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July 2015

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ffmpeg-devel.log.20150730
by burek 31 Jul '15

31 Jul '15

[00:01:40 CEST] <cone-183> ffmpeg 03wm4 07master:bf51fcd304d5: rawdec: fix mjpeg probing [00:01:54 CEST] <jnoring> Hi--what's the status of hosting for ffmpeg? I see there's a long thread on the mailing list, but not sure if there was a conclusion. [00:18:22 CEST] <cone-183> ffmpeg 03wm4 07master:4c6beaed9210: rawdec: fix mjpeg probing buffer size check [01:26:09 CEST] <cone-183> ffmpeg 03James Almer 07master:7e38340b831a: hmac: add support for SHA-2 [01:26:10 CEST] <cone-183> ffmpeg 03Michael Niedermayer 07master:865564c466c3: Merge commit '7e38340b831af8e3271f22bd66b9da14953c02af' [01:26:11 CEST] <cone-183> ffmpeg 03Michael Niedermayer 07master:8c5cc7b7fba0: avutil/hmac: support another ABI difference to the fork [01:35:55 CEST] <cone-183> ffmpeg 03James Almer 07master:e59f7cd89e7f: fate: add test vectors for HMAC SHA and SHA-2 [01:35:56 CEST] <cone-183> ffmpeg 03Michael Niedermayer 07master:277a5005b78d: Merge commit 'e59f7cd89e7f4f096f8456d63191adec2f8b1083' [01:38:20 CEST] <jamrial> michaelni: libav is about to force a major bump, so i think it's better to just add the new enums and keep ours renamed as deprecated until then, when the two projects can become synced without ABI problems [01:38:42 CEST] <jamrial> i'll send a patch [01:48:51 CEST] <cone-183> ffmpeg 03James Almer 07master:65dd6a1f84cc: rtmpproto: use AVHMAC instead of a custom implementation [01:48:52 CEST] <cone-183> ffmpeg 03Michael Niedermayer 07master:15a7a3cf0aaa: Merge commit '65dd6a1f84cc5c9a91d279e8e8b72ff7adf58d31' [02:03:47 CEST] <jamrial> michaelni: patch in the ml [02:27:12 CEST] <BtbN> "The compiler is GCC 4.2.1." [02:27:16 CEST] <BtbN> oO [02:28:39 CEST] <jamrial> lol [02:32:37 CEST] <cone-183> ffmpeg 03James Almer 07master:bc397246aa08: avutil/hmac: fix API/ABI compatibility with the fork [02:37:24 CEST] <BtbN> "and what we've used to build FFmpeg for quite a few releases now" [02:37:37 CEST] <BtbN> well, must have been realy quite a few ones [02:38:32 CEST] <jamrial> maybe this is some new mips code added in 2.7 that uses some preprocessor features that weren't available in 4.2 [02:38:53 CEST] <jamrial> fate has a handful of mips clients, but the oldest compiler used there is 4.3, and it builds fine [02:40:04 CEST] <jamrial> and really, they should update. i recall gcc 4.2 would miscompile shit all the time, at least on x86 [03:22:05 CEST] <cone-183> ffmpeg 03Michael Niedermayer 07master:363995296f65: doc/APIchanges: fill in some missing things [03:29:20 CEST] <cone-183> ffmpeg 03James Almer 07master:222d7619cea3: avfilter/vf_pnsr: fix author name [04:57:33 CEST] <cone-183> ffmpeg 03James Almer 07master:6ce3d9fc4a70: avformat/rtmpproto: free hmac context properly [05:00:16 CEST] <cone-183> ffmpeg 03James Zern 07master:88df16fe6dd1: libvpxenc: quiet coded_frame deprecation warnings [05:00:17 CEST] <cone-183> ffmpeg 03James Zern 07master:9723d439fdeb: libvpxenc: report pict_type/error via side-data [05:11:06 CEST] <cone-183> ffmpeg 03Michael Niedermayer 07master:bf2474c74f2c: avcodec/mpeg12enc: extend QP range to 28 for non linear quantizers [05:11:07 CEST] <cone-183> ffmpeg 03Michael Niedermayer 07master:f977e69dacf0: avcodec/mpegvideo_enc: Ignore QMAX if VBV constraints are exceeded [06:04:55 CEST] <rcombs> ubitux: mind reviewing this, if you've got a moment? https://ffmpeg.org/pipermail/ffmpeg-devel/2015-July/175655.html [12:24:01 CEST] <cone-638> ffmpeg 03James Almer 07master:9487ffd4c02b: rtmpproto: free hmac context properly [12:24:02 CEST] <cone-638> ffmpeg 03Michael Niedermayer 07master:dd0dbaccb74e: Merge commit '9487ffd4c02b9e261562d43735490068c0df0d4b' [12:36:38 CEST] <cone-638> ffmpeg 03Martin Storsjö 07master:bf0cef5c3a11: checkasm: Include io.h for isatty, if available [12:36:39 CEST] <cone-638> ffmpeg 03Michael Niedermayer 07master:1919827f2c2b: Merge commit 'bf0cef5c3a114df452e5476167634dd8f51eb448' [14:10:14 CEST] <BBB> do we test abi compat with fork in any systematic way? e.g. do we compile fate against libav-libraries and then run with binary libraries from ffmpeg and see if it passes? [14:11:11 CEST] <nevcairiel> it probably wouldnt work anyway =p [14:11:30 CEST] <nevcairiel> maintaining ABI compat is futile anyway, its already quite fragile as it is [14:28:43 CEST] <wm4> he's gone, but this is how we "test" it: http://git.videolan.org/?p=ffmpeg.git;a=blob;f=libavutil/utils.c;h=2e3b1e03… [14:43:33 CEST] <cone-638> ffmpeg 03Ganesh Ajjanagadde 07master:92e62f49cf74: ffmpeg: modify tty state when stderr is redirected [14:52:55 CEST] <wm4> BBB: <wm4> he's gone, but this is how we "test" it: http://git.videolan.org/?p=ffmpeg.git;a=blob;f=libavutil/utils.c;h=2e3b1e03… [14:53:15 CEST] <wm4> or does this count as trolling?P [14:53:22 CEST] <BBB> thats trolling, yes [14:53:25 CEST] <BBB> do we have functional tests? [14:53:27 CEST] <BBB> :-p [14:53:34 CEST] <wm4> not that I know of [14:54:00 CEST] <wm4> and this whole thing was just a bad hack at the start of the fork business anyway [14:54:12 CEST] <wm4> even if there was a need once (which I doubt), it's long gone [14:58:52 CEST] <ubitux> we're probably going to be able to clean up some "compatibility" stuff in the future [14:59:19 CEST] <BBB> that would be nice [14:59:39 CEST] <BBB> I mean, unless we have functional tests, I dont believe its useful [14:59:45 CEST] <BBB> it most likely doesnt work very reliably anyway [14:59:50 CEST] <BBB> which is worse than not working at all [16:03:09 CEST] <cone-638> ffmpeg 03Michael Niedermayer 07master:2d80de0be2b8: avcodec/v408enc: remove coded_frame usage [16:03:10 CEST] <cone-638> ffmpeg 03Michael Niedermayer 07master:a0123b602364: avcodec/v308enc: Remove coded_frame usage [16:22:27 CEST] <cone-638> ffmpeg 03Michael Niedermayer 07master:9ebffc5bba36: avcodec/yuv4enc: drop coded_frame usage [16:22:28 CEST] <cone-638> ffmpeg 03Michael Niedermayer 07master:77d8fd5c9c1d: avcodec/y41penc: Drop coded_frame usage [16:34:26 CEST] <cone-638> ffmpeg 03Luca Barbato 07master:5ae178539b91: http: Add the trailing endlines if they are missing [16:34:27 CEST] <cone-638> ffmpeg 03Michael Niedermayer 07master:8e67599505e3: Merge commit '5ae178539b91d25710b7bb322d156c31aea9f8bf' [16:42:19 CEST] <ubitux> rcombs: ah yeah sorry i forgot to answer [16:42:32 CEST] <ubitux> going to have a look now [16:43:33 CEST] <wm4> is it correct that incoming/system.php renders something? [16:45:02 CEST] <wm4> or did it get h4x0red [16:47:14 CEST] <ubitux> rcombs: you can have tag_name local to the sscanf if scope, right? [16:48:06 CEST] <wm4> rcombs: where's your mp4 sidx parsing patch? didn't you write one? [17:02:29 CEST] <ubitux> rcombs: looks pretty fine otherwise [17:49:46 CEST] <durandal_170> mpv uses audioconvert... in debian [17:51:15 CEST] <jamrial> durandal_170: the lavu header, not lavc's functions [17:53:57 CEST] <wm4> durandal_170: huh? it never did [17:56:40 CEST] <nevcairiel> Why is it supposed to be our job to fix downstreams [17:57:14 CEST] <nevcairiel> If they are unmaintained, stop using them [17:57:23 CEST] <nevcairiel> If they have maintainers, notify then [18:01:38 CEST] <wm4> the was also a deprecation period of 3 years or longer [18:04:09 CEST] <nevcairiel> Yeah these are really old things [18:05:31 CEST] <cone-638> ffmpeg 03Luca Barbato 07master:0c73a5a53cc9: hls: Save and forward avio options [18:05:32 CEST] <cone-638> ffmpeg 03Michael Niedermayer 07master:ba12ba859aab: Merge commit '0c73a5a53cc97f4291bbe9e1e68226edf6161744' [18:37:48 CEST] <cone-638> ffmpeg 03James Almer 07master:6877537ff321: hmac: add missing version bump and APIChanges entry [18:37:49 CEST] <cone-638> ffmpeg 03Michael Niedermayer 07master:d560c61125b3: Merge commit '6877537ff321136cf78ab03b8b91e6b6f218db5b' [18:45:48 CEST] <cone-638> ffmpeg 03Martin Storsjö 07master:764ec7014972: rtsp: Only interpret $ as interleaved packet indicator at the start of replies [18:45:49 CEST] <cone-638> ffmpeg 03Michael Niedermayer 07master:bc4bdc6d6152: Merge commit '764ec70149728be82304c163ccc4e280f1629201' [19:24:56 CEST] <cone-638> ffmpeg 03Michael Niedermayer 07release/2.5:ef905b1b27d5: avcodec/diracdec: Check slices malloc and propagate error code [19:24:57 CEST] <cone-638> ffmpeg 03Paul B Mahol 07release/2.5:ff68bf7a1bf3: avformat/nut: support WavPack [19:24:58 CEST] <cone-638> ffmpeg 03Andreas Cadhalpun 07release/2.5:74a88a509c6e: avidec: check for valid bit_rate range [19:24:59 CEST] <cone-638> ffmpeg 03Andreas Cadhalpun 07release/2.5:18189ebab3cf: libshine: fix support for shine 3.0 [19:25:00 CEST] <cone-638> ffmpeg 03Andreas Cadhalpun 07release/2.5:98aedbf9fe38: mpegaudiodec: copy AVFloatDSPContext from first context to all contexts [19:25:01 CEST] <cone-638> ffmpeg 03Andreas Cadhalpun 07release/2.5:983dc64772a0: vc1dec: use get_bits_long and limit the read bits to 32 [19:25:02 CEST] <cone-638> ffmpeg 03Andreas Cadhalpun 07release/2.5:d8c82ed321da: huffyuvdec: validate image size [19:25:03 CEST] <cone-638> ffmpeg 03Andreas Cadhalpun 07release/2.5:ffcfab882b66: snow: remove an obsolete av_assert2 [19:25:04 CEST] <cone-638> ffmpeg 03Andreas Cadhalpun 07release/2.5:5e833d958c09: Changelog update [19:38:00 CEST] <cone-638> ffmpeg 03wm4 07release/2.5:c809cf1c5b1b: rawdec: fix mjpeg probing [19:38:01 CEST] <cone-638> ffmpeg 03wm4 07release/2.5:1eb646ec9f87: rawdec: fix mjpeg probing buffer size check [20:13:46 CEST] <cone-638> ffmpeg 03Michael Niedermayer 07master:dfe8e3b3819d: avcodec/avuienc: Drop coded_frame usage [20:13:47 CEST] <cone-638> ffmpeg 03Michael Niedermayer 07master:802e51299e6d: avcodec/r210enc: Drop coded_frame usage [20:48:01 CEST] <slacker> Hi, is it possible to use harware accleration to accelrate video encoding using ffmpeg? I haven't installed proprietary drivers. VAAPI with intel intregrated graphics a good idea? [20:54:25 CEST] <llogan> that's a probably question for #ffmpeg, unless you're intending to do some development work on FFmpeg [20:55:13 CEST] <slacker> umm... I'll post it there, Thank you! [23:57:13 CEST] <cone-228> ffmpeg 03Michael Niedermayer 07master:d2077c860e56: Revert "ffmpeg: modify tty state when stderr is redirected" [00:00:00 CEST] --- Fri Jul 31 2015

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ffmpeg.log.20150730
by burek 31 Jul '15

31 Jul '15

[00:05:34 CEST] <prelude2004c> sonds good [00:05:42 CEST] <prelude2004c> what do you think is the best way to dump files into the UDP stream? [00:05:45 CEST] <prelude2004c> cat or something like that ? [00:05:53 CEST] <prelude2004c> has to be in realtime of course [00:06:04 CEST] <c_14> Whatever you have lying around. [00:06:12 CEST] <prelude2004c> i want to plya one file , one after the other or simply one file in a loop which i can do in some script like " while [ 1 ]; do , etc etc [00:06:12 CEST] <c_14> You could just use ffmpeg [00:06:30 CEST] <prelude2004c> ya the only problme is ffmpeg takes forever to startup .. i do't wnat to have lag betwen content [00:06:38 CEST] <prelude2004c> i want to basically end 1 packet and start another [00:07:08 CEST] <c_14> The problem with other solutions is going to be keeping it realtime and not faster than realtime. [00:08:01 CEST] <prelude2004c> yup [00:08:15 CEST] <prelude2004c> it is a problem.. but having ffmpeg stop and then start new session again will delay 1 - 2 seconds [00:08:23 CEST] <prelude2004c> doesn't sound like much but after 10 files.. everything will be out of sync [00:09:56 CEST] <c_14> I don't know about your PC, but ffmpeg starts pretty much instantaneously on my system even when reading from a network source. [00:16:05 CEST] <prelude2004c> just tried it [00:16:09 CEST] <prelude2004c> lost like 3 seconds of time [00:16:19 CEST] <prelude2004c> it stoped.. then started.. took 1 or 2 seconds to get going again [00:16:29 CEST] <prelude2004c> also, nothing in the UDP stream for some reason .. which sux [00:37:15 CEST] <bencc1> what format I can use to capture and read RTP packets? [00:37:25 CEST] <bencc1> I mean read with ffmpeg [00:44:13 CEST] <kepstin-laptop> really depends how you captured them, I assume you have a pcap file or something? [00:44:33 CEST] <kepstin-laptop> iirc, wireshark might have some stuff to extract some formats from an rtp stream in a pcap file [00:45:49 CEST] <kepstin-laptop> you can't really have a raw "rtp" file, because rtp normally requires being held in something that specifies packet size, it doesn't have a "raw" form. [00:46:53 CEST] <bencc1> kepstin-laptop: I'm using a simple script to save to pcap. the pcap file format is simple [00:47:07 CEST] <bencc1> just save Ethernet, IP and UDP frames and than the RTP packet [00:47:27 CEST] <bencc1> I can easily extract the RTP packet and the payload with a post processing script [00:47:52 CEST] <bencc1> how can I create a vp8 stream from the payload? [00:48:07 CEST] <bencc1> do I just need to reorder packets? [00:48:13 CEST] <kepstin-laptop> it requires knowledge of the destination format [00:48:20 CEST] <kepstin-laptop> read the specs, such as they are. [00:48:29 CEST] <bencc1> the RTP payload is already vp8 [00:48:30 CEST] <kepstin-laptop> usually the rtp encapsulation spec has enough info to reverse it [00:48:41 CEST] <bencc1> I can extract the vp8 payload [00:48:46 CEST] <bencc1> is this enough? [00:48:59 CEST] <bencc1> I also need to reorder the packets because it's send over udp [00:49:10 CEST] <kepstin-laptop> ... or you could just use wireshark to do it. [00:49:18 CEST] <bencc1> I need a script [00:49:27 CEST] <bencc1> wireshark is gui, right? [00:49:57 CEST] <kepstin-laptop> yeah, their command line stuff doesn't have the rtp stream dumping feature [00:51:08 CEST] <bencc1> any other option [00:51:31 CEST] <kepstin-laptop> it doesn't look like vp8 has a raw stream, so if you parse it out yourself you will either need to hand the individual packets directly to libavcodec or mux them to some container. [00:51:56 CEST] <kepstin-laptop> I suppose you could probably do something fun and replay the stream into ffmpeg's own rtp parsing code... [00:52:40 CEST] <bencc1> how? [00:52:46 CEST] <bencc1> by writing C code? [00:54:07 CEST] <kepstin-laptop> if you wanted to do something really crazy, use tcpreplay on the pcap file to stream it to an ffmpeg listening on localhost [00:54:17 CEST] <kepstin-laptop> not sure i'd advise actually doing that, but it might work ;) [00:55:12 CEST] <bencc1> how do I know how fast to replay? [00:57:22 CEST] <kepstin-laptop> not sure it matters, the rtp packets should have timestamps. [00:58:06 CEST] <kepstin-laptop> but there probably are some tools around already designed to extract media from rtp in pcap files. I'm just not sure where to look. [00:58:30 CEST] <kepstin-laptop> most of them would be designed for telephony stuff, so support for vp8 might be a bit iffy - that's pretty recent. [00:59:00 CEST] <kepstin-laptop> (and it's mostly used in webrtc, which mandates dtls encryption, which makes stream dumping stuff less useful...) [01:01:09 CEST] <bencc1> My pcap is raw rtp without encryption [01:01:16 CEST] <bencc1> I'm just trying to understand the logic [01:01:39 CEST] <bencc1> how do I convert raw rtp with vp8 payload to a webm file [01:01:48 CEST] <bencc1> what do I do with missing packets [01:02:25 CEST] <kepstin-laptop> if you want to know the theory of that, you want to be reading https://tools.ietf.org/html/draft-ietf-payload-vp8-16 [01:02:39 CEST] <kepstin-laptop> if you just want a tool to do it... well, i don't know of any offhand :/ [01:04:11 CEST] <bencc1> this tool does it [01:04:11 CEST] <bencc1> https://github.com/meetecho/janus-gateway/blob/master/postprocessing/janus-… [01:04:17 CEST] <bencc1> but with a custome format [01:05:59 CEST] <kepstin-laptop> that tool is probably assuming that things like packet reordering were done before the stream was saved (i'd assume janus server bits are running a jitterbuffer) [01:07:18 CEST] <kepstin-laptop> but yeah, other than that it's following the rtp vp8 spec to depacketize the vp8, then the webm specs to packetize it as webm (probably via a library) [01:07:30 CEST] <bencc1> it does reordering too [01:08:04 CEST] <kepstin-laptop> hmm? i didn't see any reordering in there, only some stuff for handling timestamp resets. [01:08:58 CEST] <kepstin-laptop> oh, i guess it does [01:09:06 CEST] <kepstin-laptop> didn't look at it that hard [01:09:48 CEST] <kepstin-laptop> so yeah, you probably could adapt that code to pcap input without too much trouble. [01:09:52 CEST] Action: kepstin-laptop is off now. [01:10:04 CEST] <bencc1> ok [04:25:54 CEST] <JimmySlice17_> would anyone be familiar with optimal x264 codec settings which adapt quantization per bitrate [05:42:39 CEST] <prelude2004c> hey everyone [05:42:44 CEST] <prelude2004c> can someone help me out ... while [ 1 ]; do ls /home/content/* > /scripts/files.txt; ffmpeg -re -f concat -i files.txt -codec copy -bsf:v h264_mp4toannexb -f mpegts udp://localhost:5000;done [05:42:47 CEST] <prelude2004c> whats wrong with that ? [05:42:54 CEST] <prelude2004c> plays just fine if i just play the file over and over [05:43:08 CEST] <prelude2004c> but soon as i use the concat with the files.txt it start to act a little crazy [05:45:12 CEST] <prelude2004c> nehhhh " -bsf:v h264_mp4toannexb " removed seems ok now [05:53:43 CEST] <prelude2004c> hey, anyone know a good resource to have someone help with some scripting? [06:17:02 CEST] <relaxed> prelude2004c: http://mywiki.wooledge.org/BashGuide [11:26:24 CEST] <Neepu> Hey! I'm looking into video & audio synchronization, do anyone know a good place to look for resources? Have very limited knowledge about the subject. Not looking for actual software doing it, but rather how it is implemented [11:30:56 CEST] <JEEB> Neepu: most sane containers have per-sample timestamps [11:31:36 CEST] <JEEB> so you have a sample (frame) of video, that has a timestamp, then you have a sample (frame) of audio and that usually is coded and after you decode it you get N samples of raw audio (so you have the start timestamp and the duration) [11:31:50 CEST] <JEEB> some containers also have the duration timestamp, like MP4 [11:32:01 CEST] <JEEB> others just have the sample's start timestamp [11:32:52 CEST] <Neepu> okay [11:33:44 CEST] <JEEB> then there's ancient stuff like AVI which just has a "sample rate" [11:34:01 CEST] <JEEB> but thankfully those are not the majority [12:09:03 CEST] <benschwarz> I want to add a timer text to a video, using a filter, I've been messing around with timecode=, but I was wondering if I'd be able to format the text to only display "Seconds.Milliseconds", eg "10.4", "3.2", etc [12:09:16 CEST] <benschwarz> The docs aren't super& great in that area [12:32:08 CEST] <raminious> Hi, I wrote a Linux shell script ( https://gist.github.com/raminious/de84575a7a5aca255c4d ) that concatenate two MP4 video file, first video file (prefix) only has Video Stream, and second mp4 file (main) has Video and Audio, it works fine on PC, but on Android default player, Audio not sync. can you take a look at my script and say what is the mistake [12:33:20 CEST] <Rumbles> Hello, I'm trying to convert some CCTV footage, stored in dav format, using ffmpeg. if I use "ffmpeg -i in.dav out.avi" the command completes, but the footage quality seems quite poor and the footage is faster than the original. I tried using the -vcodec copy flag, but that gave me an error, (H.264 bistream malformed) has anyone experience converting dav video to some other format (doesn't have to be avi, as long as something like VLC can handle it) wh [12:33:21 CEST] <Rumbles> o would care to offer some advice? [13:09:40 CEST] <KarlFranz> Rumbles: Reading the docs is good advice. [13:10:28 CEST] <KarlFranz> Rumbles: Docs say that if you don't set a bitrate, the output will use the default bitrate, which is bad as taxes. [13:11:45 CEST] <KarlFranz> That for the quality problem. [13:12:41 CEST] <KarlFranz> YOu might need to set the fps manually if you are encoding at wrong speed. [13:13:46 CEST] <chungy> also AVI is quite an ancient format and ffmpeg's defaults for it are very old codecs too. You'd probably get a significant improvement just by changing the output to *.mkv (which uses h264 by default for video) [13:14:34 CEST] <chungy> people shove more modern stuff into AVI but the support for such things is poor, you're better off moving onto a modern container. :P [13:14:56 CEST] <benschwarz> can I use `expr_int_format` within text='pts' to format the time into something like 2.3 (2 seconds, 3 tenths) [13:30:40 CEST] <Rumbles> thanks [14:05:22 CEST] <zugzwang> Is there a way to read from a file which is dynamically increasing its size? [14:05:28 CEST] <zugzwang> like a stream? [14:10:57 CEST] <DHE> best to present it as a stream. feed it via stdin or a named pipe, connect to it by network using a protocol that's for streaming, etc [14:40:35 CEST] <edoardo> Hello! [14:40:58 CEST] <edoardo> I can't seem to find a way to send custom headers to an icecast server [14:41:05 CEST] <edoardo> i'm using ffmpeg and this command: [14:41:17 CEST] <edoardo> ffprobe -headers 'Icy-MetaData:1\r\n\r\n' -print_format json -show_format -show_streams http://87.230.100.70:80/iloveradio2.mp3 [14:41:35 CEST] <edoardo> but the command hangs and outputs: [http @ 0x14a1c80] No trailing CRLF found in HTTP header. [14:42:19 CEST] <edoardo> there's a ticket but i didn't quite understand the workaround https://trac.ffmpeg.org/ticket/3268 [14:42:24 CEST] <edoardo> #3268 [14:53:29 CEST] <c_14> edoardo: ffmpeg automatically sends Icy-MetaData: 1 [14:53:31 CEST] <c_14> My version at least [14:53:48 CEST] <edoardo> c_14: it was an icecast missconfiguration, sorry [15:25:55 CEST] <prelude2004c> hey, anyone very familiar with nginx ? [15:35:01 CEST] <c_14> The people in #nginx? [15:37:59 CEST] <prelude2004c> tried that [15:38:02 CEST] <prelude2004c> no response :( [15:38:12 CEST] <prelude2004c> well nothing yet really with answers anways [15:38:18 CEST] <prelude2004c> :) thought i would ask twice [15:38:26 CEST] <prelude2004c> sort of stuck at the moment :) [15:38:55 CEST] <DHE> #nginx would be best, but if you're asking about the RTMP module that's 3rd party so even their support might be limited [15:39:36 CEST] <prelude2004c> nope.. asking about why i can't seem to be running two sessions at the same time.. eg.. server on port 8080 and port 8082 [15:39:46 CEST] <prelude2004c> the config is correct.. 8080 starts but 8082 does not [15:39:54 CEST] <prelude2004c> and there is no conflict anywhere [15:42:45 CEST] <bencoh> nginx-rtmp-module is works (or at least used to work, here) [15:43:17 CEST] <bencoh> what do you mean by "two sessions" ? http+rtmp ? [15:48:13 CEST] <prelude2004c> no like... just straight up listen 8080 and listen 8081 [15:48:21 CEST] <prelude2004c> i have both directives inside the config.. one starts the other does not [15:48:22 CEST] <prelude2004c> no errors [17:04:21 CEST] <ChazNunz> Hello - working on a ffmpeg filterchain with drawtext to timestamp a video. filter is successful at adding timecode to video. However, the video is stamped with hh:mm:ss:ff where h is hours, m = mins, s = secs, and f = frames. Is it possible to not include the frames in the timecode? Just stamp with hh:mm:ss? [17:13:49 CEST] <ChazNunz> Here's a pastebin of the drawtext filter i'm working with. [17:13:51 CEST] <ChazNunz> http://pastebin.com/nvqFXKDy [18:19:10 CEST] <Fyr> guys, how to split a FLAC having CUE? [18:20:26 CEST] <c_14> You usually use something like shnsplit. I don't think ffmpeg supports cue [18:22:43 CEST] <Fyr> =( [19:09:27 CEST] <edoardo> how reliable is the p2p tcp streaming using ffmpeg? [19:12:59 CEST] <c_14> As reliable as tcp [19:13:14 CEST] <edoardo> cool [19:13:28 CEST] <edoardo> how can i strip all the data from a stream - but the audio? [19:13:36 CEST] <c_14> -map 0:a [19:13:40 CEST] <edoardo> cool! [19:14:03 CEST] <edoardo> and it works perfectly as always [19:16:22 CEST] <edoardo> hmm, it has a lot of latency tho [19:16:32 CEST] <edoardo> any way to reduce the buffer size client side? [19:16:37 CEST] <edoardo> i'm using ffplay as a client [19:16:44 CEST] <edoardo> tcp has no options [19:17:35 CEST] <c_14> Mainly just what ffmpeg uses interally for buffering + encoder buffering. [19:18:19 CEST] <edoardo> i get ~3s of latency, the opus codec has way less latency [19:18:28 CEST] <edoardo> so there's a buffer somewhere i guess [19:19:42 CEST] <DHE> the latency won't be in TCP. that's a different kind of buffer. (unless you're getting your feed from the other side of the earth) [19:19:55 CEST] <edoardo> i'm working in local just for testing [19:20:21 CEST] <edoardo> i know it's not in tcp, but i was wondering about a buffer editable in the tcp plugin [19:20:54 CEST] <edoardo> sorta like buffer_size=size in udp [20:10:25 CEST] <astroty> Hello there, someone here could help? I have a big delay problem while streaming to a rtmp server on localhost. I noticed that the delay normally happens when someone connects to the server with the stream midsession, so there are timing differences. When they connect and then the stream starts, the delay is far lower (still 2-3 secs). [20:19:15 CEST] <astroty> latency* [20:22:44 CEST] <KarlFranz> Hey guys. When encoding video with lbvpx, I understand that -minrate and -maxrate set a floor and a ceiling for "instantaneous" bitrates, instead of constraining the average bitrate of the output only. Is that correct? [20:23:17 CEST] <JEEB> generally you never, ever use minrate [20:23:22 CEST] <JEEB> you should only care about maxrate and bufsize [20:23:32 CEST] <Nolski> ChocolateArmpits: awesome nick [20:26:02 CEST] <KarlFranz> JEEB: Ok, but that does not answer the question :-) [20:26:39 CEST] <KarlFranz> JEEB: During some tests, I have found some videos subjetively improved when using minrate, so ti has a place. [20:27:57 CEST] <JEEB> KarlFranz: in that case you are better off adjusting your general rate control mode [20:28:03 CEST] <JEEB> rather than trying to play around with minrate [20:28:40 CEST] <JEEB> and yes, maxrate+bufsize limit average bit rate over bufsize [20:29:09 CEST] <JEEB> while the actual rate control mode you set is doing the base rate control, which maxrate+bufsize is limiting [20:29:21 CEST] <JEEB> if it's ABR then it's the average rate over the whole clip [20:29:35 CEST] <JEEB> if it's constant quantizer / constant rate factor that's that [20:31:23 CEST] <KarlFranz> Ok, so setting a maxrate for X and an average for Y does not mean the encoder is gonna use X as a ceiling for limited bunches of frames while keeping Y at any cost. [20:31:33 CEST] <Chocola3> Nolski: why ? [20:32:24 CEST] <KarlFranz> I was asking because, in my tests, using a minrate didn't prevent some parts of the video from being encoded way bellow that limit. [20:34:46 CEST] <JEEB> KarlFranz: also depending on the version and the output format with libvpx the rate control for it can be quite a bit less fabulous than you'd expect [20:36:03 CEST] <JEEB> buut yeah, if you have set bit rate as your general rate control mode, then it's ABR over the whole clip (if you are doing one-pass it's a best effort guessing thing, basically). and then maxrate+bufsize limit that by trying to make sure you don't go over maxrate when you calculate average rate over bufsize [20:36:31 CEST] <JEEB> (also I have no idea how good VBV libvpx is [VBV is how the thing maxrate+bufsize do together is called]) [20:36:37 CEST] <JEEB> s/is/has/ [20:38:12 CEST] <KarlFranz> The videos don't look bad so I suppose it is ok. [20:38:28 CEST] Action: KarlFranz USES 2-PASS FOR ABR [20:40:57 CEST] <DHE> KarlFranz: black images will be encoded at whatever bitrate they end up as. [20:41:33 CEST] <DHE> the maximum settings are intended for use when you have devices with specific decoding capabilities or when you have to deal with streaming scenarios and limited size buffers. (for x264 see vbv_bufsize as well) [20:43:09 CEST] <KarlFranz> Noticed [20:50:41 CEST] <KarlFranz> Last line I have tried is: "ffmpeg -i input.x264 -c:v libvpx -b:v 1200k -minrate 1000k -maxrate 4800k -bufsize 9600k -pass 2 -an -sn out.webm". I suppose the limits could be wiser. [20:55:26 CEST] <slacker> Hi, is it possible to use harware accleration to accelrate video encoding using ffmpeg? I haven't installed proprietary drivers. VAAPI with intel intregrated graphics a good idea? [20:55:30 CEST] <slacker> 00:18 -!- jamrial [jamrial(a)201.255.122.175] has joined #ffmpeg-devel [21:00:06 CEST] <DHE> while x264 has opencl support, it's not really that great and whether it actually helps will vary [21:00:26 CEST] <DHE> AFAIK that's all the hardware acceleration there is in ffmpeg. some companies have commercial encoders that do better. [21:03:40 CEST] <slacker> DHE, Is installing an fglrx driver worth it? [21:04:42 CEST] <DHE> if you mean to get opencl support, I'm going to say No in the general case [21:07:31 CEST] <slacker> Okay. But I'm not able to get VAAPI working either. ffmpeg is build using --enable-vaapi, but the get the below error while trying to use it. [21:07:34 CEST] <slacker> http://pastebin.com/H3u7HN1b [21:08:15 CEST] <slacker> DHE, Am I right in understanding that hwaccel applied only for decoding and not encoding? [21:19:04 CEST] <DHE> slacker: currently accelerated decoding in ffmpeg is limited to VDPAU. there may be libvo support but I've not tried it myself. [21:19:21 CEST] <DHE> even then VDPAU is really only good for sending to the screen - the ordinary playback scenario [21:24:33 CEST] <slacker> DHE: That is really usefull information, thank you! I have rebuild ffmpeg from sources atleast 10 times trying to get VAAPI working, lol! [21:27:10 CEST] <slacker> I have an ATI Radeon based grphics card and I don't encode videos all that much. Just wanted video encoding on linux to be as fast as in Windows. [21:36:46 CEST] <astroty> Hello there, someone here could help? I have a latency problem while streaming to a rtmp server on localhost. I noticed that the delay normally happens when someone connects to the server with the stream midsession, so there are timing differences. When they connect and then the stream starts, the delay is far lower (still 2-3 secs). [22:28:36 CEST] <explodes> This is killing me- and I hate that it is such a niche question, but hopefully someone has some useful information for me: [22:29:41 CEST] <explodes> The video decoder in my android app is ffmpeg. Functions such as play/pause, and speed adjustment work fine. However, when the video begins, it starts between 8 and 16 seconds into the clip, not at the zero mark. [22:37:29 CEST] <explodes> I also cannot seek to position 0 [22:45:39 CEST] <ChazNunz> Hello - working on a ffmpeg filterchain with drawtext to timestamp a video. filter is successful at adding timecode to video. However, the video is stamped with hh:mm:ss:ff where h is hours, m = mins, s = secs, and f = frames. Is it possible to not include the frames in the timecode? Just stamp with hh:mm:ss? [22:45:48 CEST] <ChazNunz> http://pastebin.com/nvqFXKDy [22:50:41 CEST] <Nolski> ChazNunz: Don't think so. I've seen a bunch of other people ask that same question but never received an answer [22:50:48 CEST] <Nolski> but citation is needed [22:54:16 CEST] <ChazNunz> Bummer. Thanks for reply. [00:00:00 CEST] --- Fri Jul 31 2015

