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July 2016

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ffmpeg-devel.log.20160715
by burek 16 Jul '16

16 Jul '16

[00:51:19 CEST] <ubitux> https://blogs.dropbox.com/tech/2016/07/lepton-image-compression-saving-22-l… not pasted yet? [00:51:42 CEST] <jkqxz> nevcairiel: The code is just wrong - the allocation has to be inside the loop, because the pointers are written to the fifo and freed after reading. <http://sprunge.us/bKVj> makes it not-entirely-broken, I think? [02:25:26 CEST] <cone-908> ffmpeg 03Burt P 07master:33fc4cc7c6bd: af_hdcd: fewer false positives by ignoring code_counterC in HDCD detection [02:25:27 CEST] <cone-908> ffmpeg 03Burt P 07master:5c7457c02856: af_hdcd: only hdcd_update_info() when something changes [02:25:28 CEST] <cone-908> ffmpeg 03Burt P 07master:c41718c0d0e6: af_hdcd: detect and report encoding errors and oddities [04:41:07 CEST] <bp0> michaelni, maybe not [05:50:40 CEST] <bp0> michaelni, I submitted some trimmed versions, now about 2.3MB. Hope that is alright [07:45:52 CEST] <thardin> I am noticing I have no motivation/time to fix that mxf regression [07:48:36 CEST] <thardin> do any of the other people I've noticed contributing mxf patches work with the format in their dayjob? [09:10:04 CEST] <kode54> what the hell happened to ffurl_register_protocol? [09:10:24 CEST] <kode54> I used to register my own protocol handlers instead of compiling the stock ones in, so I could do my own file reading [09:11:29 CEST] <nevcairiel> ff* api is privaet and you should have never used it [09:12:01 CEST] <kode54> I compile and bundle my own ffmpeg anyway [09:12:02 CEST] <nevcairiel> you can implement your own reading callbacks using the avio api [09:12:06 CEST] <kode54> oh [09:12:12 CEST] <kode54> I didn't know there was an API for that [09:12:13 CEST] <nevcairiel> in a public and proper way [09:12:20 CEST] <nevcairiel> it gives you read write and seek callbacks [10:28:41 CEST] <cone-246> ffmpeg 03Carl Eugen Hoyos 07master:040b4e139b94: lavf/img2: Remove ffv1-img. [10:28:41 CEST] <cone-246> ffmpeg 03Carl Eugen Hoyos 07master:e1023aa1dd0c: lavf/img2dec: Also auto-detect (unusual) uncompressed pcx. [10:36:39 CEST] <cone-246> ffmpeg 03Clément BSsch 07master:77726d32a872: lavf/vplayerdec: Improve auto-detection. [10:38:02 CEST] <cone-246> ffmpeg 03Clément BSsch 07release/3.1:7da59005bec1: lavf/vplayerdec: Improve auto-detection. [10:39:13 CEST] <michaelni> bp0, where did you submit the trimmed versions ? [10:39:38 CEST] <michaelni> dont see any so obviously i look at the wrong plae [10:39:40 CEST] <michaelni> place [11:50:07 CEST] <marcin__> re all [11:50:44 CEST] <marcin__> i'm facing this issue (memory leak) -> http://ffmpeg.org/pipermail/libav-user/2016-July/009343.html [11:51:01 CEST] <marcin__> has anybody similar leaking problems? [13:08:56 CEST] <Rathann> marcin__: please check trac if there's a ticket open and if not, please open one [13:30:24 CEST] <marcin__> Rathann: ok, I will check then, thanks for responding [14:02:00 CEST] <cone-246> ffmpeg 03Pavel Koshevoy 07master:25ca74d314c3: Restore compatibility with powerpc-apple-darwin9-gcc-4.2.1 [14:57:25 CEST] <bp0> michaelni, I sent them as an email to ffmpeg-devel as a reply to the patch that adds the tests [14:58:08 CEST] <bp0> I received a response that they are awaiting moderation [14:59:03 CEST] <nevcairiel> sending samples directly to the ML is not really a good idea, which is probably why the filter flagged it for moderation [15:01:32 CEST] <bp0> what is the preferred way to submit samples? [15:16:57 CEST] <michaelni> bp0, upload somewhere and post url, (somewhere can be our ftp if you dont know any place) [15:36:42 CEST] <bp0> upload.fffmpeg.org? [15:38:30 CEST] <bp0> no route to host [15:38:34 CEST] <bp0> I'll figure something out [15:41:23 CEST] <Compn> you had one too many