December 2018 - Ffmpeg-devel-irc

ffmpeg-devel.log.20181230
by burek 31 Dec '18

31 Dec '18

[02:36:31 CET] <sh4rm4^bnc> hi, the configure script of ffmpeg is trying to execute this command: [02:36:32 CET] <sh4rm4^bnc> ++ sed -E -n 's/^extern AVFilter ff_([avfsinkrc]{2,5})_([a-zA-Z0-9_]+);/\2_filt$ [02:36:32 CET] <sh4rm4^bnc> /bin/sed: unrecognized option: E [02:36:59 CET] <sh4rm4^bnc> i can't find any info about what the option -E is supposed to do [02:37:54 CET] <sh4rm4^bnc> it's not even documented in the online manpages of GNU sed (i don't use GNU sed) [02:40:30 CET] <iive> -E, -r, --regexp-extended use extended regular expressions in the script (for portability use POSIX -E). sed 4.7 [02:41:08 CET] <sh4rm4^bnc> ah. could i ask about changing that to -r then ? that one is supported [02:42:23 CET] <sh4rm4^bnc> nevermind. i see latest busybox has added support for it. [02:42:40 CET] <iive> it is recommended to use posix sed :P [02:42:53 CET] <sh4rm4^bnc> yeah busybox strives to be POSIX compliant [02:42:57 CET] <sh4rm4^bnc> unlike GNU [02:43:31 CET] <sh4rm4^bnc> i also get a ton of "pr: command not found" but it seems to work anyway [02:45:37 CET] <BradleyS> iirc there was some discussion recently on making sed commands more compatible [02:45:50 CET] <BradleyS> perhaps there's more info on the ffmpeg-devel mailing list [02:49:08 CET] <sh4rm4^bnc> + pr -n -t /tmp/ffconf.XXeihohl/test.c [02:49:08 CET] <sh4rm4^bnc> ./configure: line 505: pr: command not found [02:49:31 CET] <sh4rm4^bnc> never heard of "pr" before... [02:50:38 CET] <sh4rm4^bnc> aha, seems to be POSIX. so i guess i gotta get it somehow. http://pubs.opengroup.org/onlinepubs/9699919799/utilities/pr.html [06:34:54 CET] <mostafa_s> Hi how get video thumbnail (Windows 10 ) [13:43:10 CET] <cone-582> ffmpeg 03Paul B Mahol 07master:7312e027d6d3: avfilter/af_afir: introduce AudioFIRSegment structure and use it [13:43:10 CET] <cone-582> ffmpeg 03Paul B Mahol 07master:e57053417a09: avfilter/af_afir: move allocation stuff where it belongs [13:43:10 CET] <cone-582> ffmpeg 03Paul B Mahol 07master:c343e81ffd30: avfilter/af_afir: introduce init_segment() and use it [13:43:10 CET] <cone-582> ffmpeg 03Paul B Mahol 07master:8007e8fc67e3: avfilter/af_afir: fix minp/maxp range and change default value for maxp [13:43:10 CET] <cone-582> ffmpeg 03Paul B Mahol 07master:30bf54b9f3ac: avfilter/af_afir: introduce uninit_segment() and use it [13:43:10 CET] <cone-582> ffmpeg 03Paul B Mahol 07master:1ba909fabef2: avfilter/af_afir: use segment pointer [13:43:11 CET] <cone-582> ffmpeg 03Paul B Mahol 07master:6095356d5b1e: avfilter/af_afir: use AVFrame for coeff too [13:43:12 CET] <cone-582> ffmpeg 03Paul B Mahol 07master:31c9d693aa70: avfilter/af_afir: make number of segments extendable [13:43:13 CET] <cone-582> ffmpeg 03Paul B Mahol 07master:3c3eb4930eb4: avfilter/af_afir: make part_index values per channel [13:43:14 CET] <cone-582> ffmpeg 03Paul B Mahol 07master:300dc45fdca4: avfilter/af_afir: properly split IR into multiple segments [13:43:15 CET] <cone-582> ffmpeg 03Paul B Mahol 07master:a53a9f1c8d1f: avfilter/af_afir: implement non-uniform partitioned convolution [15:36:53 CET] <spottich> re hy [15:37:03 CET] <spottich> could anybody help me by develop an windows command for convert an rtsp stream to an mjpeg stream [15:37:16 CET] <spottich> ffmpeg -y -i rtsp://stream.sdp -c copy -f hls -hls_time 3 -hls_list_size 3 -hls_wrap 10 http://localhost:8090/channel1.mjpg [16:04:40 CET] <BtbN> You should ask that in #ffmpeg [16:05:01 CET] <spottich> yeah but nobody answer ther at hours [16:05:04 CET] <spottich> thx [16:10:58 CET] <BtbN> Keep in mind that ffmpeg does not offer an http server, you will have to use apache or something to serve an hls stream. [16:24:11 CET] <spottich> ah ok [16:24:35 CET] <spottich> yes i find it out because windows hasnt an ffserver module [16:25:05 CET] <spottich> i only must place an mjpeg stream on my houseintern webside [17:27:43 CET] <durandal_1707> TRRRRRROOOOOOOOOOOLLLLLLLLLLL [17:35:00 CET] <cone-937> ffmpeg 03Alex Mogurenko 07master:e4788ae31b2e: avcodec/prores_ks: Fix luma quantization if q >= MAX_STORED_Q [17:51:38 CET] <cone-937> ffmpeg 03Paul B Mahol 07master:72270d5bd071: avfilter/af_afir: optimize code a little more [17:51:39 CET] <cone-937> ffmpeg 03Paul B Mahol 07master:ba3d983c95b2: avfilter/af_afir: stop using as much threads as stream have it [18:40:18 CET] <durandal_1707> what is mmsize for avx and what mmsize is for sse3 ? [18:41:15 CET] <atomnuker> 32 vs 16 [18:44:22 CET] <durandal_1707> looks like fftw3 have avx and is faster then our simd [18:46:21 CET] <atomnuker> yeah right [18:46:46 CET] <atomnuker> we have avx and should be faster because we don't generalize [18:50:04 CET] <durandal_1707> atomnuker: gonna send a patch? [18:51:10 CET] <atomnuker> can I see a benchmark? some old ones I've seen had lavc's fft as being faster [19:01:57 CET] <durandal_1707> atomnuker: not really, i benchmark zita_convolver effect from https://github.com/bmc0/dsp versus afir from libavfilter and my implementation is sometimes up to 2x slower, but for cases that number of min and max partitions is same my implementatio is several times faster somehow [19:02:55 CET] <durandal_1707> reason could be suboptimal partitioning on my side when minp != maxp [19:09:47 CET] <atomnuker> is afir that slow? [19:10:54 CET] <durandal_1707> atomnuker: only for minp <<< maxp [19:11:23 CET] <durandal_1707> then several fft of different sizes is used [19:11:28 CET] <cone-937> ffmpeg 03Zachary Zhou 07master:b6b3159b04af: lavfi/vaapi: Add constant VAAPI_VPP_BACKGROUND_BLACK [19:26:28 CET] <durandal_1707> i hate this troll [19:34:01 CET] <kierank> durandal_1707: who, nicholas? [19:34:30 CET] <durandal_1707> yes, bring him down! [21:05:34 CET] <cone-937> ffmpeg 03Marton Balint 07master:f9e947845f9a: avfilter/af_asetnsamples: fix last frame props [21:42:18 CET] <cone-937> ffmpeg 03Marton Balint 07master:7f42276aa91c: avformat/concatdec: always allow seeking to start [22:14:33 CET] <cone-937> ffmpeg 03Paul B Mahol 07master:ea3a980a610e: avfilter/af_afir: do not over allocate fft buffer [22:33:01 CET] <cone-937> ffmpeg 03Paul B Mahol 07master:3e8f211d8685: avfilter/vf_chromashift: remove invalid offset [00:00:00 CET] --- Mon Dec 31 2018

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ffmpeg.log.20181230
by burek 31 Dec '18