1 0

ffmpeg-devel.log.20150729
by burek 30 Jul '15

30 Jul '15

[01:46:24 CEST] <cone-787> ffmpeg 03Michael Niedermayer 07master:15eda746e738: avcodec/proresenc_anatoliy: remove coded_frame use [01:46:25 CEST] <cone-787> ffmpeg 03Michael Niedermayer 07master:9789595189e7: avcodec/utils: Set coded_frame.pict_type in generic code [03:24:36 CEST] <cone-787> ffmpeg 03Michael Niedermayer 07master:0a6aa30f7c05: avcodec/h264_refs: extend RAP heuristic to multiple PPS [03:24:37 CEST] <cone-787> ffmpeg 03Michael Niedermayer 07master:9ec17e45b28a: avcodec/h264_refs: Allow slightly larger pps_ref_count[0] in PAFF RAP detection heuristic [09:11:03 CEST] <rcombs> why does yuv410p have `.log2_chroma_h = 2`? [09:11:09 CEST] <rcombs> (AV_PIX_FMT_YUV410P) [09:11:16 CEST] <rcombs> seems like that should be 1 [09:13:45 CEST] <rcombs> michaelni: ^ seems like it's been like that since it's existed [09:13:51 CEST] <rcombs> (pixdesc.c, that is) [10:08:27 CEST] <L0RE> Does someone know "#EXT-X-KEY:METHOD=AES-128,URI=" ? [10:09:17 CEST] <L0RE> in m3u playlists? [10:09:29 CEST] <JEEB> sounds like the usual 128bit AES encryption :P [10:09:58 CEST] <JEEB> the URI is usually something that is behind some kind of authentication (possibly over HTTPS) [10:10:05 CEST] <L0RE> why does ffmpg not support it? [10:11:28 CEST] <L0RE> why iam asking is, i found out how to decode it, maybe it would be usefull for ffmpg [10:12:44 CEST] <L0RE> so i thought asking here where to put it if its wanted. (i dont know policy off ffmpg , maby drm isnt wanted in ffmpg, so i thought asking here befor making any afford) [10:13:45 CEST] <nevcairiel> its just aes encryption, its not hard to find out how to decode it =p [10:14:39 CEST] <L0RE> +nevcairiel: that why im wandered why ffmpg doenst support it... [10:15:49 CEST] <nevcairiel> it should be supported [10:16:45 CEST] <L0RE> then im to stupid to use it. can you give me an hint? [10:17:30 CEST] <L0RE> how to give ffmpg the keyfile and the salt, so the files could be decoded... [10:17:50 CEST] <nevcairiel> they keyfile is supposed to be referenced in the URI parameter [10:18:49 CEST] <L0RE> hmm, then i wonder why it is not working .. since which version should it working? [10:22:53 CEST] <wm4> just use git [10:24:43 CEST] <L0RE> o.k thanks ill try [10:24:52 CEST] <L0RE> by [11:15:29 CEST] <michaelni> rcombs, yuv410 has chroma for each 4x4 luma [11:18:45 CEST] <rcombs> oh, hmm, so it's inconsistent notation? [11:18:59 CEST] <rcombs> https://patches.libav.org/patch/27507/ looks like libav had some discussion on the topic [11:20:57 CEST] <rcombs> but the comment on the AV_PIX_FMT_YUV410P definition is pretty clear about it, so I guess it's probably fine [11:21:14 CEST] <nevcairiel> 4:1:0 should be one chroma for 2x4 pixels, shouldnt it [11:22:25 CEST] <nevcairiel> all these notations operate on a 2 rows 4 columns pixel grid [11:39:00 CEST] <michaelni> there was one description of the yuv notation from someone who knew the people in the actual comittee writing this somewhere but i cant find it anymore [11:42:51 CEST] <rcombs> apparently some people call the 1-per-4x4 format "YUV9" [11:43:23 CEST] <rcombs> nevcairiel: yeah, to keep with the common syntax, and that's also what e.g. Wikipedia says on the topic (though they don't have a specific source for it) [12:15:46 CEST] <kierank> pffft still more crashes with sliced threads [12:15:51 CEST] Action: kierank starts doing more fuzzing [13:27:36 CEST] <cone-116> ffmpeg 03Ganesh Ajjanagadde 07master:13d605e090bd: wavdec: make sample count check more precise [14:06:31 CEST] <cone-116> ffmpeg 03Michael Niedermayer 07master:f40ec7047864: avformat/wavdec: Check for data_size overflow [15:16:53 CEST] <Compn> goddamn "internet police" [15:20:55 CEST] <Compn> j-b : has anyone threatened your hosts' internet backbone about your website? [15:21:09 CEST] <Compn> i thought it happened to ffmpeg/mplayer once but i cant remember. [15:21:41 CEST] <Compn> sometimes they threaten the first host, then when that doesnt work they go bug the backbones haha [15:21:46 CEST] <j-b> Compn: yes. [15:22:05 CEST] <Compn> and your host handled it ok ? [15:22:15 CEST] <Compn> they didnt freak out and shut you down etc [15:24:05 CEST] Action: Compn afk [15:34:44 CEST] <J_Darnley> Who the heck freaks out because 1 anti-virus scan by 1 scanner from 1 company found 1 possibly infected file elsewhere on the internet? [15:38:09 CEST] <mathieu> I'd like to modify the H264 decoder and add it to ffmpeg, but I can't get it working [15:38:47 CEST] <ubitux> J_Darnley: more like 10 AV [15:39:01 CEST] <mathieu> How do I add the new decoder? [15:39:23 CEST] <mathieu> in allcodecs.c, I added a REGISTER_DECODER line [15:39:24 CEST] <J_Darnley> Are you looking for allcodecs.c? [15:40:00 CEST] <J_Darnley> Okay, then what happens when you build? [15:40:02 CEST] <mathieu> but I don't know where to put the other files (at this time, just a copy of the h264 is enough) [15:40:35 CEST] <J_Darnley> codecs go in libavcodec. [15:40:40 CEST] <mathieu> get an error: undefined symbol ff_h264_custom_decoder [15:40:55 CEST] <J_Darnley> you need to add files to the makefile if you want them built [15:41:05 CEST] <mathieu> it's the H264 codec, but I'm going to modify the decoding code [15:41:47 CEST] <mathieu> Makefile.am? [15:41:51 CEST] <J_Darnley> No [15:41:58 CEST] <J_Darnley> libavcodec/Makefile [15:42:34 CEST] <J_Darnley> No auto-tools rubbish here [15:45:16 CEST] <mathieu> okey, I'll have a look at that, thanks [16:09:49 CEST] <mathieu> I can get my own copy compiled & working, but only if i disable the original h264 [16:10:29 CEST] <BtbN> if your copy defines colliding symbols, that's not too surprising. [16:10:33 CEST] <BtbN> What are you even trying to do? [16:10:45 CEST] <mathieu> I'd like to keep both, and have my custom.c file call functions in the original h264.c [16:11:21 CEST] <mathieu> I'm going to modify the code of H264 to decode, but I need both [16:11:49 CEST] <mathieu> so I want to have one extra decoder available, running my custom code, but calling the core functions of h264 [16:12:07 CEST] <mathieu> is that possible? feasible? [16:12:49 CEST] <mathieu> Ultimately I need to create a custom gstreamer element based on the avdec_h264 decoder [16:38:05 CEST] <cone-116> ffmpeg 03Michael Niedermayer 07master:d64ba25a4dd7: ffplay: Use sws_scale to scale subtitles [17:03:23 CEST] <cone-116> ffmpeg 03Nedeljko Babic 07master:902bfa5b2208: avcodec/aacdec_fixed: Fix preparation for resampler [17:32:15 CEST] <cone-116> ffmpeg 03Michael Niedermayer 07master:f4ada6dc3ff7: ffmpeg: Implement support for seeking relative to EOF [17:50:27 CEST] <ocrete> how am I supposed to know the final format of the decoded picture when get_buffer2() is called? for example, I can't find out of the stream is interlaced or not before a frame comes out? and I need the information to setup the sink to get the frames, since if I wait for a frame to come out, it's too late as I can't change the stride once decoding has started as I undrerstand it ? [17:55:49 CEST] <nevcairiel> interlacing is of no consequence to the memory requirements for the image buffer, and all other relevant properties are well known at that time [18:01:55 CEST] <ocrete> and the heigh is incorrect in the AVFrame [18:02:09 CEST] <ocrete> it's the coded height, not the display height [18:02:21 CEST] <ocrete> so it can't be used to initialise the display device [18:02:28 CEST] <nevcairiel> during get_buffer2, thats the crucial height [18:02:35 CEST] <nevcairiel> since thats the size the buffer needs to be [18:03:06 CEST] <nevcairiel> you can get the display height in most cases from AVCodecContext->height [18:03:28 CEST] <ocrete> "in most cases" doesn't cut it.. I'm working on gst-ffmpeg, so we need something generic [18:03:48 CEST] <nevcairiel> thats as generic as it gets [18:04:04 CEST] <nevcairiel> and get_buffer2 is not a callback to initialize your display, its a callback to get a memory buffer to decode the frame into [18:04:16 CEST] <nevcairiel> if you use it for something else, you get to deal with problems from that :)( [18:04:19 CEST] <ocrete> in many hardware, you need to initialise the display to get the ram from [18:04:39 CEST] <BtbN> Is there any documentation on how to use the opencl stuff ffmpeg has? [18:04:45 CEST] <BtbN> In terms of adding support to other filters. [18:06:05 CEST] <ocrete> in particular, for anything that uses v4l2 output devices [18:07:31 CEST] <wm4> you don't need to override get_buffer2 [18:08:35 CEST] <ocrete> ? [18:08:55 CEST] <wm4> lavc will allocate the AVFrame for you [18:19:14 CEST] <ocrete> wm4: yes, if I wanted memory from malloc(), I'm trying to make it work with externally allocated memory [18:19:45 CEST] <wm4> why does it have to be externally allocated memory? [18:20:33 CEST] <ocrete> wm4: because you want to avoid a memcpy? for example, when outputting to a device that requires contiguous memory ? [18:21:45 CEST] <wm4> are you sure software decoding would be fast enough to even reach realtime on such a crappy restricted device? [18:23:40 CEST] <ocrete> wm4: yes [18:23:55 CEST] <wm4> but a memcpy kills it? [18:27:35 CEST] <cone-116> ffmpeg 03Michael Niedermayer 07master:1fc20af6af3e: avcodec/dvbsubdec: Implement display definition segment fallback from ETSI EN 300 743 V1.3.1 [18:27:36 CEST] <ocrete> well it doesn't help [18:27:50 CEST] <ocrete> wm4: you'd be surprised how fast ffmpeg's decoder is vs a memcpy [18:28:37 CEST] <BtbN> You must be using a crappy memcpy implementation then. [18:28:48 CEST] <wm4> at this point you could probably decode a frame to software memory, initialize the display, and then decode to video ram [18:28:54 CEST] <wm4> by starting again [19:10:15 CEST] <cone-116> ffmpeg 03Michael Niedermayer 07master:42209eb95547: doc/ffmpeg: Use @code [19:10:16 CEST] <cone-116> ffmpeg 03Michael Niedermayer 07master:0949869e7b80: ffmpeg_op: Print warning if duration isnt known when -sseof is used [20:33:02 CEST] <cone-116> ffmpeg 03James Almer 07master:6c87b866285f: avformat/rtmpproto: use AVHMAC instead of a custom implementation [21:34:45 CEST] <nevcairiel> f'ing mips, i hope the aac guy isnt distracted for months again trying to fix some silly mips problem [21:35:59 CEST] <jamrial> you mean the recent aac_fixed failures? [21:36:07 CEST] <nevcairiel> no [21:36:08 CEST] <nevcairiel> aac encoder [21:36:20 CEST] <nevcairiel> for some reason the encoder re-implements a lot of functions in C [21:36:24 CEST] <nevcairiel> not even asm or simd [21:36:25 CEST] <nevcairiel> just C [21:36:30 CEST] <wm4> wat [21:36:32 CEST] <nevcairiel> and it breaks occasionally when the encoder is patched [21:36:42 CEST] <nevcairiel> because the mips C isnt updated [21:36:56 CEST] <wm4> why does the aac encoder even have mips optimization [21:37:07 CEST] <nevcairiel> more importantly, why special C code [21:37:19 CEST] <jamrial> if they are plain c then they should be removed [21:37:59 CEST] <jamrial> c code that outputs optimized asm with an specific compiler is silly [21:38:10 CEST] <wm4> the code seems to have inline asm [21:40:40 CEST] <jamrial> is c99-to-89 still maintained or in development? [21:46:28 CEST] <rcombs> raise your hand if you give a shit about MIPS in 2015 [21:49:29 CEST] <jamrial> isn't it used in some smartphones? [21:49:30 CEST] Action: J_Darnley raises hand [21:49:42 CEST] <J_Darnley> I totally want to encode 4k video on my router [21:50:27 CEST] <BtbN> The new chinese loongson thing is mips [21:50:33 CEST] <BtbN> Which is also where most mips patches come from. [21:50:35 CEST] <wm4> "return AVPROBE_SCORE_MAX + 1;" [21:50:38 CEST] <wm4> sure looks funny [21:51:31 CEST] <rcombs> wat [21:52:00 CEST] <wm4> you'd think MAX is already the maximum [21:56:02 CEST] <JEEB> up to eleven, son [22:09:28 CEST] <rcombs> oh wow, it goes back a very long time [22:09:29 CEST] <rcombs> 07c4ed85 [22:09:35 CEST] <rcombs> I'd advocate for changing that [22:11:54 CEST] <wm4> heh [22:12:49 CEST] <rcombs> looks like it could have easily been a mistake [22:12:54 CEST] <wm4> it's this ffserver pseudo-protocol [22:13:00 CEST] <wm4> it's a lost cause anyway [22:30:07 CEST] <wm4> uh, int-size_t is size_t, right? [22:38:13 CEST] <nevcairiel> int-size_t wat [22:41:46 CEST] <BtbN> Is the opencl stuff realy only used in two filters? [22:43:20 CEST] <wm4> nevcairiel: typeof((int)1-(size_t)1) [22:43:30 CEST] <wm4> BtbN: probably, why not [22:44:01 CEST] <BtbN> Seems odd, it might be usefull for a lot of other filters [22:45:08 CEST] <jamrial> but someone needs to write them, and the guys behind the opencl stuff apparently only cared about those two [22:46:11 CEST] <nevcairiel> wm4: then yes, its size_t [22:46:17 CEST] <nevcairiel> although .. isnt size_t unsigned [22:46:33 CEST] <wm4> yes [22:53:33 CEST] <wm4> nevcairiel: did you ever see what crap modern mkvmerge versions write [22:53:54 CEST] <wm4> tons of tags with muxer info [22:54:46 CEST] <nevcairiel> yeah, but at least those are namespaces [22:55:10 CEST] <BtbN> I might look into adding opencl support to the color/chroma key stuff. But the two filters that support it aren't exactly simple, so finding out how to use it is quite hard. [22:55:29 CEST] <wm4> oh wait, somehow I thought this patch just used tags which mkvmerge already "standardized" [22:55:32 CEST] <wm4> I should check this [22:57:08 CEST] <nevcairiel> if you find out if there is one that mkvmerge already writes that just does this, i might not complain anymore [22:58:14 CEST] <wm4> mkvmerge writes per-track DURATION entries, but as strings [22:59:04 CEST] <wm4> http://sprunge.us/OieR [23:21:52 CEST] <Guest35713> hi! how does the chapters functionality work? can I use avpriv_new_chapter thing to write my own custom implementation of chapters? instead of chapters being define by timestamps, I want to define chapters as data offsets in the underlying bytestream. is this possible? [23:22:05 CEST] <Guest35713> s/define by/defined by [23:23:09 CEST] <wm4> why do they have to be data offsets? [23:23:16 CEST] <Guest35713> also, while I am demuxing an input file into an output files, can I tell the demuxer / codec to print the current timestamp (by using this timestamp value, I can create .cue files). [23:24:28 CEST] <Guest35713> wm4, that is how the underlying file format works, it seems [23:25:14 CEST] <Guest35713> the bytestream undergoes some internal processing before it can be given to the decoder (codec) [23:25:53 CEST] <wm4> in ffmpeg, packets must have timestamps [23:26:02 CEST] <wm4> so the data has timestamps before it goes to the decoder [23:26:21 CEST] <Daemon404> (NOPTS) [23:26:22 CEST] Action: Daemon404 runs [23:27:11 CEST] <Guest35713> wm4, oh, cool! I don't see "takdec.c" handling timestamps at all. how is this done? :) [23:27:44 CEST] <wm4> not sure what's the question at all [23:27:51 CEST] <Guest35713> in my "read_packet" function, can I access the timestamp of the last / current packet? is this possible? [23:28:03 CEST] <wm4> in tak, each packet generates a fixed number of samples I think [23:28:30 CEST] <wm4> so you always know what timestamp a packet has by adding the number of samples of the preceding elements [23:28:35 CEST] <wm4> s/elements/packets [23:29:24 CEST] <Guest35713> ah ok, AVPacket has some internal state which I need to take a look at then. [23:30:14 CEST] <Guest35713> also, I don't really know how many samples are being written out from my "read_packet" function, hopefully there will be some AVPacket field keeping track of this? [23:33:25 CEST] <cone-183> ffmpeg 03Michael Niedermayer 07master:f8b81a02c980: avformat/oggdec: ogg_read_seek: reset ogg after seeking [23:34:48 CEST] <Guest35713> wm4, I know when a new chapter starts when I am in "read_packet" function, and I now am trying to print the corresponding timestamp (so that I can make a .cue file). also, I don't really know how many samples I have written out (samples per packet are quite variable too). any hints? [23:40:01 CEST] <wm4> Guest35713: ffmpeg API users assume chapters are available after the file has been opened [23:40:24 CEST] <wm4> and I'm not sure how the format you're dealing with works [23:40:40 CEST] <wm4> if the classic demuxer/decoder separation doesn't work at all, you could move the decoder into the demuxer [23:40:46 CEST] <wm4> (though we try to avoid such things) [23:41:59 CEST] <Guest35713> wm4, the chapters are known by their data offsets (and not by any timestamps). weird but that is how it is. converting these data offsets into timestamps seems to be a real pain (after all this talk) ;( heh [23:44:12 CEST] <Guest35713> the file format can actually use any decoder (codec), so I can't move all the decoder into the demuxer. one hack -> in read_packet, I can ask the underlying decoder to give me its timestamp (or something) maybe? then I print this out to stderr / file to create a cue file? ugly but better than nothing. [23:44:41 CEST] <Guest35713> s/all the decoder/all the decoders [23:45:09 CEST] <wm4> there are parsers [23:45:18 CEST] <Guest35713> in oggdec.c there is something called ogg_calc_pts (it sounds promising) but I don't really know what it does [23:45:54 CEST] <wm4> you can look at how mp3dec.c works [23:45:59 CEST] <wm4> though it's not very intuitiv [23:46:00 CEST] <wm4> e [23:46:32 CEST] <wm4> read_packet in mp3dec.c just reads relatively arbitrary 1024 bytes [23:46:41 CEST] <wm4> and the parser rearranges this into actual mp3 packets [23:47:05 CEST] <wm4> and libavformat/utils.c automagically determines the packet timestamp by calling the parser [23:49:13 CEST] <Guest35713> wm4, sounds promising! I will take a look and come back, thanks! :) [00:00:00 CEST] --- Thu Jul 30 2015

