] <justMaku> ah, okay [16:11] <justMaku> -f segment doesn't work unfortunately [16:11] <justMaku> :D [16:12] <justMaku> sacarasc: using -f segment give me: [16:12] <justMaku> Output file #0 does not contain any stream [16:13] <sacarasc> Check the manual, I have never used it before, so I don't know what to do. [18:27] <RenatoCRON> hello people. i'm using ffmpeg to record a ipcam, using -f mjpeg, and i'm getting timeout, i guess, after 100 seconds (or so) http://i.imgur.com/oGW14Pt.png [18:27] <RenatoCRON> each squere is 20 secs [18:27] <RenatoCRON> red= no file for this time. [18:27] <RenatoCRON> command line is: [18:28] <RenatoCRON> http://pastebin.com/raw.php?i=GQpRjUbS [18:29] <RenatoCRON> I tried to add -loop 20 but it doest not 'reconnect' [18:38] <klaxa> i have never used half of those arguments [18:48] <RenatoCRON> I put a clock in front of it and i'm watching it now. it looks continuous, so no problem found. maybe this is on my side when I group by the files to show on screen. I will record for more 30mins [18:48] <RenatoCRON> http://i.imgur.com/iidqdTn.png << pretty odd way to debug, but it works! haha [19:21] <RenatoCRON> hey guys, now I know what is happing, but don't now how to solve. [19:21] <RenatoCRON> happening* [19:23] <RenatoCRON> in a 20 secs video, in video time 0, my clock marks '06:04', on 8 sec of the video, the clock is '06:16'. 16-4 = 12. this frame should only be on 12 second of the video! [19:23] <RenatoCRON> I will provide this video for you [19:28] <RenatoCRON> https://www.dropbox.com/s/gn1hql9ctaydqhk/what.mp4 [19:28] <RenatoCRON> let me try with use_wallclock_as_timestamps [19:33] <RenatoCRON> looks fine now! [22:01] <camspark> hello guys [22:02] <camspark> i was wondering if anybody here could help me out, I'm trying to live stream from cent os using ffmpeg to youtube live and I'm getting 20 second delays, I tried messing around with all the settings and nothing seems to help even --tune zerolatency [22:06] <JEEB> then something else is creating the latency. x264 with tune zerolatency most definitely should be within milliseconds by itself [22:06] <JEEB> and I have no idea how good or bad ffmpeg is at being low latency [22:08] <camspark> i could turn on teamviewer and you could take a look at my setup, perhaps you can spot something out of the ordinary [22:09] <camspark> i thought it was something with youtube live, but twitch tv has the same delay [22:09] <camspark> i know my upload speed on the vps is pretty good so it cant be a connection issue and im streaming like 10kb a second [22:10] <camspark> very small area with like 256 colors and libfdk-aac sound [22:10] <camspark> also compiled the latest ffmpeg from git and i've included everything (before i did the yum install ffmpeg old version) same delays no difference [22:11] <camspark> here is what the configuration looks like [22:11] <camspark> ffmpeg version git-2014-03-10-53eb4e0 Copyright (c) 2000-2014 the FFmpeg developers built on Mar 10 2014 10:23:36 with gcc 4.4.7 (GCC) 20120313 (Red Hat 4.4.7-4) configuration: --enable-gpl --enable-nonfree --enable-libfdk_aac --enable-libmp3lame --enable-libopus --enable-libvorbis --enable-libvpx --enable-libx264 --enable-version3 --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfaac --enable-libtheora --enable [22:12] <camspark> --enable-postproc --enable-pthreads --enable-x11grab --enable-libpulse [22:12] <camspark> perhaps this will hint you on to whats wrong, when I use --tune zerolatency, the stream fps drops from 30 to 3 or 4 [22:13] <JEEB> yes, zerolatency is actually slower than the same preset without it :P Although that sounds kind of lulzy. Anyways, I have no idea about ffmpeg and I only know that x264 itself can be quite low latency by itself [22:14] <camspark> ya alright [22:14] <camspark> i'll keep tweaking it around i guess [22:14] <brontosaurusrex> low-latency should mean less options for multithreading i imagine [22:15] <camspark> i also use preset ultrafast [22:15] <brontosaurusrex> but don't quote me [22:15] <JEEB> x264 switches from picture-based threading to slice-based [22:15] <camspark> and i used compression 51 and 0 both same delays [22:15] <JEEB> which gives you pictures quicker [22:15] <JEEB> (and is somewhat slower than picture-based threading [22:15] <JEEB> anyways, just try calculating the latency locally first, possibly without libx264 even first :P [22:15] <camspark> all i know i use VNC to that server and i get realtime desktop.. no delays at all and streaming gives me 20 second delays [22:16] <JEEB> also make sure whatever you're testing the latency isn't just doing lulzy things as well [22:16] <JEEB> in any case, I only know of x264, nothing else :P [22:16] <camspark> I used -f flv "/tmp/test.flv" the file gets generated every second no problem and when i download it it has no problems [22:17] <camspark> it something maybe with my udp? or whatever RTMP uses? [22:18] <camspark> well i have to use libx264 thats what youtube recommands what should i try instead of that vcodec? [22:20] <camspark> ill try rawvideo and report the findings [22:23] <camspark> can't use rawvideo it can't even get me 1 second of a stream it just shuts off [22:23] <camspark> frame= 11 fps=2.2 q=0.0 size= 339kB time=00:00:00.36 bitrate=756 [22:23] <camspark> bitrate=7561.5kbits/s [22:23] <camspark> people were wondering how to get good bitrate this is the way to go [22:25] <camspark> heres how my parameters look like guys [22:25] <camspark> -vcodec libx264 -qp $CONSTANT_RATE_FACTOR -preset $PRESET -profile:v baseline -g 250 -s $OUTRES \ [22:25] <camspark> CONSTANT_RATE_FACTOR="51" OUTRES="426x240" PRESET="ultrafast" [22:26] <camspark> i changed it from -cpr to -qp cause thats what fellow ffmpegers recommanded is a better compression overall [22:27] <camspark> crf* [22:28] <JEEB> if someone recommended -qp instead of -crf for libx264, they should be slapped :P [22:28] <JEEB> constant quantizer should never be used, except for development purposes [22:28] <JEEB> the only exception is when you want to do lossless compression [22:29] <JEEB> which is crf or qp 0 with 8bit x264, and qp 0 with >8bit x264 [22:30] <dv_> crf is kind of a higher level option, right? [22:30] <dv_> and qp applies to a lower level [22:31] <camspark> the guy on the forum said that both options are identical in terms of compressing speed but that qp is better quality [22:31] <JEEB> no [22:31] <JEEB> unless he meant lossless with crf 0 or qp 0 [22:31] <JEEB> where it's exactly the same [22:32] <JEEB> dv_, not... really. you are kind of right that ffmpeg/libavcodec has a generic option for constant quantizer encoding (since it's an older mode and available with pretty much everything), and crf is an encoder-specific option ("private option") [22:32] <JEEB> but they all set the same way a rate control mode in libx264 [22:32] <dv_> oh hm. I recall 0-51 being the range of constant quantization in the encoding process [22:32] <JEEB> yes, for 8bit H.264 [22:33] <dv_> so what is qp supposed to be then [22:33] <JEEB> that's a setting to set that, yes [22:33] <dv_> but qp is in the 0-69 range [22:34] <JEEB> well the range can be higher because of 1) x264's "emergency mode" and 2) bit depth, for example 10bit has +12 [22:34] <dv_> oh right, forgot about the 10bit modes [22:34] <camspark> all i know if u want good quality you use divx haha [22:34] <JEEB> 51 + 12 = 63, plus emergency mode [22:34] <JEEB> which probably is something like 69, yes [22:35] <dv_> okay, so why is setting qp instead of crf a bad idea then? [22:35] <JEEB> why would you take constant quantizers instead of actual rate control? [22:35] <dv_> ah, see, thats what I meant. crf might end up setting the qp internally [22:35] <JEEB> yes, naturally [22:35] <JEEB> all rate control modes set quantizers for all pictures [22:35] <dv_> so crf is kind of higher level :P [22:35] <JEEB> uhh, yes and no [22:36] <JEEB> constant quantizer mode just sets constant values for I,P,B slices [22:36] <JEEB> ABR (CRF or 1pass bit rate) does variable quantizers depending on the calculations of the rate control mode (+ the slice type differences) [22:37] <JEEB> and then 2pass does similar variable stuff with multiple passes to hit a specific average bit rate [22:37] <JEEB> I'm talking in x264 terms of course, where 1pass ABR and CRF share a lot of code, except the other limits the bit rate and the other uses the rate factor value [22:37] <dv_> hmm okay that makes sense [22:37] <dv_> so what forum was that camspark? [22:38] <dv_> doom9? [22:41] <dv_> JEEB: oh speaking of h264, I have a question about number of pictures. I recall that thanks to frame reordering, a decoder might have to allocate up to 17 frames just to get enough room for all the B-frames etc. but I noticed that some decoders also require sometimes about 5 or 6 extra frames just to decode one [22:41] <JEEB> I think the amount of maximum refs is either 15 or 16 [22:42] <dv_> I mean, 5 or 6 output frame buffers just to decode one. is this typical in decoders? do they need to use these extra buffers for temporary storage of something? [22:42] <JEEB> and yes, decoders will usually only output pictures in the correct PTS order [22:42] <JEEB> which is how libavcodec does it too [22:42] <dv_> I noticed this in some hardware decoders where the buffer pool is allocated once, prior to playback, and do not allow reallocation afterwards [22:42] <dv_> I had to add logic to ensure there are always at least 6 buffers free [22:42] <JEEB> so you feed it pictures until you get your first decoded pictures [22:43] <JEEB> and after that the pictures given out should be in the correct PTS-wise order [22:43] <dv_> so this amount might be caused by the POC? [22:44] <JEEB> isn't POC just something like the DTS? [22:44] <dv_> hmm I am unsure [22:44] <dv_> I am just trying to figure out how this amount (6 frames) can be explained [22:45] <JEEB> in addition to the 15 or 16 pictures? [22:45] <dv_> I didnt find anything in the spec. perhaps it is indeed hardware specific. [22:45] <dv_> yes [22:45] <JEEB> probably just random buffers if you're talking about hw decoding [22:45] <dv_> before, I made sure at least one buffer is free (= available for decoding). but then, the hw dropped some frames. [22:46] <dv_> and unfortunately, this part isnt documented [22:46] <dv_> so I reserve 5+the minimum number indicated by the hw (which can be up to 16) [22:46] <dv_> errr, 6+ [22:47] <JEEB> yeah, sounds like hw-specific stuff [22:47] <dv_> yeah... I guess I'll have to live with this magic constant [00:00] --- Tue Mar 11 2014