31 Dec '18

[01:09:39 CET] <Fiji> hi, i am trying to use CMAKE to build a project that uses ffmpeg libraries and i am getting an issue with trying to link libswscale and use the functions. my cmake file looks like this: https://hastebin.com/sogoneyige.bash , and i am getting errors like this when building: "undefined reference to `sws_getContext'" [01:12:07 CET] <oliv3> Fiji: obviously L22 seems wrong ? [01:12:25 CET] <Fiji> well ok [01:12:26 CET] <Fiji> shit lol [01:12:26 CET] <Fiji> sec [01:13:16 CET] <Fiji> even fixing that it's the same problem [01:13:57 CET] <oliv3> can't help more, i don't use cmake [01:14:06 CET] <Fiji> ok no worries, thank you for the help [08:33:21 CET] <memfrob> Hey hey hey! It's my friends! I like you guys. [08:33:30 CET] <memfrob> Matter of fact, I love you people! [08:33:47 CET] <memfrob> https://ffmpeg.zeranoe.com/forum/viewtopic.php?f=7&t=5566 [08:35:49 CET] <memfrob> My exact line is this: ffmpeg.exe -i "gay_porn_x265.mkv" -preset veryslow -c:v h264_amf -c:a aac -ac 2 -b:a 320k "gay_porn_x264.mp4" [08:36:51 CET] <memfrob> If I use `-c:a copy` instead of `-c:a aac -ac 2 -b:a 320k` the message goes away, but when ffmpeg reaches the 10 second mark, the file size is just the same. [08:39:24 CET] <memfrob> -preset doesn't do anything. ultrafast: frame= 246 fps= 14 q=-0.0 Lsize= 3782kB time=00:00:10.77 bitrate=2876.6kbits/s speed=0.603x [08:39:56 CET] <memfrob> placebo: frame= 236 fps= 14 q=-0.0 Lsize= 3646kB time=00:00:10.13 bitrate=2948.4kbits/s speed=0.584x [08:41:32 CET] <memfrob> ultrafast at 10.77 is 3.7MB but placebo at 10.13 is 3.6MB -- this is not right. [08:42:45 CET] <memfrob> `-c:a copy` with `-preset placebo` -- the message: Codec AVOption preset (Configuration preset) specified for output file blah blah blah goes away, but the results do not change. [08:43:51 CET] <memfrob> `-c:a copy -preset placebo`3834kB time=00:00:10.28 -- looks like using h264_amf doesn't accept preset options. [08:46:00 CET] <memfrob> I apologize for all the lines, I'm not the best at expressing myself sometimes. [08:55:01 CET] <memfrob> oh ffs! -crf doesn't work either [08:55:22 CET] <memfrob> so -preset and -crf are both fubar with h264_amf [08:58:45 CET] <memfrob> Using `-q 0` should be lossless but with h264_amf, I get no "Codec AVOption" warning about it. [08:59:49 CET] <memfrob> (and in that line, I forgot to add that -q 0 appears to do absolutely nothing with -c:v h264_amf) [09:03:01 CET] <memfrob> I'm really sorry for blowing up the channel. All I meant was this: -crf -preset and -q are broken with h264_amf -- My last comment until further info is requested. [09:15:40 CET] <c_14> amf doesn't support crf, it only supports constant quantization, constant bitrate, peak constrained vbr and latency constrained vbr [09:16:04 CET] <c_14> and for constant quantization, the options are -qp_i/-qp_p/-qp_b [09:18:40 CET] <c_14> ffmpeg -h encoder=h264_amf should list all that for you [09:22:24 CET] <memfrob> so the equivalent to -crf 18 is -qp_i/p/b all at 18? [09:27:25 CET] <c_14> no [09:27:38 CET] <c_14> crf and quantization parameters are on completely different scales [09:27:53 CET] <c_14> If you want to try constant quantizer you'll just have to try values to see what looks good [09:28:04 CET] <c_14> I'd probably recommend you just use peak-constrained-bitrate though [09:39:30 CET] <memfrob> -rc is default to -1 but there's 4 things to choose from, cqp, cbr, vbr_peak and vbr_latency [09:39:50 CET] <memfrob> Does -rc count from -1, 0 or 1? [09:40:02 CET] <c_14> you can use the listed names [09:40:51 CET] <memfrob> -c:v h264_amf -rc vbr_peak 5000k: Unable to find a suitable output format for '5000k' [09:41:27 CET] <c_14> -b:v 5000k [09:42:06 CET] <memfrob> oh that one! [09:42:14 CET] <memfrob> I thought you meant the one in h264_amf lol [09:46:34 CET] <memfrob> Would you say 5000K is a good balance for x264 between quality and size? [09:47:05 CET] <c_14> I've never used the amf encoder, and it also depends on the video size [09:47:10 CET] <c_14> But it doesn't sound like a terrible idea [09:49:28 CET] <memfrob> c_14, I really appreciate the help. [09:52:38 CET] <memfrob> It might be a good idea to give people a more information message, like "-crf not supported by h264_amf" [09:53:03 CET] <anshul_> Does avcodec_send_packet() and avcodec_receive_frame() works in replacement of avcodec_decode_subtitle2 [09:54:07 CET] <memfrob> I'm going to guess, "no" but I don't know. [09:54:41 CET] <memfrob> avcodec_decode_subtitle2 stands out the most of those 3. [09:55:49 CET] <anshul_> thanks [09:55:52 CET] <memfrob> https://ffmpeg.org/doxygen/4.0/group__lavc__encdec.html [09:58:40 CET] <memfrob> is avcodec_decode_subtitle2 pulled from ffmpeg? Because that function still exists in vlc and mpv: https://github.com/mpv-player/mpv/blob/master/sub/sd_lavc.c#L320 [10:00:18 CET] <memfrob> https://ffmpeg.zeranoe.com/forum/viewtopic.php?t=866 -- he posts the solution near the bottom [10:00:25 CET] <memfrob> maybe it will help your issue? [10:47:12 CET] <memfrob> Is this right? [10:48:16 CET] <memfrob> 5575kbps total bitrate x265: 4.06GB ...... 5317 total bitrate x264: 3.87GB [10:48:23 CET] <memfrob> *5317kbps [10:49:02 CET] <memfrob> the data bitrate is almost identical. 5001 for x265, 4995 for x264 [10:49:13 CET] <memfrob> how is the x264 video smaller?? [10:50:14 CET] <memfrob> somehow with a very similar bitrate, it's smaller by 200MB [11:09:10 CET] <dv_> I need some help with naming. I have a library here that has to mention hw video codecs as well as formats/standards like h264 etc. so I am looking for a terminology that is not confusing. h264 etc. are often called codecs, which is wrong though. so I was thinking of referring to the hw en/decode units as codecs and to h264 etc. as "compression format". "video format" is often also used for YUV color formats, [11:09:13 CET] <dv_> and "video standard" sounds more like PAL, NTSC, HDMI etc. [11:28:00 CET] <fella> depending on your audience i would probably rather stick with the wrong yet common naming convention where h264 is considered a codec. Why do you feel the need to clarify? [11:47:42 CET] <dv_> fella: because calling the hw unit "codec" and h264 also "codec" is confusing [15:30:34 CET] <spottich> re [15:30:48 CET] <spottich> i net help for an windows cmd [15:31:23 CET] <spottich> ffmpeg -y -i rtsp://stream.sdp -c copy -f hls -hls_time 3 -hls_list_size 3 -hls_wrap 10 http://localhost:8090/channel1.mjpg [15:31:44 CET] <spottich> i want to vonvert an rtsp stream to an mjpeg stream [15:31:56 CET] <spottich> convert sorry [15:32:14 CET] <spottich> could anybody help me for the right syntax [15:43:50 CET] <oliv3> about swscale, when i fixed all my bugs on the Open side, i thought it would work with no more modifications on Net but ran into this one. on the other side, guys on #ffmpeg told it was a security flaw/miss/error OpenBSD-side ;) [15:44:23 CET] <oliv3> (that it worked) [15:44:34 CET] <oliv3> oups [16:17:41 CET] <hifi> hey, I'm trying to figure out how to do kmsgrab from AMD discrete GPU and use Intel iGPU to hardware accelerate encoding to H.264. I can do them separately but I haven't yet found the right flags to ffmpeg to actually do that on-the-fly where work with VA-API, reason is my RX 480 can't encode 1080p60 without dropping frames but for some reason the intel iGPU can, here's the grab command line that works with [16:17:48 CET] <hifi> x264, looking for hints how to properly move the buffer ... [16:17:50 CET] <hifi> ... from AMD KMS plane over to Intel to encode: https://pastebin.com/raw/Z4DXqjjJ [16:24:45 CET] <Aerroon> i use ffmpeg to demux mkv files to mp4 files for furrther editing [16:25:12 CET] <Aerroon> i do ffmpeg -i "video.mkv" -vcodec copy -an "video.mp4" [16:25:31 CET] <Aerroon> and i have this video encoded by nvenc that's 3840x2160 [16:25:44 CET] <Aerroon> yet for some reason ffmpeg's output ends up being 3824x2178 [16:26:14 CET] <furq> Aerroon: pastebin the command line and full output [16:28:40 CET] <Aerroon> was in a bat file, so let me run it again [16:30:14 CET] <Aerroon> https://pastebin.com/Sh109B6C [16:31:12 CET] <furq> where's it showing up as 3824x2178 [16:31:21 CET] <Aerroon> in VLC [16:31:36 CET] <Aerroon> and when i import it into premiere it has these slight back bars on the side [16:32:12 CET] <Aerroon> https://cdn.discordapp.com/attachments/303861570231074816/52895841582186497… [16:32:47 CET] <furq> weird [16:33:33 CET] <Aerroon> i know [16:33:44 CET] <Aerroon> at first i thought this was obs doing it and googled it [16:33:52 CET] <Aerroon> and found that other people have had a similar issue [16:34:09 CET] <Aerroon> but then i read more into it and found out that obs doesn't seem to be involved [16:34:16 CET] <Aerroon> and i checked that my original file did not have that resolution problem [16:35:38 CET] <furq> well h264 is never really 3820 wide because it's not mod16, it's 3824 and then there are cropping flags in the video bitstream [16:36:03 CET] <furq> but that shouldn't be changed by remuxing and it shouldn't ever show up in video players [16:36:10 CET] <furq> and also 2178 is not mod16 so ??? [16:36:39 CET] <furq> i would be tempted to say vlc is broken (not for the first time) if you hadn't mentioned premiere was showing borders [16:37:02 CET] <Aerroon> i tried a test render too [16:37:40 CET] <Aerroon> and it certainly came with the borders [16:37:45 CET] <Aerroon> they're rather slight though [16:41:39 CET] <Aerroon> even the preview window in davinci resolve shows me the slight black borders [16:54:24 CET] <Aerroon> i'll test this stuff some more, i'll try to save it in a container that premiere can read to then see if the issue persists even if it doesn't go through ffmpeg, but i'm unable to try that right now [21:28:56 CET] <memfrob> Hi, is ffmpeg.pastebin.com broken? [21:29:14 CET] <furq> yeah but it's the same as pastebin.com anyway [21:29:33 CET] <memfrob> Will use pastebin.com then! [21:33:43 CET] <memfrob> I have an x264 video that is smaller than an x265 video, but the bitrate is almost the same. x265: https://pastebin.com/gKL3RyyA x264: https://pastebin.com/dAx8KKSg [21:35:10 CET] <memfrob> Also, the original source doesn't have any errors, but after re-encoding it through ffmpeg, I see this: Unsupported codec with id 100359 for input stream 2 [21:36:53 CET] <memfrob> There are no subtitles, not in the re-encoded one or the old one. [21:57:55 CET] <memfrob> My PS4 is asking me to connect to the internet to watch videos off my USB drive so hold on a second.. [21:58:24 CET] <BtbN> yay DRM [22:03:46 CET] <memfrob> Cool! All the files work on the PS4, no "unsupported media" messages or audio stretch issues, super fast video playback, etc. [22:03:51 CET] <memfrob> ffmpeg is THE BEST! [22:04:35 CET] <memfrob> And the best part is that it doesn't take very long to do all this because it can use my video card for encoding! [22:07:40 CET] <memfrob> rexxster, I see you're using a Minnesota IP address, how is it there? cold? Do you like hockey? [22:07:48 CET] <BtbN> Keep in mind that hardware encoding will yield notably worse results [22:07:57 CET] <BtbN> The only time you ever want to use it is if you can't spare the CPU time [22:08:21 CET] <memfrob> BtbN, Using -b:v 10000k (if the source bit rate is even higher) solves that issue [22:08:36 CET] <BtbN> x264 at the same bitrate will still be better [22:09:05 CET] <memfrob> hell, I could use a blu-ray and use -b:v 40000k with -quality quality (GPU prioritizes quality over speed) and it'd come close. [22:09:37 CET] <BtbN> Sure you can throw bitrate at it to compensate, but x264 will always have the better quality/bitrate ratio [22:10:46 CET] <memfrob> The real reason you're telling me this is because you're all about morals and royalty free stuff, right? [22:11:26 CET] <BtbN> ? [22:11:59 CET] <memfrob> Using the GPU for encoding, a GPU that has closed source atombios on it and all the rest would make Richard Stallman scream bloody murder. [22:12:18 CET] <BtbN> Yeah, and because I hate it so much I ported it to ffmpeg... [22:12:50 CET] <memfrob> If you hate your own work so much because it's quality is worse, why did you do it? [22:13:48 CET] <memfrob> There's always an agenda behind people's statements, you're advertising away from your own work, which I find crazy. [22:15:50 CET] <BtbN> Your usecase just isn't what hardware encoders are designed for. [22:16:49 CET] <BtbN> There are indeed only very few cases where they are useful. Usually only when the CPU isn't free to do the work instead. [22:16:54 CET] <memfrob> I can hardly tell a difference, and the reason why I can at all is because I used a slightly lower bitrate than the source. [22:18:05 CET] <memfrob> I multitask. My GPU is a R9 290X which is old but still very fast even to this day. It can do 500 things at once. Encoding a movie off the CPU bogs it down. [22:19:34 CET] <memfrob> hardware encoders are designed to encode faster than CPU, and if I can do something in hardware, I will because it's faster. [22:22:43 CET] <furq> there's a reason all the consumer video encoding asics are attached to gpus [22:22:59 CET] <memfrob> So we can use them? [22:23:03 CET] <furq> they're designed for getting a passable quality video without touching the cpu [22:23:14 CET] <furq> which is a nice feature for Pro Gameing [22:23:32 CET] <BtbN> they're not even neccesarily faster than using the CPU [22:23:35 CET] <furq> if you're just doing this to get your videos on another device then sure [22:23:39 CET] <furq> but you shouldn't use it for archival [22:23:40 CET] <BtbN> It's just that they don't use the CPU at all [22:23:50 CET] <memfrob> <furq> if you're just doing this to get your videos on another device then sure [22:23:55 CET] <memfrob> KABOOM!!!!!! [22:24:06 CET] <furq> right [22:24:49 CET] <memfrob> BtbN, If I were you, I'd be happy if someone used something I wrote. [22:24:50 CET] <BtbN> Unless your CPU is really bad, fps wise x264 and nvenc or one of the other hardware encoders won't be that far off of each other [22:25:11 CET] <BtbN> Why would I care what you use? [22:25:27 CET] <furq> BtbN: was it you who wrote the amf stuff [22:25:33 CET] <BtbN> no, nvenc [22:25:35 CET] <furq> oh [22:25:37 CET] <furq> nvm then [22:25:47 CET] <memfrob> That's not what he told me. [22:25:59 CET] <furq> i don't think you mentioned amf before he said that [22:26:07 CET] <furq> and it's generally a safe bet people mean nvenc because that's massively more popular [22:26:07 CET] <memfrob> I said atombios. [22:26:21 CET] <furq> i've never heard that word until just now [22:26:24 CET] <memfrob> I'm actually surprised BtbN isn't narcissistic. [22:28:20 CET] <memfrob> My CPU isn't the greatest btw, my GPU is much better at kicking ass and taking names. Damn companies take forever to pay. [22:28:41 CET] <memfrob> Will upgrade soon to Ryzen (time to get yelled at by Intel fanboys) [22:29:46 CET] <BtbN> The GPU is actually entirely uninvolved with the video encoding. They are terrible at it. It's dedicated silicon that does it. [22:30:26 CET] <memfrob> If it's the minions on the graphics card that do the work, it's still faster than my CPU xD [22:31:15 CET] <memfrob> Now you're arguing semantics. Can you get help? Please? [22:31:42 CET] <BtbN> I'm not arguing at all even... [22:31:54 CET] <memfrob> GPU vs dedicated silicon. [22:32:01 CET] <memfrob> You just _had_ to mention it. [22:32:20 CET] <BtbN> And in what way does it hurt you that you now know about the difference? [22:32:54 CET] <memfrob> Just making an observation. [22:33:14 CET] <memfrob> I gotta run, literally. Sun goes down in an hour. Take care everyone! [00:00:00 CET] --- Mon Dec 31 2018