1 0

ffmpeg.log.20150729
by burek 30 Jul '15

30 Jul '15

[00:00:08 CEST] Action: Nolski begins googling for acronyms [00:00:34 CEST] <klaxa> blue green red alpha [00:00:45 CEST] <klaxa> the second video is in rgb color space while the others are in yuv color space [00:00:55 CEST] <klaxa> i'm guessing that's the culprit [00:01:08 CEST] <c_14> set -pix_fmt to yuv422p when encoding that second one (or reencode to yuv422p) [00:03:42 CEST] <Nolski> Hey it worked! [00:04:10 CEST] <Nolski> That's so strange, why would it change the color space when the encoding is manually set? [00:08:10 CEST] <KarlFranz> What does the following message mean? "Non-monotonous DTS in output stream 0:0; previous: 70605251, current: 70603200; changing to 70605252. This may result in incorrect timestamps in the output file." <- When encoding an *ac3 dump from a dvd to vorbis. [00:09:03 CEST] <Nolski> KarlFranz: I'm no expert but I think it's just how the timestamps differ between encodings and sometimes it doesn't line up perfectly. I seee it all the time when encoding from one format to another [00:09:15 CEST] <Nolski> *citation needed* [00:09:42 CEST] <KarlFranz> I hope it does not cause desyncing. [00:10:09 CEST] <loop3r_> how to find out which values are k1 and k2 in matrixes http://www.ffmpeg.org/ffmpeg-all.html#lenscorrection ? [00:10:10 CEST] <KarlFranz> I am still encoding the video stream that goes with that sound, I will check when I mux it all together,. [00:13:18 CEST] <ocrete> how am I supposed to know the final format of the decoded picture when get_buffer2() is called? for example, I can't find out of the stream is interlaced or not before a frame comes out? [00:37:01 CEST] <edoceo> I've got a video that reports 10738 hours long, it's not. I process through ffmpeg to re-transcode but it still shows this errant time [00:39:51 CEST] <DHE> probably has corrupted timestamps then [00:40:07 CEST] <DHE> never seen THAT high before. 18 hours sure... [00:41:31 CEST] <c_14> Maybe try genpts? [00:42:33 CEST] <chungy> I always see that with VOB dumps from DVDs, haha... -fflags +genpts fixes it right up. Dunno if it'll help in your case. [00:46:33 CEST] <edoceo> I'll know in 8 minutes... [00:55:26 CEST] <edoceo> Dang; that did not fix my issue. Other things I could try? [00:55:44 CEST] <c_14> Input option? Output option? Both? [00:57:46 CEST] <edoceo> http://edoceo.io/paste?p=kfvYF5 [00:59:24 CEST] <chungy> move the "-fflags +genpts" to before the -i option. (and you can use "-c copy" for output to avoid re-encoding, btw) [01:00:23 CEST] <edoceo> I can't -c copy this stream cause the source audio codec is opus which confuses a down-stream consumer of this media [01:00:40 CEST] <edoceo> I'm trying the flag before -i [01:00:42 CEST] <chungy> -c:v copy then [01:00:51 CEST] <chungy> if the input and output are both VP8 anyway... heh [01:02:11 CEST] <edoceo> Our source video is not the best quality. Like this time stamp issue, among others. Before entering our processing engine I do this re-transcode to help it clean up [01:02:28 CEST] <edoceo> Otherwise the -filter_complex grid making and overlay and amix all goes to sh.. [01:04:37 CEST] <edoceo> This ffmpeg 2.2.x version works as expected, output duration is 27 minutes: http://edoceo.io/paste?p=5yWRCp [01:04:58 CEST] <c_14> Wait, is this a regression? [01:05:27 CEST] <chungy> If it works with 2.2, then yeah, it's a regression [01:05:44 CEST] <c_14> If it's a regression, open a bug on trac. [01:08:10 CEST] <edoceo> Hm, I keep getting a bunch of Captcha - and SpamBayes determined spam probability of 61.91% [01:09:31 CEST] <edoceo> Well, account is created now. I gotta do a thing, and I'll fiddle with -fflags a bit, get some repro and -report output [01:10:02 CEST] <edoceo> Well, -fflags didn't work in front of -i either. [01:10:39 CEST] <edoceo> If I upload videos to the bug tracker FTP those are not public right? And get removed/not-shared? Our sample video might be a little private [01:12:54 CEST] <c_14> They're usually attached to the bug report. Can you reproduce the issue with a small portion of the video? [01:13:48 CEST] <llogan> IIRC, the files uploaded to FTP aren't viewable by the public [01:25:47 CEST] <Nolski> I must not understand colors correctly. I'm using a simple drawtext filter but it's giving me this cannot find color error for '00)' but that isn't anywhere in the filter. [01:25:49 CEST] <Nolski> https://gist.github.com/Nolski/cee93d5c47745170057a [01:27:01 CEST] <c_14> You need to escape the ':' in the between time code thingies [01:27:37 CEST] <c_14> The last error from the drawtext filter there [01:37:02 CEST] <coops> is there a specifcation for the FFM file format anywhere? [01:42:41 CEST] <Nolski> ah right. thanks :) [03:34:53 CEST] <nardev> guys is there any quick example hoe to scale down videos only if with resolution is more than 1024 for example? [03:50:34 CEST] <c_14> you could probably just -vf scale=max(1024,x):-2 or something [04:04:27 CEST] <nardev> c_14, thank you [05:07:42 CEST] <Renari> Would anyone be willing to look at this https://gist.github.com/Renari/409f415994e9dd94930c it seems I'm having timing issues with seeking even though I'm putting -ss after the input file. In the above command intermediate2.ts is completely absent from the output file. [05:09:37 CEST] <Renari> I should mention that the output for intermediat2.ts is a 0 byte file. [05:12:29 CEST] <Renari> Is it possible to specify frame numbers when seeking instead of times? I think that might fix my issue. [05:12:56 CEST] <pzich> I think -seek and -vframes can do that? but I'm no expert [05:13:57 CEST] <pzich> nope, I'm wrong [05:14:08 CEST] <Renari> :( [05:14:37 CEST] <Renari> I don't understand why a 4 second time-window of the video is being output as a 0 byte file. [05:15:02 CEST] <pzich> anything else in the ffmpeg output for that one? does it work with 5s? [05:15:24 CEST] <pzich> oh [05:15:36 CEST] <pzich> I think the problem is that you're using -c copy [05:15:56 CEST] <Renari> It works with ts1 and ts3 though. [05:16:01 CEST] <pzich> and if your keyframes aren't where your boundaries are, it can't just copy [05:16:08 CEST] <Renari> Ah [05:16:18 CEST] <Renari> that would explain the weird timing issues I've had with ts1 and ts3 then [05:16:20 CEST] <pzich> are you sure that the other ones are exact? [05:16:38 CEST] <Renari> They are in-fact wrong based on the video timings. [05:16:50 CEST] <Renari> But work with the output. [05:18:54 CEST] <Renari> Currently trying it with a codec set. [05:19:03 CEST] <Renari> Takes a lot longer to seek though. [05:19:17 CEST] <pzich> well it has to actually encode it this time :D [05:19:56 CEST] <Renari> yup that fixed it [05:20:39 CEST] <Renari> Oh, thanks. [05:20:48 CEST] <pzich> no problem :) [12:24:23 CEST] <lilibox> hi [12:24:57 CEST] <sagax> hi! [12:25:47 CEST] <lilibox> does exist any script allows to build ffmpeg static version here? [12:26:13 CEST] <lilibox> i kept stuck on centos5 and need use the latest ffmpeg [12:26:38 CEST] <sagax> get ffmpeg archive [12:26:43 CEST] <sagax> unpack [12:26:53 CEST] <sagax> run configure in unpack dir [12:27:06 CEST] <sagax> run make and make install [12:27:11 CEST] <lilibox> with some "fork" ffmpeg already installed on system [12:27:27 CEST] <sagax> fork - removing [12:28:00 CEST] <lilibox> i am not sure if it is possible, it is used for something different [12:28:39 CEST] <sagax> try run removed and show different [12:28:43 CEST] <lilibox> soo my guess is use - create the latest ffmpeg static build and call it within full path [12:29:36 CEST] <sagax> you might configure with custom path for your self [12:30:31 CEST] <lilibox> yes, and which is the best tutorial how to create the latest ffmpeg build? any official how to? [12:30:46 CEST] <lilibox> i need static of course [12:31:31 CEST] <lilibox> mxf and h264 with sound is ok for me, at least pcm [12:31:35 CEST] <chungy> Might look at Nix and see if it runs on CentOS 5. Your biggest hurdle here is actually probably going to be the kernel; the default is 2.6.18, very ancient at this point in time. Just for Nix or other static builds to work, I wouldn't be surprised if you will be required to upgrade the kernel (which isn't too difficult in most cases) [12:32:13 CEST] <chungy> beyond that, the ./configure script is fairly self-explanatory I'd think, it has an option for static builds [12:34:54 CEST] <lilibox> chungy yes, no way to move with kernel, i must be able to do it at current state of os [12:36:42 CEST] <lilibox> ad ./configure did you speak about nix's configuration or about ffmpeg's general configuration? [12:36:44 CEST] <chungy> I would give nix a spin, but i don't know if it'll run. If it does, it's pretty easy to get going (and they probably have a pre-built ffmpeg for your needs) [12:36:51 CEST] <chungy> ffmpeg's [12:38:55 CEST] <lilibox> i must be sure that all codecs are packed in one statics binary, i am not sure if i am able to avoid situation part of my "statics" ffmpeg will call some dynamics part of existing ffmpeg which will probably criple output [12:39:29 CEST] <lilibox> that is reason i am looking for script which is able build everything iself [13:24:29 CEST] <hefest> so, when I do this: ffmpeg -y -i 2rotated.mp4 -crf 20 -vf 'transpose=2,transpose=2' -metadata:s:v:0 rotate=0 output.mp4 [13:24:34 CEST] <hefest> fps increases to 120fps [13:26:03 CEST] <hefest> original file fps is 29fps and the output.mp4 has 120fps , [13:27:21 CEST] <hefest> any way to fix this except -r 29 ? [14:04:12 CEST] <rimdeker> Any ffmpeg devs here? Just wanted to say that you guys do a hell of a job. [14:07:10 CEST] <durandal_170> most devs are on another channel [15:32:53 CEST] <mathieu> Hi, I want to copy & modify the H264 decoder [15:33:21 CEST] <mathieu> How would I make this work & include in ffmpeg? [15:33:48 CEST] <mathieu> I'd like to keep the avdec_h264 but add the modified one [15:34:51 CEST] <mathieu> is it possible to have 2 decoders simultaneously? [15:42:12 CEST] <DHE> you can have multiple decoders for the same codec. such a thing already exists for VDPAU vs software decoders [16:14:35 CEST] <well0ne> Canar you there? [16:17:43 CEST] <well0ne> here is my php solution, to my "background-process" dilemma http://privatepaste.com/58263bc365 [17:48:03 CEST] <ocrete> how am I supposed to know the final format of the decoded picture when get_buffer2() is called? for example, I can't find out of the stream is interlaced or not before a frame comes out? [18:04:15 CEST] <AndrewMock> My brain is fried. How do I apply -vf "scale=1280:533" AFTER -vf "crop=1920:800:0:140" ? [18:04:25 CEST] <AndrewMock> i know use commas somewhere [18:04:29 CEST] <AndrewMock> i just need sleep [18:08:36 CEST] <AndrewMock> makes sense now i think [18:12:46 CEST] <AndrewMock> just comma seperation unless ambigious [18:12:49 CEST] <AndrewMock> k [18:22:51 CEST] <Guest90758> I found why my command wasn't working... What was it that HEVC resolutions had to be a multiple of again? [18:22:54 CEST] <Guest90758> 4? 8? [18:23:19 CEST] <JEEB> that depends on your chroma subsampling [18:23:25 CEST] <JEEB> the format itself has no other limitations [18:24:01 CEST] <JEEB> 4:2:0 subsampling would be mod2 width and height, 4:4:4 would be mod1, 4:2:2 would be mod2 width, mod1 height [18:26:53 CEST] <Guest90758> i know what modulus is.. is that what you mean? [18:27:05 CEST] <JEEB> yes [18:27:09 CEST] <Guest90758> mod1 vs mod2 though? [18:27:16 CEST] <BtbN> h264 has a restriction of it beeing a multiple of 16 [18:27:26 CEST] <JEEB> BtbN: yes but that is of the coded region [18:27:38 CEST] <JEEB> the picture can be of whatever depending on the chroma subsampling [18:27:51 CEST] <Mavrik> Guest90758, he means that 4:2:0 has to be divisible by 2 and 4:4:4 divisible by 1 [18:27:53 CEST] <Mavrik> :) [18:28:15 CEST] <Guest90758> oh ok thx [18:28:37 CEST] <JEEB> and with HEVC the CTU can be at most 64x64, so it really doesn't make any sense to mention the coding block sizes [18:28:47 CEST] <JEEB> esp. since the padding has very close to zero cost [18:29:06 CEST] <Guest90758> so i changed my scaling to 1280x534 instead of 1280x533 and it worked... thx! [18:29:36 CEST] <JEEB> there are ways to automatize your scaling to be always within the limitations of your chroma subsampling [18:29:44 CEST] <JEEB> see the trac wiki on scaling [18:29:46 CEST] <loop3r> hello, have someone worked with ffmpeg lenscorrection I want to ask about little thing :P [18:29:50 CEST] <JEEB> and ffmpeg-all.html [18:30:09 CEST] <Mavrik> Guest90758, scale filter has a helpful "-2" constant [18:30:14 CEST] <Mavrik> e.g. "scale=1280:-2" [18:30:24 CEST] <Guest90758> oh solid [18:33:06 CEST] <Guest90758> when my h264 decoder mumbles something like "not enough frames to estimate rate; consider increasing probesize", does that mean bitrate or framerate? [18:33:51 CEST] <Guest90758> reading between the lines this means bitrate & no effect on my output bitstream. [18:36:42 CEST] <Nolski> What's the difference between the -vf flag and -filter_complex? [18:38:08 CEST] <c_14> -vf is a filterchain, -filter_complex is a filtergraph [18:38:20 CEST] <c_14> A filtergraph can contain several filterchains [18:39:23 CEST] <Nolski> ah, I see [18:41:58 CEST] Action: Nolski sighs [18:42:22 CEST] <Nolski> why does a drawtext filter kill the audio on my video that I'm applying it to? [18:52:47 CEST] <DHE> Nolski: if you're using -map to pull in filter items, be sure to explicitly pull in audio as well. -map 0:a [18:53:26 CEST] <Nolski> DHE: but if I'm not filtering any of the audio with this command do I just reference it in the -map? [18:53:37 CEST] <DHE> filter_complex will let you build a very complicated filter graph with multiple inputs and multiple outputs [18:53:57 CEST] <DHE> you're not filtering the audio, but you are explicitly building an output from built elements. you'll need to specify audio manually in this case [18:54:07 CEST] <Nolski> Ah I see [18:54:16 CEST] Action: Nolski levels up [18:54:57 CEST] <DHE> for example I'm making multiple HLS feeds (several bitrates) from a single ffmpeg instance. audio is split several ways, video is deinterlaced once and then piped through several scale filters for each input [18:59:58 CEST] <Nolski> DHE: I think I see what you're saying. I've mixed different audio streams together just fine using the amerge filter but since this is just one video with text being rendered on it, I'm a bit confused how to just carry that one stream over. [19:00:01 CEST] <Nolski> https://gist.github.com/Nolski/d9edc88da76f204441ec#file-gistfile1-txt-L26 [19:00:38 CEST] <Nolski> That stream [0:1] is being mapped using `-map` but the output doesn't have it so it breaks? [19:04:34 CEST] <Guest90758> why does my bitrate keep going up??? lol [19:11:14 CEST] <DHE> Nolski: don't use [brackets] for file inputs [19:13:08 CEST] <Nolski> DHE: not quite sure I follow [19:13:34 CEST] <DHE> -map [videoout] -map 0:a [19:26:41 CEST] <Nolski> DHE: sorry for making you spell it out for me but prior, I was never mapping the video (or [videoout]). If I assign it in the filter graph it gives the same error as with the audio [20:17:03 CEST] <AndrewMock> [h264 @ 0000000004866780] Stream #0: not enough frames to estimate rate; consider increasing probesize [20:18:30 CEST] <AndrewMock> Does that mean that the decoder can't figure out the framerate? [20:21:27 CEST] <kepstin-laptop> I think that message is actually about bitrate. Unless you're trying to figure out the bitrate of an input file, it'll just be a minor display issue. You can probably ignore it unless you have other problems. [20:24:58 CEST] <AndrewMock> well i think that it may have something to do with my apparent timing issue http://i.imgur.com/vYbsAEe.png [20:26:15 CEST] <kepstin-laptop> AndrewMock, that 'past duration too large' thing is a fairly new message in ffmpeg; it happened before but used to not print a message. It apparently has something to do with the rate control on the output encoding [20:31:36 CEST] <loop3r> who is good with lens correction filter ?:D [20:34:21 CEST] <JEEB> the person who wrote it? [20:34:38 CEST] <JEEB> I hope you read the documentation at least [20:37:26 CEST] <AndrewMock> what is rate control distortion? [20:44:30 CEST] <loop3r> JEEB: I did, I just want to ask few question about opencv in that example thats it, calm down [20:46:35 CEST] <prelude2004c> hello everyone [20:46:57 CEST] <prelude2004c> question ... how do i setup ffmpeg to allow playback of a file in a loop ? eg.. i have 10 files.. i want to play them one by one without delay [20:47:19 CEST] <c_14> https://trac.ffmpeg.org/wiki/Concatenate [20:47:48 CEST] <kepstin-laptop> prelude2004c, that sounds like something that many players can do, when given an appropriate playlist... [20:48:50 CEST] <kepstin-laptop> but yeah, you could use ffmpeg to concatenate them into a single video if you can't just play them directly. [20:49:58 CEST] <AndrewMock> I really need help with understanding what this means: http://i.imgur.com/vYbsAEe.png [20:50:06 CEST] <AndrewMock> Shall I file a bug report? [20:50:24 CEST] <AndrewMock> At least to ask for more-understandble strings? [20:51:02 CEST] <AndrewMock> some people have murmored about framerate sync and about rate control [20:51:08 CEST] <AndrewMock> but nothing explicit [20:51:43 CEST] <prelude2004c> kepstin-laptop < .. i am trying to create a stream though [20:51:43 CEST] <JEEB> loop3r: I'm calm, I just am pretty sure that most folk have no idea on such specific filters added by someone at some point for a very specific use case [20:51:51 CEST] <prelude2004c> that si why i want ffmpeg so i can output to RTMP or hls or whatever [20:52:14 CEST] <kepstin-laptop> prelude2004c, ah, gets a bit more complicated then :) [20:52:22 CEST] <prelude2004c> lol .. well yes i guess [20:52:30 CEST] <prelude2004c> i want to just say ffmpeg -i something -i something else -i somethign else.. etc etc [20:52:34 CEST] <prelude2004c> and just play them sequentially [20:52:44 CEST] <prelude2004c> without any delay from one to the next [20:52:46 CEST] <kepstin-laptop> ffmpeg can loop a single input video, but I don't know any way to loop a list of multiple videos.... [20:53:02 CEST] <prelude2004c> yup single file is not a problem.. [20:53:06 CEST] <prelude2004c> its the multiple that is a problem for me [20:53:11 CEST] <prelude2004c> i was hopping someone knew a good way to do it [20:53:15 CEST] <kepstin-laptop> in this case, you could just combine the videos into a single file in advance, of course. [20:53:23 CEST] <prelude2004c> can some player do it ( linux ) that i can just output and | > ffmpeg [20:53:31 CEST] <c_14> you can loop an infinite concat list [20:53:35 CEST] <c_14> with some bash magic [20:53:50 CEST] <prelude2004c> how would that look like ? i don't know bash very well [20:54:09 CEST] <prelude2004c> i want to basicaly get list of files ... > output | > ffmpeg to stream local udp or something [20:54:15 CEST] <prelude2004c> or i will just output to HLS [20:54:41 CEST] <c_14> ffmpeg -f concat -i <(while true; do for file in *; do printf "file %s\n" $file; done; done) [20:54:44 CEST] <c_14> something like that [20:56:05 CEST] <prelude2004c> hum.. never used that before [20:56:18 CEST] <prelude2004c> do i pipe that out after ? [20:56:26 CEST] <kepstin-laptop> that has the caveats of using the concat input format, tho; all the files have to be in a concatenatable file format (e.g. mpeg-ts) [20:56:44 CEST] <prelude2004c> or can i just continue with -f mpegts udp://localhost:xxxx [20:56:55 CEST] <c_14> you can just continue with whatever you want [20:57:05 CEST] <prelude2004c> i dont have a problem with the same format [20:57:11 CEST] <kepstin-laptop> hmm, no, that doesn't need concatenatable formats, just codecs with matching parameters, etc. [20:57:14 CEST] <prelude2004c> would be nice if files would be mpeg4 and mkv ,etc etc [20:57:33 CEST] <kepstin-laptop> I got the different concat methods mixed up [20:57:36 CEST] <prelude2004c> so your saying i can't have one mp4 , one mkv, one wmv [20:57:59 CEST] <c_14> The video/audio settings have to be the same. [20:58:11 CEST] <prelude2004c> that may be a problem since sometimes the video may be different [20:58:19 CEST] <prelude2004c> like format or even resolution [20:58:23 CEST] <prelude2004c> some videos may be different resolutons too [21:00:21 CEST] <kepstin-laptop> then your best bet is to pre-transcode them [21:01:39 CEST] <kepstin-laptop> if you pre-transcode so everything's in matching formats, c_14's trick would work, and you might even be able to get away with -c copy instead of doing an extra live encoding. depends on the player, i guess. [21:01:48 CEST] <JEEB> loop3r: if you think that randomly pm'ing people is alright, then I will just say that no - it is not [23:05:56 CEST] <prelude2004c> hey guys.. the loop function.. any way to loop a file forever? [23:10:31 CEST] <ChocolateArmpits> Hello, Anyone familiar with parsing MXF files ? [23:42:46 CEST] <prelude2004c> anyone? i can't seem to figure this out [23:45:47 CEST] <Nolski> prelude2004c: you can just write a script that keeps looping the file. [23:46:13 CEST] <Nolski> pretty sure if it's the right format you can just keep concatting the same file upon itself [23:46:24 CEST] <Nolski> or you can just make it so it loops when it lays [23:46:28 CEST] <Nolski> s/lays/plays/ [23:49:42 CEST] <prelude2004c> hum.. dont know how to do that [23:49:48 CEST] <prelude2004c> also, maybe there is a way with cat [23:49:53 CEST] <prelude2004c> eg.. have ffmpeg listen to a port [23:50:04 CEST] <prelude2004c> and then just use a script that cat file > output [23:50:13 CEST] <prelude2004c> so i i can keep sending files to the listener [23:50:15 CEST] <prelude2004c> something like that [23:50:34 CEST] <prelude2004c> i was thinking like a local UDP port and then just cat files to that udp [23:50:42 CEST] <prelude2004c> any other ideas on how to do that ? this way i can loop any file or many files [23:51:20 CEST] <c_14> ffmpeg -f concat -i <(while true; do printf "file %s\n" /path/to/file; done) [23:52:42 CEST] <prelude2004c> what if i want to introduce multiple files [23:52:51 CEST] <prelude2004c> any way to put ffmpeg as a listner that never exists ? [23:52:57 CEST] <prelude2004c> like.. for example.. i did a mkfifo test file [23:53:04 CEST] <prelude2004c> then i did ffmpeg -i fifofile [23:53:08 CEST] <prelude2004c> and it sits there and waits [23:53:15 CEST] <prelude2004c> then i add info to that file and it loads the data... but... [23:53:21 CEST] <prelude2004c> soon as the data stops , ffmpeg exists [23:53:33 CEST] <prelude2004c> any way to keep ffmpeg listening all the time and never exist so it takes any info i give it from differnet sources [23:54:25 CEST] <c_14> You could listen on udp probably. [23:55:03 CEST] <prelude2004c> right and it never exists ? like listen 239.xx.x.x .. something liek that ? [23:55:48 CEST] <c_14> ffmpeg -i udp://ip:port?listen [23:56:17 CEST] <c_14> Worked for me (for flac anyway) [23:56:54 CEST] <c_14> That might only work for some formats though. [00:00:00 CEST] --- Thu Jul 30 2015