1 0

ffmpeg-devel.log.20140309
by burek 10 Mar '14

10 Mar '14

[00:57] <cone-459> ffmpeg.git 03Diego Biurrun 07master:a80145342342: g2meet: K&R formatting cosmetics [00:57] <cone-459> ffmpeg.git 03Michael Niedermayer 07master:17a5dbacf50c: Merge commit 'a801453423420915425f902bf9f9cbe982a55d87' [01:04] <cone-459> ffmpeg.git 03Keiji Costantini 07master:b4d372e091f6: rv10: Forward error from rv10_decode_packet [01:04] <cone-459> ffmpeg.git 03Michael Niedermayer 07master:0c1353ae338b: Merge commit 'b4d372e091f6b30758db2a43a5a9fe2510ec2b13' [01:15] <cone-459> ffmpeg.git 03Keiji Costantini 07master:8c09532672d7: rv10: K&R formatting cosmetics [01:15] <cone-459> ffmpeg.git 03Michael Niedermayer 07master:2beb81522e59: Merge commit '8c09532672d7d3196ec52afefa2c96633bc17b62' [01:23] <cone-459> ffmpeg.git 03Janne Grunau 07master:61985ad72c47: arm: hpeldsp: fix put_pixels8_y2_{,no_rnd_}armv6 [01:23] <cone-459> ffmpeg.git 03Michael Niedermayer 07master:b39e89502437: Merge commit '61985ad72c47bbb668f2d3923bf5c9df83e79323' [01:50] <cone-459> ffmpeg.git 03John Stebbins 07master:0092c1dd8dac: matroskaenc: allow override of "writing application" tag [01:50] <cone-459> ffmpeg.git 03Michael Niedermayer 07master:4cb6814160d0: Merge commit '0092c1dd8dac2d9e185b58503b447a0d3fb5230d' [02:03] <cone-459> ffmpeg.git 03John Stebbins 07master:565e0c6d866c: movenc: allow override of "writing application" tag [02:03] <cone-459> ffmpeg.git 03Michael Niedermayer 07master:9827f3db88c0: Merge commit '565e0c6d866ce08d4b06427456d3d1f4fd856e9c' [02:16] <cone-459> ffmpeg.git 03Pierre Lejeune 07master:0e0cefb22216: build: Use pkg-config for openjpeg [02:16] <cone-459> ffmpeg.git 03Michael Niedermayer 07master:9c99bc1f3fb9: Merge remote-tracking branch 'qatar/master' [03:26] <cone-459> ffmpeg.git 03Marton Balint 07master:ae017c26325f: libzvbi-teletextdec: use av_dlog where possible [03:26] <cone-459> ffmpeg.git 03Marton Balint 07master:085ca7dcdbf9: libzvbi-teletextdec: split dvb packet to slices [03:26] <cone-459> ffmpeg.git 03Michael Niedermayer 07master:ea9399f8f7dc: Merge remote-tracking branch 'cus/stable' [12:57] <ubitux> wow, arm can be really awful to read sometimes when you're not used to read it [12:57] <ubitux> like, vld1.32 {d20[]}, [r0,:32], r2 [12:58] <ubitux> is this some kind of perl superset? [13:01] <wbs> * can be awful to read when you're not used to read it, news at 11 [13:02] <nevcairiel> it does look quite a bit more weird then x86 [13:02] <wbs> it has a different syntax yes - once you know what it does it's way more simple and structured to deal with [13:02] <wbs> ... afaik [13:03] <ubitux> wbs: the point was that intel for instance was more readable even when not used to read; i'm not really seriously ranting against it :p [13:04] <ubitux> not until i understand it :p [13:27] <cone-111> ffmpeg.git 03Janne Grunau 07master:4506a854a4d8: arm: vp3: remove incorrect const in ff_vp3_idct_dc_add_neon declaration [13:27] <cone-111> ffmpeg.git 03Michael Niedermayer 07master:fc1d7811ef4a: Merge commit '4506a854a4d846692ba71daeeff661dc214c8fa2' [13:31] <BBB> ubitux: I agree actually - I think the main problem is that it supports some extra things like inline shifting of GPR values when using them (similar to the mov m1, [x*8] in x86) or pre/post-increment of GPRs, and some more odd stuff, and its notation for that is "different" and you have to get used to it [13:31] <BBB> the inline shifting isn't so weird in notation [13:31] <BBB> pre/post increment is just pure garbage if you're not used to it [13:37] <cone-111> ffmpeg.git 03Janne Grunau 07master:dfe224f377be: aarch64: get_cabac inline asm [13:37] <cone-111> ffmpeg.git 03Michael Niedermayer 07master:cbee7ff0b7df: Merge commit 'dfe224f377be3e45758c69d881ca7874b82d647a' [14:07] <cone-111> ffmpeg.git 03Luca Barbato 07release/1.2:78197501572a: avidec: Let the inner dv demuxer take care of discarding [14:26] <cone-111> ffmpeg.git 03Janne Grunau 07master:634d9d8b3989: arm: get_cabac inline asm [14:27] <cone-111> ffmpeg.git 03Michael Niedermayer 07master:fa4f57399774: Merge commit '634d9d8b398982647b3d7160641198744901d8d8' [14:42] <wm4> seeking in tak seems pretty unreliable... [14:44] <nevcairiel> wasnt tak one of those formats with an optional seeking tabel [14:44] <nevcairiel> i might be confusing it [14:45] <nevcairiel> guess i was [14:45] <wm4> if it does, lavf doesn't use it [14:45] <nevcairiel> otherwise its just a vbr raw format, there is no good way to seek in those [14:45] <wm4> hm, bad [14:45] <wm4> is the format still closed? [14:46] <nevcairiel> I have been wondering if its worth implementing a memory cache with indexing for "small" audio files, small being like everything up to 100mb or something, so that you can seek in those at least [14:46] <nevcairiel> people always complain about vbr mp3 or vbr aac to me as well [14:47] <wm4> pretty much depends whether the file is on fast storage or not [14:47] <wm4> but if so, indexing "small" files would probably be pretty quick [14:51] <wm4> seems like the closed source tak decoder seems to claim it can seek sample-exact [14:52] <wm4> other than that there's a shitty SDK and a pascal source file which apparently documents the bitstream or parts of it [14:55] <wm4> there was a seek table, but it was removed later, and it appears now you're supposed to crawl the raw stream yourself, searching for a header (which contains a frame number and a crc) [14:56] <wm4> and that's where I stop caring [14:59] <wm4> nevcairiel: maybe creating an index in advance would be a nice optional feature for libavformat [14:59] <wm4> it could use the existing generic index stuff (although that looks yucky to mess with) [15:23] <cone-111> ffmpeg.git 03Janne Grunau 07master:c708b5403346: timer: use mach_absolute_time as high resolution clock on darwin [15:23] <cone-111> ffmpeg.git 03Michael Niedermayer 07master:c2923100f5a1: Merge commit 'c708b5403346255ea5adc776645616cc7c61f078' [15:23] <cone-111> ffmpeg.git 03Michael Niedermayer 07master:4159f702a7b8: avutil/timer: Fix units for x86 after c708b5403346255ea5adc776645616cc7c61f078 [15:38] <cone-111> ffmpeg.git 03Janne Grunau 07master:74cc90190574: float_dsp: fix errors in documentation [15:38] <cone-111> ffmpeg.git 03Michael Niedermayer 07master:8dfd0e62f5d3: Merge remote-tracking branch 'qatar/master' [16:39] <BBB> nevcairiel: I think that's a good idea, it could be filled in the bg while the fg thread is already serving data, and could then be format-agnostic [16:39] <BBB> nevcairiel: just needs someone doing it [21:08] <cone-111> ffmpeg.git 03Michael Niedermayer 07master:920c01adce6c: hevc: Use get_bits_long() in decode_vui() [21:08] <cone-111> ffmpeg.git 03Michael Niedermayer 07master:e15a57b67a26: Merge commit '920c01adce6c273fc043513ff237a6266e612152' [21:14] <cone-111> ffmpeg.git 03Michael Niedermayer 07master:6cc94e971933: hevc: use av_mallocz() for allocating tab_ipm [21:14] <cone-111> ffmpeg.git 03Michael Niedermayer 07master:26568c04a827: Merge commit '6cc94e971933cd38c452172bb048bf760e65cc3e' [21:41] <cone-111> ffmpeg.git 03Michael Niedermayer 07master:ed06e5d92b4c: hevc: Do not turn 32bit timebases into negative numbers [21:41] <cone-111> ffmpeg.git 03Michael Niedermayer 07master:06219db9351c: Merge commit 'ed06e5d92b4c67b49068d538461fbbe0a53a8c5e' [21:53] <ubitux> i've a hard time following the arm doc; what's the diff between "[r1,:128]" and "[r1,128]"? [21:54] <ubitux> i see "[r1,off]" and "[r1:align]" [21:54] <ubitux> what's this ",:"? it means "aligned [r1,off]" or something? [22:06] <cone-111> ffmpeg.git 03Vittorio Giovara 07master:4d33873c2990: hevc: make pps/sps ids unsigned where necessary [22:06] <cone-111> ffmpeg.git 03Michael Niedermayer 07master:eaaa5801efec: Merge commit '4d33873c2990b8d6096f60fef384f0efc4482b55' [22:07] <cone-111> ffmpeg.git 03Ben Littler 07master:c09804684a47: isom: add xd51 hdcam [22:07] <cone-111> ffmpeg.git 03Michael Niedermayer 07master:8a5397d88b8c: Merge commit 'c09804684a47e6572ee82ded5d84c8b282815da6' [22:12] <cone-111> ffmpeg.git 03Carl Eugen Hoyos 07master:eeae858596a5: isom: Support Sony XAVC in mov [22:12] <cone-111> ffmpeg.git 03Michael Niedermayer 07master:2f41b186963d: Merge commit 'eeae858596a5ce5a84797283f590a16add5704f4' [22:15] <cone-111> ffmpeg.git 03Piotr Bandurski 07master:650901239843: isom: add Radius DV YUV FourCCs [22:15] <cone-111> ffmpeg.git 03Michael Niedermayer 07master:6783b3eb761b: avformat/isom: remove duplicated line [22:15] <cone-111> ffmpeg.git 03Michael Niedermayer 07master:2030b3a4abd8: Merge commit '6509012398435252979e149ea4a73438d2107600' [22:20] <cone-111> ffmpeg.git 03Piotr Bandurski 07master:31e558fbcfec: isom: add "NO16" FourCC [22:20] <cone-111> ffmpeg.git 03Michael Niedermayer 07master:d32245e5b12c: Merge commit '31e558fbcfec8fa9fddf564ab8360384706e41c0' [22:22] <cone-111> ffmpeg.git 03Carl Eugen Hoyos 07master:073cd30fb4dc: isom: Support MPEG-2 video mov files from FPC5 [22:22] <cone-111> ffmpeg.git 03Michael Niedermayer 07master:76367f453f6f: Merge commit '073cd30fb4dc1c2a885c62540e3cfaac1f17c21b' [22:23] <cone-111> ffmpeg.git 03Piotr Bandurski 07master:db32f14c3dfd: isom: add "MNG" FourCC [22:23] <cone-111> ffmpeg.git 03Michael Niedermayer 07master:c68e53f27281: Merge commit 'db32f14c3dfd75290b57a086183d6577457beac4' [22:24] <cone-111> ffmpeg.git 03Mark Himsley 07master:360022bd3b89: isom: lpcm in mov default to big endian [22:24] <cone-111> ffmpeg.git 03Michael Niedermayer 07master:9f71e5d39e0e: Merge commit '360022bd3b894cc01ea112b275fa4c8f53881808' [22:26] <cone-111> ffmpeg.git 03Piotr Bandurski 07master:8a0bc40545a2: libspeex: support ZygoAudio (quality 10 mode) [22:26] <cone-111> ffmpeg.git 03Michael Niedermayer 07master:858d9947e4fe: Merge commit '8a0bc40545a216a49f43db5d9a42a63961333354' [22:28] <cone-111> ffmpeg.git 03Baptiste Coudurier 07master:9e71cc81f365: movdec: handle 0x7fff langcode as macintosh per the specs [22:28] <cone-111> ffmpeg.git 03Michael Niedermayer 07master:0cf6b3b5b79f: Merge commit '9e71cc81f3655cacf0f91860fba3043f13b64059' [22:30] <cone-111> ffmpeg.git 03Ben Littler 07master:fb0a988f94c3: riff: IPJ2 decodes with JPEG2000 decoder [22:30] <cone-111> ffmpeg.git 03Michael Niedermayer 07master:4aa48e467b1c: Merge commit 'fb0a988f94c3b8907610172295a7f6bf78c57105' [22:31] <cone-111> ffmpeg.git 03James Zern 07master:3e0175d89a6c: riff: add VP9 fourcc [22:31] <cone-111> ffmpeg.git 03Michael Niedermayer 07master:7a69ffaf0df4: Merge commit '3e0175d89a6c5ee8393ae05e1b40abdca1204919' [22:37] <cone-29> ffmpeg.git 03Tim Walker 07master:1d9014f0b008: movenc: use 'hev1' tag for HEVC in MODE_MOV. [22:37] <cone-29> ffmpeg.git 03Michael Niedermayer 07master:ce1ad8d775ba: Merge commit '1d9014f0b008485eac4c19d5f5e11ede59237167' [23:06] <cone-29> ffmpeg.git 03Tim Walker 07master:20b40a597cdd: movenc: write hvcC tag for HEVC. [23:06] <cone-29> ffmpeg.git 03Michael Niedermayer 07master:1b1d77ae9b38: Merge commit '20b40a597cdd4969cf1147d7c7efee2b6232524b' [23:06] <cone-29> ffmpeg.git 03Michael Niedermayer 07master:cb403b257038: avformat: fix hevcs use of golomb from avformat [23:15] <ubitux> oh, mail from qatar on ffmpeg-user. [23:16] <ubitux> http://ffmpeg.org/pipermail/ffmpeg-user/2014-March/020425.html [23:17] <ubitux> (wtf) [23:18] <JEEBsv> lol [23:18] <Mavrik> Oracle DBA. [23:19] <ubitux> it looks like a message constructed with markov chains based on indian and chinese mails from ffmpeg-user [23:29] <BBB> ubitux: I believe [r1, :128] is post-increment [23:29] <BBB> ubitux: and [r1, 128] is pre-increment [23:30] <BBB> ubitux: what that means is (in x86 notation) load [r1], then add r1, 128 (post-increment) or add r1, 128 and then load [r1] [23:30] <BBB> (pre-increment) [23:30] <ubitux> isn't this done respectively with [r1,off]! and [r1],off? [23:30] <BBB> hm maybe ! was post... argh [23:30] <ubitux> ! is pre [23:30] <BBB> lol [23:31] <BBB> ok I'm an idiot [23:31] <ubitux> [r1],off is post [23:31] <BBB> don't listen to me obviousl [23:31] <ubitux> but i don't understand how that works with that "align" parameter (the : thing) [23:32] <ubitux> so far, i understood that: "[R,off]" "R[off]", "[R,off]!" "r+=off; [R]", "[R],off" "R[off]; r+=off" [23:33] <ubitux> A B with A asm syntax and B pseudo code how i understand it [23:33] <cone-29> ffmpeg.git 03Tim Walker 07master:b6c61fb83e87: movenc: enable Annex B to MP4 conversion for HEVC tracks. [23:34] <cone-29> ffmpeg.git 03Tim Walker 07master:4f3db5d3418a: movenc: allow muxing HEVC in MODE_MP4. [23:34] <cone-29> ffmpeg.git 03Michael Niedermayer 07master:a379813ceed6: Merge commit 'b6c61fb83e876d404ac3b0b3657ebfcafdcd1926' [23:34] <ubitux> it also seems you can have mixes like "[R+off1],off2", which i guess is going to do something like "R[off1+off2]; R+=off1" [23:34] <cone-29> ffmpeg.git 03Michael Niedermayer 07master:5d5e2bd86242: avformat/hevc: Make return codes consistent and more flexible [23:34] <cone-29> ffmpeg.git 03Michael Niedermayer 07master:9563e67e0759: Merge commit '4f3db5d3418a24f3b90422e98ad75388052c4284' [23:34] <cone-29> ffmpeg.git 03Michael Niedermayer 07master:88c8e4afeaf7: avformat/hevc: fix mix of av_malloc() with free() [23:34] <ubitux> but i'm trying to get that weird ",:" thing [23:41] <cone-29> ffmpeg.git 03Tim Walker 07master:a823d0948683: matroskaenc: write private data in hvcC format for HEVC. [23:41] <cone-29> ffmpeg.git 03Michael Niedermayer 07master:38aee6408a67: Merge commit 'a823d0948683bd97dd58556b5740e434166209a8' [23:42] <BBB> ubitux: I have no idea; maybe ask wbs? [23:42] <ubitux> i hope to figure out what "vld1.32 {d0[]}, [r0,:32], r2" means by the end of the next week [23:43] <BBB> http://people.xiph.org/~tterribe/daala/neon_tutorial.pdf [23:43] <BBB> is quite useful [23:43] <BBB> (note that in their syntax, : is @) [23:43] <ubitux> ooh [23:43] <ubitux> i was trying to learn with the official man [23:43] <ubitux> but i find it extremely confusing [23:43] <ubitux> thanks for the link [23:43] <BBB> specifically slide 19 [23:44] <BBB> vld2.8 {d0, d1} [r0] load 2 d registers stride 2 8 bits from [r0] [23:45] <ubitux> yeah that part is ok [23:45] <BBB> so vld1.32 {d0[]} [r0,:32], r2 would probably be something like load consecutive units of 32 bits from r0 aligned by 32 bits (?) into d0 [23:45] <BBB> right, but then the r2... [23:45] <BBB> post-increment? [23:46] <BBB> so increment r0 by r2 after the load? [23:46] <ubitux> yes it's post increment of r2 [23:46] <ubitux> but note the ",:" [23:46] <ubitux> the ',' is for the offseting [23:46] <ubitux> and ':' is for the align [23:50] <BBB> wbs: ? ^^ [23:54] <cone-29> ffmpeg.git 03Tim Walker 07master:558b20d729bc: matroskaenc: enable Annex B to MP4 conversion for HEVC tracks. [23:54] <cone-29> ffmpeg.git 03Michael Niedermayer 07master:fb0023639ea5: Merge remote-tracking branch 'qatar/master' [00:00] --- Mon Mar 10 2014