f's [15:41:31 CEST] <Compn> bp0 [15:42:27 CEST] <bp0> :) It's true, but not when I actually tried to ftp [15:43:51 CEST] <funman> bp0: the machine is being replaced [15:44:42 CEST] <bp0> ah, I see then [15:59:01 CEST] <michaelni> bp0, ive seen people use www.wetransfer.com recently for sharing some files but you can use anything (services with less bloat & spam preferred) [16:01:27 CEST] <Dev^Null> I am sure i am in the wrong channel but there is no one in the other ones. I am trying to use ffmpeg to connect to a ELP 1280*720p camera I got the url from Ispy and I keep getting this error "method DESCRIBE failed: 451 ERROR [16:01:27 CEST] <Dev^Null> rtsp://admin@10.10.10.231/user=admin_password=_channel1=1_stream=1.sdp: Server returned 4XX Client Error, but not one of 40{0,1,3,4}" this is the syntax I am using ffmpeg -i rtsp://admin@10.10.10.231/user=admin_password=_channel1=1_stream=1.sdp -r 15 C:/DB_Videos/2013-04-30 17_18_34.703.mp4 can someone help me figure out what is wrong I am running ubuntu 16.04 and ffmpeg ver 3.1.1 [16:11:18 CEST] <bp0> https://www.dropbox.com/s/fy4r5nsr5ysweg8/hdcd-false-positive.flac?dl=1 [16:11:20 CEST] <bp0> https://www.dropbox.com/s/ifkl0x7mqhiqc4t/hdcd-encoding-errors.flac?dl=1 [16:11:23 CEST] <bp0> works? [16:33:12 CEST] <bp0> michaelni, are those dropbox links going to work for you? [16:38:30 CEST] <michaelni> bp0, yes [16:40:03 CEST] <cone-246> ffmpeg 03Steven Liu 07master:b701ec4152b8: refine the method option describe of hlsenc doc [16:41:38 CEST] <bp0> ok [17:20:42 CEST] <michaelni> bp0, hdcd files uploaded to the fate samples [17:21:40 CEST] <bp0> right on, thank you [17:22:17 CEST] <bp0> how do you feel about the added compare "grep" added to fate-run.sh? [17:23:11 CEST] <bp0> if there was another way to look for a result in the log, I didn't find it [17:26:57 CEST] <nevcairiel> testing log output is not very reliable as we generally do not guarantee it to remain stable [17:27:20 CEST] <nevcairiel> and its also not a good way for user apps to be notified [17:27:37 CEST] <nevcairiel> so maybe you identified a flaw and should add proper metadata to show the presence :) [17:29:16 CEST] <bp0> So for the HDCD filter, perhaps that summary line could be encoded into metadata? [20:35:43 CEST] <KGB> [13FFV1] 15retokromer opened pull request #21: escaping not required for markdown (06master...06patch-1) 02https://git.io/vKuwJ [21:57:58 CEST] <cone-124> ffmpeg 03Josh de Kock 07master:d52dd768a322: lavf: add libopenmpt demuxer [22:43:35 CEST] <ubitux> > NVIDIA 367.35 Released, Supports 8K H.265 Video Decoding [22:43:40 CEST] <ubitux> what an era to live in... [22:43:47 CEST] <JEEB> :D [22:47:57 CEST] <jamrial> ubitux: what gpus? and that's probably hybrid decoding [22:49:06 CEST] <ubitux> > Added support for VDPAU Feature Set H to the NVIDIA VDPAU driver. GPUs with VDPAU Feature Set H are capable of hardware-accelerated decoding of 8192x8192 (8k) H.265/HEVC video streams. [22:49:08 CEST] <ubitux> http://www.nvidia.com/download/driverResults.aspx/105343/en-us [22:49:19 CEST] <ubitux> pascal gpus probably [23:11:16 CEST] <Chloe> does ffmpeg support demuxing multiple subsongs from a single file? [23:14:47 CEST] <Rathann> Chloe: what are "subsongs"? what kind of file are you talking about? [23:18:59 CEST] <Chloe> As far as I can tell, they're just sections of an audio track which are songs, I guess it'd be something like a CD where the audio file is the CD and the songs on it are the subsongs, but I'm not entirely sure. The kind of file I'm talking about is an audio file in which there are multiple songs (note: not tracks)--in this case, a module. [00:00:00 CEST] --- Sat Jul 16 2016

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ffmpeg.log.20160715
by burek 16 Jul '16

16 Jul '16