1 0

ffmpeg-devel.log.20150728
by burek 29 Jul '15

29 Jul '15

[00:02:32 CEST] <cone-006> ffmpeg 03Tom Butterworth 07master:3ee217853a67: Support the Hap chunked frame format [00:02:33 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07master:4be1b6bef097: Merge commit '3ee217853a6741b829a2683f49c590618891b1ab' [00:12:41 CEST] <cone-006> ffmpeg 03Vittorio Giovara 07master:aaf937ee3557: hap: Add utility functions file [00:12:42 CEST] <cone-006> ffmpeg 03Vittorio Giovara 07master:abdc7e403e00: fate: Add hap-chunk ref file [00:12:43 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07master:952e9fbb4aaa: Merge commit 'aaf937ee3557bfb99c2ad298591b22a7f22ecbf7' [00:12:44 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07master:146730480265: Merge commit 'abdc7e403e003e97a0dcc4499ac319bf854aeead' [00:18:28 CEST] <philipl> rcombs: heh. I thought about that briefly as well. [00:18:44 CEST] <philipl> I guess this is what happens when you keep using a component that hasn't been maintained for 4+ years. [00:19:50 CEST] <wm4> I wonder what's the nature of this bug [00:20:04 CEST] <wm4> seems to be deep inside media processing, maybe even a demuxer or decoder bug? [00:21:18 CEST] <cone-006> ffmpeg 03Anton Khirnov 07master:db21dde3f72c: qsvdec_mpeg2: drop an incorrect comment [00:21:19 CEST] <cone-006> ffmpeg 03Alexandra Hájková 07master:b5c1c16247ab: asfdec: do not align Data Object when Broadcast Flag is set [00:21:20 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07master:4ecc9a2f31bf: Merge commit 'db21dde3f72c2331653399bdb8745350f015d847' [00:21:21 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07master:e1296b5fa4e5: Merge commit 'b5c1c16247ab7d166c84eaf4564e49a1535fdaaf' [00:40:01 CEST] <cone-006> ffmpeg 03John Högberg 07master:b7040e67ec18: h264: fix AVDISCARD_NONKEY for some interlaced content [00:40:02 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07master:ea7e4f8e6aa2: Merge commit 'b7040e67ec18259ca634a0e29d98469b3484a87c' [00:44:14 CEST] <philipl> wm4: CVEs are not published yet? or am I missing something [00:49:39 CEST] <wm4> no idea [00:51:01 CEST] <philipl> wm4: are you going to attempt to get that vaapi/egl craziness working in mpv? [00:53:36 CEST] <wm4> philipl: most likely I will, but currently I'm worried that the drivers on my test system are not new enough [01:02:21 CEST] <philipl> wm4: That's Intel for you. Always need to be running master. [01:14:56 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07master:360cebc9c82c: avcodec/mpegvideo_enc: move ff_side_data_set_encoder_stats() call to after the vbv retry code [02:11:07 CEST] <cone-006> ffmpeg 03James Almer 07master:5d95f32bff86: avocdec/mpeg12dec: fix mpeg_xvmc hwaccel flag [02:18:21 CEST] <jamrial> michaelni: ^ not sure if you prefer to add an AV_ prefixed version of the flag since it's deprecated, like you did with the vdpau one [02:28:00 CEST] <michaelni> is the flag needed/used? [02:28:46 CEST] <jamrial> no idea, but it broke compilation of a fourth of our fate clients [02:29:13 CEST] <jamrial> ok, not a fourth but a couple dozen [02:29:35 CEST] <michaelni> i must have missed it as it wasnt build locally [02:30:21 CEST] <jamrial> yeah, configure shows X11_extensions_XvMClib_h as a dep for that decoder [02:33:42 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07master:27294bc203a4: avcodec/mpegvideo_enc: export per frame PSNR through side data [02:33:43 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07master:1ca042a14b93: avcodec/snowenc: Export quality/pict_type/PSNR stats through side data [02:33:44 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07master:cdb0225fa9e6: ffmpeg: Use side data instead of coded_frame for error[] values [03:21:49 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07release/2.5:f1a4af4dc27c: avformat/wavdec: Increase probe_packets limit [03:21:50 CEST] <cone-006> ffmpeg 03Rodger Combs 07release/2.5:9e1a818a2df1: avformat/wavdec: Increase dts packet threshold to fix more misdetections [03:21:51 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07release/2.5:962ad32891c8: avcodec/put_bits: Update size_in_bits in set_put_bits_buffer_size() [03:21:52 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07release/2.5:e83d99863418: avcodec/mpegvideo_enc: Update the buffer size as more slices are merged [03:21:53 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07release/2.5:4ac8b94a7321: avcodec/put_bits: Assert that there is enough space left in skip_put_bytes() [03:21:54 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07release/2.5:8d98fbef1655: avcodec/bitstream: Assert that there is enough space left in avpriv_copy_bits() [03:21:55 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07release/2.5:a61759596dc7: ffmpeg_opt: Set the video VBV parameters only for the video stream from -target [03:21:56 CEST] <cone-006> ffmpeg 03Luca Barbato 07release/2.5:6b2163caabad: h264: Make sure reinit failures mark the context as not initialized [03:21:57 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07release/2.5:fd989ff9ca4d: swresample: Check the return value of resampler->init() [03:21:58 CEST] <cone-006> ffmpeg 03Andreas Cadhalpun 07release/2.5:42c54f8f4dd9: libopenjpegenc: add NULL check for img before accessing it [03:21:59 CEST] <cone-006> ffmpeg 03Ganesh Ajjanagadde 07release/2.5:3dc303a05192: swresample/dither: check memory allocation [03:22:00 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07release/2.5:f569422d669f: avformat/mxfenc: Accept MXF D-10 with 49.999840 Mbit/sec [03:22:01 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07release/2.5:1728aa3442d7: Revert "avformat/rtpenc: check av_packet_get_side_data() return, fix null ptr dereference" [03:22:02 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07release/2.5:de943a3460a8: swresample/swresample: Cleanup on init failure. [03:22:03 CEST] <cone-006> ffmpeg 03Nedeljko Babic 07release/2.5:3dd5c6a4ba80: libavutil/softfloat: Fix av_normalize1_sf bias. [03:22:04 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07release/2.5:177014a49437: avcodec/atrac3plusdec: consume only as many bytes as available [03:22:05 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07release/2.5:5ebae729b9a9: avcodec/alsdec: Check for overread [03:22:06 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07release/2.5:cdd594e24ba1: avcodec/adpcm: Check for overreads [03:22:07 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07release/2.5:057d6e1cd620: avformat/ffmdec: Check ffio_set_buf_size() return value [03:22:08 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07release/2.5:062f63e46cd8: avcodec/jpeg2000dec: Check that coords match before applying ICT [03:22:09 CEST] <cone-006> ffmpeg 03Simon Thelen 07release/2.5:fb22fc0c46bb: libavutil/channel_layout: Correctly return layout when channel specification ends with a trailing 'c'. [03:22:10 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07release/2.5:a2a545d8db8e: avdevice/lavfi: do not rescale AV_NOPTS_VALUE in lavfi_read_packet() [03:22:11 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07release/2.5:f5a73058b0f2: avcodec/h264_slice: Use AVFrame diemensions for grayscale handling [03:22:12 CEST] <cone-006> ffmpeg 03Andreas Cadhalpun 07release/2.5:3c99f8b4ada2: takdec: ensure chan2 is a valid channel index [03:22:13 CEST] <cone-006> ffmpeg 03Deliang Fu 07release/2.5:bda906a19561: avformat: Fix bug in parse_rps for HEVC. [03:22:14 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07release/2.5:7d03a9156da3: ffmpeg_opt: Check for localtime() failure [03:22:15 CEST] <cone-006> ffmpeg 03Andreas Cadhalpun 07release/2.5:7a12379463b5: mov: abort on EOF in ff_mov_read_chan [03:22:16 CEST] <cone-006> ffmpeg 03Andreas Cadhalpun 07release/2.5:a1879d347e62: libopenjpegdec: check existence of image component data [03:22:17 CEST] <cone-006> ffmpeg 03Andreas Cadhalpun 07release/2.5:b4d265e2dcd9: arm: only enable setend on ARMv6 [03:22:18 CEST] <cone-006> ffmpeg 03Andreas Cadhalpun 07release/2.5:e4e3b14bba12: vp9: change type of tile_size from unsigned to int64_t [03:22:19 CEST] <cone-006> ffmpeg 03Andreas Cadhalpun 07release/2.5:27d50fb2d52b: vp8: change mv_{min,max}.{x,y} type to int [03:22:20 CEST] <cone-006> ffmpeg 03Andreas Cadhalpun 07release/2.5:da13957525a7: sonic: set avctx->channels in sonic_decode_init [03:22:21 CEST] <cone-006> ffmpeg 03Andreas Cadhalpun 07release/2.5:58a0dc1bdf67: h264: er: Copy from the previous reference only if compatible [03:22:22 CEST] <cone-006> ffmpeg 03wm4 07release/2.5:3903a60d4bee: avio: fix potential crashes when combining ffio_ensure_seekback + crc [03:22:23 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07release/2.5:568c1b8166b1: ffmpeg: Free last_frame instead of just unref [03:22:24 CEST] <cone-006> ffmpeg 03Andreas Cadhalpun 07release/2.5:dd141a50ccb3: postproc: fix unaligned access [03:22:25 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07release/2.5:fc0df14de1a6: avcodec/dpxenc: implement write16/32 as functions [03:22:26 CEST] <cone-006> ffmpeg 03Andreas Cadhalpun 07release/2.5:5464da890566: matroskadec: validate audio channels and bitdepth [03:22:27 CEST] <cone-006> ffmpeg 03Andreas Cadhalpun 07release/2.5:a5bc5dd49ae1: matroskadec: check audio sample rate [03:22:28 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07release/2.5:7ff8f9a604f0: swr: Remember previously set int_sample_format from user [03:22:29 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07release/2.5:f9dbe8ab2a03: swscale/rgb2rgb_template: Disable shuffle_bytes_2103_c on big endian [03:22:30 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07release/2.5:17b7262b4f9d: swscale/rgb2rgb_template: Implement shuffle_bytes_0321_c and fix shuffle_bytes_2103_c on BE [03:22:31 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07release/2.5:7085f68c14ba: swscale/rgb2rgb_template: Fix signedness of v in shuffle_bytes_2103_c() [03:22:32 CEST] <cone-006> ffmpeg 03Sebastien Zwickert 07release/2.5:8f067955332d: vda: unlock the pixel buffer base address. [03:22:33 CEST] <cone-006> ffmpeg 03James Almer 07release/2.5:3528822f969e: swscale/x86/rgb2rgb_template: add missing xmm clobbers [03:22:34 CEST] <cone-006> ffmpeg 03James Almer 07release/2.5:1fc2a1f3367c: swscale/x86/rgb2rgb_template: fix signedness of v in shuffle_bytes_2103_{mmx,mmxext} [03:22:35 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07release/2.5:98d916300afe: ffmpeg: Do not use the data/size of a bitstream filter after failure [03:22:36 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07release/2.5:bf44ab5181d4: swscale/swscale_unscaled: Fix rounding difference with RGBA output between little and big endian [03:22:37 CEST] <cone-006> ffmpeg 03Andreas Cadhalpun 07release/2.5:755c3222e2c6: wmavoice: limit wmavoice_decode_packet return value to packet size [03:22:38 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07release/2.5:d661cfad02af: avcodec/pngdec: Only allow one IHDR chunk [03:22:39 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07release/2.5:0628dfd99427: avcodec/pngdec: Require a IHDR chunk before fctl [03:22:40 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07release/2.5:6a71e748b3a5: avcodec/pngdec: Check values before updating context in decode_fctl_chunk() [03:22:41 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07release/2.5:c7cb598bdd8a: avfilter/vf_transpose: Fix rounding error [03:22:42 CEST] <cone-006> ffmpeg 03James Zern 07release/2.5:68650eda9522: vp9/update_prob: prevent out of bounds table read [03:22:43 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07release/2.5:58070590b1fc: avcodec/h264_slice: Use w/h from the AVFrame instead of mb_w/h [03:22:44 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07release/2.5:1b15487e903b: avcodec/aacsbr: check that the element type matches before applying SBR [03:22:45 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07release/2.5:dec0316c0f3b: avcodec/aacsbr: Assert that bs_num_env is positive [03:22:46 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07release/2.5:e9deb5581075: avcodec/rawenc: Use ff_alloc_packet() instead of ff_alloc_packet2() [03:22:47 CEST] <cone-006> ffmpeg 03Andreas Cadhalpun 07release/2.5:29a97823ba7c: wmalosslessdec: avoid reading 0 bits with get_bits [03:22:48 CEST] <cone-006> ffmpeg 03Andreas Cadhalpun 07release/2.5:dcfd83fb8e3c: wmalosslessdec: reset frame->nb_samples on packet loss [03:22:49 CEST] <cone-006> ffmpeg 03Chris Watkins 07release/2.5:4f4ff23ccb88: oggparsedirac: check return value of init_get_bits [03:22:50 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07release/2.5:d137bb92b298: avcodec/mpegvideo: Clear pointers in ff_mpv_common_init() [03:22:51 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07release/2.5:80e091853d1e: avcodec/utils: use a minimum 32pixel width in avcodec_align_dimensions2() for H.264 [03:22:52 CEST] <cone-006> ffmpeg 03Anton Khirnov 07release/2.5:abe503329de0: bytestream2: set the reader to the end when reading more than available [03:22:53 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07release/2.5:3489693051a7: avcodec/alac: Clear pointers in allocate_buffers() [03:22:54 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07release/2.5:1cb19f4f08d5: avcodec/sanm: Reset sizes in destroy_buffers() [03:22:55 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07release/2.5:95d8912bfdc5: avcodec/pthread_frame: check avctx on deallocation [03:22:56 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07release/2.5:3e167497c78f: ffmpeg: Fix cleanup with ost = NULL [03:22:57 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07release/2.5:40f5222e9c17: ffmpeg: Fix crash with ost->last_frame allocation failure [03:22:58 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07release/2.5:3953c497e275: avformat/mov: Fix deallocation when MOVStreamContext failed to allocate [03:22:59 CEST] <cone-006> ffmpeg 03Zhang Rui 07release/2.5:d819a2c9869c: avutil/fifo: Fix the case where func() returns less bytes than requested in av_fifo_generic_write() [03:23:00 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07release/2.5:46236b0bae23: swscale/utils: Clear pix buffers [03:23:01 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07release/2.5:b7c8d072ea1b: avcodec/pthread_frame: clear priv_data, avoid stale pointer in error case [03:23:02 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07release/2.5:384f6801a2ed: avfilter/af_aresample: Check ff_all_* for allocation failures [03:23:03 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07release/2.5:8696762b9a3b: avcodec/rv34: Clear pointers in ff_rv34_decode_init_thread_copy() [03:23:04 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07release/2.5:c72c1b871602: avcodec/diracdec: Check for hpel_base allocation failure [03:23:05 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07release/2.5:1166d9245ab1: avcodec/vp8: Fix null pointer dereference in ff_vp8_decode_free() [03:23:06 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07release/2.5:be369f6e4034: avcodec/vp8: Check buffer size in vp8_decode_frame_header() [03:31:21 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07release/2.5:ad03cef4208b: Update for 2.5.8 [05:29:14 CEST] <cone-006> ffmpeg 03Chris Spencer 07master:9c41126e1b9f: avdevice/decklink: Fix build error caused by a change in the SDK. [06:27:06 CEST] <jamrial> looks like we hit a c99-to-c89 bug [06:27:20 CEST] <jamrial> "Assertion failed: index_is_unique(&struct_array_lists[rec.parent->data.sal_idx], sai->index), file convert.c, line 1601" [06:27:39 CEST] <jamrial> this is with the vs2012 fate client [08:45:23 CEST] <lomer_> hey people ffmpeg -i input.mkv -vcodec copy -acodec copy -ss 00:04:00 -t 00:04:00 output.mkv (Delay relative to video: -651ms) Delay relative to video a mistake? [13:17:41 CEST] <cone-787> ffmpeg 03Michael Niedermayer 07master:accc91642b60: avcodec/h264_slice: do not skip 2nd field if first was not skipped [13:35:26 CEST] <cone-787> ffmpeg 03Michael Niedermayer 07master:c3fdf536f6c0: avcodec/h264_slice: Also check sei_recovery_frame_cnt for skip_frame nokey [13:57:11 CEST] <cone-787> ffmpeg 03Michael Niedermayer 07master:07558ad5824a: avcodec/libdcadec: exss is used only under ifdef, thus mark it as potentially unused [15:39:32 CEST] <unknown33> using AVOption in a new demuxer (being written) causes a segfault in "avformat_open_input" -> "av_opt_set_defaults" code path. is there some setup function I need to call before using AVOption stuff? [15:41:33 CEST] <lglinskih> kierank: ping [15:48:28 CEST] <durandal_170> unknown33: you have avclass in private context? [15:49:35 CEST] <nevcairiel> yeah make sure the context starts with an AVClass [15:54:19 CEST] <unknown33> durandal_170, nevcairiel thanks! that was it (I missed it) [15:56:19 CEST] <kierank> lglinskih: pong [15:58:54 CEST] <lglinskih> kierank: new version of test for draw_horiz_band: https://github.com/lglinskih/FFmpeg/commit/ac8c8b3673e8bca7af8ea3a5bed1c49c… [15:59:23 CEST] <lglinskih> but it works only for decoding in one thread [16:00:20 CEST] <lglinskih> I don't know how correctly test decoding in multiple threads [16:01:56 CEST] <unknown33> is ffplay == avplay? it seems that it has support for handling chapters (page up / /page down keys), how do I ensure that my demuxer exposes the chapter information, so that ffplay can use it? [16:10:28 CEST] <BtbN> avplay is libav [16:11:50 CEST] <kierank> lglinskih: hmm [16:16:03 CEST] <kierank> I think it's ok though [16:16:12 CEST] <kierank> you should submit it and work on seek test fixes if you haven't already [16:17:25 CEST] <lglinskih> ok [16:25:13 CEST] <durandal_170> unknown33: see avpriv_new_chapter [16:26:45 CEST] <unknown33> durandal_170, will do, thanks! [16:33:35 CEST] <cone-787> ffmpeg 03Ivan Uskov 07master:0b8b18b4fbbd: libavcodec/hevc_mp4toannexb_bsf.c: Optional argument "private_spspps_buf" to avoid extradata modification. [18:52:30 CEST] <jamrial> since libav is pushing for a generalized major bump, anything that could be done on our end to take advantage of the occasion? [18:52:37 CEST] <jamrial> what could use cleaning? [18:53:55 CEST] <nevcairiel> I'm sure there is a few APIs that no longer exist in libav but were kept in ffmpeg that could get another look [18:54:18 CEST] <nevcairiel> and in that thought, I would be happy already if we follow those removals and don't "postpone" half of them [18:54:49 CEST] <jamrial> they posted a patch to postpone most of them again, lol [18:55:15 CEST] <nevcairiel> a bunch of those would ease quite a bit of user confusion as they remove duplicate old APIs, like the old frame allocation and handling stuff [18:55:34 CEST] <nevcairiel> having only the new API will make understanding the API easier [18:56:11 CEST] <jamrial> send a reply to the thread then [18:56:12 CEST] <nevcairiel> I suppose they had a specific timeframe in mind, and only dropped those before that [18:58:11 CEST] <durandal_170> we should drop -deinterlace and audioconvert [18:59:40 CEST] <jamrial> agree. at least the latter [19:00:10 CEST] <nevcairiel> those are incoming from libav removals, at least [19:01:23 CEST] <nevcairiel> half of those are API cleanups and not functionality removals, so they should go in no question [19:01:31 CEST] <jamrial> no, audioconvert is not on libav's side [19:01:38 CEST] <nevcairiel> i'm sure someone will complain loudly about audioconvert or deinterlace [19:01:53 CEST] <nevcairiel> hm i must've misread them, there was one patch removing something audioconvert [19:02:12 CEST] <jamrial> it was a lavu header [19:02:17 CEST] <durandal_170> they removed them before [19:02:24 CEST] <jamrial> not lavc's stuff [19:02:31 CEST] <nevcairiel> ah [19:04:34 CEST] <jamrial> and whoever's using lavc audioconvert should have moved to swr or avr long ago [19:04:57 CEST] <nevcairiel> same with deint really [19:05:09 CEST] <nevcairiel> same with everything in those things deprecated for over 2 years [19:19:28 CEST] <philipl> why would anyone use -deinterlace instead of properly configuring a filter? [19:27:40 CEST] <wm4> is -deinterlace a ffmpeg.c option? [19:28:29 CEST] <durandal_170> yes [19:29:02 CEST] <durandal_170> its awfull deinterlacer IIRC [19:29:05 CEST] <wm4> and it uses the ancient lavc deinterlacer or what? [19:30:20 CEST] <rcombs> no, it just inserts yadif [19:30:45 CEST] <durandal_170> Hmm, you sure? [19:30:57 CEST] <wm4> maybe it was changed at one point [19:31:17 CEST] <wm4> Libav wants to drop the ancient lavc deinterlacer soon [19:31:21 CEST] <wm4> I bet ffmpeg will keep it [19:31:31 CEST] <rcombs> I just read the relevant ffmpeg_filter.c code [19:31:54 CEST] <rcombs> oddly, though, it comes _after_ autorotate [19:32:21 CEST] <durandal_170> didn't libav already removed it? [19:32:39 CEST] <rcombs> (shouldn't autorotate be a filter instead of an ffmpeg[_filter].c option?) [19:32:55 CEST] <rcombs> filter or lavfi thing or something [19:33:13 CEST] <Compn> i thought it was changed to yadif, but a while ago i cant remember... [19:33:45 CEST] <rcombs> 4257b804 [19:34:03 CEST] <rcombs> "ffmpeg: Replace -deinterlace (which was broken by the buffer ref stuff) with yadif injection", march 2013 [19:34:31 CEST] <durandal_170> ah [19:35:38 CEST] <durandal_170> then audioconvert is left [19:38:06 CEST] <rcombs> AFAICT, lavc's audioconvert is currently deprecated public API and isn't used anywhere internally [19:38:59 CEST] <rcombs> so it should just be a matter of removing the .c and .h from the tree and the Makefile, and then the symbols from the .v [19:39:45 CEST] <rcombs> eww, namespace failure here [19:45:59 CEST] <nevcairiel> its not about the difficulty of removing it [19:46:14 CEST] <nevcairiel> but some certain leader liking to keep it for "compatibility" =p [19:55:25 CEST] <cone-787> ffmpeg 03Michael Niedermayer 07master:8c2f00d5909d: ffmpeg.c: remove all remaining coded_frame uses [19:56:39 CEST] <Compn> could just post a patch to remove it [19:56:46 CEST] <Compn> instead of complaining so much about things [19:56:52 CEST] <Compn> wheres the fun in that ? :P [19:58:16 CEST] <durandal_1707> its fun to remove old/unused/bad code [20:13:25 CEST] <jamrial> Compn: we'll after the bump is done/merged :p [20:35:46 CEST] <cone-787> ffmpeg 03Nedeljko Babic 07master:fee7c42bf45f: avcodec/aac_fixed: Fix a bug in spectral_to_sample() [20:54:56 CEST] <cone-787> ffmpeg 03Ivan Uskov 07master:947c2aa45677: libavcodec/qsvenc.c delay in 1 microsecond replaced to more appropriate 500 microseconds [21:10:31 CEST] <jamrial> nevcairiel: there's no sha512 in libav :p [21:11:27 CEST] <nevcairiel> your task is clear [21:11:52 CEST] <nevcairiel> does ffmpeg have them? [21:13:04 CEST] <jamrial> yes, and ffmpeg hmac supports them [21:13:42 CEST] <nevcairiel> oh well [21:51:23 CEST] <cone-787> ffmpeg 03Ivan Uskov 07master:9f543e01afe3: libavcodec/qsvdec.c delay in 1 microsecond replaced to more appropriate 500 microseconds [22:30:26 CEST] <cone-787> ffmpeg 03Martin Storsjö 07master:d75b55635a02: dxva2/d3d11va: Set _WIN32_WINNT to 0x0602 instead of 0x0600 [22:30:27 CEST] <cone-787> ffmpeg 03Michael Niedermayer 07master:77c3908d5cd0: Merge commit 'd75b55635a02444c2f188c26e431a1cec992babe' [22:39:04 CEST] <cone-787> ffmpeg 03Martin Storsjö 07master:c6e0829e3696: configure: Don't force _WIN32_WINNT to an older version if targeting winphone/winrt [22:39:06 CEST] <cone-787> ffmpeg 03Michael Niedermayer 07master:7a0e085ec5fe: Merge commit 'c6e0829e3696c894d24c8088c3ee30a9c17534c2' [23:17:41 CEST] <cone-787> ffmpeg 03Michael Niedermayer 07master:47424b865aa6: avcodec/avcodec: define CODEC_FLAG based on AV_CODEC_FLAG instead of duplicating the numbers and comments [00:00:00 CEST] --- Wed Jul 29 2015