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ffmpeg-devel.log.20181229
by burek 30 Dec '18

30 Dec '18

[00:53:49 CET] <cone-695> ffmpeg 03Michael Niedermayer 07master:fccba32b4c4f: tools: add target_dec_fate.sh [02:28:32 CET] <ShadowOfThePast> hello guys [02:28:58 CET] <ShadowOfThePast> i'm losing my mind tracing an ENOMEM issue in libavformat [02:29:53 CET] <ShadowOfThePast> is there any sorcery involved in the av_fast_realloc function? [02:31:03 CET] <ShadowOfThePast> i added some logging and it returns 1114144 in its size argument at the call site i'm interested in [02:32:37 CET] <ShadowOfThePast> it does so hundreds of times so that it should exceed the memory of my box pretty quickly, but it only finally errors out after ~1400 of those 1 MB allocations (my box has 384 MB of ram, runs Linux 2.6.34 and has no paging from what i know, because it runs off a 32 mb nand flash) [02:33:26 CET] <ShadowOfThePast> the particular function that calls it in my case is mov_try_read_block in mov.c [02:34:00 CET] <ShadowOfThePast> i'm reading a cenc-encrypted mp4 file and it uses this function while parsing saiz atoms [02:50:45 CET] <iive> ShadowOfThePast, if i read the code correctly, it just allocates 6% bigger buffer (1/16 of the target size). [02:51:24 CET] <ShadowOfThePast> yea alright, that explains why it returns that odd number instead of 1024*1024 [02:52:42 CET] <ShadowOfThePast> i'm really mind-blown though how it works, because those blocks aren't ever freed (until it runs out of memory and stops playback) [02:53:05 CET] <ShadowOfThePast> they're only freed in mov_free_encryption_index, which isn't called unless the context is closed [02:54:00 CET] <ShadowOfThePast> and like i said, my file has 1559 of those atoms so it should be taking a ton of memory (stupidly, because the data itself is only like 50 bytes every time) [02:54:41 CET] <iive> how big is this file? [02:54:53 CET] <ShadowOfThePast> 500 MB [02:55:17 CET] <iive> so... the atoms are not really 1mb each? [02:55:30 CET] <ShadowOfThePast> yea they're just a couple of bytes [02:55:47 CET] <ShadowOfThePast> that read_block function just allocates at least a meg every time no matter what [02:56:31 CET] <iive> oh, i see. mov probably allocates max size, then reallocs to the real size. but fast-realloc never shrinks [02:58:02 CET] <iive> sorry, I'm not familiar with mov. does atoms have size parameter? [02:59:37 CET] <ShadowOfThePast> yes they do, but in this case it reads a part of the atom only, where the size of that part is specified by an integer in the atom [02:59:45 CET] <ShadowOfThePast> check out the mov_read_saiz function [03:02:02 CET] <ShadowOfThePast> a realloc to shrink it would make sense, but i don't see that happening anywhere for the auxiliary_info_sizes buffer [03:10:54 CET] <iive> so, the problem is in mov_try_read_block() [03:12:37 CET] <iive> ShadowOfThePast, wb [03:12:44 CET] <ShadowOfThePast> thanks, net died [03:12:59 CET] <iive> just put an av_realloc() before the return. [03:16:15 CET] <iive> if i read the code correctly, it should be "size" bytes long. [03:17:00 CET] <ShadowOfThePast> yea i was under the same impression [03:17:02 CET] <ShadowOfThePast> i'll try that now [03:17:40 CET] <iive> i guess that this function starts with 1mb buffer, just in case... [03:18:31 CET] <ShadowOfThePast> i'll know whether it helped in 46 minutes [03:20:20 CET] <iive> and it stays at 1mb [03:20:30 CET] <iive> well, tell me when i come back. [03:20:32 CET] <iive> gtg [03:20:40 CET] <ShadowOfThePast> aight [03:21:26 CET] <iive> don't forget to open ticket and send patch :D [03:27:33 CET] <iive> who wrote that function?... [03:28:34 CET] <iive> anyway... [03:28:39 CET] <iive> n8 ppl [04:15:01 CET] <ShadowOfThePast> shrinking the buffer did indeed fix the issue, the stream plays through fine now [04:41:17 CET] <ShadowOfThePast> there we go [04:41:20 CET] <ShadowOfThePast> good night [10:57:17 CET] <j-b> 'morning [00:00:00 CET] --- Sun Dec 30 2018

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ffmpeg.log.20181229
by burek 30 Dec '18