1 0

ffmpeg.log.20140309
by burek 10 Mar '14

10 Mar '14

[01:14] <bove> Can I set swscale flags from the command line? [01:15] <Jack64> I finally made it work :D [01:35] <NID> Is it possible to record an application using pulse for audio? [01:35] <NID> I'm trying to figure out how to do this [01:37] <klaxa> ffmpeg -f alsa -i pulse -c:a flac out.flac [01:38] <klaxa> that will record audio from your default recording device [01:38] <klaxa> you can set your default recording device to a monitor device [01:38] <NID> Right, but is it possible to record a stream, for example from an application? [01:39] <klaxa> theoretically yes, probably not without delay to your soundcard [01:39] <klaxa> someone had a similar usecase: http://klaxa.eu/record%20game%20audio.svg [01:40] <klaxa> if you don't need to listen to the audio while it is being recorded you can skip the loopback module [01:40] <NID> Yeah I don't need to listen to it [01:40] <klaxa> then you can create a nullmodule, redirect the application's audio to the null-sink and record from the monitor of the null-sink [01:41] <NID> How would I go about doing that? [01:42] <klaxa> with "pacmd load-module module-null-sink" you can load a null-module [01:42] <NID> Ah nevermind, I figured it out [01:42] <NID> Thanks :) [01:42] <klaxa> with pavucontrol you can set the streams in the gui [01:42] <klaxa> :) [01:43] <Jack64> klaxa: made it work :D [01:44] <Jack64> klaxa: check out the working script http://pastebin.com/F9dkuMQ7 it has some comments on the top to help explain [01:45] <klaxa> oh i see what you are doing now... i think generating a video that updates in sync with the original that only contains the scoreboard might be easier [01:46] <klaxa> this seems a bit roundabout on it... [01:46] <Jack64> wow never thought of that -.- [01:46] <klaxa> depending on how complex the scoreboard is, you could probably even do it better with styled subtitles [01:46] <Jack64> well some times it's 29-0 [01:46] <Jack64> haha [01:47] <klaxa> do you have a screenshot of the scoreboard? [01:47] <Jack64> generated on demand with PHP [01:47] <Jack64> so no.. [01:47] <klaxa> can you generate one and show it to me? [01:47] <Jack64> ok just a sec [01:47] <klaxa> using aegisub might actually be a lot easier than this... [01:47] <klaxa> and way less prone to errors [01:48] <Jack64> where can I paste the photo? [01:48] <Jack64> imageshack requires login -.- [01:48] <klaxa> dunno, imgur? [01:48] <klaxa> http://imgur.com/ [01:49] <Jack64> http://imgur.com/JKp9IUR [01:49] <Jack64> the colors are still not right [01:49] <Jack64> but that's not my part to do xD [01:50] <Jack64> the team names are on a local sql db [01:50] <Jack64> and my friend made an HTML interface [01:50] <Jack64> pretty cool [01:51] <klaxa> ah well... using styled subtitles would be possible, gradients *might* be an issue, but given you probably already put a lot of time and effort into this, changing it now would be bad [01:51] <Jack64> ah, and the white parts on the edges are not part of the image [01:51] <Jack64> well yea it's all kind of hacked together [01:51] <Jack64> just for internal use [01:52] <klaxa> well if it works now, that's great, if it doesn't, i would think about generating one video to overlay over everything [01:52] <klaxa> would highly simplify the script too [01:52] <Jack64> he was sick of wasting hours in power director doing the same thing every week [01:52] <klaxa> things that can be automated, should be automated [01:52] <Jack64> yea it would, but doing it like this was also an exercise for me it was fun [01:52] <klaxa> so you can spend more time on being productive [01:52] <Jack64> you're right on that [01:53] <Jack64> I had to write C code to patch into ffprobe and get the keyframe for the splits eheh [01:54] <Jack64> and then I do it all in RAM because splits are instantaneous [01:54] <Jack64> so maybe there's not a big performance penalty for doing it like this instead of a movie [01:54] <klaxa> you could have used libavformat, or just used mkv as a intermediate container with mkvtoolnix [01:55] <Jack64> but that's definitely the better solution. Generate a movie from the png images [01:55] <klaxa> uh... i mean with mkvtoolnix's mkvinfo to get keyframes [01:55] <klaxa> that's what i do for splitting videos at least [01:55] <Jack64> yea [01:57] <Jack64> actually another crappy performance hit if you see on the script is because of the way I'm adding the splash screen at the end I have to reprocess the whole video for adding just a 3 second overlay.. [01:57] <Jack64> .. at the beginning [01:57] <klaxa> mkv and mkvsplit with concatenation :x [01:57] <Jack64> yea that'd probably have been better [01:58] <Jack64> hey it's the first time I tried to do this :P [01:58] <klaxa> you can also use the concat demuxer to concatenate videos [01:58] <klaxa> http://trac.ffmpeg.org/wiki/How%20to%20concatenate%20%28join,%20merge%29%20… [01:58] <Jack64> yea that's what I use [01:58] <Jack64> that's at the end of the script [01:59] <klaxa> that's not a lot of overhead then [06:18] <Jack64> klaxa: I'm going nuts over here it fixed it in some scripts, others it didn't -.- I think I'm gonna have to rewrite it the way you said making a movie of all the overlays [07:39] <Snowleaksange> WWWWWWWWAAAAAAAAAAAAAAHHHHHHHHHHHHHHHH my ffmpeg prog is leaking mem =( [07:40] <Snowleaksange> av_free_packet() av_frame_free() [07:40] <Snowleaksange> what more do you want from me avcodec_decode_video2 ? [07:43] <Snowleaksange> seriously ffmpeg youre my only hope [08:33] <Snowleaksange> avcode_decode_video2 decode_frame ff_thread-get_buffer ff_get_buffer video_get_buffer av_buffer_poo_get av_buffer_allocz av_malloc [08:34] <Snowleaksange> got like 70 meg alloced there that keep growing at map changes :/ [09:34] <Snowleaksange> hmm think my problem is not calling avcodec_clsoe.. only problem is .. it crashes [10:17] <jarainf> jarainf lol [10:46] <batence> "movie=logo.png, scale=200:37 [watermark]; [in][watermark] overlay=10:10 [out]" - how to change these pixels 200:37 in percentage, because when i convert video with big resolution ... watermark comes small, but when convert video with small resolution - watermark comes very big, so is there a solution to scale watermark in percentage of video resolution ? [10:59] <charred> Hello, I'm attempting to compile openlase (following the guide here) https://github.com/marcan/openlase/wiki/Installation ... and I'm getting an error libavresample/avresample.h file not found. Some googling said the ffmpeg configure I had to add an --enable-avresample flag, but I have no idea how to do this [10:59] <Snowleaksange> thought my ffmpeg upgrade went smooth till i realized only 90% mark. now another day trying to fix mem leak [10:59] <charred> Anyone have an idea? [11:24] <Snowleaksange> hmm [11:26] <Snowleaksange> i think my memory leak is from not calling avcodec_close... but avcodec_close crashes.. i call avformat_open_input with a filename, then open the codecs. to close im trying to iterate the nb_streams and call avcodec_close() if stream->codec->codec. but this crashes for png codec on freeing avctx->internal->to_free->metadata [11:40] <Snowleaksange> hmm.. not calling av_frame_free() makes avcodec_close() not crash... and uses a lot less mem [11:58] <Mavrik> Snowleaksange, why don't you just use valgrind to see what's leaking exactly? [12:02] <JEEB> charred, when you compile ffmpeg you will have to manually enable avresample in its configuration script, which is --enable-avresample [12:22] <Snowleaksange> i did Mavrik. avcodec_decode_*. Now trying to fix it. [12:25] <Snowleaksange> well its really not even a leak anymore, as how to not crash avcodec_close [12:25] <Snowleaksange> in the case of loading png files, if i dont call av_frame_free then avcodec_close doesnt crash [12:26] <Snowleaksange> the leak was because i wasnt calling avcodec_close [12:34] <Mavrik> Snowleaksange, mhm, compiling ffmpeg with debug symbols and using gdb helps [12:54] <luc4> Hello! I'm using ffmpeg to dump metadata from a video file. I used: ffmpeg -i INPUT -f ffmetadata FFMETADATAFILE. What I get is no data in the output file. But ffprobe returns creation_time : 2012-10-18 16:26:59 in metadata. Maybe -f ffmetadata is not sufficient? [19:08] <hannes2> hey, asking for a windows-bound friend: what is the easiest way to make her able to run "ffmpeg -i image%04d.png -vcodec libx264 -crf 18 -preset veryslow -pix_fmt yuv420p out.mkv" on windows 7? [19:08] <hannes2> last time i tried to install x264 on someone's windows machine it ended up with a mess [19:09] <sacarasc> http://ffmpeg.zeranoe.com/builds/ [19:09] <hannes2> thanks! [19:09] <sacarasc> Windows builds with libx264 all in. [19:09] <sacarasc> Grab the static one, it's the simplest. [19:09] <hannes2> nice, that looks fantastic [19:11] <hannes2> you made my/our day, thanks! [19:12] <sacarasc> :) [21:20] <viddy> Hello. I have a file which has a change in the audio stream layout like 10~ minutes in. It seems ffmpeg mostly cares about the stream-layout when it starts, so when a new audio stream is introduced suddenly it ends up being dropped (not ending up in the encoded file). Can anyone point me in the direction of a solution to this? Can it be done? Thanks. [21:56] <fp> Trying to decode frames in sync between two files, i.e. get frame 0 from file 1 and then frame 0 from file 2 to do some calculations on the YCbCr-data. Then frame 1 and so on... [21:56] <fp> Using decoder-loop from ffmpeg/doc/examples/demuxing_decoding.c as my example. See http://pastebin.com/kPmczqbB Completely lost, any help would be greatly appreciated! [22:03] <Mavrik> ok? [22:19] <IamTrying> ffmpeg -i ITR.wav -ss 00:00:12 -t 00:00:17 sara1.wav Why its not cutting the WAV file between 12 and 17 seconds but making it 17 seconds in length [22:19] <Mavrik> IamTrying, because that's what you're telling it to do. [22:19] <Mavrik> -t is length, not end position [22:20] <IamTrying> Mavrik, i have a 1 hour WAV file for IVR message. how can i cut the audio starts at 00:00:12 and ends in 00:00:17 ? [22:22] <Mavrik> again. [22:22] <Mavrik> -t is length of a clip. [22:22] <Mavrik> you do know basic maths right? [22:22] <Mavrik> set -t to length you want. [22:23] <IamTrying> Mavrik, -t 00:00:07 works but its not accurate the audio has Milliseconds too, how can i do in the -t Millisecond adjust? [22:23] <klaxa> use 00:00:06.123 instead [22:23] <IamTrying> OK [22:23] <klaxa> i think that's even documented in the manpage [22:24] <Mavrik> mhm. [22:24] <Mavrik> :) [22:24] <IamTrying> Thank you Mavrik , klaxa [22:24] <klaxa> you can also use -to to specify a position rather than a duration [22:32] <IamTrying> How can i merg "goodbye" in the sara5.wav? "Unfortunately nobody is available to join, please call back later or send us an email" = $ ffmpeg -i ITR.wav -ss 00:00:42 -t 00:00:7.500 sara5.wav but i need to add "GoodBye" in sara5.wav from -ss 00:01:55. [22:35] <klaxa> http://trac.ffmpeg.org/wiki/How%20to%20concatenate%20%28join,%20merge%29%20… [22:36] <IamTrying> Thank you klaxa works [00:00] --- Mon Mar 10 2014