1 0

ffmpeg.log.20150728
by burek 29 Jul '15

29 Jul '15

[00:00:44 CEST] <Nolski> explodes: by using that project's code, you are violating US copyright law [00:01:11 CEST] <explodes> Nolski Firstly, I'm not using it. Secondly, dang. [00:02:06 CEST] <Nolski> explodes: right, not gonna sue you or anything ;) It's normally a misconception that a lot of people on github make. Normally I create an issue letting them know. [00:02:23 CEST] <explodes> Nolski Is there any way to use ffmpeg in android legally? [00:02:45 CEST] <Nolski> android is open source right? [00:03:12 CEST] <explodes> yea, I don't know enough about the relevant licenses to understand what that implies, though [00:06:10 CEST] <explodes> One person in ##android-dev mentioned if I was going to use FFMPEG was to leave out the "parts" covered by the license, can I do video playback without having to adhere to the GPL license? [00:06:35 CEST] <explodes> Really my other options are to go through some other party and pay them to use their video player, which is no fun for me [00:07:42 CEST] <Nolski> explodes: so the GPL essentially says any code that you use it in, you must keep that code open source. [00:07:50 CEST] <Nolski> So if your code is open source, then you can use the GPL :) [00:08:05 CEST] <Nolski> even if your code is MIT, BSD, Apache or whatever. [00:08:16 CEST] <Nolski> If you modify ffmpeg itself, then you must submit those changes upstream [00:09:48 CEST] <c_14> No you don't. [00:10:02 CEST] <explodes> Is FFMpeg entirely GPL? [00:10:07 CEST] <c_14> You just have to release the modified sources in the same channel as the binaries. [00:10:16 CEST] <Nolski> ^ right, my bad [00:10:26 CEST] <c_14> explodes: it is if you use --enable-gpl [00:10:30 CEST] <c_14> Otherwise it's LGPL [00:11:10 CEST] <explodes> LGPL still requires the project to be open source, correct? [00:11:58 CEST] <c_14> If you modify the source you still need to publish the changes to the ffmpeg source under the LGPL, but you can link proprietary code against LGPL code [00:12:12 CEST] <explodes> ok [00:12:54 CEST] <explodes> So I can include ffmpeg builds that aren't built with --enable-gpl as long as I do not modify the source of the ffmpeg library [00:13:50 CEST] <c_14> If you distribute binary versions of ffmpeg you also need to distribute source versions of ffmpeg, but you do not need to release source versions of your program that links against the LGPL ffmpeg [00:16:11 CEST] <explodes> What is a good way to distribute source versions of ffmpeg? [00:16:31 CEST] <explodes> I have a github- can I just provide a fork? [00:17:03 CEST] <c_14> You can usually just put a link to the ffmpeg sources on your download page/ your github readme [00:17:57 CEST] <c_14> ianal though [00:25:33 CEST] <Nolski> c_14: think you could take a quick look at this. adding amix ends up breaking the command https://gist.github.com/Nolski/5877cee63b2f37be3fd6 [00:26:09 CEST] <pzich> looks like you have a trailing colon in your filters [00:26:17 CEST] <pzich> what was the actual error? [00:26:45 CEST] <Nolski> pzich: refresh, I put the output in a comment [00:27:00 CEST] <c_14> You probably need to actially label the pads [00:27:09 CEST] <pzich> like [0:a][1:a]amix... [00:27:13 CEST] <c_14> ye [00:28:05 CEST] <pzich> also, it looks like your first video input doesn't have an audio track? [00:29:11 CEST] <Nolski> pzich: possibly. this is a script so it's overlaying two videos and I want it to overlay the audio as well if there is any (on either video) [00:29:54 CEST] <pzich> I'm not sure if there's a simple way to write this to only mix audio if both inputs have audio, you may need to pre-check that and branch in your code [00:30:18 CEST] Action: Nolski sighs [00:30:28 CEST] <Nolski> I figured that'd be the case. [00:31:11 CEST] <Nolski> not even sure how to figure out how to check that [00:31:48 CEST] <pzich> ffprobe or ffmpeg can tell you the streams in an input [00:31:56 CEST] <Nolski> with -i? [00:31:57 CEST] <pzich> so you need to parse that and then decide how to filter [00:32:18 CEST] <pzich> I don't think you use -i for ffprobe, but yes for ffmpeg [00:33:23 CEST] <Nolski> ah I see [00:33:51 CEST] <pzich> ffprobe also has some different formatting outputs that may make it easier to parse [00:35:25 CEST] <pzich> e.g. `ffprobe <movie> -show_streams -of json` [01:08:51 CEST] <Nolski> pzich: wait I'm a bit confused. ffmpeg is showing that there is a stream for audio but then errors when I try to use it [01:08:56 CEST] Action: Nolski gets paste ready [01:09:51 CEST] <Nolski> https://gist.github.com/Nolski/da11224bf650ba0059a5 [01:10:05 CEST] <Nolski> pzich: ^ [01:16:50 CEST] <blurider> I've been asking for a while but haven't gotten a response. If ffmpeg's compiled with neon instructions, will it work on non-neon systems? [01:17:00 CEST] <c_14> probably not [01:17:08 CEST] <blurider> c_14: Okay, thanks a lot! [01:27:37 CEST] <edoceo> I've got a source video with an invalid duration: 10757 hours. I've transcoded with ffmpeg but it still shows this invalid duration [01:28:13 CEST] <edoceo> Here is the command and output: http://edoceo.io/paste?p=W6OdWN [01:45:31 CEST] <pzich> Nolski: look at the inputs of your second file, Input #1 has a video and audio stream, Input #0 has just video [01:47:16 CEST] <Nolski> pzich: hm... I see [01:47:45 CEST] <Nolski> I was looking at the original files and not the ones i was running ffmpeg on [01:49:59 CEST] <Nolski> ooooh wait. When you scale a video, it doesn't keep the audio track in the output does it? [01:59:10 CEST] <c_14> It should. Depends on if you have a map or not. [04:29:49 CEST] <Mista-D> Has anyone been succesfull in building FFmpeg with QuickSync support on Linux ?? having some build difficulties [08:31:50 CEST] <lomer> hey people ffmpeg -i input.mkv -vcodec copy -acodec copy -ss 00:04:00 -t 00:04:00 output.mkv (Delay relative to video: -651ms) [08:33:09 CEST] <lomer> Delay relative to video a mistake? [09:35:08 CEST] <ruukasu> I have a rip of a blu-ray including "BDMV" and "CERTIFICATE" dirs. at BDMV/STREAM there are .m2ts files, the first two of which each contain two episodes of a show, and the markers for chapters as well as splitting it into two episodes are somewhere else in BDMV [09:35:23 CEST] <ruukasu> is there any way I can convert this strange format into MKVs without transcoding anything? [09:35:41 CEST] <ruukasu> I believe the video in m2ts is h.264 so it should be possible [09:36:45 CEST] <ruukasu> (I'd also like to keep all the audio and subtitle tracks as well as chapter marks, no idea which files those come from) [09:38:03 CEST] <KarlFranz> If it was a DVD, I would use mencoder to extract each track and then mkvmerge to put it all together in an .mkv [09:38:12 CEST] <KarlFranz> But I am nto familiar with blue-rays [09:38:27 CEST] <ruukasu> I think it's a very similar system [09:38:38 CEST] <ruukasu> the weird part is two episodes per title/track/whatever [09:38:52 CEST] <KarlFranz> I have seen weirder things. [09:39:08 CEST] <ruukasu> not sure if there's some kind of chapter-like mark splitting the two episodes [09:39:12 CEST] <KarlFranz> The problem is gonna be if it is an encrypted blu-ray or something like that. [09:39:49 CEST] <ruukasu> nah, I'm viewing it perfectly fine in mpv on gnu/linux, so I'm pretty sure the encryption is gone [09:40:16 CEST] <KarlFranz> Great. [09:40:52 CEST] <ruukasu> if I give mpv the entire directory it has chapter marks, but if I give it the m2ks file directly it does not, so I think the chapter marks come from elsewhere [09:40:55 CEST] <ruukasu> this is strange [11:11:07 CEST] <chungy> Is there a way to set the flag-default property in mkv streams? [11:11:25 CEST] <chungy> I can edit them after-the-fact with mkvpropedit, but it'd be nicer if I can set it directly from ffmpeg. [11:11:54 CEST] <termos> I'm trying to compile ffmpeg with --enable-decklink but --disable-sdl, then I get a linker error saying libdl.so.2: error adding symbols: DSO missing from command line. Is this a common problem? [11:20:09 CEST] <KarlFranz> chungy: I use mkvmerge for that, and it has an option for it. [11:20:26 CEST] <KarlFranz> heh, it looks like I own stocks on mkvmerge or something of the sort. [11:21:37 CEST] <chungy> stocks? [11:23:18 CEST] <KarlFranz> Because I think I have been recommending that software for every mkv related operation since I arrived here. [11:23:40 CEST] <KarlFranz> YOu can also use mkvpropedit for changing the flags of already existing mkv files. [11:31:11 CEST] <chungy> yeah I plan on doing that if ffmpeg can't do it directly. [12:47:11 CEST] <satinder_786> hi I am receiving video udp from network , store it in ring buffer for 10 mins then transmit after 10 mins but my cpu is badly effected . My program use 100% cpu . how I can resolve that issue [12:47:13 CEST] <satinder_786> ?? [13:44:17 CEST] <satinder> hi I am receiving video udp from network , store it in ring buffer for 10 mins then transmit after 10 mins but my cpu is badly effected . My program use 100% cpu . how I can resolve that issue?? [15:45:15 CEST] <kubast2> Hey ,I need my audio to be compressed[my moto g doesn't have sdcard slot]. Which audio codec would you recommend to me? libopus ,libfdk_aac ,libvorbis?[flac would be obvious choice for audio quality ,but I need my audio codec to be space+quality efficient]. [15:45:57 CEST] <JEEB> for the least effort (something you don't necessarily need to build yourself) you should pick either vorbis or opus [15:46:11 CEST] <JEEB> opus is better in really low bit rates, and vorbis is already quite good at 128-192 range [15:47:02 CEST] <sagax> hiall! [15:47:13 CEST] <sagax> please, help me [15:47:38 CEST] <sagax> howto make ffserver streaming with idv3 description [15:47:41 CEST] <sagax> ? [15:48:19 CEST] <sagax> i read official docs, but not finded information about this [15:54:37 CEST] <Fyr> how to convert audio/video only from moment I choose? [15:54:42 CEST] <Fyr> (to moment I set) [15:55:57 CEST] <Fyr> the documentation lacks of it. [15:56:15 CEST] <JEEB> see the documentation of parameters -ss and -t [15:56:22 CEST] <Fyr> ok [15:56:26 CEST] <JEEB> https://www.ffmpeg.org/ffmpeg-all.html [15:56:48 CEST] <Fyr> it's where I'm grepping. [15:57:23 CEST] <JEEB> then you should be able to find those two :) [15:58:10 CEST] <Fyr> I tried "time." [16:14:42 CEST] <sagax> ffserver with idv3 it's impossibe? [16:14:46 CEST] <sagax> impossible* [16:17:30 CEST] <Fyr> is there a raw audio format? [16:17:43 CEST] <BtbN> wav? [16:17:50 CEST] <Fyr> wav is a container. [16:17:57 CEST] <BtbN> for raw pcm. [16:18:10 CEST] <Fyr> with tags [16:18:19 CEST] <j3imi> pcm_s16le? [16:18:23 CEST] <Fyr> is there one without them? [16:18:40 CEST] <BtbN> What are you trying to achive? [16:18:43 CEST] <c_14> -f s16le etc [16:18:49 CEST] <Fyr> ok, thanks [16:19:16 CEST] <j3imi> https://trac.ffmpeg.org/wiki/audio%20types [16:19:24 CEST] <Fyr> thanks [16:19:45 CEST] <Fyr> I'm trying to understand what format Audacity uses. it uses ffmpeg to import WAV. =) [16:20:17 CEST] <BtbN> propably whatever format is in the wav file, unless it supports only one [16:52:05 CEST] <Fyr> is it possible to use only opencl without involving CPU? [16:52:22 CEST] <DHE> no. only certain features are available as opencl [16:52:23 CEST] <Fyr> or at least workload ~1% of CPU? [16:52:28 CEST] <Fyr> =( [16:52:42 CEST] <DHE> ffmepg only supports a few filters. libx264 supports some lookahead features [16:52:54 CEST] <DHE> but for x264 I find opencl only provides ~20% performance improvement by offloading [16:53:10 CEST] <JEEB> wow [16:53:19 CEST] <JEEB> you actually get some performance out of that PoC feature? [16:53:36 CEST] <JEEB> you must have really low latency between your CPU and the thing you're running the OpenCL stuff on [16:53:37 CEST] <Fyr> it's not about performance. [16:53:43 CEST] <Fyr> it's about temperature. =)) [16:53:44 CEST] <DHE> not a lot [16:53:57 CEST] <DHE> at the end of the day, the CPU remains hot and the GPU has seen tougher workloads [16:54:21 CEST] <JEEB> it's a proof of concept feature and it requires fast bandwidth/low latency to be effective [16:54:28 CEST] <JEEB> so I'm kind of surprised :P [16:54:47 CEST] <Fyr> when I'm using openCL-optimized programs, I observe <1% of CPU workload and ~0% of GPU workload. [16:54:54 CEST] <Fyr> I compared ffmpeg and flacCL. the latter is faster than the formar at ~50 times. [16:54:55 CEST] <sagax> howto make ffserver+ffpeg streaming with IDv3 description? [16:55:51 CEST] <Fyr> flacCL can make use of only GPU without burdening CPU. [16:57:55 CEST] <DHE> JEEB: it depends on the input video and settings as well. I just re-ran it with some quick tests. -preset slow gave an improvement, default gave worse performance... [16:58:18 CEST] <DHE> and my last test was over a year ago so I don't remember the specifics. either way... [17:03:58 CEST] <sagax> when ffmpeg started, howto i change metadata? [17:05:19 CEST] <sagax> dynamicle change metadata for ffmpeg, this need for some player might showing title track and other parameters [17:07:07 CEST] <sagax> i find key -metadata for set metadata options, but it's method is static [17:08:06 CEST] <sagax> when i have stream, title must be changing dynamicle, without stoping ffmpeg process [17:08:40 CEST] <sagax> what i can do with this problem? [17:16:21 CEST] <loop3r_> hello, I am trying to remove fisheye effect using ffmpeg, I was googling some time, I found that there are lenscorrection in ffmpeg and defish0r plugin, but I can't get it working, do you know where can I find working examples ?:P [18:24:20 CEST] <Nolski> loop3r_: I'd try looking and seeing if you can find one of the creators or an irc channel for that plugin [18:28:38 CEST] <ocrete> in _getbuffer2(), the height in the AVFrame is not the size of the final image, but seems to include the padding (it give 1088 for a 1080p video), is there some way to get the final image's size? [18:37:40 CEST] <Nolski> why would amix screw up how a video is rendered? [18:42:49 CEST] <loop3r_> Nolski: ok I will try :P [18:45:07 CEST] <durandal_170> Nolski: use pan [18:47:00 CEST] <Nolski> durandal_170: sorry I've never used pan before. The examples seem a bit complex. I'm just looking to have the output of an overlay filter also contain both audio streams in the end product as well [18:49:54 CEST] <Nolski> durandal_170: I think the filter would just be `[0:1][1:1]pan=stereo` if I wanted to mix both channels into one? [19:08:21 CEST] <Eduardo_1> Hello, someone knows how to do 2 pass encoding using nvenc? [19:08:49 CEST] <Nolski> ah wait, if you don't map the audio, it just kinda writes it all over the video... [20:12:17 CEST] <aeikum> hi folks. i'm trying to call libavcodec from wine in order to decode wma v1 & v2 audio into pcm. it's mostly going well, but the decoded audio is corrupt. i think this is because i'm missing the extradata that the decoder expects [20:12:38 CEST] <aeikum> the WAVEFORMATEX that the applications pass in has a cbSize of 0, so there is no extradata to pass in from the format [20:13:01 CEST] <aeikum> however, applications do provide some data with each buffer they submit: https://msdn.microsoft.com/en-us/library/windows/desktop/microsoft.directx_… [20:13:31 CEST] <aeikum> i was wondering if that looks like the correct data, and if so, how to pass that in as extradata after the context has been initialized [20:14:01 CEST] <aeikum> happy to provide data dumps, etc if that would help [20:16:40 CEST] <aeikum> here's an example of an input format and a buffer: http://pastebin.com/MT3ruPch [20:21:05 CEST] <aeikum> also worth mentioning, i get many of these errors: [wmav2 @ 0x65ff15a0] overflow (1866 > 1864) in spectral RLE, ignoring [20:28:30 CEST] <aeikum> oh, just found the comments in >libavformat/xwma.c> about creating some fake extradata. i'll give that a shot [20:47:38 CEST] <aeikum> hey, it works! :D [20:55:47 CEST] <JEEB> 'grats [20:59:39 CEST] <loop3r_> does Dennedy visit irc sometime ? [21:01:14 CEST] <llogan> loop3r_: if that's his nick, then i haven't ever seen him here. [21:04:04 CEST] <loop3r_> llogan: unfortunately I am not sure his nickname on IRC, I wanted to ask about frei0r plugin, https://github.com/ddennedy he is owner of the repo [21:06:35 CEST] <llogan> is that now the official repo? [21:08:03 CEST] <llogan> i think that's a mirror [21:08:20 CEST] <loop3r_> ouu [21:08:24 CEST] <llogan> git://code.dyne.org/frei0r.git [21:09:11 CEST] <llogan> http://frei0r.dyne.org/ [21:09:47 CEST] <loop3r_> Ive visited dyne.org [21:10:03 CEST] <loop3r_> contributors are the same [21:10:32 CEST] <loop3r_> so it doesnt matter at all [21:10:57 CEST] <llogan> if you have questions for them, then use their mailing list [21:11:04 CEST] <loop3r_> I wanted to ask about fisheye effect and lenscorrection, becouse I can't find working example [21:11:50 CEST] <llogan> or #dyne irc.dyne.org:9999 [21:12:10 CEST] <loop3r_> oo this is a really good idea! [21:12:31 CEST] <loop3r_> thanks i will try to join #dyne there [21:13:56 CEST] <llogan> loop3r_: ffmpeg has a lenscorrection filter [21:16:17 CEST] <loop3r_> llogan: I tried http://www.ffmpeg.org/ffmpeg-all.html#lenscorrection but I am not native english speaker, I think I miss something I am doing something wrong there, I thought defish0r might be easier [21:17:37 CEST] <llogan> an example in the docs would be nice; especially showing usage of the mentioned opencv calibration sample [21:27:54 CEST] <loop3r_> llogan: could you summarize to me, how to pass my fisheyed movie through ffmpeg with lenscorrection, I will find more specific by myself just in big summarazie :P [21:30:21 CEST] <llogan> i don't know. i guess if you're lazy you may be able to use blender to give you the k1 and k2 values [21:32:36 CEST] <loop3r_> llogan: I Have to do it from cmd, i will try it all once again [21:32:43 CEST] <loop3r_> anyway thanks for help! [23:09:47 CEST] <Nolski> Hey, think any of yall could help take a crack at this problem? I am overlaying two videos and normally this command works but I'm getting this funky error and it's creating this video thats all garbled up and barely recognizable. [23:09:49 CEST] <Nolski> https://gist.github.com/Nolski/739936423464e7cd1adf [23:22:48 CEST] <klaxa> well first of all, using -preset with -c:v huffyuv doesn't make sense afaik [23:23:15 CEST] <klaxa> also, the only error i see is the out of memory error [23:23:49 CEST] <klaxa> Nolski: ^ [23:23:51 CEST] <Nolski> Right, i was thinking maybe that's related [23:24:39 CEST] <klaxa> other than that the command looks fine to me [23:26:25 CEST] <klaxa> did you try a different codec? [23:26:42 CEST] <klaxa> also, how is ram usage while ffmpeg is running? [23:26:52 CEST] <Nolski> Not bad at all [23:27:23 CEST] <Nolski> klaxa: cpu spikes a bit but memory is fine. I have over 8GB left free [23:27:36 CEST] <klaxa> huh, that's weird [23:28:02 CEST] <Nolski> Yeah, it's a bit strange because the videos start off as webm [23:28:32 CEST] <Nolski> and I encode them with huffyuv because otherwise the end quality looks like crap [23:32:08 CEST] <coops> I've a ffserver time shifting setup that's happily working, but I would like to extract information about the ffm files that it's buffering to. (e.g. their duration, start time, any gaps, etc). Is there a tool to do this, and if not, where can I read about the specs for FFM files? [23:53:14 CEST] <Nolski> klaxa: it turns out the issue is coming from when I concat videos (as part of the overlay process) [23:53:23 CEST] <Nolski> does anything about this command look fishy to you? https://gist.github.com/Nolski/aeb50d0dc9b9558841a6 [23:53:36 CEST] <Nolski> Normally I use filters but I figured it wouldn't be all that different like this [23:53:43 CEST] <klaxa> depends on the content of loop.txt [23:54:05 CEST] <klaxa> and the files referenced within [23:54:06 CEST] <Nolski> All the videos should have the same codec They're all part of the same orginal video just cut up into 3 pieces with the middle one having the overlay [23:54:10 CEST] <klaxa> if the files have different codecs it will probably not work [23:54:11 CEST] <klaxa> ah [23:54:17 CEST] <klaxa> sounds good [23:54:48 CEST] <Nolski> but the middle video gets all screwed up (upside down, major discoloration, etc) after the concat [23:55:05 CEST] <Nolski> so maybe it is a codec thing and I need to re-encode using a filter? [23:56:30 CEST] <klaxa> i don't know, that has never happened to me before [23:56:40 CEST] <klaxa> can you play the individual files without problems? [23:56:58 CEST] <Nolski> klaxa: yup. Prior to the concat, everything plays fine [23:57:12 CEST] <Nolski> it's just that middle video after the concat that gets all screwed up [23:57:47 CEST] <klaxa> can you pastebin ffprobe for all three files? [23:59:35 CEST] <Nolski> klaxa: sure thing: https://gist.github.com/Nolski/6c22e7766689fa1d7b5d [23:59:59 CEST] <Nolski> that middle one says bgra [00:00:00 CEST] --- Wed Jul 29 2015