30 Dec '18

[02:22:53 CET] <sandreas> Could anyone provide an example for creating a fingerprint with ffmpeg using the chromaprint extension? I did not find anything in the docs... [02:26:09 CET] <sandreas> Docs: https://www.ffmpeg.org/ffmpeg-formats.html#chromaprint-1 [07:27:29 CET] <hungrywolf> I am trying to build omxplayer, it involves building ffmpeg [07:27:47 CET] <hungrywolf> The process saw an error [07:27:55 CET] <hungrywolf> I have cleared the console by mistake [07:28:16 CET] <hungrywolf> I am not sure if it stopped at ffmpeg build or omxplayer build itself [07:28:36 CET] <hungrywolf> can some one tell me if I can verify that ffmpeg is built properly [07:28:46 CET] <hungrywolf> and if I can, how should I do that? [08:38:11 CET] <th3_v0ice> How much complexity does drawtext filter introduces? [15:23:54 CET] <geomobb> Hello Is there anyway to pause when in -nodisp mode? When the display is open you can just hit "p" to pause the audio/video stream. I want to pipe that in using subprocess.communicate("p") using the python subprocess module [15:24:59 CET] <geomobb> (using ffplay) [15:37:37 CET] <Sandreas> Could someone give me an example, how to create an audio fingerprint with "chromaprint" muxer? (https://www.ffmpeg.org/ffmpeg-all.html#chromaprint-1) [16:22:32 CET] <Sandreas> Found it: ffmpeg -i "input.mp3" -f chromaprint fingerprint.txt [00:00:00 CET] --- Sun Dec 30 2018

1 0

ffmpeg-devel.log.20181228
by burek 29 Dec '18

29 Dec '18

[00:22:58 CET] <cone-847> ffmpeg 03Carl Eugen Hoyos 07master:17aed996bc35: lavf/vividas: Do not increase extradata_size after allocation. [00:22:59 CET] <cone-847> ffmpeg 03Carl Eugen Hoyos 07master:01db03f15848: lavf/vividas: Support demuxing on big-endian hardware. [00:23:00 CET] <cone-847> ffmpeg 03Carl Eugen Hoyos 07master:1a6cca19898c: configure: Make sure libpostproc can be found if -rpath-link doesn't work. [01:04:13 CET] <jamrial> BBB: can you take a look at the av1 frame split patch whenever you have time? [01:04:24 CET] <jamrial> you don't need to understand how cbs works internally, just know that you pass it a packet and it returns an array of obus with filled values plus size and data pointers to the start of the revelant obu [01:05:19 CET] <jamrial> much like in the vp9 superframe split, i'm using those values to create packet references pointing to different offsets in the buffer [01:10:15 CET] <BBB> ok [01:10:19 CET] <BBB> so it's obu splitter basically? [01:11:47 CET] <jamrial> not into individual obus, into groups of obus that form a frame [01:12:08 CET] <jamrial> so it could be a frame obu, or a frame header obu + one or more tile group obus [01:14:33 CET] <jamrial> seq header is included with the first frame if present, any unrelated obu is included if present in between related obus (like padding or whatever in between frame header and tile groups and weird things like that), and any unrelated trailing obu in included with the last frame [01:15:28 CET] <jamrial> i could use the writing features from cbs to actually create entire new packets with cherry picked obus, but that's slow and this bsf is meant to be used with a decoder, much like the vp9 one [01:17:43 CET] <atomnuker> what's it meant to be fed by? shouldn't containers take care of that? [01:18:32 CET] <BtbN> Well, users tend to do weird things [01:18:36 CET] <BtbN> like, not using containers [01:20:19 CET] <jamrial> atomnuker: you feed it a temporal unit, which can contain several frames [01:21:03 CET] <jamrial> it's only useful for a decoder for frame multi threading purposes [01:23:55 CET] <atomnuker> ah, I see [11:41:48 CET] <durandal_1707> in all internet there is single ambisonic download, ignoring youtube untagged crap [12:27:01 CET] <cone-293> ffmpeg 03Paul B Mahol 07master:cfa7709d05b1: avcodec/wavpack: fix decoding of files with many channels [12:55:24 CET] <cone-293> ffmpeg 03David Bryant 07master:db109373d87b: avformat/wvdec: detect and error out on WavPack DSD files [14:05:51 CET] <cone-293> ffmpeg 03Carl Eugen Hoyos 07master:8d422e7fef97: Makefile: Allow "make clean" to delete compat/atomics/pthread/stdatomic.o [00:00:00 CET] --- Sat Dec 29 2018