[00:11:55 CEST] <zhanshan> any screen capture command line professionals awake? [01:30:19 CEST] <DHE> zhanshan: just say what you need [01:30:58 CEST] <medya> I am currently using ffmpeg to conver AVI to MP4. I pass some random video codec stuff to it, and I am not happy with the quality, I have some mp4 videos with better quality and less size, is there any way to copy their Video Codec measurements ? [01:31:22 CEST] <medya> currently I am using ffmpeg -i $file -acodec libfaac -b:a 128k -vcodec mpeg4 -b:v 2500k -flags +aic+mv4 [01:31:37 CEST] <medya> is there a way to replace 128K and 2500K with exact numbers of another video file [01:31:46 CEST] <medya> or copy another video files exact format and preset ? [01:32:11 CEST] <DHE> mpeg4 is old. why not h264? [01:32:36 CEST] <DHE> you need a build of ffmpeg with libx264, but that's pretty common [01:37:00 CEST] <medya> DHE I am open to suggestion can you give me a command to convert my AVI files to a modern format [01:37:15 CEST] <medya> @DHE how many parameters are there to be passed [01:45:17 CEST] <c_14> medya: https://trac.ffmpeg.org/wiki/Encode/H.264 https://trac.ffmpeg.org/wiki/Encode/AAC [01:46:29 CEST] <medya> can u give it to me in one command [01:46:40 CEST] <medya> it confuses me when I have to decide both video and audio stuff [01:46:51 CEST] <c_14> what audio codec is your input? [01:46:56 CEST] <medya> I dont care [01:47:00 CEST] <medya> just a normal one [01:47:10 CEST] <medya> like a old digital camera [01:47:15 CEST] <medya> how can I find out [01:47:53 CEST] <c_14> ffmpeg -i file -c:v libx264 -crf 23 -c:a aac -b:a 260k out.mp4 [01:47:55 CEST] <c_14> you can use that [01:48:02 CEST] <medya> thanks [01:48:07 CEST] <medya> those colons in the options scare me [01:48:13 CEST] <medya> what does c:v or b:a mean [01:48:24 CEST] <medya> I never seen a linux command option with colon in it [01:48:28 CEST] <c_14> If your input is aac or mp3 (check with ffprobe) you can use -c:a copy to spare the conversion [01:48:41 CEST] <catchin> medya: https://trac.ffmpeg.org/wiki/Encode/MP3 [01:48:43 CEST] <c_14> -c is short for -codec :a selects applies the option to audio streams [01:48:46 CEST] <c_14> :v applies it to video streams [01:49:04 CEST] <medya> why there is a colon [01:49:25 CEST] <medya> what does 260K represent [01:49:32 CEST] <medya> if I lower than number my file will be smaller? [01:49:55 CEST] <c_14> ffmpeg has a (rather complex) stream selection system that uses colons https://ffmpeg.org/ffmpeg.html#Stream-specifiers-1 , that syntax gets applied everywhere [01:49:56 CEST] <medya> and what does crf mean [01:50:01 CEST] <c_14> 260k is the bitrate of the audio stream [01:50:14 CEST] <c_14> Lowering it will make the output smaller (though at that bitrate audio won't eat that much) [01:50:29 CEST] <c_14> crf is short for constant rate factor and is the quality knob for libx264 [01:50:42 CEST] <c_14> higher is lower quality, lower is better quality [01:50:43 CEST] <medya> so crf maens 23 frames per second? [01:51:17 CEST] <c_14> no [01:51:35 CEST] <c_14> It's a quality knob [01:51:54 CEST] <c_14> It affects the size of the output video [01:52:00 CEST] <medya> I see. [01:52:30 CEST] <medya> is there a way to copy exact v:c and a:c of another video [01:52:58 CEST] <c_14> not unless it's written into the metadata of the input video [01:53:57 CEST] <medya> I did ffmpeg -i on a video which I like its quality and gives me this : Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 encoder : Lavf56.25.101 Duration: 00:02:00.01, start: 0.000000, bitrate: 2134 kb/s Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1280x720, 2132 kb/s, 30 fps, 30 tbr, 1200k tbn, 60 tbc (default) Metadata: [01:54:07 CEST] <medya> can I steal this exact format [01:56:43 CEST] <c_14> not unless the video was encoded at constant bitrate [01:56:54 CEST] <c_14> using a crf around 23 will most of the time do what you want [01:57:16 CEST] <medya> ok thank you so much [01:57:56 