1 0

ffmpeg-devel.log.20150727
by burek 28 Jul '15

28 Jul '15

[00:14:56 CEST] <cone-084> ffmpeg 03Michael Niedermayer 07master:d41dceb14e06: avcodec/dvbsubdec: Add option to select when to computer clut (always/never/"if needed") [00:14:57 CEST] <cone-084> ffmpeg 03Michael Niedermayer 07master:33c4fc0a2d70: doc/decoders: Add entry for dvbsub and document compute_clut [00:31:52 CEST] <iive> "to computer"? [00:41:00 CEST] <jamrial> iive: he clearly meant compute, so typo [00:47:41 CEST] <iive> it's probably too late to amend [05:00:58 CEST] <cone-621> ffmpeg 03Reynaldo H. Verdejo Pinochet 07master:2ea642ff4b73: ffserver: move decl to start of func [05:00:58 CEST] <cone-621> ffmpeg 03Reynaldo H. Verdejo Pinochet 07master:c75bc268a289: ffserver: simplify assignment with ternary [05:00:58 CEST] <cone-621> ffmpeg 03Reynaldo H. Verdejo Pinochet 07master:b16b8c815c2c: ffserver: drop superfluous else clause [11:02:09 CEST] <ubitux> nevcairiel: btw, the iphone seems to actually be able to decode everything at 240fps [11:02:28 CEST] <ubitux> i see something between 700 to 1100µs [11:02:41 CEST] <ubitux> so about 1ms per frame [11:04:23 CEST] <nevcairiel> what resolution was that at? [11:04:29 CEST] <ubitux> 720p [11:04:40 CEST] <nevcairiel> i guess thats in the realm of possibility then [11:04:54 CEST] <ubitux> but maybe i'm not timing the good thing, let me check again [11:05:22 CEST] <nevcairiel> recent hardware decoders can do 720p at like 1000 fps raw decoding performance [11:05:39 CEST] <nevcairiel> but mobile devices may be memory bandwidth constrained [11:05:46 CEST] <nevcairiel> but 720p at 240 fps seems plausible [11:16:23 CEST] <__gb__> a HW decoder that claims to support H.264 HP @ L5.1 can perfectly do 720p240 [11:17:54 CEST] <nevcairiel> they often dont claim 5.1 officially, since that also includes 4k which they often choke on, although no clue what the iphone says [11:21:43 CEST] <ubitux> anyway, with ffmpeg hwaccel that seems not possible for some reason [11:21:57 CEST] <nevcairiel> works for dxva2 on windows [11:22:01 CEST] <nevcairiel> no experience with anything else [11:22:01 CEST] <nevcairiel> :D [11:22:17 CEST] <ubitux> a call to the decode function takes about 6ms on iOS with the vt accel [11:22:38 CEST] <ubitux> which is at least twice too slow [11:22:46 CEST] <ubitux> :( [11:23:16 CEST] <__gb__> apple claims HP @ L4.2 for iPhone 6+, but some people indicated it could support 4K with "low" bitrates (< 50 mbps) [11:24:31 CEST] <__gb__> based on that, I'd say you should be fine for 720p120 (and up to 140) [11:24:51 CEST] <__gb__> so, if you can't sustain 720p120, then there is room for improvement :) [11:25:24 CEST] <ubitux> the same hw is able to 720p240; i want to achieve the same with ff :p [11:26:36 CEST] <__gb__> ok, fair comparison, but the minimum should be 720p120 I'd say :) [11:34:42 CEST] <__gb__> what kind of video are you testing? raw or with some container that directly exposes avcC data? [11:35:54 CEST] <__gb__> are you measuring only the VT calls? e.g. if you don't have "native" avcC data, maybe some time is spent to actually generate avcC compatible bits for VT [11:35:58 CEST] <nevcairiel> 240 fps sounds like these slowmo iphone videos, so mov/mp4 [11:41:05 CEST] <ubitux> yeah slowmo iphone video; the code i'm looking at is abusing AVFoundation [11:41:26 CEST] <ubitux> playback is at 30Hz, but it's decoding 8 buffers anyway [11:46:15 CEST] <zugzwang> Hi. Could someone point to where can I learn how to add a custom codec to ffmpeg? [11:46:32 CEST] <BtbN> read other encoders/decoder. [11:46:54 CEST] <zugzwang> Is it too difficult? [11:47:06 CEST] <zugzwang> no documentation besides the source code right? [11:47:09 CEST] <BtbN> I'm just not aware of any written documentation. [11:47:30 CEST] <zugzwang> ok [11:48:38 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07master:c719b7a42b3e: ffserver: add (), fix order of operations [11:57:26 CEST] <cone-006> ffmpeg 03Anton Khirnov 07master:f3bd3810d274: qsvdec_*: add missing CODEC_CAP_DR1 [11:57:27 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07master:d12be9ed70f5: Merge commit 'f3bd3810d274a7f51b5925fc3d2fc33e8043a5d4' [12:22:16 CEST] <rcombs> doesn't the 6+ supposedly have some sort of HEVC support as well [12:23:05 CEST] <nevcairiel_> only kind of [12:23:24 CEST] <nevcairiel_> they dont expose it directly, only use it for their video chat thingy iirc [12:25:13 CEST] <rcombs> it was listed on the tech specs page at launch (though only in the FaceTime section, yeah), but seems to have been removed since [12:27:42 CEST] <cone-006> ffmpeg 03Henrik Gramner 07master:65c148015270: checkasm: Modify report format [12:27:43 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07master:b940145c67db: Merge commit '65c14801527068fcaf729eeffc142ffd4682a21a' [12:41:15 CEST] <cone-006> ffmpeg 03Alexandra Hájková 07master:9e8627a1ff92: asfdec: interpret the first flag in an asf packet as length flag [12:41:16 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07master:90696ef36881: Merge commit '9e8627a1ff9207b9e272d248da2e1bd0cc6fe2fe' [12:43:55 CEST] <wm4> that reminds me, what is the VideoToolbox support doing? [12:50:58 CEST] <cone-006> ffmpeg 03Martin Storsjö 07master:6d3081e6c374: doc: Remove the now unnecessary remark about PATH and link.exe [12:50:59 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07master:46f01e413320: Merge commit '6d3081e6c374ff7da12b07ed33d1662be1b32dbc' [13:00:12 CEST] <cone-006> ffmpeg 03Martin Storsjö 07master:e4015b00d4e9: configure: Simplify, remove an unnecessary intermediate variable [13:00:13 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07master:a74e3fc18b56: Merge commit 'e4015b00d4e9e40dc1693a018edd51bf7a04993e' [13:07:57 CEST] <cone-006> ffmpeg 03Martin Storsjö 07master:2192ff84dd72: configure: Default to armasm for --toolchain=msvc when targeting arm [13:07:58 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07master:d9f3efd3d611: Merge commit '2192ff84dd720968108bc1ca54e239f4c94eb61d' [13:18:43 CEST] <cone-006> ffmpeg 03Martin Storsjö 07master:616b409c8f1e: configure: Check MSVC defines for identifying hardfloat [13:18:44 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07master:8324d427c21a: Merge commit '616b409c8f1e4fa568908212c01f6530da8d2e71' [13:32:18 CEST] <cone-006> ffmpeg 03Martin Storsjö 07master:60a21b3d81c1: configure: Check for _M_ARMT to detect thumb when using MSVC [13:32:19 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07master:d06ea6e5ce51: Merge commit '60a21b3d81c1a11cf5a08950eadd4e84ca2e597c' [13:48:27 CEST] <Compn> zugzwang : http://wiki.multimedia.cx/?title=FFmpeg_codec_HOWTO , but its possibly outdated :( [15:06:03 CEST] <ubitux> valgrind seems not happy with the fixed aac; but that's known, right? (i think i saw it mentioned on the ml) [15:21:21 CEST] <kierank> hmmm no ludmila [15:59:32 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07master:42aa02418e43: avformat/mpegts: Use DVB_TELETEXT timestamp heuristic also for DVB subtitles [16:14:24 CEST] <durandal_1707> anybody against pushing acrossfade? [16:55:38 CEST] <kurosu> are the other fixed point decoders (ac3, anything from imgtech, basically) ok either? [16:55:54 CEST] <kurosu> I remember fixing some stuff in eac3, but I never ran valgrind on those [17:25:30 CEST] <cone-006> ffmpeg 03Shivraj Patil 07master:71aede3ced76: avcodec/mips: MSA (MIPS-SIMD-Arch) optimizations for VP9 bilinear functions [17:50:06 CEST] <JEEB> is duration - start_time the effective duration of the clip as long as start_time isn't negative? in which case those are leading samples etc? [18:16:27 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07master:c8c86b8f9b8c: avformat/mpegts: Replace silent cliping of language_count by asserts [18:26:36 CEST] <nevcairiel> JEEB: duration should not include the start_time [18:26:49 CEST] <nevcairiel> its always the duration [18:26:51 CEST] <JEEB> ok [18:27:18 CEST] <nevcairiel> so duration+start_time is the highest timestamp in the filer [18:27:19 CEST] <nevcairiel> -r [18:27:22 CEST] <nevcairiel> theoretically anyway [18:27:43 CEST] <JEEB> yea, makes sense [18:55:38 CEST] <BtbN> michaelni, what's the policy on backporting stuff to stable branches? I'd like to backport the nvenc performance fix to 2.7. (commit 9f4bff834c) [19:12:03 CEST] <michaelni> BtbN, bugfixes should be ok [19:17:41 CEST] <BtbN> It increases the encoding performance drastically, and makes some 4K stuff possible in realtime that formerly it was too slow for. [19:18:41 CEST] <nevcairiel> technically its not a stable-worthy change [19:19:01 CEST] <BtbN> yeah, that's why i'm wondering. [19:19:02 CEST] <Compn> BtbN : are you going to test the patch before commit with the release ? [19:19:08 CEST] <Compn> if so , its fine with me [19:19:11 CEST] <michaelni> nevcairiel, about ff_alloc_packet2(), should i add a flag argument or can the patch be applied or you prefer something else ? [19:19:46 CEST] <BtbN> Compn, yeah, i'd do some extended testing with it. At least as much as my hardware allows. [19:20:05 CEST] <BtbN> Can't test h265 stuff [19:20:24 CEST] <Compn> BtbN : i think "fixing" 4k is an important enough fix to backport :) [19:21:32 CEST] <BtbN> it's not an overly complicated functional change. Basicaly it's just waiting a bit longer before reading frames from the gpu [19:22:01 CEST] <nevcairiel> michaelni: i still find simply null-ing the context for a behaviour change quite odd api, but its internal api so i don't really care that much, if you think its fine just push it [19:22:37 CEST] <BtbN> Though i wonder if anybody actualy using nvenc doesn't use master anyway [19:29:13 CEST] <cone-006> ffmpeg 03Claudio Freire 07master:59216e0525a5: AAC Encoder: clipping avoidance [20:19:13 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07master:e36db49b7b31: avcodec: Add a min size parameter to ff_alloc_packet2() [20:19:14 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07master:0dbfb5386f1e: avcodec/internal: Deprecate ff_alloc_packet() in favor of ff_alloc_packet2() [20:19:15 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07master:9fe873bec8b6: avcodec/utils: do not use internal->byte_buffer when little downsizing is expected [20:19:16 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07master:af5f707e46f0: avcodec/v410enc: do not use internal->byte_buffer [20:22:58 CEST] <jamrial> michaelni: why deprecate? it's internal [20:27:04 CEST] <michaelni> i can revert if preferred but i thought people want consistency [20:30:02 CEST] <BtbN> I don't see a problem with deprecating internal functions. Helps cleaning up. [20:30:33 CEST] <michaelni> yep that was the idea [20:41:19 CEST] <cone-006> ffmpeg 03Vittorio Giovara 07master:a67b67944aa9: ac3enc_template: Use the correct context field [20:41:20 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07master:c7074375724e: Merge commit 'a67b67944aa9e6e794934d15f9fd9a9cf7173e09' [20:51:23 CEST] <cone-006> ffmpeg 03Vittorio Giovara 07master:03eb55741427: wmv2enc: Check memory allocation [20:51:24 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07master:59b009411fcc: Merge commit '03eb55741427c6608f63972c105e565ca0ba4f15' [21:14:17 CEST] <cone-006> ffmpeg 03Vittorio Giovara 07master:4b6b1082a739: lavc: Deprecate avctx.me_method [21:14:18 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07master:0b6f092ed31b: Merge commit '4b6b1082a73907c7c3de2646c6398bc61320f2c6' [21:52:53 CEST] <cone-006> ffmpeg 03Paul B Mahol 07master:4a2836eaf33b: avfilter: add acrossfade filter [21:57:18 CEST] <philipl> Oh, stagefright. [21:57:19 CEST] <philipl> http://arstechnica.com/security/2015/07/950-million-android-phones-can-be-h… [22:10:05 CEST] <BtbN> Well, better than the cars that can be hacked and remote controled by malicious radio broadcasts. [22:10:33 CEST] <philipl> There' [22:10:43 CEST] <philipl> s probably at least one car that's vulnerable to this one :-P [22:11:10 CEST] <J_Darnley> Wait... isn't that a video decoding lib? What's it doing with text messages? [22:11:42 CEST] <JEEB> previews [22:11:47 CEST] <J_Darnley> oh, an MMS [22:11:58 CEST] <JEEB> and yes [22:20:14 CEST] <cone-006> ffmpeg 03Vittorio Giovara 07master:7c6eb0a1b7bf: lavc: AV-prefix all codec flags [22:20:15 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07master:94d68a41fabb: Merge commit '7c6eb0a1b7bf1aac7f033a7ec6d8cacc3b5c2615' [22:53:16 CEST] <cone-006> ffmpeg 03Vittorio Giovara 07master:def97856de60: lavc: AV-prefix all codec capabilities [22:53:17 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07master:444e9874a75c: Merge commit 'def97856de6021965db86c25a732d78689bd6bb0' [22:56:18 CEST] <lglinskih> kierank: hi! About draw_horiz_band-test: now it works for h263 and huffyuv (yuv420p and yuv422p), but only for decoding in one thread [22:58:34 CEST] <wm4> lglinskih: what happens with multiple threads? [23:03:18 CEST] <lglinskih> wm4: mmm... I use only one buffer to store results of draw_horiz_band... as I understand draw_horiz_band is called from different threads in which different frames are decoded [23:04:06 CEST] <wm4> so it's not known yet whether the lavc API works in multithreaded mode [23:07:43 CEST] <lglinskih> wm4: it works! it really corrupts my buffer=) [23:08:33 CEST] <wm4> I'm not convinced whether it can be used [23:08:37 CEST] <wm4> the API I mean [23:08:43 CEST] <wm4> (in multithreaded mode) [23:20:20 CEST] <cone-006> ffmpeg 03Vittorio Giovara 07master:059a934806d6: lavc: Consistently prefix input buffer defines [23:20:21 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07master:29d147c94dd4: Merge commit '059a934806d61f7af9ab3fd9f74994b838ea5eba' [23:28:56 CEST] <rcombs> philipl: the first time I saw anything about that, they were calling the vuln itself "stagefright" and I got mad about dumb naming collisions [23:34:27 CEST] <cone-006> ffmpeg 03Vittorio Giovara 07master:b94ec30428b9: lavc: Update version and APIchanges [23:34:28 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07master:e3ec2cde2c39: Merge commit 'b94ec30428b9696f99b08055735689623fe63954' [23:45:46 CEST] <cone-006> ffmpeg 03Tom Butterworth 07master:43dd004747fa: hap: Move some per-stream setup into decoder init rather than per-frame [23:45:47 CEST] <cone-006> ffmpeg 03Michael Niedermayer 07master:f01e3c5d000d: Merge commit '43dd004747fa697396b47d034a80e069facbea09' [00:00:00 CEST] --- Tue Jul 28 2015

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ffmpeg.log.20150727
by burek 28 Jul '15