1 0

ffmpeg.log.20181228
by burek 29 Dec '18

29 Dec '18

[00:13:23 CET] <octet> sorry, I lost connection. did anyone reply to my inquiry about ASS to SRT subs? [00:24:53 CET] <pink_mist> no [00:28:20 CET] <octet> I guess I will just use "sed -i 's/size="36"/size="20"/' *.srt" for now [00:40:03 CET] <beandog> octet, what are you trying to do? [00:40:51 CET] <octet> trying to extract some SSA/ASS subtitles, and create SRTs with proper size as the original [00:41:00 CET] <beandog> ah [00:41:06 CET] <beandog> good luck. :) [00:41:18 CET] <octet> my media center forces transcode with SSA subs, so I need SRT [00:41:30 CET] <beandog> what's your source? [00:41:38 CET] <beandog> can you grab it from open subtitles? [00:41:52 CET] <octet> no, it is embedded in an mkv [00:41:56 CET] <beandog> lame [00:42:16 CET] <octet> yeah, my hacked solution will work fine for now though [00:48:07 CET] <fantastictoys> when i convert from flac to opus, i find that my file has been truncated. how can i fix this? [00:50:18 CET] <relaxed> fantastictoys: pastebin.com your command and output [00:59:21 CET] <fantastictoys> command was "ffmpeg -i 01-06\ Shake\ It\ Off.flac 01-06\ Shake\ It\ Off.opus" output is here https://ptpb.pw/kXPT [01:22:37 CET] <relaxed> looks like a decoding error. Is the track complete? [01:28:15 CET] <relaxed> fantastictoys: see if this decodes it correctly, flac -d 01-06\ Shake\ It\ Off.flac [01:34:33 CET] <fantastictoys> relaxed: lol the source file is incomplete [01:36:41 CET] <fantastictoys> relaxed: thanks for your help [05:32:06 CET] <NGTmeaty> Hey guys. https://trac.ffmpeg.org/ticket/1291 doesn't seem to be fixed. Can anyone else verify as well? [06:25:00 CET] Last message repeated 1 time(s). [09:31:08 CET] <tempuser> For every 5 seconds that go by, I want just 1 frame extracted. What's the command for that? [09:31:36 CET] <tempuser> I'm trying ffmpeg -i video.mp4 -framerate 1/120 image-%03d.png [11:04:42 CET] <pagios> hi all, i did a modprobe v4l2loopback and i got /dev/video0 how can i get /dev/video1 , 2 ,3 ,4 ? [11:22:29 CET] <relaxed> modprobe v4l2loopback devices=5 [11:22:55 CET] <relaxed> is this for webrtc? [11:33:53 CET] <pron> so,i got video like this 720x576 [SAR 64:45 DAR 16:9] and if add png overlay it gets stretched, i suspect it is related to sar, but i am not entirely sure how to fix it properly [11:34:00 CET] <pron> any hints/tips? [11:36:16 CET] <pron> what i tried was to scale overlay width i_w/(64/45):in_h but i am relay unsure if it is correct way [11:38:55 CET] <relaxed> what is the frame size of the png? [11:40:44 CET] <pron> relaxed: Stream #0:0: Video: png, rgba(pc), 122x193, 25 tbr, 25 tbn, 25 tbc [12:05:05 CET] <relaxed> pron: bc -l <<<'122*(64/45)/(122/193)' [12:07:08 CET] <pron> relaxed: can you explain me what is that? [12:11:13 CET] <relaxed> width of png * (SAR you want to match) / ( pngwidth/pngheight) = final png height [12:12:56 CET] <pron> i tought its only widh of pixels that is affected by non square pixel thing [12:14:29 CET] <relaxed> hmm, it's early. maybe I'm wrong :) [12:14:36 CET] <pron> so what i did was i resized ping to 122/(64/45):193 [12:14:43 CET] <pron> png [12:15:01 CET] <pron> i just wanted to understand if i am not messing up with sar par and what not [12:15:15 CET] <pron> since i dont fully understand those things [12:15:29 CET] <pron> i do think i get the idea =) [12:15:39 CET] <pron> and the result looked fine ;D [12:52:35 CET] <skident> Hi there [12:53:24 CET] <skident> Is it possible to change AVFilter options without re-creating the whole filtergraph& [12:53:26 CET] <skident> Is it possible to change AVFilter options without re-creating the whole filtergraph? [12:56:30 CET] <durandal_1707> skident: no, use filter commands [12:56:42 CET] <durandal_1707> if filter have such support [12:57:02 CET] <durandal_1707> skident: in what filter(s) are you interested? [12:57:31 CET] <skident> acompressor, agate, equalizer, alimiter [12:58:48 CET] <skident> durandal_1707: are there any list of commands for each filter? [12:59:01 CET] <durandal_1707> equalizer supports commands [12:59:24 CET] <durandal_1707> skident: they are documented in usual filter documentation [12:59:29 CET] <skident> I see [13:00:08 CET] <durandal_1707> http://ffmpeg.org/ffmpeg-filters.html#toc-equalizer [13:01:16 CET] <skident> durandal_1707: Thanks for link. [13:02:44 CET] <skident> durandal_1707: I have a problem with re-creating a new filtergraph and replace an old one with new one. After replacing filtergraphs there are some audio clipping for less than 1 sec., but I can hear it [13:04:00 CET] <durandal_1707> skident: you cannot simply replace filtergraph and expect no clippings with realtime filtering [13:04:33 CET] <skident> yeah, I see. But how to do it smoothly?! [13:05:53 CET] <durandal_1707> filter audio with both filtergraphs from beginning and do crossfade when switching to another one? [13:07:03 CET] <skident> For example, I use only filters with zero latency and I could try to filter the last audio frame through both filtergraphs (old and new) and then merge these buffers somehow to avoid clips [13:08:06 CET] <durandal_1707> yes, but this is at level above ffmpeg [13:08:48 CET] <skident> durandal_1707: do you mean, I need a third filtergraph with only one filter 'crossfade' and pass both buffers through it to get a "normalized" audio? [13:09:31 CET] <durandal_1707> no, ffmpeg cant do what you seek, also with 0 latency you can not genereally do what you seek [13:10:47 CET] <durandal_1707> generally when doing new filtergraphs, you loose old state of old filtergraph [13:11:07 CET] <skident> so what then does crossfade? Isn't it smooth two audio streams? [13:11:52 CET] <durandal_1707> yes, but if you gonna use it, then you should use yet another filtergraph which complicates stuff [13:12:01 CET] <skident> :) [13:12:48 CET] <durandal_1707> and sure can not be 0 latency [13:13:23 CET] <skident> could you please explain why 0 latency is bad in this case? [13:13:49 CET] <durandal_1707> it is not bad, you just can not do it without latency [13:14:48 CET] <skident> I meant to say, that I push N samples into my filtergraph and then I receive the same amount of filtered samples [13:15:01 CET] <durandal_1707> also what parameters you need to change with what filters? as I may implements commands for filters missing them [13:16:04 CET] <durandal_1707> skident: yea, but some filters may introduce unwanted delay, i fixed one such example recently [13:16:05 CET] <skident> Because some of filters like 'loudnorm' can't work like this. Such filter waits until some amount of samples will be pushed and then can do its stuff [13:16:21 CET] <durandal_1707> yes, i know [13:16:40 CET] <skident> so, I meant that I don't use such filters [13:16:48 CET] <durandal_1707> ok [13:17:16 CET] <durandal_1707> skident: how big is your input amount of samples? [13:17:48 CET] <skident> something like 1600 samples, 2 channels [13:18:18 CET] <durandal_1707> so 1600 samples per channel for 44100? [13:18:24 CET] <skident> 48000 [13:18:33 CET] <skident> fltp format [13:22:39 CET] <skident> I'd like to change almost all parameters of 'acompressor' [13:25:35 CET] <durandal_1707> if parameters are changed abruptly some artifacts may happen [13:27:20 CET] <durandal_1707> so some interpolation might be needed [13:28:57 CET] <skident> can you suggest any interpolation algo? [13:29:57 CET] <durandal_1707> you just change parameters to final value in small step across timeline [13:33:12 CET] <skident> sounds good. Seems like it will cause some amount of re-creation filtergraph. [13:34:40 CET] <durandal_1707> well, i meant commands non parameters, parameters as commands [13:35:20 CET] <durandal_1707> any recreation of filtergraph in middle of processing will cause artifacts [13:36:04 CET] <skident> oh, sure. But for as You said and I also checked, only 'equalizer' supports commands [13:37:50 CET] <skident> oh, sure. But as You said and I also checked, only 'equalizer' supports commands [13:39:07 CET] <durandal_1707> yes, adding commands to acompressor seems rather trivial [13:39:57 CET] <durandal_1707> also you can already use ladspa filter with commands just look for theirs variants of compressor or limiter [13:40:16 CET] <durandal_1707> if you are on linux [13:40:54 CET] <skident> unfortunately, macos and windows ( [13:45:02 CET] <durandal_1707> i think macos can use ladspa [13:49:00 CET] <skident> durandal_1707: thank you, for your help. [13:52:09 CET] <pron> can i use nvenc together with -filter_complex? [13:53:57 CET] <BtbN> sure, but be aware what kind of frames each filter takes [13:55:30 CET] <pron> i am intrested and adding png overlay [13:59:54 CET] <BtbN> If you're not using hardware frames, there is nothing different there [17:39:32 CET] <microcolonel> Hmm [17:39:51 CET] <microcolonel> I have no idea what the individual components of encoding cost for e.g. baseline H.264 [17:40:19 CET] <microcolonel> is there a way to take shortcuts transcoding MJPEG to a genuine video format? [17:54:58 CET] <pi-> Could anyone critique this code? https://paste.pound-python.org/show/7tkwvakxrHNti4gLn344/ [17:55:23 CET] <pi-> I'm guessing it is pretty horrible, but it seems to work. [17:55:45 CET] <pi-> The line I contributed was: $FFMPEG -i "$SRC" -i LAST_FRAME.mp4 -filter_complex "[0:v] [1:v] concat=n=2:v=1 [v]" -map "[v]" "$DST" [17:56:04 CET] <pi-> Because the `$FFMPEG -f concat -safe 0 -i list.txt -c copy "$DST"` line was glitching the target vid. [17:56:39 CET] <pi-> But have I got that `-filter_complex "[0:v] [1:v] concat=n=2:v=1 [v]" -map "[v]"` part down right? [17:57:33 CET] <furq> pi-: shorter than what [17:58:21 CET] <pi-> furq: ah that error message is obsolete, tx [17:58:36 CET] <furq> if you just want to repeat the last frame then there are easier ways to do it [17:58:44 CET] <furq> https://ffmpeg.org/ffmpeg-filters.html#framesync [17:58:55 CET] <furq> maybe overlay with eof_action=repeat or something [17:59:28 CET] <furq> lots of filters support framesync so there's probably something even easier [18:00:04 CET] <pi-> I did have another technique before, but it was slooooow: https://paste.pound-python.org/show/ef7Us9FUjNBNVBkEmubh/ [18:00:45 CET] <furq> well if you're using filters instead of the concat demuxer then both ways reqire you to reencode the entire video [18:01:17 CET] <pi-> I wish I could find an ffmpeg engineer to rewrite my Python-wrapping-ffmpeg lib. It feels really unstable. [18:01:27 CET] <furq> that other paste is basically what i was suggesting [18:01:32 CET] <furq> i'd be surprised if it was much slower [18:01:35 CET] <pi-> I don't suppose you might be interested... furq? [18:01:46 CET] <furq> not today [18:03:08 CET] <pi-> well, if any time in the future, do get back to me. What I have at the moment is a hack job. I would love to have someone who knows what they are doing look over it. [18:45:45 CET] <jinboli> Hello, does anyone know if OpenMAX supports hardware encoding on Android? I notice OpenMAX partially supports hardware encoding on Android. [18:47:39 CET] <Mavrik> ? [18:50:57 CET] <pink_mist> jinboli: did ... you just answer your own question in the same breath as you asked it? [18:51:17 CET] <Mavrik> Yeah, I'm not quite sure what that was [19:08:33 CET] <jinboli> Sorry for the confusion. I'm new to ffmpeg. I'm trying to implement hardware encoding for video call on an Android app. I don't quite understand what "partially supported" mean. And I don't quite understand the difference between "standalone" and "hardware input" in the FFmpeg API Implementation Status [19:10:50 CET] <Mavrik> Android has MediaCodec APIs for encoding [19:10:54 CET] <Mavrik> OpenMAX isn't for app use [19:16:49 CET] <furq> jinboli: mediacodec will use a hardware encoder if there's one available [19:18:52 CET] <mrskman> Hello! Is there any way how to select a video stream by bitrate/resolution? I have a dash stream input and I don't want to map the best quality. Problem is that streams in the dash manifest are randomly ordered with every request so I can't map them by index. [19:19:51 CET] <jinboli> But it says on the FFmpeg API Implementation Status table that MediaCodec only support hardware decoding on Android. So the table is not up to date? [19:21:28 CET] <furq> jinboli: i meant mediacodec in general, not the ffmpeg implementation of it [19:22:02 CET] <jinboli> Ok, thanks a lot [20:19:32 CET] <Bombo> hi [20:21:04 CET] <Bombo> i want to record video from v4l device + alsa audio, when i do "ffmpeg -thread_queue_size 1024 -framerate 25 -video_size 720x576 -c:v libx264 -f v4l2 -i /dev/video0 -f alsa -i hw:3 out.mkv" i get "unknown decoder 'libx264'" [20:21:55 CET] <Bombo> i want to ENcode the thing [20:22:14 CET] <Bombo> i don't get why it says DEcoder [20:39:29 CET] <Bombo> ok got the right order now i think: ffmpeg -thread_queue_size 1024 -framerate 25 -video_size 720x576 -f v4l2 -i /dev/video0 -thread_queue_size 1024 -f alsa -i hw:3 -c:v libx264 -preset superfast -crf 23 -vf yadif -c:a libmp3lame -b:a 128k -t 00:00:10" out.mkv [20:39:36 CET] <Bombo> seems to work ;) [00:00:00 CET] --- Sat Dec 29 2018

1 0

ffmpeg-devel.log.20181227
by burek 28 Dec '18

28 Dec '18

[12:34:58 CET] <cone-658> ffmpeg 03Paul B Mahol 07master:aece1eb1e978: avfilter/af_headphone: use av_log2() [12:34:58 CET] <cone-658> ffmpeg 03Paul B Mahol 07master:b5e0a0fe3acb: avfilter/af_sofalizer: use av_log2() [12:57:12 CET] <cone-658> ffmpeg 03Paul B Mahol 07master:ff0b4d5af78f: avfilter/af_afir: remove not needed operations [14:36:52 CET] <cone-658> ffmpeg 03Paul B Mahol 07master:f266d2ac90c8: avfilter/af_afir: remove not needed nb_samples from filter context [15:53:54 CET] <hans_> https://ffmpeg.org/bugreports.html is talking about connecting via FTP to upload.ffmpeg.org , but upload.ffmpeg.org resolves to 88.191.250.119 , and 88.191.250.119 isn't responding on port 21, and port 21 is the default port for FTP, and https://ffmpeg.org/bugreports.html doesn't say anything about using a non-standard port to connect [15:54:35 CET] <JEEB> yea small uploads go to the trac and it seems like we're now utilizing the videolan sample storage for samples [15:57:22 CET] <hans_> think i could get away with submitting a youtube-dl command in the "how to reproduce" section? [15:59:37 CET] <hans_> ("videolan sample storage" as a google search query didn't yield any useful results after a quick search) [16:02:02 CET] <JEEB> most likely since the youtube-dl should be reproducable as long as it doesn't need credentials [16:02:43 CET] <hans_> thanks, i'll TIAS, and no, it doesn't require credentials [16:56:06 CET] <kierank> durandal_1707: do you understand the point of michaelni making fuzzed streams faster to decode [17:05:40 CET] <durandal_1707> kierank: no [17:05:51 CET] <kierank> ok good [17:05:58 CET] <kierank> it's not just me who thinks its crazy [17:09:13 CET] <atomnuker> definitely not [17:23:15 CET] <durandal_1707> how to get uniform partitioned convolution not delay everything by partition size? [18:05:58 CET] <cone-501> ffmpeg 03Paul B Mahol 07master:dbf43ace214f: afilter/af_afir: remove invalid delay [22:40:40 CET] <durandal_1707> bye folks [23:01:03 CET] Action: BradleyS gives due [23:01:31 CET] <BradleyS> silly textual, don't mind me [23:36:40 CET] <cone-847> ffmpeg 03gxw 07master:d86f698e3864: avcodec/mips: [loongson] optimize theora decoding in vp3dsp. [00:00:00 CET] --- Fri Dec 28 2018