CEST] <medya> another question, is there any way to make timelapse video out of a long video, like if you have 1 hour long, can you make a 1 minute long fast video out of it [02:01:15 CEST] <c_14> ffmpeg -i video -vf setpts=PTS/60,fps=30 out.mkv [02:01:59 CEST] <medya> why this one is mkv instead of mp4? [02:02:05 CEST] <c_14> doesn't matter [02:02:07 CEST] <c_14> I just always use mkv [02:02:08 CEST] <medya> what is -vf do [02:02:15 CEST] <c_14> specifies video filters [02:02:25 CEST] <medya> what is the video filter in this case [02:02:57 CEST] <c_14> setpts takes the timestamps of the input video and divides them by 60 (makes the video 60 times faster so that 1 hour fits in 1 minute), the fps filter then drops frames [02:04:37 CEST] <medya> makes sense, and why did you have to add fps=30 [02:04:43 CEST] <medya> how does that option help [02:05:59 CEST] <c_14> the fps filter drops frames, otherwise you'd still have all the frames just with very small delays between them and the output would be unnecessarily large and most of the frames would be dropped on display anyway [02:06:31 CEST] <medya> so the video would not be 1 minute it will be 10 hours ? [02:06:38 CEST] <medya> like opposite of shorteninig it [02:06:40 CEST] <medya> right? [02:06:51 CEST] <medya> is there a place to see list of all the video filters [02:08:16 CEST] <c_14> no, it'd be 1 minute. You know those flip picture books? It'd be like taking a really thick one of those and getting rid of pages you wouldn't be able to see anyway because they're flipping by too fast [02:08:19 CEST] <c_14> https://ffmpeg.org/ffmpeg-filters.html [02:08:24 CEST] <c_14> ^has all the filters and what they do [02:08:31 CEST] <medya> thank you so much [02:08:36 CEST] <medya> I feel so much better now [02:08:39 CEST] <medya> that I undrestand it [02:08:58 CEST] <medya> do you know if I can combine 3 videos into one big video? [02:09:07 CEST] <medya> like add them one after another in one video [02:09:18 CEST] <c_14> yep https://trac.ffmpeg.org/wiki/Concatenate [02:12:01 CEST] <medya> c_14++ [02:13:42 CEST] <medya> c_14 why I cant find stepts in https://ffmpeg.org/ffmpeg-filters.html [02:13:58 CEST] <c_14> https://ffmpeg.org/ffmpeg-filters.html#setpts_002c-asetpts [02:14:01 CEST] <c_14> It's right there [02:14:50 CEST] <medya> ah you are right, its weird chrome's Ctrl+F doesnt work for stepts [02:15:07 CEST] <c_14> setpts? [02:15:09 CEST] <c_14> one t too many [02:15:38 CEST] <medya> oops [02:15:51 CEST] <medya> you are right :) sorry typo [02:15:54 CEST] <medya> now it works [02:19:04 CEST] <medya> is there a way to get the embeded subtitle in mp4 file exported to a text file [02:19:16 CEST] <c_14> ffmpeg -i mpv out.srt [02:19:22 CEST] <c_14> s/mpv/mp4/ [02:19:29 CEST] <thebombzen_> medya: you could try ffmpeg -i input.mp4 -map 0:s out.srt [02:19:39 CEST] <thebombzen_> actually you don't even need -map 0:s scratch that [02:19:56 CEST] <medya> c14 so ffmpeg is smart enoguh if I make the extension srt it will make it subtitle? [02:20:04 CEST] <c_14> yes [02:20:08 CEST] <medya> sweeet ! [02:21:16 CEST] <medya> how can I cut the first 10 minute of a video of 30 minute. ( I only care the first 10 miute I dont want the rest) [02:21:39 CEST] <c_14> ffmpeg -i video -c copy -t 10:00 out.mkv [02:22:52 CEST] <medya> wow c_14 you are like a ffmpeg nina [02:22:56 CEST] <medya> ninja* [02:23:00 CEST] <medya> you should write a book about it [02:23:28 CEST] <medya> how can I get what is the current length of a video [02:23:42 CEST] <c_14> use ffprobe and check the duration tag [02:24:01 CEST] <klaxa> what's non-current lenght of a video? :^) [02:24:08 CEST] <klaxa> *length [02:24:28 CEST] <c_14> klaxa: you know, those videos that change length depending on the phase of the moon [02:24:35 CEST] <klaxa> oic [02:25:02 CEST] <klaxa> like those videos that are stored are 23.976 fps but are