28 Jul '15

[00:15:38 CEST] <TikityTik> what happens when i do ffmpeg -i thefile.mp4 out.mp4 [00:15:54 CEST] <TikityTik> does it encode at all or just copies the data as is? [00:15:57 CEST] <c_14> encodes [00:16:40 CEST] <TikityTik> does it try to keep the same quality and bitrate? [00:17:08 CEST] <c_14> nope [00:17:14 CEST] <c_14> just uses whatever the default is [00:20:32 CEST] <well0ne> Hi [00:20:34 CEST] <well0ne> Hey Canar [00:20:39 CEST] <well0ne> I solved it [00:20:46 CEST] <well0ne> with php and ffmpeg [00:22:03 CEST] <well0ne> wanna see? [00:28:47 CEST] <chungy> TikityTik: use -c copy if you want to copy [00:29:13 CEST] <chungy> and by default it only does one video stream, one audio, and one subtitle. can use -map to control those further :P [00:29:13 CEST] <TikityTik> chungy: and it will copy the quality for all channels? [00:29:24 CEST] <chungy> "-c copy" just copies the streams, not encode [00:29:30 CEST] <chungy> it'd be identical to the original [00:30:11 CEST] <TikityTik> chungy: thanks so much [00:30:21 CEST] <TikityTik> for some reason OBS doesn't make headers properly for recordings [00:30:54 CEST] <chungy> If you have multiple video/audio/sub streams and you want them all, "-map 0" is handy too [00:31:01 CEST] <chungy> that forces it to copy everything :P [00:31:46 CEST] <TikityTik> chungy: it will copy attachments as well? [00:32:21 CEST] <chungy> unsure. make a test. [00:32:51 CEST] <c_14> not by default [00:32:58 CEST] <c_14> if you want to map everything just use -map 0 [00:33:01 CEST] <chungy> if it's a very large file, you might try using "-t 10" to limit it to the first 10 seconds or something. Gives a good hint of how things turn out [00:33:44 CEST] <TikityTik> what's wrong with just stopping early? [00:34:43 CEST] <c_14> If you use q or ^C, nothing afaik. [00:36:11 CEST] <chungy> I think two ^Cs makes it abort before a valid file is written out, but generally, same effect. [01:09:29 CEST] <TikityTik> how do you know what bufsize to use? [01:16:57 CEST] <c_14> https://trac.ffmpeg.org/wiki/EncodingForStreamingSites [01:17:02 CEST] <c_14> Check the -bufsize header [01:20:58 CEST] <TikityTik> c_14: does that apply to videos instead? [01:23:46 CEST] <c_14> It shouldn't matter [01:23:59 CEST] <c_14> bitrate * time_for_buffer [01:24:11 CEST] <c_14> Though I'm not sure the bufsize option applies to audio [02:12:30 CEST] <TikityTik> I just realized you can make gifs with ffmpeg [02:12:34 CEST] <TikityTik> WOW [09:21:02 CEST] <user_> Hello [09:26:27 CEST] <user_> I have a problem with a transport stream copy (hls,applehttp). The copy is not edible with avidemux. I want only merge/cut segments and I don't want do a re-encoding. I googled already a lot but can't find a hint. http://pastebin.com/gt4AaQYi Any suggestion how to proceed to get the recording edible ? [09:32:45 CEST] <user_> the raw copy is barely play-able with VLC. VLC lose audio after a wile [09:37:39 CEST] <user_> I tried to remux the .ts to matrosca but the audio stream is not correct recognized. (error in the pastebin) [09:39:29 CEST] <user_> The transport stream support of avidemux (2.6.8) is bad. No edit possible. [09:41:53 CEST] <user_> "Error parsing AAC extradata, unable to determine samplerate" is the error when I try to remux to mkv. [09:43:23 CEST] <user_> Any suggestion how to force the known ACC key data (sample rate, channels etc.)? [09:45:50 CEST] <user_> http://pastebin.com/n9dQfYEZ is paste to ffmpeg.pastebin.com [09:46:24 CEST] <user_> Would a short segment of the ts stream help ? [09:47:17 CEST] Action: KarlFranz checks [09:47:47 CEST] <satinder> hi , I have problem with ffmpeg when I streaming a HD video file from my pc to network . Acually I make a udp stream of recorded HD file but when I play that udp on vlc that is not playing proper . anybody help me please ?? [09:48:40 CEST] <KarlFranz> YOu are going to need to define the problem better than that. [09:48:58 CEST] <satinder> my command is following : ffmpeg -i video_hd.ts -vcodec copy -acodec copy -f mpegts udp://@227.40.50.60:1234 [09:49:12 CEST] <satinder> video streaming is to much fast [09:49:38 CEST] <satinder> due to this reason video not playing proper on the network [09:50:26 CEST] <satinder> please help ?? [09:51:32 CEST] <user_> Come on satfinder, this is not the way to ask for help. You have to provide more details on a pastebin. [09:52:25 CEST] <satinder> ok sir [09:52:25 CEST] <satinder> I will [09:53:05 CEST] <KarlFranz> user_: You are sure the original stream source is healthy, are you? [09:53:29 CEST] <user_> Define healthy please [09:54:16 CEST] <user_> I can upload a couple of seconds ? [09:55:13 CEST] <KarlFranz> I usually use mkvmerge for creating *.mkv files from streams, but I doubt it is a problem in ffmpeg's muxing in this case. [09:56:41 CEST] <KarlFranz> user_: "Healthy; the file is not broken" [09:57:03 CEST] <user_> Well, the goal is to get 3-4 stream segment merged and cut without re-encoding. The modification on the .ts need to be just enough to make avidemux happy [09:59:04 CEST] <user_> The stream comes from a content distribution network (CDN) that deliver to a flash stream player application. [09:59:20 CEST] <satinder> Actually I want receive a udp and record it and the transmit it after some delay (10 mins delay) there have any parameter in ffmpeg for receive udp with delay [09:59:40 CEST] <user_> The frame rate is dynamic in a wide range. 5 to 30 fps. [10:03:51 CEST] <user_> I will upload 10 seconds dropcanvas [10:04:47 CEST] <KarlFranz> user_: Have you tried setting the sample_rate by hand? If you are sure about the sample rate of the input file it could do the trick. [10:04:59 CEST] <KarlFranz> I think it is the -ar option for audio. [10:05:19 CEST] <user_> the input file sample rate is variable. [10:05:33 CEST] <KarlFranz> My bad. [10:05:36 CEST] <user_> sorry, [10:05:46 CEST] <user_> not the audio part [10:11:27 CEST] <user_> Here are 20 seconds = 1.8 mb http://dropcanvas.com/um9f0 [10:16:01 CEST] <satinder> Hi but I want 10 mins delay [10:16:15 CEST] <satinder> input to output [10:21:38 CEST] <user_> mediainfo (output also in the pastebin) recognize the audio stream as: Format : AAC / Format version : Version 4 / Format profile : LC / Bit rate mode : Variable / Sampling rate : 16.0 KHz [10:22:58 CEST] <KarlFranz> Damn, it is behind cloudflare. Wait a second. [10:23:20 CEST] <user_> the tor user.... [10:26:19 CEST] <user_> mplayer output: AUDIO: 16000 Hz, 1 ch, floatle, 47.5 kbit/9.28% (ratio: 5937->64000) [10:29:01 CEST] <user_> I think the 'floatle' is the problem. [10:32:51 CEST] <user_> @satfinder: no delay in this dimensions without tmp storage in a file [10:45:22 CEST] <user_> This is the avidemux error with the ffmpeg stream copy: http://ibin.co/2A3WNSLPWjvj [10:50:17 CEST] <KarlFranz> Is the ts file supposed to show a static image while playing music? Because that is what I get. [10:50:46 CEST] <KarlFranz> Plus there is no known duration for the file as far as my player is concerned. [10:51:25 CEST] <user_> no, normally not. but I can't upload stuff with people in it without permissions [10:53:02 CEST] <KarlFranz> Using mkvmerge on the ts works with no problems. [10:53:12 CEST] <KarlFranz> I get a 21s *mkv file [10:53:27 CEST] <KarlFranz> The tracks are jus copied with no recoding. [10:53:44 CEST] <user_> What command line ? [10:54:14 CEST] <KarlFranz> mkvmerge -o test.mkv test_ts.ts :-) [10:54:45 CEST] <KarlFranz> You might want to add some metadata, like a tittle or something like that, but the basic just works for me. [10:55:24 CEST] <KarlFranz> You might want to specify the fps of the video manually. [10:56:08 CEST] <user_> I try this. One moment [11:02:01 CEST] <satinder> ok , thank you for giving response [11:02:43 CEST] <satinder> one another thing which I want discuss with [11:03:03 CEST] <satinder> I receive udp in a temp file [11:03:32 CEST] <satinder> after 10 mins I transmit that file with ffmpeg [11:04:12 CEST] <satinder> but during making the udp from .ts file the transmission rate is very fast [11:04:42 CEST] <user_> KarlFranz: cut and merge works with the "mkvmerge -o test.mkv test_ts.ts" files. PERFECT. Thank you. :-D [11:04:43 CEST] <satinder> due to this reason my video is scattering at the receiver end [11:06:17 CEST] <user_> So, to take home: ffmpeg's mkv muxer is buggy. [12:03:59 CEST] <zugzwang> Is there a way to read from a named pipe in windows? [12:04:13 CEST] <zugzwang> ffmpeg -i \\.\pipe\pipename does not work [13:20:48 CEST] <satinder> Hi how we can calculate the bit rate of video udp on network ?? [14:20:15 CEST] <BtbN> record it for X seconds, devide size by X [14:41:21 CEST] <DHE> I think wireshark has a statistic view you can use in the GUI... [15:11:15 CEST] <TikityTik> having troubles with -ss and bit stream coying [15:11:50 CEST] <TikityTik> if i try input seaking, it isn't accurate seeking. And when I use output seeking, it freezes on the first frame of the video for a second. [15:20:01 CEST] <DHE> that does sound right if the video doesn't start with a keyframe. stream copying must start with a keyframe so you can either start with the nearest keyframe (inaccurate start) or have the video frozen for a moment (accurate start, bad codec initialization) [15:22:36 CEST] <TikityTik> DHE: is it bad to force keyframes at every second? [15:23:21 CEST] <DHE> that's an encoding thing. if you're starting with an existing video and not going to reencode it, you're outta luck [15:23:47 CEST] <DHE> as for forcing keyframes, it has a negative impact on bitrate [15:24:19 CEST] <TikityTik> DHE: how can i get the subtitles/ass video filter to work with seeking? [15:29:09 CEST] <elmargol> I have to reencode some gopro videos. Is it safe to use -crf and a maxrate at the same time? [15:35:38 CEST] <DHE> elmargol: those are mutually exclusive. try -crf first, it'll probably do okay for you [15:35:56 CEST] <DHE> higher number for lower bitrate, lower number for better image quality [15:41:03 CEST] <TikityTik> elmargol: you probably want to set -bufsize and -b:v if you're going to use -crf [15:42:20 CEST] <TikityTik> DHE: why isn't the ass filter working? [15:42:28 CEST] <TikityTik> is it because i'm using -to? [15:44:37 CEST] <elmargol> DHE, for a 720p 50fps video what is a "normal" filesize / minute? [15:44:55 CEST] <elmargol> If I use crf 23 the files are HUGE [15:46:11 CEST] <elmargol> If I do crf 23 i get rates 10.000+ for 720p isn't that insane? [15:46:12 CEST] <TikityTik> Greater than 1500 kbps if you're using 50 fps [15:46:29 CEST] <TikityTik> you could try 15000 kbps [15:52:10 CEST] <DHE> elmargol: I would ballpark 500 kilobytes/second for "looks quite good", but it will vary by content [15:52:15 CEST] <DHE> TikityTik: dunno, I don't do subtitles [15:52:22 CEST] <TikityTik> I got it working [15:52:48 CEST] <TikityTik> my fontconfig was broken [16:05:29 CEST] <Fjorgynn> must I compile my own version of ffmpeg for debian? [16:11:35 CEST] <elmargol> Fjorgynn, I'd compile yes [16:11:44 CEST] <elmargol> use a fresh checkout from git [16:13:16 CEST] <Fjorgynn> any good compile instructions? [16:20:42 CEST] <elmargol> apt-get build-dep <- to install all tools needed [16:22:04 CEST] <elmargol> http://www.zoharbabin.com/build-and-install-ffmpeg-and-x264-on-debian-squee… <- found on google [16:29:35 CEST] <termos> I'm trying to compile ffmpeg with support for decklink capture cards and the ffmpeg program can read from it just fine. But when I try to open it with the C api it can't find the format with av_find_input_format. Is there some linking that's wrong? [16:43:43 CEST] <Fjorgynn> wierd [16:49:57 CEST] <Mista-D> Problems building FFmpeg 2.7.2 with libmfx : http://pastebin.com/J0CUHpYe [18:03:09 CEST] <wizbit> is it possible to use ffmpeg to check the integrity of mp3 / flac and ogg files? [18:09:52 CEST] <durandal_1707> what you mean by integrity? [18:10:22 CEST] <durandal_1707> flac have checksums others not [18:13:53 CEST] <wizbit> i guess i could just use these tools: mp3val / flac -s -t / ogginfo -v [18:24:29 CEST] <Nolski> Is there a way to ensure that all ffmpeg commands are run with lossless encoding? [18:25:59 CEST] <Nolski> when I do -c:v with a lossless encoding I always get this error for opening the encoder for output stream 0:0 [18:45:20 CEST] <durandal_1707> wizbit: that can do ffmpeg too.... [21:13:15 CEST] <ocrete> in _getbuffer2(), the height in the AVFrame is not the size of the final image, but seems to include the padding (it give 1088 for a 1080p video), is there some way to get the final image's size? [21:14:27 CEST] <JEEB> like the "width" and "height" parts of AVFrame? [21:14:36 CEST] <JEEB> http://ffmpeg.org/doxygen/trunk/structAVFrame.html [21:20:11 CEST] <bjvl> Is ffmpeg supporting m3u8 with mp4 fragments? [21:20:38 CEST] <BtbN> you mean DASH? [21:21:06 CEST] <BtbN> Is HLS with mp4 files even well defined? [21:21:20 CEST] <JEEB> you can just look at the spec and take a wild guess [21:22:15 CEST] <edoceo> I've got a video that ffprobe reports as " Duration: 3579:11:45.21, start: 0.000000, bitrate: 0 kb/s" [21:22:24 CEST] <edoceo> I'm sure it's not that long [21:22:39 CEST] <llogan> ffmpeg -i input -f null - [21:22:46 CEST] <llogan> then refer to time= in console output [21:22:55 CEST] <llogan> is it the same? [21:23:38 CEST] <bjvl> BtbN: I've a situation with m3u8 but mp4 fragment emitted by the server instead of ts [21:24:06 CEST] <bjvl> bjvl: but ffmpeg try to probe for mpegts [21:24:47 CEST] <BtbN> what kind of server? [21:25:04 CEST] <edoceo> llogan: I see this "[null @ 0x12369b0] Encoder did not produce proper pts, making some up" [21:25:31 CEST] <edoceo> A lot of DTS / PTD invalid/dropping messages [21:25:59 CEST] <edoceo> Then: frame= 233 fps=0.0 q=0.0 Lsize=N/A time=00:02:39.37 bitrate=N/A [21:26:39 CEST] <edoceo> When I transcode this webm source into an mkv container, the invalid duration persists [21:26:53 CEST] <edoceo> How can I get ffmpeg to fix it? [21:28:26 CEST] <edoceo> the ffprobe: http://edoceo.io/paste?p=fRbHju [21:29:27 CEST] <llogan> what about your encoding command and output? [21:29:41 CEST] <llogan> also, try a build from current git master before doing anything. [21:30:15 CEST] <llogan> that's quite the configure. why do gentoo users always have such an odd configure? [21:32:26 CEST] <c_14> Because the ebuild lets you turn everything on/off and there's no setting for use whatever configure defaults to [21:33:04 CEST] <llogan> huh. sounds dumb. [21:33:19 CEST] <c_14> It could be much, much better. [21:34:39 CEST] <bjvl> BtbN: The http server inside Canon legria series camcoder [21:35:07 CEST] <blurider1> If I compile ffmpeg with --enable-neon, will it work on cpus without neon support? [21:35:35 CEST] <edoceo> llogan: I also use the static from from John's website, same issue as my Gentoo box [21:36:04 CEST] <BtbN> c_14, useflags are either on or off, and cause some configure setting to be enabled or disabled [21:36:12 CEST] <BtbN> it's also trying to make the build as reproducible as possible. [21:36:24 CEST] <BtbN> So, rule out all magic auto-detection [21:36:41 CEST] <llogan> and non-neckbeards. [21:38:37 CEST] <edoceo> So, Gentoo users are neckbeards? [21:39:07 CEST] <llogan> i forgot about LFS, so I guess not. [21:42:56 CEST] <edoceo> Here's a pastebin showing the command and output: http://edoceo.io/paste?p=W6OdWN [21:43:12 CEST] <edoceo> Even after the processing the time is: =3579:11:44.91 [22:13:38 CEST] <edoceo> llogan: ^^^ see anything obvious in my paste? [22:20:46 CEST] <Nolski> is there a flag i can give ffmpeg to just overwrite video files if they exist without asking? [22:20:52 CEST] <c_14> -y [22:25:46 CEST] <Nolski> thanks c_14 :) [22:28:51 CEST] <z3bra> Hi [22:29:23 CEST] <z3bra> I request your help, ô ffmpeg gurus [22:30:03 CEST] <z3bra> chrome decided not to read my webms. Some guy working on chrome told me that's because my video has the metadata "alpha_mode" set to 1 [22:30:13 CEST] <z3bra> I'm now trying to remove it [22:30:46 CEST] <z3bra> I already tried -metadata alpha_mode=0 and -vf il=alpha_mode=none [22:30:50 CEST] <z3bra> both unsuccessful [22:31:12 CEST] <c_14> Is it global metadata or stream metadata? [22:31:21 CEST] <z3bra> stream [22:31:36 CEST] <c_14> -metadata:s:0 alpha_mode= [22:31:45 CEST] <c_14> 0 [22:32:44 CEST] <z3bra> nope, still set to 1 [22:33:06 CEST] <z3bra> http://sprunge.us/IdBP [22:33:41 CEST] <z3bra> the command I used to record the webm is [22:33:44 CEST] <z3bra> ffmpeg -f x11grab -s 1440x900 -an -i :0.0 -c:v libvpx -b:v 5M -crf 10 -quality realtime text.webm [22:35:46 CEST] <relaxed> z3bra: it doesn't happen here, can you see if it happens using http://johnvansickle.com/ffmpeg/ ? [22:35:59 CEST] <c_14> ^ [22:36:08 CEST] <relaxed> or build a recent version [22:37:29 CEST] <z3bra> give me a second [22:38:42 CEST] <z3bra> looks better [22:40:50 CEST] <loop3r_> hello, I am trying to remove fisheye effect using ffmpeg, I was googling some time, I found that there are lenscorrection in ffmpeg and defish0r plugin, but I can't get it working, do you know where can I find working examples ?:P [22:48:16 CEST] <z3bra> it worked well, thanks everyone! [22:48:41 CEST] <z3bra> now I need to figure out what's wrong with my ffmpeg [22:49:16 CEST] <relaxed> z3bra: could just be an outdated libvpx [22:49:35 CEST] <z3bra> 1.3.0 [22:49:39 CEST] <z3bra> what's the latest? [22:49:53 CEST] <relaxed> my builds have 1.4.0-865 [22:50:01 CEST] <z3bra> okay [23:41:30 CEST] <DHE> ffmpeg logging to syslog... good idea? (I'd need to add it, but I can do that) [23:43:22 CEST] <pzich> with normal ffmpeg log level? seems sorta verbose [23:56:50 CEST] <DHE> pzich: maybe set a minimum loglevel. I would probably choose anything above the live encoder stats output [23:57:22 CEST] <Nolski> does the overlay filter also apply to audio? [23:58:01 CEST] <c_14> Nolski: no [23:58:27 CEST] <Nolski> c_14: is there a way I can overlay the audio and video? [23:58:39 CEST] <c_14> Define overlay audio [23:58:57 CEST] <DHE> https://ffmpeg.org/ffmpeg-filters.html#amix [23:59:00 CEST] <DHE> I'm guessing this is what you want [23:59:28 CEST] <Nolski> so I have videoA on the bottom and videoB also has an audio track that I want the user to hear when output video is played [23:59:58 CEST] <c_14> amix is probably what you want [23:59:58 CEST] <explodes> Hey guys. There is no license information in this whole project: https://github.com/nxtreaming/FFmpegAndroid - I'm trying to avoid violating the GPL. Does anyone have any remarks as to whether or not my app will need to conform to the GPL if I include this library in my project? [00:00:00 CEST] --- Tue Jul 28 2015

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ffmpeg-devel.log.20150726
by burek 27 Jul '15

27 Jul '15

[00:00:33 CEST] <nevcairiel> I had the same feeling when I was working on decoding at some point [00:00:43 CEST] <nevcairiel> just delaying access to the frame and it becomes much faster [00:02:40 CEST] <BtbN> hm, can't think of a good name for that parameter [00:02:48 CEST] <nevcairiel> delay [00:03:38 CEST] <BtbN> hm, yeah [00:04:29 CEST] <philipl> nevcairiel: aren't the decoders conceptually stateless? [00:05:13 CEST] <nevcairiel> from the API sure, but if you access the frame too early you probably enforce an internal sync of some sort, which blocks the parallelism [00:06:12 CEST] <philipl> a fair point [00:06:14 CEST] <nevcairiel> basically, you get a frame handle back for the resulting frame, but the decoder is still working on it, so if you delay it 1-2 frames... more speed [00:06:30 CEST] <philipl> Yes, makes sense. Basic pipelining [00:07:07 CEST] <nevcairiel> i read some tips from nvidia on that, they basically explained it with decoding being in multiple stages, and if you do it that way, you can have different frames in the different stages [00:07:15 CEST] <nevcairiel> which makes sense [00:07:20 CEST] <philipl> fur sure. [00:07:46 CEST] <cone-201> ffmpeg 03Paul B Mahol 07master:c1fa846d0c23: avfilter: add sidechain compress audio filter [00:07:47 CEST] <cone-201> ffmpeg 03Paul B Mahol 07master:62aab839f805: avfilter/Makefile: fix order of framestep filter [00:07:52 CEST] <nevcairiel> encoding probably works similar [00:08:03 CEST] <nevcairiel> so keeping a buffer before accessing the result makes it faster [00:08:27 CEST] <philipl> Yeah. [00:21:57 CEST] <BtbN> philipl, https://github.com/BtbN/FFmpeg/commit/a67f9ac874495a616f9d88a9b266e0ec8ccff… [00:22:45 CEST] <philipl> BtbN: default to 0? [00:23:02 CEST] <BtbN> that'd be the old slow behaviour [00:23:30 CEST] <philipl> Sure, but you want to keep that as the default? [00:24:05 CEST] <BtbN> I think it makes more sense this way around [00:24:26 CEST] <BtbN> propably also puts less stress on the hardware [00:25:17 CEST] <BtbN> Nobody would ever add some delay, people only want less delay. [00:25:32 CEST] <BtbN> So having it in the fast mode by default seems better [00:26:27 CEST] <philipl> BtbN: Hmm. [00:26:34 CEST] <philipl> The default doesn't seem to acting as if its zero [00:26:46 CEST] <BtbN> hm? [00:26:50 CEST] <philipl> I'm confused. [00:26:54 CEST] <nevcairiel> the default is INT_MAX [00:26:57 CEST] <nevcairiel> it says so [00:27:04 CEST] <philipl> oh, sorry. I read it wrong. [00:27:11 CEST] <philipl> That's why I was confused. [00:27:18 CEST] <philipl> I thought you defaulted it to zero [00:27:44 CEST] <philipl> No wonder we were both so confused. [00:30:09 CEST] <BtbN> I'll do some testing with it myself, but i think this should be fine, if you have no objections, i'll push it once i'm done. [00:30:20 CEST] <philipl> Looks good to me, works as advertised. [00:33:26 CEST] <philipl> BtbN: From a quick check, I think the hardware is happy once the delay is at least 3 frames. [00:33:34 CEST] <cone-201> ffmpeg 03Anton Khirnov 07master:96dca089b19b: qsvdec: move qsv_process_data() from qsvdec_h264 to the common code [00:33:35 CEST] <cone-201> ffmpeg 03Michael Niedermayer 07master:a822c3bd7974: Merge commit '96dca089b19bf55293136277f5b081c56146e10e' [00:36:27 CEST] <philipl> nvidia-settings has a video engine utilization percentage these days. With delay 0 it says 22% for my example, and with max delay it's 77% [00:41:57 CEST] <BtbN> philipl, even with my kepler card, transcoding a simple test clip, 1080p, i get 60 fps with -delay 0, and 90 without [00:42:04 CEST] <BtbN> (all nvenc settings on default) [00:42:43 CEST] <cone-201> ffmpeg 03Anton Khirnov 07master:f89f78c1c563: lavc: add a HEVC mp4->annex B bitstream filter [00:42:45 CEST] <cone-201> ffmpeg 03Michael Niedermayer 07master:15bcbc9d3b88: Merge commit 'f89f78c1c563d98f10ee1d7e1ed67c9f9e03b741' [00:43:05 CEST] <durandal_1707> libav bugtracker forbids to peek at crashes how neat... [00:44:19 CEST] <BtbN> what? [00:50:06 CEST] <kierank> durandal_1707: huh? [00:50:57 CEST] <philipl> BtbN: And I got 99% utilization from a 4k encode. [00:52:34 CEST] <durandal_1707> kierank: it even doesn't list it I guess one need to register [00:54:13 CEST] <durandal_1707> this .needs_fifo sucks for audio when 2nd stream reaches eof [00:56:08 CEST] <durandal_1707> and using fifo directly is pita because one needs to allocate frame to read samples [01:00:04 CEST] <cone-201> ffmpeg 03Timo Rothenpieler 07master:9f4bff834c47: avcodec/nvenc: Delay frame output to increase encoding speed [01:05:03 CEST] <philipl> managed to crash my nvidia driver with all my tests :-) [01:06:02 CEST] <philipl> nevcairiel: did you ever establish whether your ucNumDeltaPocsOfRefRpsIdx fix was needed? [01:07:26 CEST] <cone-201> ffmpeg 03Anton Khirnov 07master:bf52f773913c: lavc: add Intel libmfx-based MPEG2 decoder. [01:07:27 CEST] <cone-201> ffmpeg 03Michael Niedermayer 07master:bb619f41bea8: Merge commit 'bf52f773913cf74bdf0d2c8c2cb4473fa1b7801e' [01:09:12 CEST] <BtbN> Is that the one from libav? So, there are two now? [01:15:58 CEST] <philipl> weeeeeeeeeeeeeeeeeeeeeeeeeee [01:17:22 CEST] <wm4> business as usual [01:20:05 CEST] <BtbN> philipl, also getting 99% utilization with 1080p at ~90 fps now [01:35:35 CEST] <cone-201> ffmpeg 03Anton Khirnov 07master:41d47ea85fb4: lavc: add Intel libmfx-based HEVC decoder. [01:35:36 CEST] <cone-201> ffmpeg 03Michael Niedermayer 07master:508b79e6c1dc: Merge commit '41d47ea85fb4ad9cfb5c2dc808a46bc1d57f3986' [01:56:07 CEST] <ckh_> Hey guys. I saw your post that you need hosting / server space. I'd like to help (and the higher ups said okay). I'm colleen from Facebook. You can hit me up at ckh(a)fb.com [01:57:52 CEST] <BtbN> There's currently a discussion ongoing on the ffmpeg-devel mailing list, about choosing the new host, it's propably best to post there [01:58:31 CEST] <BtbN> More options are definitely still welcome, the current host was kinda rushed because stuff had to move fast [02:00:39 CEST] <ckh_> Ah. But they currently have options? Thats good news :) [02:01:10 CEST] <BtbN> Yes, a few people offered sponsored or free hosting options [02:01:56 CEST] <ckh_> Ah. I suggested internally that we foot the bill for you guys to use EC2, GCE, whatever. [02:02:42 CEST] <ckh_> I gotta run to an airport. Ill check out the mailing list at some point. Thanks :) [02:04:16 CEST] <cone-201> ffmpeg 03Ivan Uskov 07master:1defff85cb2c: libavcodec/h264_mp4toannexb_bsf.c: Optional argument "private_spspps_buf" to avoid extradata modification. [02:16:32 CEST] <cone-201> ffmpeg 03Ivan Uskov 07master:684b7038436f: libavcodec/qsvdec_h264.c: using "private_spspps_buf" argument for av_bitstream_filter_filter() to avoid failure after decoder re-initialization. [02:31:14 CEST] <jamrial> BtbN: no, not two fortunately this time :p [02:31:23 CEST] <jamrial> libav's was basically not merget at all [04:16:55 CEST] <cone-201> ffmpeg 03Michael Niedermayer 07master:e01153839451: avformat/ipmovie: Factor init_audio() out [04:16:56 CEST] <cone-201> ffmpeg 03Michael Niedermayer 07master:8e0fdb03a2f1: avformat/ipmovie: Parse&handle late audio init [04:16:57 CEST] <cone-201> ffmpeg 03Michael Niedermayer 07master:c3361b3a8786: avformat/ipmovie: Fix late audio detection [05:40:58 CEST] <cone-201> ffmpeg 03James Almer 07master:ec7f04c13a6e: avcodec/texturedsp: fix undefined shift [06:00:15 CEST] <cone-201> ffmpeg 03Anshul Maheshwari 07master:379ddc6af44f: avformat/hlsenc: Add WebVtt support in hls [07:51:54 CEST] <cone-201> ffmpeg 03James Almer 07master:844bef578ee0: avcodec/x86: add missing colon to labels [07:51:55 CEST] <cone-201> ffmpeg 03James Almer 07master:d9e10af54784: x86/vf_interlace: add missing colon to labels [07:51:56 CEST] <cone-201> ffmpeg 03James Almer 07master:4d2c014a8f0a: x86/float_dsp: add missing colon to labels [07:51:57 CEST] <cone-201> ffmpeg 03James Almer 07master:f37a5dcb55d1: swresample/x86: add missing colon to labels [10:01:57 CEST] <nevcairiel> philipl: that fix is definitely needed for dxva2, but apparently for nothing in fate [11:21:02 CEST] <nevcairiel> philipl: as in, i can definitely still reproduce failure on my GTX960 without it on a certain sample [11:21:13 CEST] <nevcairiel> (just tried again this minute to make sure) [12:04:40 CEST] <cone-084> ffmpeg 03Michael Niedermayer 07master:4d0d55cd623b: checkasm: Use LOCAL_ALIGNED [12:04:41 CEST] <cone-084> ffmpeg 03Michael Niedermayer 07master:ce466275f8d6: Merge commit '4d0d55cd623bcd504867f948849380f6b4060b4d' [13:57:17 CEST] <durandal_1707> have someone also been contacted by some gif guys? [13:57:35 CEST] <BtbN> gif guys? [13:57:58 CEST] <Daemon404> is that some sort of perverted cult [13:59:08 CEST] <durandal_1707> folks that create gifs for you to enjoy [13:59:55 CEST] <BtbN> Aren't those sites now using mp4 videos anyway? [14:00:12 CEST] <BtbN> even though they invented the new format gifv for it [14:00:20 CEST] <BtbN> (which is mp4 with a diffrent file extension) [14:04:00 CEST] <ubitux> isn't it webm? [14:04:37 CEST] <BtbN> also possible, but it's just something common with their own file extension [14:06:27 CEST] <Daemon404> it's both [14:06:31 CEST] <Daemon404> it serves it depending on browser [14:06:49 CEST] <Daemon404> 'gifv' is to make it easier for normal people to understand [14:07:32 CEST] <durandal_1707> "normal" [14:07:50 CEST] <phh> ubitux: ^^^ [14:08:46 CEST] <ubitux> :( [14:08:59 CEST] <ubitux> durandal_1707: anyway yeah i've been contacted as well [14:16:08 CEST] <Compn> durandal_1707 : are the gif people asking you for favors or job offers? [14:17:26 CEST] <durandal_1707> they are asking to ban Compn from this channel :/ [14:18:42 CEST] <wm4> lol [14:41:09 CEST] <cone-084> ffmpeg 03Michael Niedermayer 07master:4b90dcb84935: avcodec/dvbsubdec: Compute default CLUT based on bitmap analysis [15:07:13 CEST] <BBB> durandal_1707: who is they? [15:09:25 CEST] <durandal_1707> nobody wants ban that was joke [15:09:56 CEST] <BBB> oh ok [16:15:00 CEST] <michaelni> philipl, BtbN, theres a nvenc pull req https://github.com/FFmpeg/FFmpeg/pull/140 [16:15:26 CEST] <michaelni> just saw it [16:16:10 CEST] <michaelni> can you review? should i apply/merge it ? [16:19:46 CEST] <wm4> close it and say patches go to the mailing list [16:20:29 CEST] <BtbN> This PR looks strange. While he is right that "2Pass" supports seems to have improved, i'm not sure of the way he implements that. [16:21:15 CEST] <nevcairiel> judging from the commit in the PR, it was never explicitly forbidden in the code [16:21:18 CEST] <BtbN> if i'm not mistaken, 2Pass mode allways worked with all presets, and only displayes a warning [16:21:19 CEST] <nevcairiel> just issued a warning message [16:21:28 CEST] <BtbN> exactly [16:21:29 CEST] <nevcairiel> which makes his commit way too complex [16:21:35 CEST] <nevcairiel> just remove the warning and voila [16:31:40 CEST] <BtbN> his commit even breaks the twopass tristate option [17:29:59 CEST] <cone-084> ffmpeg 03Michael Niedermayer 07master:e0031ca29a47: avcodec/avrndec: Fix cropping [17:44:04 CEST] <cone-084> ffmpeg 03WereCatf 07master:7106dfd94568: avcodec/nvenc: 2-pass mode works even with non-low latency presets now [17:46:29 CEST] <j-b> WereCatf? [17:51:41 CEST] <BtbN> guy on github [17:52:18 CEST] <BtbN> michaelni, the PR should have auto-closed once the commit reached github [18:04:01 CEST] <liquidee> hello everyone. Has anyone ever used av_format_control_message? I am trying to implement a way to send control messages to devices like xv to control their behaviour. Is it possible to get AVFormatContext of the specific device inside a filter? [18:04:17 CEST] <michaelni> BtbN, ohh ok didnt realize [18:04:49 CEST] <liquidee> Just looking for a clear and proper way to implement this [18:13:09 CEST] <durandal_1707> ubitux: is nlmeans filter going to have temporal option? [18:21:22 CEST] <ubitux> durandal_1707: i read papers, and they were saying it was pointless, so i didn't plan to [18:26:28 CEST] <ubitux> should i? [18:56:55 CEST] <philipl> nevcairiel: Hmm. I've played around with and without your patch and the Sintel file you said showed the problem but I never saw any difference in vdpau [19:06:59 CEST] <cone-084> ffmpeg 03Niklesh 07master:f21fcf60eca9: movtextdec.c: Add support for fontsize [21:42:51 CEST] <BBB> does anyone here think we should just rewrite ffplay in straight opengl? [21:43:03 CEST] <BBB> (random idea; probably far-fetched) [21:45:57 CEST] <BtbN> dealing with opengl context creation on all platforms? Nah [22:05:21 CEST] <nevcairiel> there is light-weight libs for that [22:05:28 CEST] <nevcairiel> more light-weight than sdl [22:05:39 CEST] <nevcairiel> the added problem is audio then [22:06:17 CEST] <BtbN> aren't there audio output filters for almost everything already? [22:14:08 CEST] <wm4> doesn't libavdevice have opengl support? [22:14:55 CEST] <wm4> but seriously, using a muxer API for video output [22:15:07 CEST] <wm4> this shit should just be removed and replaced by a proper API [22:17:06 CEST] <cone-084> ffmpeg 03Steve Lhomme 07master:c5327df83839: force WINAPI_FAMILY to WINAPI_FAMILY_DESKTOP_APP to use DVXA [22:17:07 CEST] <cone-084> ffmpeg 03Michael Niedermayer 07master:29a43bd588b0: avformat/hlsenc: Fix wording in comment [22:39:08 CEST] <cone-084> ffmpeg 03Michael Niedermayer 07master:dc07e576ce72: avformat/isom: Use AVRN codec to handle AVDJ [22:43:09 CEST] <durandal_1707> wm4: with one from libmpv? [22:43:54 CEST] <wm4> durandal_1707: no, libmpv doesn't have a video output API [22:44:08 CEST] <wm4> and mpv's internal one isn't that great either [22:44:29 CEST] <wm4> just go design something reasonable instead of coercing it into a muxing API or whatever [22:45:30 CEST] <durandal_1707> can't because awaiting libav [22:47:04 CEST] <wm4> ? [22:54:16 CEST] <cone-084> ffmpeg 03Steve Lhomme 07master:9b4b96c0dee9: force WINAPI_FAMILY to WINAPI_FAMILY_DESKTOP_APP to be able to use dxva.h [22:54:17 CEST] <cone-084> ffmpeg 03Michael Niedermayer 07master:508c815d4db0: Merge commit '9b4b96c0dee90d260891ec765b28f6ca15594340' [23:04:12 CEST] <kierank> libav was going to merge it into libavformat [23:18:35 CEST] <cone-084> ffmpeg 03Martin Storsjö 07master:0cff125200ab: configure: Only add -FIstdlib.h for msvc/icl if necessary [23:18:36 CEST] <cone-084> ffmpeg 03Michael Niedermayer 07master:867686997cba: Merge commit '0cff125200ab53fa3ae70d85b4f614f269fe3426' [23:30:23 CEST] <cone-084> ffmpeg 03Martin Storsjö 07master:5fd553d31272: configure: Only redefine inline to __inline for msvc if necessary [23:30:24 CEST] <cone-084> ffmpeg 03Michael Niedermayer 07master:27e61a716c89: Merge commit '5fd553d31272d5ed42a7a5a0ecaab7b3452da83a' [23:57:50 CEST] <cone-084> ffmpeg 03Martin Storsjö 07master:342b0ba5f93b: configure: Only redirect strtoll to _strtoi64 if necessary [23:57:51 CEST] <cone-084> ffmpeg 03Michael Niedermayer 07master:dcc540db6ee0: Merge commit '342b0ba5f93b09b1d0c2597db44605300e6fcc53' [00:00:00 CEST] --- Mon Jul 27 2015