1 0

ffmpeg-devel.log.20140308
by burek 09 Mar '14

09 Mar '14

[00:12] <cone-388> ffmpeg.git 03Keiji Costantini 07master:84cfce9f9980: matroskadec: K&R formatting cosmetics [00:12] <cone-388> ffmpeg.git 03Michael Niedermayer 07master:b3d9ab182929: Merge commit '84cfce9f99805a144df684417e166c1ae6f95421' [00:15] <llogan> Compn: you got the author incorrect [00:18] <wm4> Compn stole the copyright for a 1 line patch [00:21] <cone-388> ffmpeg.git 03Diego Biurrun 07master:f69befe5eefe: matroskadec: cosmetics: Fix "attachement" vs. "attachment" typo [00:21] <cone-388> ffmpeg.git 03Michael Niedermayer 07master:fabf69f37abe: Merge commit 'f69befe5eefef12172a6479dd9cef3c01bddec7a' [00:40] <cone-388> ffmpeg.git 03Derek Buitenhuis 07master:f87a6e500bcd: fic: Properly handle skip frames [00:40] <cone-388> ffmpeg.git 03Michael Niedermayer 07master:6eb08783a978: Merge remote-tracking branch 'qatar/master' [01:10] <llogan> roger should read what he types: "Basically, you can accept input dshow input into you program that uses, see [[Using libav*]] to know what libav does." [01:11] <llogan> i didn't know he was a writer for Zero Wing [01:47] <cone-388> ffmpeg.git 03Michael Niedermayer 07master:d1122b7ce5f5: avcodec/wmadec: initialize max_exponent to valid values [01:58] <Compn> llogan : i had author in a previous commit but i had to checkout a new tree 3 or 4 times so it got lost in the shuffle :) [01:59] <llogan> i don't think he'll care [02:02] <Compn> llogan : wm4 was wondering why you'd make him send his patch to the list instead of committing it yourself :P [02:02] <Compn> but i guess if you dont commit much , we shouldnt be making you do things :) [03:57] <JodaZ> i am getting [adts @ 0x2b57100] Encoder did not produce proper pts, making some up. [03:57] <JodaZ> in the middle of some encoding [03:58] <JodaZ> its not true tho, it comes from libavformat/mpegtsenc.c line 1253, where a new packet is created, and has size and data copied from the old one, but pts/dts not, which it feeds back to av_write_frame, which results in that error [05:34] <cone-836> ffmpeg.git 03Michael Niedermayer 07master:b918d6e2e637: avcodec/vorbis: return proper error codes from ff_vorbis_len2vlc() [05:34] <cone-836> ffmpeg.git 03Michael Niedermayer 07master:742d8601031f: avcodec/vorbis: fix decoding of single element huffman trees [09:55] <ubitux> wm4: btw, for using the api, lavfi device is a good example as well [09:55] <ubitux> easier to extract from ffmpeg or ffplay [09:56] <ubitux> also, you have working doc/examples available [09:57] <wm4> ah, there are filtering examples [14:19] <cone-853> ffmpeg.git 03Carl Eugen Hoyos 07master:6208bb965d30: Set codec_tag in dshow device, needed to distinguish between YUV and YVU. [14:19] <cone-853> ffmpeg.git 03Don Moir 07master:24e2a82f75ab: Make 32bit RGB dshow input opaque. [15:01] <cone-853> ffmpeg.git 03Nicolas George 07master:ea6825fd090e: lavc/pthread: copy packet side data. [15:01] <cone-853> ffmpeg.git 03Nicolas George 07master:c37bbe54f400: lavd/xv: take aspect ratio into account. [15:01] <cone-853> ffmpeg.git 03Nicolas George 07master:bc6901c94944: lavfi/af_atempo: clear references before returning error. [15:01] <cone-853> ffmpeg.git 03Nicolas George 07master:97e87e09c841: lavf: add subfile protocol. [15:01] <cone-853> ffmpeg.git 03Nicolas George 07master:8a670f52a56b: lavf/concatdec: more reliable test for absolute URLs. [15:01] <cone-853> ffmpeg.git 03Nicolas George 07master:6bffa83bbeac: lavf/concatdec: reindent after last commit. [15:01] <cone-853> ffmpeg.git 03Michael Niedermayer 07master:b28c37156782: Merge remote-tracking branch 'cigaes/master' [15:19] <cone-853> ffmpeg.git 03Nicolas George 07release/2.1:910fd77967b0: lavfi/af_atempo: clear references before returning error. [15:19] <cone-853> ffmpeg.git 03Nicolas George 07release/2.2:7d995cd1b8e2: lavfi/af_atempo: clear references before returning error. [16:21] <cone-853> ffmpeg.git 03Michael Niedermayer 07release/0.11:fdf57ad2ff74: avcodec/dsputil: fix signedness in sizeof() comparissions [16:21] <cone-853> ffmpeg.git 03Michael Niedermayer 07release/0.11:ceff07dfb94a: avfilter/vf_fps: make sure the fifo is not empty before using it [16:21] <cone-853> ffmpeg.git 03Michael Niedermayer 07release/0.11:66fcda7e7043: avcodec/mjpegdec: Add some sanity checks to ljpeg_decode_rgb_scan() [16:21] <cone-853> ffmpeg.git 03Michael Niedermayer 07release/0.11:ec39abcd3702: avcodec/truemotion2: Fix av_freep arguments [16:21] <cone-853> ffmpeg.git 03Michael Niedermayer 07release/0.11:b650d5376c82: avcodec/ffv1enc: update buffer check for 16bps [16:21] <cone-853> ffmpeg.git 03Michael Niedermayer 07release/0.11:2289bffa312d: avcodec/parser: reset indexes on realloc failure [16:21] <cone-853> ffmpeg.git 03Michael Niedermayer 07release/0.11:3b03778d3578: avfilter/ff_insert_pad: fix order of operations [16:21] <cone-853> ffmpeg.git 03Michael Niedermayer 07release/0.11:82746c468b10: avcodec/avpacket/av_packet_split_side_data: ensure that side data padding is initialized [16:21] <cone-853> ffmpeg.git 03Michael Niedermayer 07release/0.11:683c92adde01: avutil/opt: initialize ret [16:21] <cone-853> ffmpeg.git 03Michael Niedermayer 07release/0.11:84c84204b312: avcodec/jpeglsdec: check err value for ls_get_code_runterm() [16:21] <cone-853> ffmpeg.git 03Michael Niedermayer 07release/0.11:3bcf733b5a9c: cmdutils: update year [16:21] <cone-853> ffmpeg.git 03Michael Niedermayer 07release/0.11:912b7fe922bc: dnxhdenc: fix mb_rc size [16:21] <cone-853> ffmpeg.git 03Michael Niedermayer 07release/0.11:c598a46821f0: avcodec/vmnc: Check that rectangles are within the picture [16:21] <cone-853> ffmpeg.git 03Michael Niedermayer 07release/0.11:b66a0dce117a: avcodec/vc1: reset fcm/field_mode in non advanced header parsing [16:21] <cone-853> ffmpeg.git 03Justin Ruggles 07release/0.11:236b7b16ebcc: samplefmt: avoid integer overflow in av_samples_get_buffer_size() [16:22] <cone-853> ffmpeg.git 03Michael Niedermayer 07release/0.11:c23597ec7eae: avcodec/wmalosslessdec: fix mclms_coeffs* array size [16:22] <cone-853> ffmpeg.git 03Michael Niedermayer 07release/0.11:12770701856a: avformat/mpegtsenc: Check data array size in mpegts_write_pmt() [16:22] <cone-853> ffmpeg.git 03Michael Niedermayer 07release/0.11:3a893335bde2: avcodec/msrle: use av_image_get_linesize() to calculate the linesize [16:22] <cone-853> ffmpeg.git 03Michael Niedermayer 07release/0.11:bc07ec7b2f97: avcodec/snow: split block clipping checks [16:25] <cone-853> ffmpeg.git 03Timothy Gu 07release/0.11:2a45007f3408: configure: use pkg-config to detect libbluray [16:25] <cone-853> ffmpeg.git 03Michael Niedermayer 07release/0.11:aa3378b579f2: Merge remote-tracking branch 'TimothyGu/release/0.11' into release/0.11 [16:48] <cone-853> ffmpeg.git 03Michael Niedermayer 07release/1.0:d7ce013de58f: avcodec/truemotion2: Fix av_freep arguments [16:48] <cone-853> ffmpeg.git 03Michael Niedermayer 07release/1.0:55ad69e851ae: avcodec/ffv1enc: update buffer check for 16bps [16:48] <cone-853> ffmpeg.git 03Michael Niedermayer 07release/1.0:6f9ae27f5136: avcodec/parser: reset indexes on realloc failure [16:48] <cone-853> ffmpeg.git 03Michael Niedermayer 07release/1.0:2448c1867686: avfilter/ff_insert_pad: fix order of operations [16:48] <cone-853> ffmpeg.git 03Michael Niedermayer 07release/1.0:ba59ab596462: avcodec/avpacket/av_packet_split_side_data: ensure that side data padding is initialized [16:48] <cone-853> ffmpeg.git 03Michael Niedermayer 07release/1.0:f83f09bfcef5: avutil/opt: initialize ret [16:48] <cone-853> ffmpeg.git 03Michael Niedermayer 07release/1.0:855fe6544792: avcodec/jpeglsdec: check err value for ls_get_code_runterm() [16:48] <cone-853> ffmpeg.git 03Michael Niedermayer 07release/1.0:584e15d64395: avutil/log: skip IO calls on empty strings [16:48] <cone-853> ffmpeg.git 03Michael Niedermayer 07release/1.0:e571010c5c68: cmdutils: update year [16:48] <cone-853> ffmpeg.git 03Michael Niedermayer 07release/1.0:617fecbe0f9d: dnxhdenc: fix mb_rc size [16:48] <cone-853> ffmpeg.git 03Michael Niedermayer 07release/1.0:87c203ba2684: avcodec/vmnc: Check that rectangles are within the picture [16:48] <cone-853> ffmpeg.git 03Michael Niedermayer 07release/1.0:5b38967745e9: avcodec/vc1: reset fcm/field_mode in non advanced header parsing [16:48] <cone-853> ffmpeg.git 03Justin Ruggles 07release/1.0:0e551aa966ef: samplefmt: avoid integer overflow in av_samples_get_buffer_size() [16:48] <cone-853> ffmpeg.git 03Michael Niedermayer 07release/1.0:b9bf17c55ac2: avcodec/wmalosslessdec: fix mclms_coeffs* array size [16:48] <cone-853> ffmpeg.git 03Michael Niedermayer 07release/1.0:7ad6b66370e5: avformat/mpegtsenc: Check data array size in mpegts_write_pmt() [16:48] <cone-853> ffmpeg.git 03Michael Niedermayer 07release/1.0:54b61181ed55: avcodec/msrle: use av_image_get_linesize() to calculate the linesize [16:48] <cone-853> ffmpeg.git 03Michael Niedermayer 07release/1.0:d7071f676571: avcodec/ansi: fix integer overflow [16:48] <cone-853> ffmpeg.git 03Michael Niedermayer 07release/1.0:a0d15a6271a7: avcodec/snow: split block clipping checks [16:48] <cone-853> ffmpeg.git 03Michael Niedermayer 07release/1.0:e59fe7895813: avcodec/utvideoenc: fix slice_bits size [16:58] <cone-853> ffmpeg.git 03Timothy Gu 07release/1.0:cea57b7fe05b: configure: use pkg-config to detect libbluray [16:58] <cone-853> ffmpeg.git 03Michael Niedermayer 07release/1.0:8418de295a9a: Merge remote-tracking branch 'TimothyGu/release/1.0' into release/1.0 [21:52] <Jack64> guys, can anyone tell me why in this script http://pastebin.com/n6emmH4P the concatenation of files overlaid3.mp4 and overlaid4.mp4 isn't in sync ? there seems to be about 1 second duplication of frames, but all others are concatenated seamlessly except those 2 [23:01] <Jack64> any dev online? [23:13] <BBB> Jack64: that's a user question, not a dev question [23:13] <BBB> Jack64: a dev question is "I just wrote this patch, can someone review it?" or something along those lines [23:18] <ubitux> BBB: which mc function i should start with? [23:19] <ubitux> put4 & put8? [23:20] <ubitux> also, i see no vp9 namespace for those, is it really ok? [23:22] <BBB> ubitux: x86 have vp9 namespace, right? [23:22] <BBB> ubitux: the c ones don't b/c well, they're static, so it doesn't matter [23:23] <BBB> ubitux: I'd start with sth simple, so yeah put4/8/16/32/64 is quite a good idea, and then try the subpel mc ones [23:23] <BBB> and the avg ones [23:23] <BBB> that should give a massive speedup [23:24] <ubitux> <@BBB> ubitux: x86 have vp9 namespace, right? // doesn't seem so, but i may be missing something [23:24] <ubitux> ok, will start with those [23:24] <BBB> hm really? [23:25] <BBB> maybe I forgot [23:25] <BBB> sorry [23:25] <BBB> ah crap you're right the vp9 namespace is missing for the mc functions [23:25] <BBB> sorry [23:26] <BBB> maybe we should add that [00:00] --- Sun Mar 9 2014

1 0

ffmpeg.log.20140308
by burek 09 Mar '14

09 Mar '14

[00:57] <lindenle> hi trying to decode h264 in a mp4 container and i get crop values invalid error. I assume this means I have setup my decoder incorrectly? [00:57] <lindenle> http://ffmpeg.org/doxygen/2.1/doc_2examples_2decoding_encoding_8c-example.h… - working from that example but trying to change the input format [01:07] <JodaZ> lindenle, well, that examples reading is kinda made specifically for mpeg(2), you should look at another example [01:07] <lindenle> JodaZ: oh, can you point me to a better one? [01:07] <lindenle> JodaZ: im totally new to libavcodec [01:10] <JodaZ> well, possibly the filtering_video.c in the same directory where you may ignore most of the filtering stuff [01:10] <JodaZ> i am new to this too :) [01:11] <lindenle> JodaZ: cool ill look thanks [01:18] <lindenle> Anyone else have some example fro decoding h264 stream from mp4 container? First question is do i need to pick the stream out somehow? [01:19] <lindenle> ok so i assume demux with avformat convert with avcodec mux with avformat? [02:41] <lindenle> does anyone know what the function img_convert has been renamed to? [02:41] <lindenle> oh no, looks like it is in swscale [03:05] <lindenle> Can i use the same context for muxing and demuxing or do i need two of them? [03:31] <iive> lindenle: two of them. [03:31] <lindenle> iive: thanks, that is what i figured [03:32] <iive> it's kind of hard to imagine how you could use single context for two completely different operations on two different files ;) [03:42] <JodaZ> is anyone here who knows his way around the code? [03:43] <JodaZ> i am getting [03:43] <JodaZ> [adts @ 0x2b57100] Encoder did not produce proper pts, making some up. [03:43] <JodaZ> in the middle of some encoding, and i think thats an error [03:45] <JodaZ> as in libavformat/mpegtsenc.c line 1253, a new packet is created, and has size and data copied from the old one, but pts/dts not, which it feeds back to av_write_frame, which results in that error [03:59] <lindenle> should i be using add_stream or av_new_stream in 2.1? I can't seem to find av_new_stream in the docs at: http://ffmpeg.org/doxygen/2.1/ [04:00] <lindenle> but on a mac with n2.1.3 i dont see add_stream in the header files... [04:01] <lindenle> ah avformat_new_stream [04:01] <lindenle> nm [04:27] <Snowleaksange> hrm @ libavdevice missing reference to sdl [04:28] <Snowleaksange> i didnt enable sdl! [04:28] <Snowleaksange> cant figure out how to disable either [04:29] <Snowleaksange> hmm possibly --disable-index=sdl [04:29] <Snowleaksange> err --disable-indev=sdl [04:30] <Snowleaksange> oh nope didnt match [04:31] <Snowleaksange> --disable-outdev=sdl ! [04:31] <Snowleaksange> if this works everyone will think im the compiler genius [04:49] <Snowleaksange> b00m [04:49] <Snowleaksange> i rule the earth w my intelligence [05:50] <Jack64> anyone home? :) [05:51] <Jack64> I got a question, I'm splitting a movie in a few parts (2 or 3), overlaying an image and then stitching them back together and sometimes I get a pause, like the stitching isn't done right.. how can I fix this? [05:53] <Jack64> anyone? [05:55] <Zeranoe> Jack64: !pb [05:56] <Zeranoe> Woops, paste your command+full output [05:56] <Jack64> there's no error but ok [05:56] <Jack64> ffmpeg -y -i /tmp/ram/$infile -t $intime0 -c copy $workingdir/smallfile0b.mp4 -ss $intime0 -c copy $workingdir/smallfile0e.mp4 </dev/null >/dev/null 2>/var/log/ffmpeg.log [05:56] <Jack64> ffmpeg -y -i $workingdir/smallfile0e.mp4 -t $intime1 -c copy $workingdir/smallfile1b.mp4 -ss $intime1 -c copy $workingdir/smallfile1e.mp4 </dev/null >/dev/null 2>/var/log/ffmpeg.log [05:57] <Jack64> ffmpeg -y -i $workingdir/smallfile1e.mp4 -t $intime2 -c copy $workingdir/smallfile2b.mp4 -ss $intime2 -c copy $workingdir/smallfile2e.mp4 </dev/null >/dev/null 2>/var/log/ffmpeg.log [05:57] <Jack64> ffmpeg -y -i $workingdir/smallfile2e.mp4 -t $intime3 -c copy $workingdir/smallfile3b.mp4 -ss $intime3 -c copy $workingdir/smallfile3e.mp4 </dev/null >/dev/null 2>/var/log/ffmpeg.log [05:57] <Jack64> 4 part split [05:57] <Jack64> 5 actually [05:57] <Jack64> sorry [05:58] <Zeranoe> Whats your stitch command [05:58] <Jack64> then I just process smallfile0b,smallfile1b,smallfile2b,smallfile3b and smallfile3e [05:58] <Jack64> and stitch them back together [05:58] <Jack64> with concat [05:58] <Jack64> 1 time out of 3 it's not doing it right [06:00] <Zeranoe> You mean you rerun the same command and get a different result? [06:00] <Jack64> no [06:00] <Jack64> I mean like when I concat those files I said, transitions between 1 and 2 and 3 are perfect and then 3 and 4 breaks for a split second then resumes [06:01] <Zeranoe> Jack64: Can you confirm that 3 and 4 don't have any leading gaps [06:01] <Jack64> yes [06:01] <Jack64> they're split [06:01] <Jack64> from the originaç [06:01] <Jack64> original* [06:02] <Jack64> and I even have a script that looks for the key frame so it can be split at exactly that point [06:02] <Jack64> actually I use ffprobe to get the closest keyframe to any mm:ss position [06:02] <Jack64> and use that as split point [06:03] <Jack64> so when I stitch it back together it should work [06:03] <Zeranoe> Jack64: Can you try just concating the 3 and 4? [06:03] <Jack64> yes, the problem persists [06:03] <Zeranoe> So it's those files, not the concat [06:03] <Jack64> but how, if those files are generated from the same input as the others? [06:03] <Zeranoe> What frame type is the first frame in 3 and 4 [06:04] <Jack64> how do I check that? [06:05] <Zeranoe> ffprobe -show_frames perhaps? Check if they are different than 1 and 2 [06:05] <Zeranoe> I'm shooting in the dark here by the way [06:05] <Jack64> well the question I'm really asking is whether key frames change when you overlay something and output a different video [06:06] <Jack64> cause that may be the problem [06:06] <Jack64> I can't test that easily because those are temp files generated in ram [06:06] <Jack64> and I have to run a long script to get to those [06:06] <Jack64> takes a while [06:06] <relaxed> save the overlay until you concat [06:06] <Jack64> can't [06:06] <Jack64> that's why I'm splitting in the first place [06:07] <Jack64> different overlays for each part [06:07] <Zeranoe> Why are you feeling null into ffmpeg btw [06:09] <Jack64> check documentation [06:09] <Jack64> it's cause I'm running it from PHP [06:09] <Jack64> actually bash script dinamically generated by PHP [06:09] <Jack64> kind of a mess really [06:10] <Zeranoe> feeding* [06:10] <Jack64> but I'm pretty close [06:10] <Jack64> just have to fix that damn break [17:47] <usrnix> Hi, using https://help.ubuntu.com/community/Webcam#FFmpeg to record from webcam, but when I hit ctrl+c it seems to cut off a big chunk of footage off the end; anything wrong with this command? [17:50] <usrnix> Figure out that it's actually skipping.. weird, I counted and I was fast and skipped about six numbers o.o is just a generic HP webcam [17:51] <usrnix> Soz for spam, just seems to be rubbish vpx, ogg was fine [18:27] <q_bit> Can someone confirm that a simple use like this will dump the data out of a prores file without filtering? I need to avoid all the color shifts, incorrect use of 601 etc on some input files I use. ffmpeg -i SMPTE-VideoRange-Bars.mov -t 1.0 -vcodec rawvideo raw.yuv I'm hoping that this output yuv file will be unfiltered or changed from the 709 based 422 that was put into the prores file. [18:58] <Styles_> Is it possible to truncate an audio file at a frame boundary? [21:02] <troulouliou_dev> hi can somabody explain me how av_read_frame works internally ? is it not possible to read only frame from a specified stream ? [21:02] <troulouliou_dev> it seems like i need to read all the frames and filter with pkt.stram_index == . [21:02] <Mavrik> mhm [21:03] <Mavrik> av_read_farme just calls out to the muxer for your format [21:03] <troulouliou_dev> Mavrik, so it can be a fram from any streams ? [21:03] <Mavrik> of course [21:03] <Mavrik> because that's how files are saved [21:03] <Mavrik> how do you expect to magically read file/stream while skipping data you haven't demuxed yet? [21:04] <troulouliou_dev> Mavrik i want a raw access to a frame to analyse it; the only way is to read all frames and discard it until i get a correct frame ? [21:05] <Mavrik> of course [21:05] <troulouliou_dev> Mavrik, but how does seek works then ? it reread everything ? [21:05] <Mavrik> how do you think you/ffmpeg would even know what is a "correct frame" if it didn't demux it? [21:05] <Mavrik> yes. [21:05] <Mavrik> troulouliou_dev, I suggest you look up on how the format you're reading is structured [21:06] <troulouliou_dev> Mavrik, for mkv for instance the clusters have some metadata i tought an index was built in memory [21:06] <Mavrik> troulouliou_dev, if you don't find the fact that you have to read the file to see what you have logical [21:06] <troulouliou_dev> Mavrik, no its ok i just didn't know that :) [21:06] <Mavrik> you don't have location for _each_ packet in that index :) [21:06] <Mavrik> you just have approximate location <-> time mapping [21:06] <troulouliou_dev> Mavrik, thanks at least it is confirmed :) [21:06] <Mavrik> you still have to read file and separate out streams and frames [21:06] <Mavrik> troulouliou_dev, demuxing is very light on cpu anyway :) [21:07] <Mavrik> it's just reading markers and sizes into memory [21:07] <Mavrik> troulouliou_dev, actually decoding the frames is expensive [21:07] <troulouliou_dev> Mavrik, but seek frame will read the first frame at a position for any stream ? i m somehow confused [21:08] <Mavrik> yes [21:08] <troulouliou_dev> Mavrik, ha sorry there is a stream index in it [21:08] <Mavrik> mhm :) [21:08] <Mavrik> troulouliou_dev, file is a sequence of frames with a stream index [21:08] <Mavrik> so when you tell it "jump to that byte" [21:08] <Mavrik> there's really nothing else it can do... it grabs first packet it encounters and demuxes it [21:09] <troulouliou_dev> Mavrik, but frames are stil somehiow consecutive i guess [21:09] <Mavrik> it cannot how file is stored - file can easily be VVVVAAAASAAAVVV or VAVAVAVAVASS or VAAAVAAAVAAA [21:09] <troulouliou_dev> so it is easier for demuxer to lpay a file [21:09] <Mavrik> most formats don't really force ordering of streams :) [21:11] <troulouliou_dev> Mavrik, there is a commadn then to set the position to the beginning of the file [21:11] <troulouliou_dev> whatever the first fram eis [21:11] <troulouliou_dev> the first frame is ? [21:11] <Mavrik> didn't really catch that question :) [21:11] <Mavrik> what command? [21:12] <troulouliou_dev> Mavrik, is there a function to reset the position to the first frame, in the state before the first av_read_frame [21:43] <Jack64> guys, can anyone tell me why in this script http://pastebin.com/n6emmH4P the concatenation of files overlaid3.mp4 and overlaid4.mp4 isn't in sync ? there seems to be about 1 second duplication of frames, but all others are concatenated seamlessly except those 2 [21:51] <Jack64> anyone? [22:05] <Jack64> guys, can anyone tell me why in this script http://pastebin.com/n6emmH4P the concatenation of files overlaid3.mp4 and overlaid4.mp4 isn't in sync ? there seems to be about 1 second duplication of frames, but all others are concatenated seamlessly except those 2 [22:05] <Jack64> please help me out, it's the only thing that's not working -.- [22:47] <Zeranoe> Jack64: At this point you might want to try the mailing list [22:49] <klaxa> yeah probably, i started looking at the script and i'm a little confused as to what it does [22:49] <Jack64> Zeranoe: is splitting and merging that tricky? come on -.- [22:49] <klaxa> maybe because it's simply too long? [22:49] <Jack64> I can explain what it does [22:49] <Mavrik> Jack64, yes, yes it is. [22:49] <klaxa> can you break it down to the command that doesn't work, and/or provide some sample files to generate the output? [22:49] <Mavrik> Jack64, especially with formats like mp4. [22:49] <Jack64> it's pretty straightforward, except the comments are in portuguese [22:50] <Jack64> the origin file is a MOV recording from an iPad [22:50] <Jack64> it's a part of a recording of an indoor soccer game [22:50] <Jack64> so I feed it the goal times [22:50] <klaxa> oh i see you are using -c copy [22:50] <Jack64> and it makes the scorebar overlay [22:50] <klaxa> maybe some i-frames are skewed [22:50] <klaxa> can you try re-encoding just as a test? [22:50] <klaxa> that might be an issue [22:51] <Jack64> I can try but it takes a while cause the script's pretty long [22:51] <klaxa> just encode to like mpeg1video with a low bitrate [22:51] <klaxa> but i think it's worth a shot [22:51] <Jack64> sure I can try that [22:51] <Jack64> what modifications would I make? instead of libx264 I use mpeg1video? [22:51] <klaxa> better than being puzzled and not knowing what to do [22:51] <klaxa> wait [22:52] <Jack64> yea I guess lo [22:52] <Jack64> lol* [22:52] <klaxa> wait a sec [22:52] <Jack64> ok.. [22:53] <klaxa> it will speed things if you replace libx264 with mpeg1video, the important part is to use a codec for the -t -ss segments, the ones where you use -c copy [22:53] <klaxa> replace the ones where you copy the codec by something that encodes stuff [22:53] <Jack64> why? [22:53] <Jack64> the only place where I copy is when I split the files [22:54] <Jack64> then I always need encoding because I'm changing the video track [22:54] <Jack64> then I use copy for the concat as well [22:54] <klaxa> yeah use encoding at the point where you split [22:54] <Jack64> those are the only copy calls [22:54] <klaxa> because that's where stuff could break shit because of copying the code [22:54] <klaxa> *codec [22:54] <Jack64> that'll make splitting a 20min video take some time instead of instantaneous [22:55] <Jack64> with copy it's instantaneous (cause it's a RAMDisk) [22:55] <Jack64> but sure, if it'll work, I don't care [22:55] <Jack64> what puzzles me is why some work and some don't, because the origin file is the same! [22:56] <Jack64> I split infile, then I take the last part of that split and split it again [22:56] <Jack64> and so on [22:56] <Jack64> so technically it's the same origin file [22:56] <Jack64> except just an excerpt [22:57] <klaxa> yes, but the beginning of a file you split has to be an I-Frame if you are copying the codec [22:57] <Jack64> OH! [22:57] <Jack64> I thought it had to be split by KeyFrame [22:57] <klaxa> as in, the first frame of the resulting split file has to start with an i-frame [22:57] <klaxa> yes an i-frame is a keyframe [22:58] <Jack64> oh [22:58] <Jack64> then it's already doing that [22:58] <Jack64> see the ms=xxxxx ? [22:58] <Jack64> at the top? [22:58] <Jack64> before the script is generated, php runs ffprobe to look for the nearest keyframe [22:58] <klaxa> ah [22:58] <Jack64> nearest to the input time (goal time) [22:58] <klaxa> hmm... [22:58] <klaxa> no idea then, sorry [22:58] <klaxa> gotta go afk [22:59] <Jack64> it's really weird [22:59] <Jack64> thanks for your help anyway klaxa [22:59] <Jack64> I might try that mpeg1video bit just to see if anything changes [23:00] Action: Jack64 calls on all hard-core ffmpeg'ers to conjure their wisdom in his aid [23:03] <Jack64> Zeranoe: no ideas? [23:13] <batence> hello there [23:13] <batence> I have a question, is somebody here? [23:13] <batence> :) [23:14] <another> just ask [23:14] <batence> good [23:16] <batence> "movie=logo.png, scale=200:37 [watermark]; [in][watermark] overlay=10:10 [out]" - how to change these pixels 200:37 in percentage, because when i convert video with big resolution ... watermark comes small, but when convert video with small resolution - watermark comes very big, so is there a solution to scale watermark in percentage of video resolution ? [23:18] <Jack64> batence: maybe you should have generate different watermarks with different resolutions to fit each type of video? [23:18] <Jack64> then just overlay the appropriate one? [23:18] <Jack64> seems to me the easiest solution [23:19] <Jack64> otherwise you'll have to stretch a small image (distorting output) or shrink a large one (compressing at CPU cost) [23:20] <batence> I just run video site [23:20] <batence> everything must me complete for users :) [23:20] <batence> *be [23:21] <PuffyBSD> hey im not sure whats going on , i have been trying to get ffmpeg to sync video and voice together [23:22] <PuffyBSD> i notice i get alot of frame drops too when doing a screencast if i launch a program or anything.. im not sure for openbsd s16le is good for the audio device as it works but its probably not the one.. i have no idea what openbsd uses or whatever name it is for the device for the mic [00:00] --- Sun Mar 9 2014