1 0

ffmpeg.log.20181227
by burek 28 Dec '18

28 Dec '18

[04:02:52 CET] <memfrob> 'low errbody! Quick question. What difference is there between native and external libraries/codecs besides the fact that one is internal and other is external? [04:03:58 CET] <memfrob> There is a difference in file size (re-encoded via ffmpeg) when comparing internal vs. external as well as slightly different algorithm? [04:04:16 CET] <memfrob> question mark because I don't know for sure. [04:05:35 CET] <pink_mist> absolutely different algorithms [04:05:52 CET] <pink_mist> otherwise there wouldn't be a point to link to the external one [04:06:33 CET] <pink_mist> whether the external one has a better algorithm or not probably is a case by case thing [04:06:51 CET] <memfrob> how can internal be "true" to the real codec then if it uses a different algorithm? [04:07:26 CET] <pink_mist> because the codec isn't defined by the implementation (whether external or internal) [04:08:27 CET] <memfrob> does ffmpeg have an internal x264 codec or only external? [04:09:10 CET] <memfrob> pink_mist, you're a really knowledgeable guy with this stuff, your help is very appreciated. [04:09:22 CET] <pink_mist> I'm really not that knowledgeable [04:09:26 CET] <pink_mist> but I try [04:10:24 CET] <memfrob> i think it does because -c:v x264 works but -c:v x265 does not work [04:10:45 CET] <pink_mist> x264 is a specific implementation of h.264 ... and I think there are other alternative implementations that ffmpeg can use, but I don't know if any of them are internal ... either way, the x264 implementation is one of if not the best h.264 implemenations that exist [04:10:49 CET] <memfrob> (x265 must have lib before it, and lib* means external if i have that right) [04:11:17 CET] <pink_mist> x264 is also external [04:11:37 CET] <Hello71> modern video formats have very broad specifications, there are many ways to implement encoders [04:11:39 CET] <memfrob> ah, so just because lib is not there before it, doesn't mean it's internal. [04:11:47 CET] <pink_mist> yeah [04:12:06 CET] <memfrob> Hello71, would you say that's a good thing or a bad thing? [04:12:09 CET] <Hello71> the first mp3 encoders were complete garbage [04:12:33 CET] <Hello71> which is part of the reason why some people still think that lossless is always better [04:12:46 CET] <memfrob> lame is good for what it does -- i know it's not the point [04:14:12 CET] <Hello71> lame is great but its first release was 5 years after the mp3 format itself [04:14:18 CET] <kepstin> well, the point is that lame is *much* better than the initial mp3 encoders, in part because it uses different algorithms for parts of the encoding. [04:15:00 CET] <kepstin> since for many codecs, you can implement the encoder in different ways as long as a reference decoder (or a decoder written to the specification) can decode it. [04:15:29 CET] <memfrob> so perhaps if the specs weren't so broad, those first 5 years wouldn't have been such a mess? [04:15:33 CET] <memfrob> similar to the ACPI spec.. [04:15:48 CET] <Hello71> unfortunately, people only recently re-realized that it's good for the encoding to be open, but the decoding should be bit-exact [04:15:56 CET] <memfrob> interesting how linux and windows both handle power management completely different :P [04:16:12 CET] <kepstin> if the encoder had been specified rather than the format/decoder, we'd have been stuck with a bad format until someone made a completely new different format [04:16:12 CET] <Hello71> not really, backwards compatibility is important [04:16:16 CET] <kepstin> instead, mp3 got better [04:16:34 CET] <Hello71> this is the time of portable mp3 players [04:16:46 CET] <Hello71> it would be impossible to popularize if you needed to upgrade every year to play new files [04:17:02 CET] <Hello71> er, this was the time [04:17:03 CET] <memfrob> Hello71, that's where commercializing comes into play. [04:17:15 CET] <kepstin> heh, my first mp3 player (it was an iriver 128mb flash player) actually supported vorbis. [04:17:42 CET] <memfrob> businesses help prevent legacy APIs that are total crap to continue being used everywhere. [04:17:57 CET] <Hello71> hm... I thought lame was better than libvorbis [04:18:23 CET] <memfrob> hold on... what i meant was.. businesses allow crappy APIs to continue being crap, and shove it down everyone's throat. [04:19:16 CET] <Hello71> memfrob: in general, external libraries are called libx in ffmpeg [04:20:22 CET] <kepstin> external libraries (encoders and decoders) are used typically when there are no internal encoder/decoder in ffmpeg, or when there is an internal encoder/decoder but it's missing features or lacking in performance compared to an external library. [04:20:31 CET] <memfrob> backwards compatibility isn't always a good thing, look at BIOS and x86. [04:20:46 CET] <memfrob> 1980s and still going! and not in a good way! [04:22:13 CET] <memfrob> Hello71, so I was half right about the libx thing [04:22:21 CET] <memfrob> not _always_ but in general. [04:23:10 CET] <kepstin> i think they always are, but in some cases there's aliases that allow the name without lib to work; and if the external library is the only encoder, you can just say the codec name and ffmpeg will pick the only available encoder. [04:23:30 CET] <memfrob> wouldn't it be best to fix that? i'm sure it'd take like 5 minutes to go through the code and remove all the codecs that do not start with "lib" and point to internal libraries. [04:24:03 CET] <kepstin> (this confused some people recently - "opus" used to select libopus, but an internal opus encoder was added recently so you have to say "libopus" to get the external one now) [04:24:09 CET] <memfrob> see!!!! [04:24:47 CET] <memfrob> kepstin should get commit rights if he doesn't already :P [04:25:04 CET] <kepstin> don't really need them, I have a few patches in already... [04:25:41 CET] <kepstin> but anyways, the thing where it selects an available encoder when someone uses a codec name is intentional, and would break backwards compat if it was changed, so :/ [04:25:50 CET] <kepstin> (still not recommended tho) [04:26:23 CET] <memfrob> in that case, using x265 should select libx265, not say "no codec: x265" [04:26:31 CET] <kepstin> x265 isn't a codec [04:26:39 CET] <kepstin> the codec name in ffmpeg for that is 'hevc', iirc [04:26:46 CET] <memfrob> x264 is a codec though..? [04:26:50 CET] <pink_mist> no [04:26:58 CET] <memfrob> ok well why the heck does x264 work but x265 doesn't? [04:26:59 CET] <kepstin> x264 is an *encoder implementation* [04:27:09 CET] <kepstin> must be an alias for x264, i dunno why :/ [04:27:36 CET] <pink_mist> I don't know why either, but it's still neither internal, nor the name of a codec; it's just an implementation of a codec [04:28:10 CET] <pink_mist> afaik the best implementation of that codec [04:28:42 CET] <memfrob> does anything else even code h.265 besides x265? [04:29:00 CET] <memfrob> *encode [04:29:04 CET] <kepstin> there's reference and closed-source and hardware encoders for h265/hevc [04:29:10 CET] <pink_mist> some graphics cards can have hardware encoding capability for hevc I believe [04:29:13 CET] <pink_mist> yeah [04:29:17 CET] <pink_mist> and ffmpeg can use those [04:29:44 CET] <memfrob> oh yeah that's right! that's why using gpu-accel h.265 encoding will be different than using x265 directly [04:29:56 CET] <kepstin> it's not gpu-accel, it's hardware encoder [04:30:17 CET] <memfrob> hardware encoder provides gpu acceleration capability. [04:30:35 CET] <kepstin> it's not accelleration, that implies that an existing software implementation is made faster [04:30:42 CET] <kepstin> rather, it's a completely different encoder implementation [04:31:01 CET] <pink_mist> memfrob: it doesn't accelerate the x265 implementation; it's just its own (likely shitty) implementation, that'll likely be fast [04:31:08 CET] <memfrob> i didn't think it was _that_ different [04:31:25 CET] <memfrob> but jeeze, so it's completely unrelated to x265 [04:31:30 CET] <pink_mist> yes [04:31:37 CET] <kepstin> the hardware encoder is just a black box on the gpu die, you send it raw frames, it gives you back encoded frames. [04:31:39 CET] <memfrob> does that mean you can encode h.265 movies without the x265 codec even installed? [04:31:59 CET] <Hello71> you can use the reference implementation... [04:32:08 CET] <kepstin> nobody's patient enough for that ;) [04:32:12 CET] <Hello71> (although nobody does except for testing purposes) [04:32:33 CET] <memfrob> patient enough for what? [04:32:49 CET] <kepstin> patient enough to wait for the reference (research) encoder implementation to finish encoding something [04:33:18 CET] <memfrob> i'm a bit lost, does that statement have anything to do with using a hardware encoder without x265 on the system? [04:33:51 CET] <kepstin> the reference encoder is a different software encoder that you could use instead of x265 [04:33:58 CET] <kepstin> even without a hardware encoder [04:34:21 CET] <memfrob> i'm talking about using the hardware encoder. there still needs to be a layer that talks to the GPU driver to tell it to encode something. [04:34:37 CET] <memfrob> (for AMD cards this is called VCE) [04:36:05 CET] <memfrob> oh, i see.. ffmpeg has vaapi support which does that. [04:36:24 CET] <memfrob> vaapi is completely separate from x265 -- vaapi comes from mesa [04:36:54 CET] <memfrob> https://trac.ffmpeg.org/wiki/Hardware/VAAPI -- this answers that question [04:38:45 CET] <memfrob> am i talking too much? i had a very lonely christmas. [05:10:49 CET] <memfrob> Mom!! FFmpeg is lying to me! Stream mapping: [05:10:49 CET] <memfrob> Stream #0:0 -> #0:0 (h264 (native) -> h264 (h264_amf)) [05:11:09 CET] <klaxa> native decoder [05:11:11 CET] <klaxa> not lying [05:11:13 CET] <memfrob> ohh [05:11:16 CET] <memfrob> DEcoder [05:11:56 CET] <klaxa> :) [05:12:27 CET] <memfrob> so ffmpeg can have native decoders without native encoders, that's interesting. [05:13:06 CET] <memfrob> i'm deducting from that, that it's easier to write decoders than encoders. [05:13:23 CET] <pink_mist> much, usually [05:14:01 CET] <memfrob> i'm testing out h264_amf in windows -- non-AMD cards should be able to play this back right? [05:19:12 CET] <klaxa> h264 is h264, it shouldn't matter which encoder produced it [06:15:12 CET] <memfrob> people say the hardware encoders are kinda goofy, if it's ok to the naked eye, why is that... a problem? [06:15:31 CET] <memfrob> (minus licensing/proprietary/royalty crap) [06:17:19 CET] <memfrob> i don't even think you guys care so much about that. fedora for example doesn't even have ffmpeg in it's own repos. [07:08:35 CET] <mixfix41> hmm looking into this audio forwarding with ssh socat and pulse, on my regular desktop ffplay is known to stutter a bit song or video, but if i play it on the same processor as a raspberry 2 then the file will play smoothly as if i was using vlc on t he desktop [07:09:43 CET] <mixfix41> interesting [07:09:53 CET] <mixfix41> and firefox and programs play good anyway [07:10:08 CET] <mixfix41> so it is narrowed to programs also mplayer and ffplay [07:10:15 CET] <mixfix41> maybe ill have to try ou t more desktops [07:41:53 CET] <memfrob> "regular desktop" "but if i play it on the same processor as a raspberry 2" ... your desktop has the same processor as the rpi 2? [07:43:21 CET] <memfrob> mixfix41, x86 and ARM handle floating point differently, as well as the distribution on the rpi 2 has differences in it besides it being compiled for arm. [07:44:01 CET] <memfrob> different versions of ffmpeg perhaps, or pulse, etc. as well as a different kernel audio driver. [07:59:17 CET] <mixfix41> it could be a pulse issue , its just so present with the x86 it stutters a little bit but its bearable listening to this [08:00:18 CET] <mixfix41> just a funny thought since if it were mplayer and ffplay playing good just that would be with the other applications [08:01:09 CET] <mixfix41> though the raspberry also especially i think if i play it from the chromebook the issues vary also [08:01:36 CET] <mixfix41> though the chromebook is wifi even yet the processor is stronger than the rasp [08:01:53 CET] <mixfix41> though the processing doesnt really take much al beit [08:03:19 CET] <memfrob> one more "though" out of you, mister!!! [12:45:21 CET] <th3_v0ice> Is there a way to force the muxer to write the packets to the output even if the new packet PTS is less then the old packets PTS? [12:46:54 CET] <JEEB> umm [12:47:04 CET] <JEEB> that sounds completely normal with b-frames f.ex. [12:47:14 CET] <JEEB> just make sure your DTS is <= PTS at all times [12:47:35 CET] <th3_v0ice> JEEB: Sorry, I mean DTS [12:47:52 CET] <JEEB> 100% depends on the muxer [12:48:03 CET] <JEEB> if the muxer is telling you to f off, then you can't really do much about that [12:48:11 CET] <JEEB> also if possible, just fix timestamps on your side :P [12:48:17 CET] <th3_v0ice> For some UDP input, server is basically cycling the timestamps. [12:48:38 CET] <th3_v0ice> But after some period I will end up with a number that is bigger then INT64_MAX. [12:48:49 CET] <th3_v0ice> What can I do in that case? Close and reopen the output context? [12:48:58 CET] <JEEB> larger than INT64_MAX? [12:49:10 CET] <th3_v0ice> Continuous UDP streaming. [12:49:15 CET] <JEEB> even with MPEG-TS that takes QUITE the while [12:49:20 CET] <JEEB> with 90k time base [12:49:42 CET] <JEEB> because 33 bits is 26.5 hours [12:50:05 CET] <th3_v0ice> Ok, but what to do when it happens? [12:50:29 CET] <JEEB> if that really does happen, use a container where that can happen and I think in the packet signal a discontinuity [12:51:20 CET] <th3_v0ice> Does MPEG-TS support that? [12:51:53 CET] <JEEB> I'd say yes. also https://www.ffmpeg.org/doxygen/trunk/group__lavf__decoding.html seems to recommend avformat_flush in that case too [12:52:14 CET] <JEEB> ah no [12:52:23 CET] <JEEB> that's for input AVFormat contexts :) [12:53:22 CET] <th3_v0ice> Well the input cycling is done by the server that is delivering the packets. [12:53:40 CET] <th3_v0ice> So my best bet is to signal a packet discontinuity? How can I do that? [12:57:33 CET] <th3_v0ice> Its interesting that there is live_flv demuxer which supports packet discontinuity, but there is no such thing documented in the muxers. [14:19:43 CET] <tommy``> guys i've a software that allow to trim video with a specific command but i think it's not compatible with last ffmpeg version (i think) [14:19:50 CET] <tommy``> the lines is: '"#{encbin}" -ss #{startt} -sn -i "#{inpath}/#{input}" -vframes #{lenf} "#{prefix}/#{output}[#{startf}-#{endf}]-%05d.jpg"' [14:20:01 CET] <tommy``> i would like to change some parameters and save in .mp4 [14:20:14 CET] <tommy``> i know that to trim video i've to use something like "ffmpeg -i input.mp4 -ss 01:10:27 -to 02:18:51 -c:v copy -c:a copy output.mp4 [14:20:30 CET] <tommy``> but i dunno how to change the syntax before [14:22:55 CET] <tommy``> the tokens are: https://paste.ofcode.org/gVVvuf2kBvJsvLrA7dXWzP [14:27:04 CET] <Blacker47> tommy``, i can't help, but the syntax above will output selected frames recompressed to single jpeg-files (for every frame). i will not say it is itself "trimming". even if there is second step to put single images to a new video together it is not the best idea to recompress it twice. [14:27:36 CET] <tommy``> i've deleted that all the sintax between the ' ' [14:28:04 CET] <tommy``> without any specific syntax ffmpeg trim and made about 40 .jpg [14:32:22 CET] <Blacker47> tommy``, there is no one way to "trim" video files in ffmpeg. no idea why your software stops to work but it is not because you have to change the syntax to the bottom example. you have to stop using your software or contact the maintainer for a fix. the bottom syntax btw. will have problems if you want to trim your videos frame-wise but it is encoded with P,B-frames, because of the "copy" ffmpeg can cut only on I-frames then... [14:32:46 CET] <tommy``> oh ok [14:33:10 CET] <tommy``> i'm alazing the plugin that allow that and checking how ffmpeg works in it [14:33:53 CET] <tommy``> Blacker47 https://github.com/TypesettingTools/Aegisub-Motion/wiki/Trim-Configuration#… I'm reading here now [15:16:01 CET] <hans_> i'm trying to cut out a very specific part of a longer video, but the first video frame on the cut out version first appear 2 seconds into the audio cut out, any idea why? here's the command i'm using: ffmpeg -i "in.mkv" -c:v copy -c:a copy -ss 00:11:32 -t 10.8 -y test.mkv [15:16:54 CET] <hans_> (in the source, video frames are present for the 2 seconds that they're not present in the result) [15:17:22 CET] <hans_> when trying to view the result in VLC, the first 2 seconds are just audio. [15:18:42 CET] <hans_> well, "ffmpeg version git-2017-12-25-613f789", guess i could try updating [15:24:05 CET] <hans_> updated to "ffmpeg version N-92820-gd702600350 built with gcc 8.2.1 (GCC) 20181201", same issue. [15:48:08 CET] <pron> hey any ideas why my overlayed watermark png is stretched out and how to fix it? [15:50:41 CET] <hans_> https://ffmpeg.org/bugreports.html is talking about connecting via FTP to upload.ffmpeg.org , but upload.ffmpeg.org resolves to 88.191.250.119 , and 88.191.250.119 isn't responding on port 21 [15:51:14 CET] <hans_> (and port 21 is the default port for FTP, and https://ffmpeg.org/bugreports.html doesn't say anything about using a non-standard port to connect) [16:13:02 CET] <hans_> i added the video cutting issue to the bugtracker - https://trac.ffmpeg.org/ticket/7640 [16:37:56 CET] <klaxa> cutting with copy is only possible at keyframes [16:38:06 CET] <klaxa> this is not a bug but an inherent property of how video encoding works [21:38:06 CET] <octet> I am trying to convert embedded SSA/ASS subs to external SRT subs for a group of video files. here is the command I have: https://pastebin.com/u3eHYssS . How can I make it look like the original subs? At the moment, it outputs to boldface and size 3 which is quite large compared to the original ASS/SSA subtitles [21:42:02 CET] <Mavrik> eh? [21:42:42 CET] <octet> size 36* <font face="Open Sans Semibold" size="36"><b>Blah Blah</b></font> [21:58:58 CET] <pikapika> Hi [21:59:24 CET] <pikapika> Is there a way to make certain types of changes 'in place' in a file, ie without separate input and output file? [21:59:33 CET] <pikapika> Changes such as editing meta data etc [22:01:43 CET] <durandal_1707> no [22:28:26 CET] <dingwat> Is there an option similar to -pattern_type glob for ffplay? I have a directory of png frames I'd like to play but they're named by date rather than sequentially [23:53:17 CET] <fantastictoys> when i convert from flac to opus, i find that my file has been truncated [00:00:00 CET] --- Fri Dec 28 2018

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ffmpeg-devel.log.20181226
by burek 27 Dec '18