1 0

ffmpeg.log.20150726
by burek 27 Jul '15

27 Jul '15

[00:21:41 CEST] <dks-> hi all.. please, how can i embed subtitle in a video? [00:22:02 CEST] <KarlFranz> Do you mean t hardburn it [00:22:03 CEST] <KarlFranz> ? [00:22:13 CEST] <KarlFranz> Or is softburning enough for you? [00:22:28 CEST] <dks-> hardburn [00:22:32 CEST] <durandal_1707> Read fine documentation [00:22:48 CEST] <dks-> i already did [00:22:57 CEST] <dks-> but no success [00:23:33 CEST] <dks-> https://trac.ffmpeg.org/wiki/HowToBurnSubtitlesIntoVideo [00:23:36 CEST] <KarlFranz> If you have tried something and it did not work, you are supposed to tell people what you tried. [00:23:38 CEST] <dks-> i did this [00:23:51 CEST] <dks-> yeah, sorry [00:24:34 CEST] <dks-> i downloaded from http://johnvansickle.com/ffmpeg/ [00:25:28 CEST] <KarlFranz> What does "no success" mean? Do you get an error message? [00:26:17 CEST] <dks-> ffmpeg starts and ends the process [00:26:21 CEST] <dks-> no error message [00:26:31 CEST] <dks-> but not subtitle embed [00:27:05 CEST] <dks-> wow [00:27:11 CEST] <dks-> i found an error: [00:27:12 CEST] <dks-> [Parsed_subtitles_0 @ 0x3d800c0] Error decoding: Invalid data found when processing input (ignored) [00:27:12 CEST] <dks-> [srt @ 0x3d88500] Invalid UTF-8 in decoded subtitles text; maybe missing -sub_charenc option [01:43:15 CEST] <dks-> why my subs are skipping some lines? [01:44:37 CEST] <h1113> never had that [01:45:11 CEST] <h1113> copy your command line [01:49:26 CEST] <dks-> h1113 /opt/ffmpeg/ffmpeg -i black.avi -vf subtitles=black.srt -sub_charenc ISO-8859-1 -codec:v libx264 -crf 23 -codec:a copy black-embutida.mp4 [01:51:56 CEST] <h1113> I don't get any missing lines issues, but i always use .ass format subtitles [01:52:01 CEST] <h1113> maybe the problem is in the .srt? [01:52:11 CEST] <dks-> maybe [01:52:26 CEST] <dks-> how can i convert srt to ass? [01:52:37 CEST] <dks-> can i use ffmpeg? [01:52:59 CEST] <h1113> don't bother, i doubt that's the issue [01:53:04 CEST] <dks-> ffmpeg -i input.srt output.ass ? [01:53:12 CEST] <h1113> open the .srt with notepad and see if the lines are actually there [01:53:26 CEST] <dks-> it's opened now [01:53:36 CEST] <dks-> i was comparing [01:54:39 CEST] <dks-> i get this error : [01:54:39 CEST] <dks-> [srt @ 0x40f7a00] Invalid UTF-8 in decoded subtitles text; maybe missing -sub_charenc option [01:54:40 CEST] <dks-> Error while decoding stream #0:0: Invalid data found when processing input [01:54:44 CEST] <h1113> .srt syntax is pretty human-readable.... just look for the missing line and see if there's anything wrong with it [01:54:57 CEST] <h1113> does it have non-english characters? [01:55:06 CEST] <dks-> yes [01:55:11 CEST] <dks-> brazilian portuguese [01:55:26 CEST] <dks-> 23.94.51.252/a/black.srt [01:55:32 CEST] <dks-> http://23.94.51.252/a/black.srt [01:59:46 CEST] <h1113> on windows? [02:00:07 CEST] <h1113> there should be a specific option for this [02:00:12 CEST] <h1113> but very easy workaround [02:00:19 CEST] <h1113> open with windows notepad -> save as -> UTF-8 [02:00:27 CEST] <dks-> okay [02:03:55 CEST] <h1113> ffmpeg -sub_charenc LATIN1 -i black.srt a.ass [02:03:57 CEST] <h1113> that worked [02:04:07 CEST] <h1113> im off. [02:05:27 CEST] <dks-> Unrecognized option 'sub_charenc'. [02:05:27 CEST] <dks-> Error splitting the argument list: Option not found [02:06:05 CEST] <dks-> worked for you? [02:24:33 CEST] <dks-> i found a way to run all commands in a linux box.. iconv -f iso-8859-1 -t utf-8 input.srt > output.srt [02:24:37 CEST] <dks-> :D [03:23:40 CEST] <WereCatf> Does anyone here use ffmpeg with nvenc? I can't find any documentation on how to actually make use of it with ffmpeg, even if I have built ffmpeg with support for it [03:59:23 CEST] <WereCatf> I can basically get nvenc working with ffmpeg on the command-line, but I want to use HEVC. Looking at the sourcecode I need to somehow pass AV_CODEC_ID_H265 in avctx->codec->id and I just don't know how. I'm really not an expert with ffmpeg and this is really frustrating [11:56:45 CEST] <pragomer1> what is the best / an actual ppa for ffmpeg under ubuntu 14.04 ? [11:58:37 CEST] <BtbN> the one linked on ffmpeg.org [12:00:00 CEST] <pragomer1> omg.. there is one on your site.. sorry.. really didnt find this.. thank you [12:00:45 CEST] <pragomer1> do I understand it right.. that in ubuntu 14.04 ffmpeg isnt in the repos.. but mplayer1 is.... in 15.04 ffmpeg is back.. but mplayer1 is gone (and mplayer2 that isnt useful is there..) ? [12:00:54 CEST] <BtbN> ubuntu still is using libav. [12:01:12 CEST] <BtbN> As debian switched back to ffmpeg, ubuntu also will with the next release. [12:02:50 CEST] <pragomer1> that really.. mm... but let me say both mplayer1 and ffmpeg are so so so important programs.. I am looking forward for both to be in the ubuntu repos instead of forked things [12:03:17 CEST] <BtbN> mplayer is important? [12:04:33 CEST] <pragomer1> yes.. often need the mencoder... converting videos to png mencoder is often a bit better than ffmpeg... [12:11:07 CEST] <pragomer1> another question: I want to convert a bluray video folder (subfolder "stream" contains m2ts-files) to mkv.. would this be the best way : [12:11:13 CEST] <pragomer1> http://privatstrand.dirkschmidtke.de/2012/05/09/videos-verlustfrei-von-m2ts… [12:11:56 CEST] <pragomer1> and what does the 3rd line of these 3 mean? [12:12:19 CEST] <JEEB> pragomer1: If you absolutely need to use mencoder for something, I recommend you make absolutely sure that you can't do it with ffmpeg or mpv's encoding thing [12:12:39 CEST] <BtbN> I'm not sure if ffmpeg is able to decrypt a bluray. [12:12:51 CEST] <pragomer1> its not a crypted blueray [12:14:32 CEST] <pragomer1> BtbN: so these lines are a good way to do it? [12:15:05 CEST] <pragomer1> JEEB: so you would prefer ffmpeg before mplayer? what do you mean by mpv's enconding thing? [12:15:18 CEST] <BtbN> no idea, to my knowledge all blurays are full of drm and aacs encryption, never seen one that wasn't. [12:15:31 CEST] <JEEB> BtbN: he probably already decrypted or downloaded a rip [12:15:38 CEST] <JEEB> not like those are rare [12:16:16 CEST] <pragomer1> its a blueray with my own videos [12:16:35 CEST] <JEEB> pragomer1: mplayer is a pile of steaming bollocks and the less you use it the better. so yes, I would prefer ffmpeg to mencoder any day :P [12:16:49 CEST] <JEEB> and mpv is the least retarded of the mplayer family at this point [12:17:06 CEST] <JEEB> it has new encoding support, so if ffmpeg doesn't cut it you can look at that [12:18:32 CEST] <pragomer1> :-) I thought mplayer and ffmpeg would be created by the same crew or parts of them.. sorry I am just a normal user and thought so.. and its my 2nd time I am using an irc chat in my life :-) [12:19:25 CEST] <pragomer1> how could I append 2 video parts.. would I append the m2ts files or the converted mkv ? [12:19:27 CEST] <JEEB> yes, there was a lot of incestious relationships between ffmpeg and mplayer. and that is not necessarily a good thing. like mplayer having a copy of ffmpeg within its goddamn source base and using _private_ APIs) [12:20:19 CEST] <JEEB> also mplayer2 actually did improve on mplayer, but it did remove the godawful mencoder part. then with mpv a new encoding feature was added that was somewhat less retarded [12:20:43 CEST] <pragomer1> ok... ffmpeg really seems more gentle :-) [12:22:24 CEST] <pragomer1> so I will try it with ffmpeg converting m2ts to mkv.. like I wrote.. last question.. how do I append 2 files.. m2ts or mkv? and how? [12:23:23 CEST] <JEEB> for some formats you can use the concat protocol (like MPEG-TS), but I usually use the concat filter [12:23:58 CEST] <pragomer1> mm.. I get an error converting the first file: [12:24:00 CEST] <pragomer1> Application provided invalid, non monotonically increasing dts to muxer in stream 2: 93052 >= 93051 [12:24:00 CEST] <pragomer1> av_interleaved_write_frame(): Invalid argument [12:29:52 CEST] <pragomer1> that is the error: http://pastebin.com/97XPRDbF [14:00:49 CEST] <active8> ffmpeg -i file.mp4 -filter_complex "[0:v]setpts=2.0*PTS[v];[0:a]atempo=0.5[a]" -map "[v]" -map "[a]" half-speed.mp4 wprks pretty good for half speed and doing it twice gives me quarter speed but [14:01:12 CEST] <active8> -filter_complex "[0:v]setpts=1.5*PTS[v];[0:a]atempo=0.75[a]" -map "[v]" -map "[a]" [14:01:52 CEST] <active8> for 3/4 speed... the video audio sync quickly comes unglued with the video lagging the audio [14:23:41 CEST] <DHE> 0.75 is 3/4 speed, but 1.5 is 2/3 speed [14:41:07 CEST] <active8> DHE I thought that might be the case but couldn't see where I calculated that proportion wrong [14:49:33 CEST] <DrSlony> Hey, I have an MP3 but I need to put that into a video format so facebook will allow one to upload it. What is hte easiest way to put this MP3 into a video container? [14:56:00 CEST] <BtbN> i don't think they will allow it without an actual video [14:56:52 CEST] <DrSlony> yeah sohow do i use ffmpeg to generate a black video for the duration of the audio? [15:02:08 CEST] <DrSlony> or even better - when i use ffplay on the audio i get a waveform animation thing (what is that called?), would be neat if i could have that as the video track [15:02:38 CEST] <JEEB> you can use the filters to create that [15:03:01 CEST] <JEEB> pretty sure if you search the trac and the documentation on it, there will be examples [15:05:51 CEST] <DrSlony> this? http://www.ffmpeg.org/ffmpeg-filters.html#histogram [15:06:30 CEST] <DrSlony> trying it now [15:06:30 CEST] <JEEB> that seems to be a histogram of the video [15:06:40 CEST] <JEEB> so if you have no video, that kind of doesn't help :D [15:09:27 CEST] <JEEB> http://www.ffmpeg.org/ffmpeg-filters.html#avectorscope http://www.ffmpeg.org/ffmpeg-filters.html#showspectrum http://www.ffmpeg.org/ffmpeg-filters.html#showwaves [15:09:31 CEST] <JEEB> these seem more like it [15:28:42 CEST] <active8> So, JEEB. It doesn't mention what all can be done with that filter out, like whether it automatically goes into the chain with the audio so the output can be, say, an mp4 or if you have to multiplex the two, afterwards [15:30:42 CEST] <active8> i don't quite get the mapping they do in the example [15:30:56 CEST] <active8> i think it's some kinda mapping [15:31:47 CEST] <active8> 'course you'd use ffmpeg, not ffplay [15:36:12 CEST] <DrSlony> lost connection to my irc server [15:36:19 CEST] <DrSlony> whoever it was who helped me, thank you [15:37:22 CEST] <JEEB> active8: you'd create the input with lavfi and then map the output of that filter chain to be an input for encoding [15:37:27 CEST] <JEEB> then you'd encode the video and copy over the audio [15:38:51 CEST] <DrSlony> JEEB that sounds like advice for me [15:53:53 CEST] <DrSlony> i have the showspectrum part working when i use ffplay, but im having no luck getting it saved into a video using ffmpeg [15:53:56 CEST] <DrSlony> ffmpeg -lavfi amovie=foo.mp3,asplit[out0],showspectrum=mode=combined:color=intensity:slide=2:scale=cbrt[out1] /tmp/foo.mp4 [15:56:02 CEST] <DrSlony> ffmpeg -i foo.mp3 amovie=foo.mp3,asplit[out0],showspectrum=mode=combined:color=intensity:slide=2:scale=cbrt[out1] -c:v libx264 -r 24 -preset slower -crf 23 /tmp/foo.mp4 [16:09:02 CEST] <DrSlony> this seems a step closer [16:09:04 CEST] <DrSlony> ffmpeg -f lavfi -i amovie=foo.mp3,showspectrum=mode=combined:color=intensity:slide=2:scale=cbrt -c:v libx264 -r 24 -preset fast -crf 23 /tmp/foo.mp4 [16:23:13 CEST] <DrSlony> that seems to result in a valid video being saved, but with no audio. How do I put the audio into it all in one step? [16:25:18 CEST] <JEEB> add another -i after the amovie -i , and then set -c:a copy [16:25:25 CEST] <JEEB> that should do it [16:30:35 CEST] <DrSlony> i ran a quick 10 second test, seems to work! than you JEEB! [16:30:40 CEST] <DrSlony> ffmpeg -f lavfi -i amovie=foo.mp3,showspectrum=mode=combined:color=intensity:slide=1:scale=cbrt -i foo.mp3 -c:a copy -c:v libx264 -r 24 -preset fast -crf 23 /tmp/foo.mp4 [18:00:19 CEST] <liquidee> has anyone ever used av_format_control_message? [20:00:11 CEST] <wiak> cropping 4096x2160 to 3840x2160, what should i use? crop or cropdetect? [20:00:54 CEST] <JEEB> cropdetect is for random automation when you don't know how big the crop will be [20:00:58 CEST] <JEEB> crop is when you actually want to crop [20:01:09 CEST] <JEEB> basically cropdetect outputs the information on how much it thinks you have to crop [20:07:22 CEST] <wiak> um something like ffmpeg -i A001_C014_01217G.4k.y4m -vf "crop=3840:2160:128:128" outputgirl2.y4m [20:07:22 CEST] <wiak> ? [20:16:02 CEST] <wiak> JEEB was that correct? or am i doing it wrong [20:26:57 CEST] <ChocolateArmpits> wiak: it should be 3840:2160:128:0 [20:27:11 CEST] <ChocolateArmpits> you're not cropping vertical, are you? [20:30:48 CEST] <wiak> i just want top remove 128 at top and bottom [20:30:49 CEST] <ChocolateArmpits> The fourth input describes vertical offset [20:30:54 CEST] <wiak> ooh [20:31:18 CEST] <ChocolateArmpits> but the resolution change you described implies horizontal crop [20:31:59 CEST] <ChocolateArmpits> Are you really sure what you're doing ? [20:32:11 CEST] <wiak> nope [20:34:12 CEST] <ChocolateArmpits> ok, let's figure this out, could you host a snapshot of your video to determine what and how much cropping you need ? [20:37:10 CEST] <wiak> this is gonna take a little while uploding [20:38:27 CEST] <wiak> ChocolateArmpits FYE: http://awesome.nwgat.ninja/misc/out.png [20:39:56 CEST] <ChocolateArmpits> wiak: so you have to only crop horizontally, not vertically [20:40:41 CEST] <ChocolateArmpits> Then 3840:2160:128:0, it will crop 128 pixels from the left side and 128 pixels from the right side [20:41:48 CEST] <wiak> thanks [20:41:49 CEST] <wiak> :) [20:47:14 CEST] Action: wiak trows some norwegian milk chocolate @ ChocolateArmpits [20:47:29 CEST] <ChocolateArmpits> no problem [20:47:37 CEST] <wiak> meanwhile http://i.imgur.com/RheCUPd.png [00:00:00 CEST] --- Mon Jul 27 2015

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