1 0

ffmpeg-devel.log.20140307
by burek 08 Mar '14

08 Mar '14

[00:03] <cone-884> ffmpeg.git 03Luca Barbato 07master:5b977c1d4fd9: mxf: Return meaningful errors [00:04] <cone-884> ffmpeg.git 03Michael Niedermayer 07master:39f5003a07db: Merge commit '5b977c1d4fd99f325049d48fb8f0a3240d50eef8' [00:15] <cone-884> ffmpeg.git 03Rainer Hochecker 07master:b66382101cff: dxva2: Increase maximum number of slices for mpeg2 [00:15] <cone-884> ffmpeg.git 03Michael Niedermayer 07master:b7c1f6f9cc75: Merge commit 'b66382101cff33e2ce66500327a90d0a105eedeb' [00:33] <cone-884> ffmpeg.git 03Derek Buitenhuis 07master:d67dbfa7c195: libx265: Support API version 9 [00:33] <cone-884> ffmpeg.git 03Michael Niedermayer 07master:0707a34dd817: Merge remote-tracking branch 'qatar/master' [00:46] <cone-884> ffmpeg.git 03Hendrik Leppkes 07master:276839b8de7f: dxva2_h264: set the correct ref frame index in the long slice struct [00:46] <cone-884> ffmpeg.git 03Hendrik Leppkes 07master:8b2a130d3fe6: dxva2_h264: add a workaround for old intel GPUs [00:49] <michaelni> nevcairiel, ok, thanks, applied [00:54] <cone-884> ffmpeg.git 03Michael Niedermayer 07master:64e4bd7414fc: doc/APIchanges: update versions & hashes [11:02] <nevcairiel> this locked trac database thing is stupid [11:05] <ubitux> and annoying [11:06] <nevcairiel> if its really a backup, someone should move it into EU sleepy times [11:07] <nevcairiel> and make it faster! [11:12] <wm4> obviously the server should just run a script that deletes the lock every other minute [11:13] <wm4> or just restart the server altogether to be sure [11:13] <nevcairiel> when a backup process locks the database, such a lock is somewhat meaningful [11:13] <nevcairiel> as long as it unlocks again once its done [11:13] <nevcairiel> and doesnt take half an hour to backup o.o [11:13] <wm4> oh I see, so the backup process is locking it [11:14] <nevcairiel> thats what someone claimed is happening [11:37] <nevcairiel> i'm starting to wonder if the theory isnt wrong [11:37] <nevcairiel> its still happening [12:08] <michaelni> nevcairiel, ubitux is that db locked thing still happening ? which url shows it ? [12:15] <anshul> michaelni,no lock is not now, it gives lock when i try to login [12:18] <ubitux> michaelni: you just need to try to login to trigger the lock when that happens [12:19] <nevcairiel> michaelni: seems its gone again [12:19] <nevcairiel> i just tried to open a issue link as spammed by the bot [12:20] <nevcairiel> (but i am logged in, so that may trigger it) [12:53] <cone-388> ffmpeg.git 03Patrice Clement 07master:93d216d37a3f: dict: K&R formatting cosmetics [12:53] <cone-388> ffmpeg.git 03Michael Niedermayer 07master:774d0c9dbaf5: Merge commit '93d216d37a3f95190ecb9d51cf72f54ea4e04ec7' [12:58] <cone-388> ffmpeg.git 03Martin Panter 07master:5b2ad78f97d4: rtmppkt: Handle extended timestamp field even for one-byte header [12:58] <cone-388> ffmpeg.git 03Michael Niedermayer 07master:d540e7577a4c: Merge commit '5b2ad78f97d43299adcb038c04346999fe9b196c' [13:10] <cone-388> ffmpeg.git 03Martin Panter 07master:48cfb5f41140: rtmppkt: Add a descriptive comment [13:10] <cone-388> ffmpeg.git 03Michael Niedermayer 07master:94ec27131c7a: Merge commit '48cfb5f4114048383a4ee85b2f500eefce24cad2' [13:49] <cone-388> ffmpeg.git 03Martin Panter 07master:8bbb02ae4f44: rtmppkt: Properly handle extended timestamps when writing [13:49] <cone-388> ffmpeg.git 03Michael Niedermayer 07master:9baef60ae400: Merge commit '8bbb02ae4f44f1cddc1ce1e74a71fa2022e93da3' [14:14] <cone-388> ffmpeg.git 03Martin Storsjö 07master:77eed91fab9b: rtmppkt: Rename the ts_delta field to ts_field [14:14] <cone-388> ffmpeg.git 03Michael Niedermayer 07master:4fc339faea11: Merge remote-tracking branch 'qatar/master' [14:30] <plepere> BBB : if punpcklbw/punpckhbw is slower than movhlps+pmovzxbw, should I do 2 different versions of the functions ? [14:30] <plepere> one for SSSE3, one for SSE4 [15:01] <plepere> actually, the punpck is faster. :P [15:16] <Skyler_> plepere: in my experience pmov* is basically useless except for sign-extension [15:16] <plepere> ok [15:17] <Skyler_> you'd think it would be faster in some case, but often it's reliably slower [15:18] <Skyler_> (I really only have wild guesses as to why) [15:24] <ubitux> i've written 3 lines of arm asm [15:24] <ubitux> and already triggered an objdump bug [15:24] <ubitux> awesome. [15:24] <ubitux> (and same in gdb ofc) [15:24] <ubitux> "all versions of objdump that I have seem to disassemble 'vshll.u16' instruction incorrectly" [15:24] <ubitux> great.. [15:25] <Skyler_> what do they disassemble it as? [15:26] <ubitux> stuff like "vshll.u16 q6, d26, #6" becomes "vshll.u16 d12, q13, #6" [15:26] <Skyler_> oh, geez, so they just get the registers wrong completely [15:26] <ubitux> yeah they swapped wrongly [15:27] <Skyler_> though isn't d26 == q13 and q6 == d12? [15:27] <ubitux> d12/d13 are q6 and d26/d27 are q13 [15:27] <Skyler_> oh, I see, you mean it displays wrong because it should show {Q, D} not {D, Q} [15:27] <ubitux> well it displays wrongly but the "wrong" version makes no sense and isn't assemblable [15:28] <ubitux> q6 is d12+d13, and d26+d27 are q13 [15:32] <ubitux> - val = get_xbits(gb, nbits); [15:32] <ubitux> + val = get_xbits(gb, nbits); [15:32] <ubitux> lol [15:32] <ubitux> so much better [15:39] <plepere> it's all in the indentation [15:50] <plepere> isn't it possible to assign a string in ASM preprocessor ? [15:50] <plepere> %assign %%lod movd [15:50] <plepere> for instance [15:55] <Skyler_> %define is what you use for that, not %assign, I think [15:55] <Skyler_> %assign is for variables [15:57] <plepere> ok [15:57] <plepere> thanks [15:58] <plepere> works ! [15:58] <plepere> :D [16:08] <cone-388> ffmpeg.git 03Hendrik Leppkes 07master:930f67b71294: dxva2_vc1: fix intensity compensation condition with interlaced fields [16:08] <cone-388> ffmpeg.git 03Hendrik Leppkes 07master:61ff0431ba8e: dxva2_vc1: include the start code in wMBbitOffset [17:47] <ubitux> BBB: i'll probably start doing the vp9 mc for arm, since i don't know how they work [17:47] <ubitux> any suggestion or whatever information about those before i start? [19:58] <troulouliou_dev> http://bitbin.it/Ky30I5LP [19:58] <troulouliou_dev> a small bug correction patch for the demux example [20:00] <llogan> troulouliou_dev: patches should be sent to ffmpeg-devel@ffmpeg.org...otherwise we tend to forget [20:00] <troulouliou_dev> llogan, ok [20:01] <llogan> troulouliou_dev: and please use git format-patch, or git send-email if you can [20:47] <wm4> wow requiring ML interaction for such trivial patches [20:55] <Compn> i could have sent it to the list or committed it [20:55] <Compn> i just wasnt here at the moment wm4 :P [20:57] <Compn> i guess i can commit it now [20:57] <Compn> good time to install git i think [20:58] <Compn> see how long this takes [21:11] <Compn> ok cloning the websites [21:11] <Compn> now how to developer access the actual repo [21:14] <Compn> anyone remember pls pm me :) [21:14] <llogan> websites? [21:14] <Compn> ffmpeg-web and rtmpdump-web [21:14] <llogan> for that trivial patch? [21:14] <Compn> no [21:14] <Compn> just because i had to reinstall git [22:40] <Compn> michaelni : you there to help me with dumb git question ? [22:43] <michaelni> Compn, maybe ;) [22:44] <Compn> michaelni : do you remember how to set git up so ican commit to ffmpeg ? [22:44] <Compn> i have a fresh install and forgot the steps [22:46] <michaelni> maybe http://www.ffmpeg.org/git-howto.html [22:46] <michaelni> and if not then it should be fixed [22:46] <Compn> is the developer repo git url the same as the read git repo ? [22:46] <Compn> :) [22:47] <michaelni> push urls differ from anon read urls [22:47] <michaelni> normally [22:47] <Compn> ok , wheres the push url ? [22:49] <michaelni> it should be in the git howto and maybe download page but its possibly not [22:49] <michaelni> for web its gil@ffmpeg.org:ffmpeg-web [22:49] <michaelni> feel free to add it [22:50] <michaelni> if its missig [22:50] <Compn> michaelni : i have forgotten the url, whats in your .git/config ? [22:51] <michaelni> for web: url = gil@ffmpeg.org:ffmpeg-web [22:53] <Compn> not for web [22:53] <Compn> i know web [22:53] <Compn> i mean ffmpeg itself [22:56] <michaelni> githowto says: git clone git@source.ffmpeg.org:ffmpeg [23:03] <Compn> ehe [23:06] <Compn> heh [23:06] <Compn> git clone git://source.ffmpeg.org/ffmpeg is what the git-howto says [23:10] <Compn> oh theres two [23:10] <Compn> humm [23:12] <wm4> michaelni: are you sure you want him to push _anything_ to the git repo [23:14] <michaelni> Compn, what do you want to push to the repo ? [23:14] <Compn> michaelni : just that patch fixing typo in the doc [23:16] <michaelni> you can also send me a pull req or give me a url of that git patch. [23:16] <michaelni> if you push directly make sure you dont make a mistake [23:17] <michaelni> always do a "git push origin master --dry-run" before actual push [23:17] <Compn> its on ffmpeg-devel already :P [23:17] <Compn> the patch [23:18] <beastd> doesn't mean you can't fuck up the push [23:18] <nevcairiel> in some crazy oddball patch format [23:18] <Compn> i know [23:18] <beastd> look at the dryrun output closely [23:18] <Compn> ive committed to ffmpeg before, and followed dry run instructions beastd :) [23:18] <Compn> just cant remember how i setup last time [23:20] <beastd> just put the url there michaelni mentioned, i cannot think of problems besides ssh key auth [23:20] <Compn> i'm guessing its ssh key auth problem [23:22] <cone-388> ffmpeg.git 03compn 07master:84bccae8e9ac: doc: fix typo [23:22] <Compn> pfft [23:22] <Compn> i guess i had to restart my git program for it to work [23:22] <Compn> thanks guys :) [23:24] <wm4> ubitux: I should have read that post more closely, I thought he wanted to _use_ libavfilter [23:25] <Compn> wm4 : you doubted my skills? :P [23:25] <beastd> Compn: Do you have the priv key for the corresponding pub key you gave for ffmpeg.git repo on that machine at a location where ssh will find it? [23:25] <Compn> beastd : ^^ i just committed, thanks for helping. :) [23:25] <Compn> and no , i dont know what was wrong. [23:26] <beastd> Compn: Oh ok. That commit came so fast that I doubted it was you who pushed it ^^; [23:26] <Compn> or how i fixed it [23:26] <beastd> That always leaves an uneasy feeling IMHO [23:26] <Compn> url = git@git.videolan.org:ffmpeg.git [23:27] <Compn> is whats in my .git/config [23:27] <beastd> url = git@source.ffmpeg.org:ffmpeg # that is what i have, and i never had any problems with push [23:37] <Compn> i dont even want to know what kind of voodoo is going on with the two git urls :) [23:53] <Compn> michaelni : i guess i dont have any wisdom to add to the git-howto , only 'if its not working, try turning it off and on again' :) [00:00] --- Sat Mar 8 2014