27 Dec '18

[10:47:17 CET] <cone-590> ffmpeg 03Paul B Mahol 07master:530fc345ec73: avfilter/af_sofalizer: add fltp sample format support [10:47:17 CET] <cone-590> ffmpeg 03Paul B Mahol 07master:8e4e2c9e40b1: avfilter/af_sofalizer: use float constants [10:47:17 CET] <cone-590> ffmpeg 03Paul B Mahol 07master:60e9007be2e9: avfilter/af_sofalizer: fix typo in comments [10:47:17 CET] <cone-590> ffmpeg 03Paul B Mahol 07master:92ed9316bba1: avfilter/af_sofalizer: fix regression after 7ea4b928a264 [10:47:17 CET] <cone-590> ffmpeg 03Paul B Mahol 07master:9ce96a744d6f: avfilter/af_headphone: fix regression after 7c201e420 [12:15:48 CET] <cone-590> ffmpeg 03hwrenx 07master:7499d0b0159f: lavc/libdavs2: enable multithread [12:22:30 CET] <cone-590> ffmpeg 03vkot 07master:3ee735901e08: avformat/hls.c: Properly free prev_segments dynarray after playlist parsing [12:32:29 CET] <cone-590> ffmpeg 03Adrian Guzowski 07master:c822d70c3d2d: Fix usage of temp_file flag in hls_flags option. [12:32:30 CET] <cone-590> ffmpeg 03Steven Liu 07master:cef857da481e: avformat/hlsenc: remove unused variable to fix compiler warning [14:40:30 CET] <jamrial> j-b: https://aomedia.googlesource.com/aom/+/9b1252eab0616d2c1f6d7990c6256441c0b6… they reverted it [14:40:56 CET] <j-b> jamrial: wow [14:41:02 CET] <j-b> I'm a bit surprised. [14:42:46 CET] <jamrial> it was a pretty bad break, affecting the codec control IDs which are used by everyone. it wasn't like the previous breaks where they removed random unused defines inherited from vpx [14:43:42 CET] <j-b> jamrial: reminds me that we should make it official that dav1d API will be frozen soon. [14:44:07 CET] <j-b> There is still a new call needed IIRC and the timestamp discussion, but then it should be frozen [16:09:34 CET] <jamrial> durandal_1707: wtf was that reply? [17:21:15 CET] <kierank> michaelni: your 6s timeout is too large [17:21:21 CET] <kierank> need to make it smaller [17:35:59 CET] <ramiro> I got a crash on fate-4xm-1 due to a buffer not being aligned enough for avx's clear_blocks(). [17:36:07 CET] <ramiro> I'm too lazy to send a patch, but here's the fix: https://gist.github.com/ramiropolla/86497b62a66e8dc4cbe179db6a7d62b8 [17:36:53 CET] <ramiro> (also in the diff is a similar change for dnxhddec.c, but since it's the first element in the struct it will be correctly aligned anyways) [17:50:28 CET] <JEEB> ramiro: I guess you don't have git send-email configured? :) [17:50:55 CET] <JEEB> also I think we had some macro for the alignment [17:52:28 CET] <JEEB> I remember lately some patch changed some numeric magical value to the alignment requirement value which then differs between AVX2/AVX etc [17:53:21 CET] <jamrial> the blockdsp api requires 32byte alignment, so that patch is correct [17:53:42 CET] <JEEB> yes, I am just trying to quickly git grep for the macro [17:53:55 CET] <JEEB> because I remember there was a single macro that would get bumped by new SIMDs [17:55:02 CET] <JEEB> I know av_cpu_max_align is a thing in cpu.h, but I did remember something else being around, too [17:55:30 CET] <jamrial> there's the padding define, but that one is currently 64 [17:55:55 CET] <JEEB> yes, because AVX2 [17:55:58 CET] <JEEB> (or 512?) [17:56:45 CET] <jamrial> 512 :p [17:57:13 CET] <jamrial> STRIDE_ALIGN is 16, 32 or 64 depending on configure time options and host [17:57:21 CET] <JEEB> yup [17:59:28 CET] <cone-665> ffmpeg 03Ramiro Polla 07master:4d34205bec70: avcodec/4xm: fix block alignment [17:59:28 CET] <cone-665> ffmpeg 03Ramiro Polla 07master:c268712895f6: avcodec/dnxhddec: fix block alignment [18:01:12 CET] <JEEB> anyways, my comments mostly stemmed from the thing that I notice every now and then that we have missed alignment requirements and have to bump things in various places. so in a way when such a thing happens my first reaction is not to go for the actual currently required alignment, but instead the current max SIMD alignment requirement (if such exists like STRIDE_ALIGN), since that way every time that [18:01:18 CET] <JEEB> gets bumped everything gets bumped as well [18:01:18 CET] <JEEB> I hope this sort of logic makes sense :P [18:06:59 CET] <kierank> that was the idea a few years ago but mru didn't like it [18:07:06 CET] <kierank> forcing arm users to align to 32-bytes [18:08:14 CET] <ramiro> JEEB: dumping a patch on IRC is faster than git send-email :) [18:08:21 CET] <ramiro> jamrial: thanks! [18:08:27 CET] <atomnuker> you know at configure time the arch so you can figure the max padding too [20:29:21 CET] <cone-665> ffmpeg 03Lauri Kasanen 07master:8dd9df9ecd25: swscale/output: Altivec-optimize float yuv2plane1 [21:59:21 CET] <cone-665> ffmpeg 03Paul B Mahol 07master:8443462e4a56: avfilter/af_afir: remove unused variable [23:08:04 CET] <cone-665> ffmpeg 03Paul B Mahol 07master:d702600350de: avfilter/af_afir: remove dead store variable [00:00:00 CET] --- Thu Dec 27 2018

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ffmpeg.log.20181226
by burek 27 Dec '18

27 Dec '18

[02:09:01 CET] <oliv3> Hi there, any reason why swscale fails using SWS_FAST_BILINEAR on NetBSD with a mprotect() error ? [02:11:14 CET] <oliv3> I get [swscaler @ 0x6f90befb6000] mprotect failed, cannot use fast bilinear scaler when sws_getContext is called [02:11:39 CET] <oliv3> It works with SWS_BILINEAR, though [02:15:09 CET] <iive> oliv3, the fast one might be the version where it uses executable code modification. [02:15:54 CET] <iive> aka, it has the assembly precompiled, but it modifies the code to put some constants in there, like width/height of the image, or the ratios. [02:18:36 CET] <oliv3> I don't have this error on OpenBSD, and if some privileges are required (which I thought could be), running as root doesn't help either [02:19:51 CET] <oliv3> And to me, Open is way stricter than Net :) [02:23:27 CET] <oliv3> Simply put, FAST_BILINEAR works on all Linuces, FreeBSD, OpenBSD; it's the first time I encounter this error on NetBSD [02:24:26 CET] <oliv3> Anyhow, using plain SWS_BILINEAR would be ok for my case, I was just wondering why/how this wouldn't work on Net.. [02:24:42 CET] <iive> allowing to modify executable code is more dangerous for root ;) [02:24:53 CET] <oliv3> yeah sure :) [02:25:16 CET] <oliv3> then, strange this works with OpenBSD :) [02:25:26 CET] <oliv3> even in userland [02:26:08 CET] <BtbN> OpenBSD just lacks the security feature to protect against it [02:26:08 CET] <oliv3> and Open does have mprotect(2) [02:26:20 CET] <BtbN> It clearly is not working then [02:29:34 CET] <oliv3> Ok, I'm AFK but still there, will be back later [02:29:55 CET] <oliv3> Will ask to friends @ :) [02:30:18 CET] <oliv3> Thanks ! [22:13:23 CET] <fSharp> hello, youtube-dl rejects the download for the video name being too long. is there a parameter that I can get it ignored? [22:14:33 CET] <JEEB> I think youtube-dl does downloading 100% by itself. it may call ffmpeg.c (the command line API client) afterwards [22:15:58 CET] <fSharp> so how can I overwrite this length limit? [22:17:33 CET] <JEEB> try to make sure where the issue lies - if it's youtube-dl itself you go ask youtube-dl developers [22:41:41 CET] <fella> fSharp: try 'youtube-dl -o short.name URL' [22:42:26 CET] <fella> .. and/or share that url .. i'm curious now [22:45:01 CET] <fSharp> fella, yes, I was waiting for the download to end to see if it worked..I used -o with a different addition; youtube-dl -o "$HOME/Videos/%(somename)s-%(videoID)s.%(mp4)s" 'https://vimeo.com/videoID' [22:45:32 CET] <fSharp> found the solution here; https://askubuntu.com/questions/923814/how-to-fix-youtube-dl-error-unable-t… [22:47:24 CET] <fSharp> fella, your solution works, too, and much shorter :( [22:47:28 CET] <fSharp> oops :) [22:47:31 CET] <fSharp> thank you! [22:47:46 CET] <fella> lol - output filename actually means filename not path -- 'Kind of counter-intuitive.' [22:48:42 CET] <fella> fSharp: yw - still curious about that video you dl'd since i never ran into this problem [22:49:32 CET] <fSharp> fella, https://vimeo.com/9690833 [22:49:43 CET] <fella> ty :) [22:51:29 CET] <fella> download] Destination: 'HER NAME' Official Music Video - TayF3rd & Mario C [Direct [22:51:33 CET] <fella> r - Mitch Lemos]-9690833.mp4 [22:51:53 CET] <fella> doesn't seem to be that long though!? [22:52:35 CET] <fSharp> seems it is for my installed youtube-dl [22:53:16 CET] <fella> have you tried with '--restrict-ascii' [22:53:49 CET] <fella> have you tried with '--restrict-filenames' [22:53:52 CET] <fella> sry [22:53:53 CET] <fSharp> hm, no [22:54:51 CET] <fSharp> fella, that didn't work, either [22:54:57 CET] <fella> kk [23:17:56 CET] <fSharp> fella, thank you again, for your help, take care! [00:00:00 CET] --- Thu Dec 27 2018

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