1 0

ffmpeg.log.20140307
by burek 08 Mar '14

08 Mar '14

[10:58] <aterra> Hi I'm trying to stream a v4l2 device using ffserver, but I'm having a error in the codec configuration (libx264) which I can't figure out, when I use ffmpeg from the command line to encode into a file there is no issues, however when using ffserver I get "error while opening encoder for output stream #0:0", I have pasted a minimum working example of the stream part of the ffserver configuration file as well as the ffmpeg command on http://pastebin.com/tHW [10:58] <aterra> gBZZs [10:58] <aterra> http://pastebin.com/tHWgBZZs [10:58] <steinchen> hi all.. may someone provide me a cmdline to convert hd 1080p into 576i? when i try any setting we have distortion in pans [11:04] <steinchen> interlaced must be TFF in mpeg2 9mbit HQ [11:10] <sine9> How does one use mixed frame rate clips correctly? my resulting framerate will be 25 and I shoot with 25, but when bringing in clips that are 29 and 23 etc, whats the best way to handle this. I am using Ae but ffmpeg is in my toolbox, just havent needed to use it yet [11:50] <wix> Hello, I'm currently using ffmpeg to transcode and send a video stream to youtube (rtmp/flv/x264). I would like to know if there is a way to tell ffmpeg to exit when it's not sending data anymore (displayed size is not growing anymore). Thanks for your help ! [11:55] <relaxed> wix: write a script that checks the size from time to time, which then run $(killall -INT ffmpeg) when the size no longer increases. [11:56] <relaxed> sine9: ffmpeg will drop or duplicate frames to achieve the desired frame rate. [11:56] <wix> ok thanks relaxed, so there is no other way [11:57] <relaxed> you can set -t $time.in.seconds if you know the output duration before hand [12:13] <sine9> relaxed ok so i can take a clip that is 5 seconds at 29 fps and set the time for the same but with 25 fps and it will do the rest [12:15] <relaxed> sine9: yes, you may want to do it lossless: ffmpeg -i 29fps.input.mkv -r 25 -c:a flac -c:v libx264 -crf 0 output.mkv [12:17] <relaxed> lossless except for the dupped/dropped frames, that is ;) [12:51] <sine9> thanks [12:53] <sine9> so flac for the audio lossless [12:53] <sine9> but x264 with 0 compression ? [12:55] <sacarasc> x264 with crf 0 is lossless. Unless you have to change pixel format. [12:55] <sacarasc> Or change resolution, etc. [12:56] <sine9> cool thanks [15:05] <flufx> Cannot compile vf_noise.c from git here: clang says "error: index register is 32-bit, but base register is 64-bit" [15:46] <flufx> Thank you! [15:48] <sacarasc> Sorry, that was for me for a different channel. :D [15:49] <flufx> lol, that was exacly what I was looking for! [15:49] <sacarasc> Heh. [17:20] <Guiri> I noticed that x264 can use opencl if --opencl is passed as a runtime argument. Is this argument automatically passed or would I have to use --x264-params '--opencl'? [17:21] <Guiri> With regard to the libx264 encoder, that is [18:24] <Wzee> Hello! I hope someone might shed some light on this... I'm attempting to run ffmpeg to do HLS input at 29.97 frames, with image2 output, the issue being that multiple duplicate images are created on connection, anywhere from one dup, to ~80 frames. Could this be because it's connecting mid stream with no iframe? [18:40] <Trasp> Okay, I'm trying mix two audio files into one, not so they're played after eachother, but at the same time. What is the command I should look up? Stupid question maybe, but still.. [18:49] <Trasp> solved [18:51] <Wzee> Same here! -vsync 0 skips the duplicated frames in the output from image2. [19:15] <troulouliou_dev> hi why do i have an empty codec_name in AVCodecContext while codec_id is set and correct ? [19:37] <JodaZ> wtf °.° [19:59] <troulouliou_dev> http://bitbin.it/Ky30I5LP [19:59] <troulouliou_dev> <troulouliou_dev> a small bug correction patch for the demux example [21:49] <Mista_D> can't use "-ss 275 -i ..." for some reason, -ss 276 and -ss 274 work fine. Error is: "ffmpeg: libavformat/utils.c:1828: ff_gen_search: Assertion `pos_limit <= pos_max' failed." Core dumped... any work of advice? [21:49] <Mista_D> word of advice (: [21:52] <rjp421> which os and build version? [23:20] <wodim> hello, i created a new .mkv file by copying the video stream of a file and the audio stream of another file. both files have the same length, but the audio starts desynchronising as the movie goes on. i suspect it's because the fps in both source movies are different. any hints? [23:38] <lindenle> How do i link to libffmpeg.so in mac? [23:39] <lindenle> or is it called dylib? [23:42] <JEEB> FFmpeg contains no libffmpeg [23:42] <lindenle> JEEB: oh so i just link to the differen libs lik libavcodec.dylib [23:42] <lindenle> ? [23:42] <JEEB> so go look at whomever provides that library and then go poke them :) [23:43] <JEEB> did you even check what libraries you have? :P [23:43] <lindenle> JEEB: yes [23:43] <lindenle> JEEB: i was looking for libffmpeg... my bad [23:43] <JEEB> well then you should know if you have libffmpeg or something else [00:00] --- Sat Mar 8 2014

1 0

ffmpeg-devel.log.20140306
by burek 07 Mar '14

07 Mar '14

[02:02] <cone-739> ffmpeg.git 03wm4 07master:636273d3d4a8: http: handle ICY in presence of chunked transfer encoding [02:09] <cone-739> ffmpeg.git 03Lukasz Marek 07master:f495fbe76a26: lavf/avio: fix ffurl_alloc error checks [02:09] <cone-739> ffmpeg.git 03Lukasz Marek 07master:4ba6a534dc94: lavf/http: return error on seeking to negative postion [02:09] <cone-739> ffmpeg.git 03Lukasz Marek 07master:1aa262f460c0: lavf/http: return error from seek on invalid whence [02:09] <cone-739> ffmpeg.git 03Lukasz Marek 07master:2475fdbd047d: lavd/avdevice: always free detected devices on error [02:09] <cone-739> ffmpeg.git 03Michael Niedermayer 07master:c832bf0c38a3: Merge remote-tracking branch 'lukaszmluki/master' [02:11] <michaelni> nevcairiel, should i apply the 2 dxva2_h264 patches or is there something to wait for ? ( the discussion doesnt clearly say apply it nor dont apply it) [03:37] <cone-739> ffmpeg.git 03Lukasz Marek 07master:bba7b6fc4105: Revert "lavu/buffer: add release function" [04:41] <Zeranoe> Looks like the downmix/loss of LFE is getting more attention: http://ffmpeg.zeranoe.com/forum/viewtopic.php?t=1851 [08:19] <saigono> nevcairiel: Thank you! Your advice was very helpful :) [10:05] <cone-884> ffmpeg.git 03Michael Niedermayer 07release/2.1:d87ac93bcad3: avformat/oggparsevorbis: dont use invalid granules [10:05] <cone-884> ffmpeg.git 03Michael Niedermayer 07release/2.2:124c78fd4449: avformat/oggparsevorbis: dont use invalid granules [12:48] <cone-884> ffmpeg.git 03Anton Khirnov 07master:bba2a7cc5f7c: lavfi: use the correct filter context for logging an error. [12:48] <cone-884> ffmpeg.git 03Michael Niedermayer 07master:b76059058aa6: Merge commit 'bba2a7cc5f7c7aaa32a938f3d4edd9f555f39cdb' [13:08] <cone-884> ffmpeg.git 03Anton Khirnov 07master:599b81ca9a8e: lavfi: add shuffleplanes filter [13:08] <cone-884> ffmpeg.git 03Michael Niedermayer 07master:2fb0027c267d: Merge commit '599b81ca9a8e04a27ddad94af462171d16063167' [13:23] <plepere> hmm. having trouble on a simple refactor in ASM : if I replace "r12q" by "rfilterq", I have an "invalid combination of opcode and operands" error. does this ring a bell to anybody ? [13:24] <plepere> and this error comes on mov and sub instructions. :/ [14:08] <cone-884> ffmpeg.git 03Anton Khirnov 07master:e1f2987b1011: FATE: add tests for the shuffleplanes filter [14:08] <cone-884> ffmpeg.git 03Michael Niedermayer 07master:7011aab5018a: Merge commit 'e1f2987b10112489f6da5501d4c8735a798c9e3f' [15:00] <cone-884> ffmpeg.git 03Diego Biurrun 07master:3bfdee00cd92: x86: dcadsp: Fix linking with yasm and optimizations disabled [15:00] <cone-884> ffmpeg.git 03Michael Niedermayer 07master:2d99de66b7f1: Merge commit '3bfdee00cd92ff07c364d4901c4aefda32780756' [15:18] <cone-884> ffmpeg.git 03Anton Khirnov 07master:713d3f98c8b0: vf_shuffleplanes: fix the type of the mapping indices [15:18] <cone-884> ffmpeg.git 03Michael Niedermayer 07master:d168729004a9: Merge remote-tracking branch 'qatar/master' [15:20] <j-b> 1m [15:20] <j-b> oops [16:00] <plepere> BBB : should I expand all the macros in the functions ? [16:05] <BBB> plepere: for what? [16:05] <BBB> I don't think so, just take 1-2 functions and disassemble the result and see if that's what you think it should be (regardless of the macros) - i.e. "can I make this faster" [16:05] <plepere> I see [16:06] <BBB> macros should do no more than make your life (and thus the final binary) better [16:06] <BBB> I just want to make sure the macros you're using indeed serve that purpose, think of it as a socratic experiment, nothing more [16:07] <plepere> a socratic experiment ? [16:08] <plepere> oh and while I'm talking to you : I can't do the pmullw that you're suggesting. I need to do a pmulhw +pmullw + unpack to full 32bit + pack back to 16bit. [16:10] <plepere> basically, I can't stop at pmullw. [16:40] <plepere> ok, I've sent my patch. Again. [16:42] <plepere> BBB : I know that the fullpel isn't good for you as it is, but that's how it's done in openHEVC, and I'd rather validate things here, merge the ASM back into OpenHEVC and then do the weighting (or qpel_hv, but it will be tricky) [16:51] <ubitux> plepere: please remove all those "//////////" [17:07] <cone-884> ffmpeg.git 03James Almer 07master:99b4da73c8fd: lavd/Makefile: Add fbdev_common.h to SKIPHEADERS [17:07] <cone-884> ffmpeg.git 03Michael Niedermayer 07master:ffe7e7c195c3: avfilter/vf_shuffleplanes: Fix flags / add AV_OPT_FLAG_FILTERING_PARAM [17:20] <cone-884> ffmpeg.git 03sfan5 07master:69ead9af7519: libx265: Use ctx->vui. instead of ctx-> for some options A recent change in libx265 moved some options such as sar_width into a 'vui' struct. [19:15] <BBB> plepere: wait why 32bit? [19:15] <BBB> plepere: for 8bit, which part requires 32bit? [19:17] <BBB> and I'll review patch "soon", hopefully finish this weekend [20:01] <Zeranoe> Any comments on the downmix/LFE loss topic: http://ffmpeg.zeranoe.com/forum/viewtopic.php?f=17&t=1851 [20:03] <llogan> i don't see an actual question in that thread [20:05] <Zeranoe> llogan: I think hes suggesting to make -ac 2 automatically downmix with those options [20:07] <llogan> submitting a patch would be the best way for comments [20:36] <cone-884> ffmpeg.git 03James Almer 07master:9e0e1f906743: x86/dsputil: add emms to ff_scalarproduct_int16_mmxext() [20:59] <burek> I guess you've all heard about this, but it's nice to hear it again :) http://letterboxd.com/blots/film/the-fifth-estate/ [20:59] <burek> Favourite line: We can use FFmpeg and throw it into Final Cut. [20:59] <burek> :) [21:01] <burek> oh.. this "news" is 2 months old :/ https://twitter.com/FFmpeg/status/422566363467505664 [21:02] <burek> anyway +beer :) [21:41] <ubitux> "dict: K&R cosmetic refactoring" [21:41] <ubitux> lol [21:42] <kierank> where did all these cosmetic people come from [21:42] <kierank> i've seen 2 in the last few days [21:42] <ubitux> i don't know :') [21:46] <llogan> "maybe they were born with it. maybe it's maybeline." [21:46] <wm4> maybe someone is trolling diego? [21:47] <wm4> it's the only logical explanation I can come up with [21:49] <ubitux> well, they have 3 guys full time on cosmetics, it seems they're going to have 2 new guys [21:49] <ubitux> ready for http://cdn.memegenerator.net/instances/500x/46616414.jpg @_@ [21:49] <ubitux> michaelni: i started checking the hashes & stuff, 'will confirm the APIChanges tonight [21:50] <kierank> ubitux: 3 guys? [21:51] <ubitux> diego/luca/koda [21:51] <kierank> ah koda [21:54] <llogan> the term is "beautician" [21:54] <llogan> One skilled in giving cosmetic treatments. [21:55] <wm4> ubitux: koda seems to make functional cleanups, though [21:55] <ubitux> lol [21:55] <cone-884> ffmpeg.git 03Andreas Cadhalpun 07master:64b6164b721c: Correct the FSF address for two avisynth files to '51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA' [21:55] <cone-884> ffmpeg.git 03Andreas Cadhalpun 07master:eeb3baf7f725: Fix spelling error 'Inconsistant -> Inconsistent' [21:55] <cone-884> ffmpeg.git 03Andreas Cadhalpun 07master:9898bd9a82ad: Fix spelling errors in texi files: accomodate -> accommodate allows to -> allows one to choosen -> chosen compability -> compatibility explictly -> explicitly overriden -> overridden specifed -> specified Trasmission -> Transmission [21:56] <wm4> ubitux: and only diego apparently does nothing but cosmetics [21:56] <ubitux> if you want to be honest, he does some stuff to the build system sometimes [22:04] <cone-884> ffmpeg.git 03Diego Biurrun 07master:84bf88172061: configure: Split x86 SIMD architecture extensions into separate list [22:04] <cone-884> ffmpeg.git 03Michael Niedermayer 07master:85d2b0987bae: Merge commit '84bf8817206126dab3c9abf6055b593389bcb241' [22:12] <cone-884> ffmpeg.git 03Diego Biurrun 07master:d48430c36794: build: Let the SVQ3 decoder depend on the H.264 decoder [22:12] <cone-884> ffmpeg.git 03Michael Niedermayer 07master:1618f162a9da: Merge commit 'd48430c367947a64647c6959cf472f2c01778b17' [22:28] <cone-884> ffmpeg.git 03Matthieu Bouron 07master:e118bb1a3388: mxf: Introduce ff_mxf_get_samples_per_frame [22:28] <cone-884> ffmpeg.git 03Michael Niedermayer 07master:2265212396c1: Merge commit 'e118bb1a33889d4df56f28975b4fd0793b4f5c32' [22:31] <michaelni> nevcairiel, ok if i apply the 2 dxva2_h264 patches or should i wait ? [22:31] <michaelni> you are the author of the patches [22:32] <JEEBsv> t [22:57] <cone-884> ffmpeg.git 03Matthieu Bouron 07master:5b930092c3af: mxf: Set audio packets pts [22:57] <cone-884> ffmpeg.git 03Michael Niedermayer 07master:3ce858141309: Merge commit '5b930092c3afd2ae01f1c8aa7fb405911d6ad416' [23:14] <cone-884> ffmpeg.git 03Matthieu Bouron 07master:570af382eea9: mxf: Handle identification metadata [23:14] <cone-884> ffmpeg.git 03Michael Niedermayer 07master:5a4852bc4d64: Merge commit '570af382eea902afe09f3562e5e1b483981cca7e' [23:33] <nevcairiel> michaelni: if noone complains that old GPUs suddenly fail at decoding unless the calling app sets the workaround. IMHO the benefit of newer GPUs starting to work is bigger [23:39] <Compn> nevcairiel : what old cards fail ? [23:40] <nevcairiel> any intel GPUs before Sandy Bridge, and possibly Sandy Bridge GPUs if the calling application is a bit stupid :) [23:40] <Compn> as long as vlc has the workaround , said patches are ok to me :P [23:40] <Compn> only intel gpu then? not nvidia or ati stuff ? [23:40] <nevcairiel> (since SNB supports both way) [23:40] <nevcairiel> no, only intel [23:40] <Compn> oh, yes then i'd say apply [23:40] <nevcairiel> the work around is like two lines, one if and one assignment [23:41] <Compn> scared me i thought you wanted to break nvidia :P [23:41] <Compn> ehe [23:50] <cone-884> ffmpeg.git 03Paul B Mahol 07master:f06f6daaf853: mxf: Parse random index pack [23:50] <cone-884> ffmpeg.git 03Michael Niedermayer 07master:618d2262d71e: Merge commit 'f06f6daaf8538eb8ceeb690b761f1256771b6ba6' [00:00] --- Fri Mar 7 2014

1 0

ffmpeg.log.20140306
by burek 07 Mar '14

07 Mar '14

[00:17] <jangle> ok, I get source listings for ffprobe in gdb, but not accurate listings for ffplay.... [00:20] <jangle> when I put a breakpoint on main when running in gdb, the source listing given is for the yuv2rgb.c source file&. [00:21] <klaxa> the mp4 container supports mp3 according to wikipedia btw [00:25] <JodaZ> hmm, isn't the inuilt aac codec pretty shit? [00:25] <sybariten> thanks llogan , that command line solved my task! [00:36] <jangle> for future reference, my issue is caused by the fact that sdl is used in ffplay, so a bp needs to be set on SDL_main in order to stop at the "main" defined in ffplay.c [00:36] <jangle> thanks all [00:36] <jangle> wc [03:05] <rhuang> Hello all, I am using ffmpeg's null filter in the following command: ./ffmpeg -i Megamind.avi -vf null Megamind_null.avi [03:05] <rhuang> and I am wondering why it still modifies the audio output? [03:05] <rhuang> *video [03:07] <rhuang> According to the documentation, http://ffmpeg.org/ffmpeg-filters.html#null, it just "Pass the video source unchanged to the output.". [03:09] <llogan> rhuang: that's a video filter [03:10] <llogan> what ar you trying to do? why are you using null? [03:10] <rhuang> I made a type, I meant to say it modified the video. [03:10] <rhuang> *typo [03:11] <rhuang> I want to pass the video unchanged from the source to target. I am actually developing my own videofilter, I started with the null filter as a base and add my code to it, so I am playing with it to make sure I understand what it does. [03:26] <iive> rhuang: try looking up -vcodec copy or whatever the current syntax is. [03:52] <rhuang> Thanks [04:11] <Dark-knight> ducks ducks ducks ducks [04:17] <amigojapan> Hi, I concatenated this video http://www.youtube.com/watch?v=xln-sGv5W1Q using this command d:\ffmpeg\bin\ffmpeg.exe -f concat -i mylist.txt -c copy o5.avi as you can [04:17] <amigojapan> see the mouths are out of sync with the soud, eventhought the original videos have the mouth synced... I can tell you that the sound files that where originally mixed [04:17] <amigojapan> into the videos has slightly shorter sound files than the video.... it seems like these sound files were stuck together ignoring the timming of the videos... in the [04:17] <amigojapan> original videos, this is most noticible in the last video, where the whisper voice mouth is totally out of sync in the concatenated video, but it is in sync in the [04:17] <amigojapan> original,here are links to the original videos http://pastebin.com/4tAvMRqQ [04:22] <relaxed> amigojapan: does it happen at the very beginning? [04:23] <amigojapan> relaxed: what do you mean? at the beginning the sound seems to be in sync, it seems to go out of sync each video I concatenate [04:23] <relaxed> add -async 1 and see if that fixes it [04:24] <amigojapan> thanlx relaxed , I will try it out [04:29] <amigojapan> relaxed: it made no difference, but I did notice this message in ffmpeg's output, soudns related: [concat @ 026a3f00] Estimating duration from bitrate, this may be inaccurate [05:10] <amigojapan> ok relaxed , I fixed it, I had to specify the duration of the animations in the text file that it uses for concatenating the videos [05:19] <JodaZ> :/ can i use a different timebase than 1/90k in mpegts or is that fixed? i want to reuse pts from my input so i don't need to do any rounding or anything [12:17] <Stormer_> Hello, i know this might be noobish, but can someone tell how i compile ffmpeg to my ubuntu server? maybe someone have the codes ready.. i tried the Ubuntu Compile guide.. i get program "ffmpeg" is currently not installed [12:18] <Stormer_> though after copy and paste each section [12:18] <Stormer_> i did install ffmpeg as in the last section [12:19] <JEEB> did you make sure that your new ffmpeg binary is a) in the PATH b) if yes, did you once try to close the terminal and then re-open it [12:20] <JEEB> although b) wouldn't generally be needed in such a case [12:20] <JEEB> because you seemingly didn't have the binary at all [12:21] <Stormer_> going to ~/bin [12:21] <Stormer_> i do see fffmpeg folder there [12:21] <JEEB> that's not the question [12:22] <JEEB> the question is if your installed ffmpeg binary itself if is in the active PATH [12:22] <JEEB> echo $PATH to show PATH [12:24] <JEEB> (otherwise just call your ffmpeg binary with the full path to it) [12:25] <Stormer_> how i do that [12:25] <JEEB> what is the --prefix you set when you configured? [12:26] <JEEB> you can check the config.log file that's written where you ran configure as well for it [12:26] <Stormer_> i followed this, so you have all the --prefix over there [12:27] <Stormer_> https://trac.ffmpeg.org/wiki/UbuntuCompilationGuide#libopus [12:27] <JEEB> no, I want yours exactly [12:27] <JEEB> of the ffmpeg itself [12:28] <JEEB> your last configuration line is at the top of the config.log file [12:28] <JEEB> as well as probably in your terminal's history [12:29] <Stormer_> http://pastebin.com/2Cz3kk3V [12:29] <Stormer_> this is what i found in ffmpeg/config.log [12:29] <JEEB> thankfully the prefix was the first setting [12:29] <JEEB> --prefix=/home/enc3/ffmpeg_build [12:30] <JEEB> you can thus call the binary with /home/enc3/ffmpeg_build/ffmpeg if you have installed it [12:30] <JEEB> that's the full path to the binary [12:30] <JEEB> you can also check PATH (echo $PATH in terminal) to see if that directory is in your PATH [12:30] <Stormer_> in enc/ffmpeg_build/ cant find ffmpeg [12:31] <Stormer_> enc3* [12:31] <JEEB> ugh [12:31] <JEEB> so it uses a separate bindir? [12:31] <JEEB> basically scroll the first line of config.log [12:31] <Stormer_> i dunno, what it did use using that trial [12:31] <JEEB> or wait [12:31] <Stormer_> haha [12:32] <JEEB> first check /home/enc3/ffmpeg_build/bin/ffmpeg [12:32] <JEEB> if that's not there [12:32] <JEEB> then go through that first line that's cut [12:32] <JEEB> further right that is [12:32] <Stormer_> only x264 inside bin [12:32] <JEEB> ok, then go further right-wise on that first line of config.log [12:32] <JEEB> because the line doesn't end there [12:33] <Stormer_> the full path for ffmpeg is [12:33] <Stormer_> enc3/bin/ffmpeg [12:34] <JEEB> ok, then you should be able to call it with the full path to it, /home/enc3/bin/ffmpeg [12:35] <Stormer_> so you mean /home/enc3/bin/ffmpeg -i ... [12:35] <Stormer_> ? [12:35] <JEEB> well just try running it first :P [12:35] <JEEB> see if you get version output etc. [12:35] <Stormer_> this is the rest of the line btw [12:35] <Stormer_> http://pastebin.com/PiX3Hbba [12:35] <JEEB> yeah, it sets a separate bindir [12:36] <Stormer_> its a directory it says [12:36] <JEEB> wat [12:36] <Stormer_> yeah what you mean by running it? [12:36] <JEEB> --bindir=/home/enc3/bin means that there should be a ffmpeg binary there :P [12:37] <JEEB> not a directory [12:37] <Stormer_> ffmpeg ffmpeg-snapshot.tar.bz2 vsyasm x264 yasm ytasm [12:37] <Stormer_> everything inside enc3/bing [12:37] <Stormer_> bin* [12:37] <JEEB> wait [12:37] <JEEB> that looks normal [12:38] <JEEB> are you sure the ffmpeg there is a directory? [12:38] <Stormer_> yeah and ffmpeg is bluish cant run [12:38] <Stormer_> says directory [12:38] <JEEB> what says? [12:38] <Stormer_> yeah [12:38] <Stormer_> i try [12:38] <Stormer_> ./ [12:38] <Stormer_> it has blue color [12:38] <Stormer_> i can go inside it [12:38] <Stormer_> there are config.log etc. [12:38] <JEEB> uhh [12:38] <JEEB> what the hell happened there :P [12:39] <Stormer_> configure is a binary inside [12:39] <Stormer_> haha dude im telling ya [12:39] <Stormer_> i dunno haha [12:39] <JEEB> do you have a symlink to your ffmpeg source code in there? [12:39] <JEEB> try removing that and re-running make install where you ran ffmpeg's configure [12:39] <Stormer_> i never ran ffmpeg configure [12:40] <JEEB> then what was that config.log? [12:40] <Stormer_> it has many files inside [12:40] <JEEB> you are saying one thing and seemingly have done another. Sync your both sides [12:40] <Stormer_> and found that config log inside the ffmpeg dir [12:40] <JEEB> then you did not build ffmpeg at all, am I correct? [12:40] <Stormer_> srsly never ran configure [12:41] <Stormer_> after installing the other thing [12:41] <Stormer_> i lastly did this one [12:41] <Stormer_> http://pastebin.com/Hb5PTmfq [12:41] <JEEB> you're clearly running ffmpeg's configure there [12:42] <Stormer_> ye i did run it btw [12:42] <Stormer_> oh true [12:42] <Stormer_> the comman did run configure.. [12:42] <Stormer_> my bad.. [12:42] <Stormer_> just checked the code i pasted in [12:42] <Stormer_> true true.. [12:42] <Stormer_> so where is ffmpeg binary? [12:43] <JEEB> it should be in bindir as well after your make install, but if you had a symlink to your ffmpeg source dir there I have no idea what the installation part of the makefile did [12:43] <JEEB> do you still have the output of the make install part around? [12:44] <JEEB> also lol at running make distclean there [12:45] <JEEB> why do you even have config.log then... although that's just a minor detail so fuck that [12:45] <JEEB> anyways, for future reference: never run anything after "make" ing something unless the compilation finished SUCCESSFULLY [12:45] <Stormer_> yeah everything was successfully [12:45] <Stormer_> that why i took the next part [12:46] <Stormer_> and pasted in [12:46] <Stormer_> is it possible if you can give me ffmpeg compiling codes? [12:46] <Stormer_> so i can run it over.. [12:46] <Stormer_> would really appreciate it .. [12:46] <JEEB> anyways, pastebin ls -alh /home/enc3/bin [12:47] <Stormer_> http://pastebin.com/nnsBZxCf [12:48] <JEEB> you have clearly missed a thing somewhere because you have just put the ffmpeg source code into the bindir [12:48] <JEEB> move that away from there [12:48] <JEEB> then go to that dir [12:49] <JEEB> re-run the configure script, re-run make (if you have multiple cores, do make -jN where N is the amount of cores you have) [12:49] <JEEB> check that both run successfully :P [12:49] <Stormer_> move what away? [12:49] <JEEB> the fucking directory [12:49] <Stormer_> haha [12:50] <Stormer_> ok if i delete it? [12:50] <JEEB> do you have the ffmpeg source code somewhere else? [12:50] <JEEB> because it seems like you have by mistake run the grabbing and extraction commands in the bindir :P [12:50] <JEEB> also fuse... what the fuck have you been doing there [12:51] <Stormer_> dude i havent been touching anything [12:51] <Stormer_> tell that to the guide [12:51] <Stormer_> haha [12:51] <JEEB> no, the guide doesn't do that [12:51] <JEEB> I'm pretty sure of that [12:52] <Stormer_> anyway i moved ffmpeg now out of enc3/bin [12:52] <JEEB> ok [12:52] <Stormer_> should i paste in the last code i sent you? [12:52] <Stormer_> http://pastebin.com/Hb5PTmfq [12:52] <JEEB> just re-run the configure line and see if you get any errors [12:52] <JEEB> if you don't, start make'ing [12:52] <JEEB> I mean, in the source code dir [12:53] <Stormer_> ERROR: libfdk_aac not found [12:54] <JEEB> well, no wonder then you didn't get anything [12:54] <JEEB> back to building and making sure that fdk-aac is installed in a spot where you're pointing the ffmpeg configure to look for it [12:55] <Stormer_> yes sir [12:55] <JEEB> basic reading comprehension does go a long way ;) [12:55] <JEEB> btw [12:55] <JEEB> what things do you even need in ffmpeg? [12:56] <JEEB> generally regarding encoding side of things [12:57] <Stormer_> streaming channels [12:57] <JEEB> no [12:57] <JEEB> I mean, container, audio and video formats [12:57] <Stormer_> i need x264 aac [12:57] <JEEB> ok [12:57] <JEEB> then you need fdk-aac [12:57] <Stormer_> ye [12:58] <Stormer_> the fuck fdk aac didnt install [12:58] <JEEB> as always [12:58] <Stormer_> bcz didnt have unzip installed on the server [12:58] <Stormer_> hahaha [12:58] <JEEB> see its configure output [12:58] <JEEB> hah [12:58] <Stormer_> but now its installing [12:58] <JEEB> just use the git repo? [12:58] <Stormer_> haha [12:58] <JEEB> but maybe you should just follow the "tutorial" :P so I don't have to do extra work in telling you certain things [12:58] <JEEB> and yes, don't fire commands blindly [13:01] <Stormer_> true true [13:01] <Stormer_> anyway now seems like ./configure did run [13:01] <Stormer_> http://pastebin.com/f4X3fYEE [13:01] <Stormer_> ewith no errors [13:02] <JEEB> might have wanted to paste some of the stuff higher up too [13:02] <JEEB> since the found libraries IIRC are listed there [13:03] <Stormer_> didnt get to see higher than that [13:03] <JEEB> but yeah, /seems/ to have finished without errors [13:03] <JEEB> if you're in a gui terminal, set your history to more lines :P [13:03] <JEEB> or even "unlimited" [13:07] <JEEB> now first of all when you finish make'ing, you should have a run'able ffmpeg binary in the directory you ran make in [13:07] <JEEB> and then after you make install, it should copy the things to your bindir [13:14] <Stormer_> yeah now after the config comman [13:15] <Stormer_> im ran make install [13:15] <Stormer_> i ran* [13:15] <JEEB> always run just make first [13:15] <Stormer_> yeah first make [13:15] <Stormer_> then make install [13:15] <JEEB> ok, and you checked that you could run the ffmpeg binary after make'ing? [13:16] <Stormer_> .. no :( [13:18] <JEEB> well, in any case make install should have shown in its log that it copied the ffmpeg binary :P [13:19] <Stormer_> yeah didnt check, went to grab a redbull.. [13:19] <Stormer_> but its still installing [13:19] <JEEB> what [13:19] <Stormer_> so hopefully its working [13:20] <Stormer_> its done now [13:20] <Stormer_> yay it does have ffmpeg binary [13:21] <Stormer_> in enc3/bin [13:21] <Stormer_> should i run it? [13:21] <JEEB> no, you should practice zen meditation first [13:22] <Stormer_> haha [13:22] <Stormer_> i do some yoga [13:22] <JEEB> also given how long running the make install command took, you most certainly seemingly didn't run "make" first and then "make install" [13:22] <JEEB> because most of the heavy lifting is done by the first one [13:23] <JEEB> and then the second just copies things over [13:23] <Stormer_> working :D [13:33] <Stormer_> Jeeb you there? [13:40] <Stormer_> How can i make it so ffmpeg encode couple channels with a script [13:40] <Stormer_> and starts on startup [13:40] <Stormer_> so no need to run the command each time [13:43] <Hello71> >encode couple channels [13:43] <Hello71> wat? [13:43] <Stormer_> yeah for example ffmpeg -i ... [13:43] <Stormer_> runes one channel [13:43] <Stormer_> runs* [13:43] <Stormer_> how can i run multiple ffmpeg commands [13:53] <JEEB> Stormer_, someone would have to write an initialization script that can also log it and restart it in case of failures :P [13:57] <Stormer_> damit [13:57] <Stormer_> forget about the restart in failure [13:57] <Stormer_> how just to run multiple ffmpeg commands [13:58] <Stormer_> without ffmpeg blocking the terminal [14:10] <Stormer_> http://pastebin.com/22Ajg19G [14:10] <Stormer_> and the info on bottom just keep rolling, [14:11] <Stormer_> cant add command etc. without ctrl +t which terminate the command [14:11] <Hello71> that's a shell question. [14:28] <Stormer_> huh? [15:53] Action: infojunky waits [15:57] <JodaZ> how do i convert a time in seconds to a pts time_base units? [16:12] <Mavrik> JodaZ, time_in_seconds * time_base [16:13] <JodaZ> i think its actually time_in_seconds / time_base, but the time_base is an AVRational, and the seconds i have as an int [16:14] <gigi> hi all [16:14] <gigi> how do i connect with a chat program ?? I am in an web-interface right now [16:25] <gigicarlo> Hello ! [16:25] <gigicarlo> Anybody here ???? just checking - first time here... [16:56] <Stormer_> Hello [16:56] <Stormer_> oh he left.. [18:11] <Xeta> When I use avformat_open_input to open a URL to a remote audio file, does FFMPEG start threads in the background to start streaming and buffering the data? Or does each individual call to av_read_frame actually fetch the next packge from the source? [18:11] <Xeta> I'm trying to figure out if I should do my own temporary buffering before i send the data to OpenAl for playback. [19:12] <Zeranoe> Are there any noise removal filters that are LGPL? owdenoise, and hqdn3d are both GPL [19:22] <gigicarlo> Anybody there ?? [19:41] <firemanxbr> hey guys ffmpeg have support for VPU ? [19:45] <firemanxbr> for example: [19:45] <firemanxbr> http://imx.solid-run.com/wiki/index.php?title=CuBox-i_Frequently_Asked_Ques… [19:45] <firemanxbr> anybode help me about this it ? [19:47] <llogan> yes [19:48] <llogan> what is your actual question? [19:48] <llogan> what are you trying to do? [19:49] <firemanxbr> llogan, I'm using Fedora 20 in my cubox-i4pro [19:50] <firemanxbr> llogan, one board armv7hl [19:51] <firemanxbr> llogan, my objective is run this device with my webcam logitech c920 [19:51] <firemanxbr> llogan, and send live stream to youtube live [19:51] <firemanxbr> llogan, I used in my laptop complety sucess, but in my arm device no :( [19:52] <firemanxbr> llogan, I believe don't more cpu avaliable [19:52] <firemanxbr> llogan, my ffmpeg version is 2.1.4 [19:52] <firemanxbr> llogan, this rpm: ffmpeg-2.1.4-1.fc20.armv7hl [19:53] <firemanxbr> llogan, how-to use my VPU with ffmpeg ? [19:54] <llogan> there is no "VPU" encoding support in ffmpeg itself [19:56] <firemanxbr> humm [19:56] <firemanxbr> llogan, and if I use gstreamer is possible ? [19:57] <llogan> i know nothing of gstreamer. [19:57] <Zeranoe> firemanxbr: Probably not [19:57] <firemanxbr> llogan, thankz for help [19:57] <firemanxbr> Zeranoe, humm [19:58] <firemanxbr> Zeranoe, thnkz [19:58] <Zeranoe> Your dvice has no CPU? [20:00] <firemanxbr> Zeranoe, his have [20:00] <firemanxbr> Zeranoe, but I using for encoding 720p [20:01] <firemanxbr> Zeranoe, I need more power process :D [20:01] <firemanxbr> one error always present in my tests is: ALSA xrun buffer [20:02] <firemanxbr> I hate this error :) [20:03] <firemanxbr> llogan, okay [20:10] <firemanxbr> llogan, http://ur1.ca/grwbk [20:16] <Zeranoe> firemanxbr: "Please don't paste your scripts" You need to paste the output [20:17] <firemanxbr> Zeranoe, sorry, understood now :D [20:19] <suseROCKs> Hi folks, I've tried a thousad diffrent ways after googling to figure out how to convert an m4a to mp3 with no such luck. Always get back "no such file or directory" yet the filename is properly input [20:19] <firemanxbr> Zeranoe, http://ur1.ca/grwdx [20:19] <Zeranoe> suseROCKs: Whats command are you running [20:24] <suseROCKs> Zeranoe, here's the one I tried the most... [20:24] <suseROCKs> ffmpeg -i inpuut *.m4a -acodec libmp3lame -ab 128k.m4a acodec libmp3lame -ab 128k output TeenageWasteland.mp3 [20:25] <Zeranoe> wat [20:25] <Zeranoe> whats your input file name, the m4a file [20:25] <suseROCKs> I tried both *.m4a and (fullname).m4a [20:27] <Zeranoe> ffmpeg -i (fullname).m4a -c:a libmp3lame -b:a 128k TeenageWasteland.mp3 [20:29] <suseROCKs> Zeranoe, You're my hero! Many thanks! [20:31] <llogan> firemanxbr: why are you upscaling from 640x480 to 1280x720? what is the output of: ffmpeg -f v4l2 -list_formats all -i /dev/video0 [20:32] <firemanxbr> llogan, my webcam is full hd (logitech c920) [20:32] <firemanxbr> llogan, I'm geting this output [20:33] <firemanxbr> llogan, http://ur1.ca/grwgi [20:34] <Zeranoe> firemanxbr: You need to set your input side [20:34] <Zeranoe> size* [20:34] <firemanxbr> llogan, but in my command I used get set resolutions: http://fpaste.org/83106/94132994/ [20:34] <Zeranoe> Your not using your webcam at 1080p, but are upscaling [20:35] <firemanxbr> Zeranoe, I seted [20:35] <Zeranoe> set -s before you call your input video [20:36] <Zeranoe> -s 1920x1080 -i /dev/video0 [20:38] <firemanxbr> Zeranoe, but same erro: http://ur1.ca/grwhm llogan [20:38] <Zeranoe> At least your using the correct resoution [20:39] <Zeranoe> you might want to check out http://trac.ffmpeg.org/ticket/615 [20:40] <firemanxbr> Zeranoe, humm great info, thnkz [20:57] <llogan> firemanxbr: maybe you can stream copy the h264 video from the webcam instead of re-encoding it [20:57] <llogan> if the bitrate isn't prohibitive [20:57] <llogan> for streaming [20:58] <firemanxbr> llogan, I believe no, please see this requirements: https://support.google.com/youtube/answer/2853702?hl=en [20:58] <firemanxbr> llogan, about 720p or 1080p is not compatible with my webcam :( [21:03] <llogan> ffmpeg -f v4l2 -input_format h264 -video_size 1280x720 -i /dev/video0 -codec:v copy -f flv output [21:03] <llogan> i don't know if it will work but it's worth a try [21:09] <firemanxbr> llogan, okay I'm trying... [21:10] <Zeranoe> llogan: Why not -c:v [21:11] <llogan> so users know what c stands for [21:11] <llogan> no other reason, really. [21:12] <firemanxbr> llogan, http://ur1.ca/grwqp [21:13] <firemanxbr> llogan, ALSA xrun buffer no more present, but other error in this output [21:14] <llogan> console outputs without the commands are useless [21:17] <firemanxbr> llogan, http://ur1.ca/grws8 [21:18] <llogan> commands without the console outputs are useless [21:18] <Zeranoe> lol [21:18] <firemanxbr> llogan, sorry again, window error in my xchat :D [21:24] <firemanxbr> llogan, my output no stop, ever and ever with errors: [flv @ 0x8b9f0] Non-monotonous DTS in output stream.... [21:25] <Zeranoe> firemanxbr: techincally that isn't an error [21:26] <firemanxbr> Zeranoe, but my stream no sucess, with this error :( [21:33] <firemanxbr> llogan, youtube live reporting for me: Please use a keyframe frequency of 2 seconds or less. Keyframes are not being sent often enough, this will cause buffering. Current keyframe frequency: 11.0 seconds. GOP sizes can be wrong if you have ingestion errors. [21:39] <firemanxbr> thnkz for help [21:47] <llogan> firemanxbr: then you'll have to re-encode i guess. but your hardware is too underpowered for your desired video_size and framerate. [21:48] <llogan> you can see if opencl support in x264 makes a measurable difference. i don't know if it will with your system. [21:49] <llogan> i don't know how much of a difference lookahead makes anyway [22:10] <lindenle> Is there a simple example of re-encoding a file using the shared library and c-code? [22:11] <lindenle> http://ffmpeg.org/doxygen/trunk/encoding-example_8c-source.html [22:11] <lindenle> nm' [23:11] <onefix_work> I'm having problems trying to repair a video stream...it had problems with it when the program was on, so I'm not thinking that I can fix it completely. It was a video captured from a HDHomeRun and I get "ac-tex damaged" errors. The only think I want is to minimally get the occasional pixel error as opposed to the current dropped frames that I'm seeing now. [23:12] <onefix_work> What is the best way to transcode this stream? I don't care about the time it takes either... [23:18] <Hello71> "when the program was on" [23:18] <Hello71> what. [00:00] --- Fri Mar 7 2014

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