played back at 25 fps [02:25:29 CEST] <klaxa> th-thanks differing national standards [02:25:47 CEST] <klaxa> television is a silly thing [02:31:01 CEST] <thebombzen_> medya: you could use ffprobe to doublecheck the duration. if you want you coudl use "ffprobe input_file -show_format | grep duration" [02:31:13 CEST] <thebombzen_> there's some syntax to take it directly. I don't know what it is tho so I use grep [02:31:21 CEST] <thebombzen_> I wonder if if c_14 knows [02:32:37 CEST] <c_14> ffprobe input_file -show_entries format=duration [02:32:57 CEST] <c_14> -loglevel quiet -of flat [02:34:35 CEST] <klaxa> fucking lexicon. train a network [02:34:57 CEST] <klaxa> might not be a bad idea actually [02:35:14 CEST] <klaxa> https://xkcd.com/810/ [02:37:04 CEST] <thebombzen> haha klaxa today I signed up for the arch forums [02:37:31 CEST] <thebombzen> the captcha was "what is the output of date +%V$(uname)|sha1sum" or something like that [02:37:40 CEST] <thebombzen> imagine if a bot could answer that question [02:38:17 CEST] <c_14> Well it could, at least until someone invents a captcha that tries installing a rootkit [02:39:12 CEST] <klaxa> (Robots: no sandboxes!) [02:41:06 CEST] <medya> c_14 thanks this command looks like a charm ffprobe inputfile.mp4 -show_entries format=duration -v quiet| grep "duration" [02:44:57 CEST] <thebombzen> does anyone know if ffmpeg supports libdaala? I can't find it in ./configure but that doesn't mean there's no way to do it [02:45:13 CEST] <c_14> not yet [02:45:40 CEST] <c_14> atomnuker was working on a decoder a while back afaik, not sure how far that got [02:46:24 CEST] <thebombzen> well I definitely feel like libdaala codec in libavcodec would be easier than writing a native decoder. [02:46:39 CEST] <thebombzen> it doesn't quite seem worth it to write a native decoder because the bitstream format isn't frozen [02:46:57 CEST] <c_14> yeah, it was mostly just an experiment/base to work on [02:50:32 CEST] <medya> c_14 when you said to make a time lapse ffmpeg -i video -vf setpts=PTS/60,fps=30 out.mkv [02:50:44 CEST] <medya> if I wanna make my time lapse 6 seconds how do I change the number [02:50:53 CEST] <medya> do I just change PTS/6 and not touch fps? [02:51:24 CEST] <c_14> if the input is 1 hour long it would be PTS/600 [02:51:39 CEST] <medya> ok the PTS is the only thing will change [02:52:12 CEST] <c_14> yep [02:52:16 CEST] <medya> gtocha thanks [02:53:30 CEST] <c_14> It's basically input_time/x = length_you_want [02:55:02 CEST] <medya> so what is mkv and why do you use that ? if you name the output file exention mkv, will ffmpeg automaticly put some default format and codec for me [02:55:47 CEST] <c_14> mkv is matroska, I use it because it supports basically every codec under the sun and isn't quite as braindead as mp4 in certain things [02:56:04 CEST] <c_14> It will choose default codecs if you don't specify any on the commandline (but afair those should be the same as for mp4) [02:57:23 CEST] <medya> cool, is there a way to make the setpts smarter, like if there 1 hour of video and first 20 minute is not much movement, dont use that one, and only put frames with more movement ? [02:57:44 CEST] <medya> like choose the frames with more contrast and change with other frames [02:57:56 CEST] <c_14> setpts doesn't have motion detection or anything like that, you can use the select filter with the scene variable [02:58:26 CEST] <medya> so there is already a motion detector filter embeded ? [02:58:59 CEST] <c_14> It's not motion detection [02:59:11 CEST] <c_14> It's basic scene detection, i.e. how much changes between frames [02:59:44 CEST] <medya> so do you happen to know how it would be in one command [03:01:19 CEST] <c_14> something like -vf select='gt(scene\,0.2)',fps=30 [03:01:27 CEST] <c_14> you'll probably have to play around with that though [03:06:07 CEST] <medya> so I add -vf select='gt(scene\,0.2)' to the already setps? [03:06:20 CEST] <c_14> nah, without the setpts [03:06:35 CEST] <medya> like this? ffmpeg -i videofile -vf select='gt(scene\,0.2)' -vf setpts=PTS/60,fps=30 out.mkv [03:06:50 CEST] <medya> but how would the scence0.2 know to make 6 secnds [03:06:55 CEST] <medya> I want one hour in 6 seconds [03:11:54 CEST] <c_14> That's the issue, the select filter will drop all frames that don't match that. What you could do is add a minimum/maximum to the select like -vf select='gt(scene\,0.2)*gt(prev_selected_t\,0.001)+lt(prev_selected_t\,0.002)' [03:12:00 CEST] <c_14> that should work [03:12:05 CEST] <c_14> assuming my math didn't go bonkers [03:15:03 CEST] <c_14> you can adjust the time values if you want [03:15:28 CEST] <c_14> eh, I think my math might be wrong [03:15:31 CEST] <medya> what do those numbers means, which ones are time values [03:15:46 CEST] <medya> what is 0.2 what is 0.001 .... what is gt what is lt [03:15:53 CEST] <c_14> The 0.001 should be ~550 and the 0.002 650 or so [03:16:29 CEST] <c_14> The 0.2 (for the scene) is a value between 0 and 1 which determines the likelihood that this frame begins a new scene 0 being very unlikely and 1 being very likely [03:16:46 CEST] <c_14> The gt prev_selected number is at least how many seconds to space frames apart [03:17:00 CEST] <c_14> The lt prev_selected number is the maximum distance between two frames [03:17:54 CEST] <c_14> So it should try and select frames that start new scenes, but will make sure that the whole video is on average the length you want [03:40:33 CEST] <medya> sorry c_14 my comperut froze, is there a way to make ffmpeg not to use 100% of my cpu ? [03:40:59 CEST] <c_14> you can set -threads [03:41:05 CEST] <DHE> most people want it to so processing is done as fast as possible. [03:41:11 CEST] <c_14> if you're on a unix system you can set the niceness [03:41:18 CEST] <medya> I am on ubuntu [03:41:26 CEST] <medya> how can I set niceness or -threads [03:41:29 CEST] <DHE> prefixing the command with "nice" will run at reduce priority [03:41:37 CEST] <medya> oh [03:41:38 CEST] <medya> cool [03:41:53 CEST] <DHE> still uses 100% CPU, but will yield to other applications more [03:44:13 CEST] <medya> @c_14 so you were saying -vf select='gt(scene\,0.2)*gt(prev_selected_t\,0.001)+lt(prev_selected_t\,0.002)' The 0.001 should be ~550 and the 0.002 650 or so [03:44:21 CEST] <medya> what did you mean The 0.001 should be ~550 and the 0.002 650 or so [03:44:54 CEST] <c_14> -vf select='gt(scene\,0.2)*gt(prev_selected_t\,550)+lt(prev_selected_t\,650)' [03:45:12 CEST] <medya> what does 550 and 650 do for me [03:45:31 CEST] <c_14> minimum and maximum distance between frames in seconds [04:09:32 CEST] <medya> and 0.2 is ? [04:09:46 CEST] <medya> and how would I make sure 1 hour video becomes 6 second [04:09:47 CEST] <medya> or 10 second [04:09:54 CEST] <medya> how would I adjust the gt and lt [04:09:59 CEST] <medya> to make whatever length I want [04:11:27 CEST] <c_14> original_length / wanted_length = x (in seconds) [04:11:31 CEST] <c_14> then just adjust up and down [04:12:44 CEST] <medya> so I put x for gt and then I add x+a small number for lt ? [04:13:00 CEST] <c_14> x-a for gt and x+a for lt [04:14:35 CEST] <klaxa> c_14: say, how did we fail the written math exams again? :/ [04:16:38 CEST] <c_14> wrong questions [04:17:12 CEST] <klaxa> yeah probably [04:24:35 CEST] <thebombzen> how do I do a transcode without mapping any metadata? [04:24:43 CEST] <c_14> -map_metadata -1 [04:24:48 CEST] <thebombzen> oh thanks [04:28:38 CEST] <medya> c_14 in what does * do in the command you gave me "-vf select='gt(scene\,0.2)*gt(prev_selected_t\,550)+lt(prev_selected_t\,650)'" [04:30:15 CEST] <c_14> * is the equivalent of an AND and + is the equivalent of an OR [04:30:46 CEST] <c_14> logical AND and logical OR [04:31:35 CEST] <medya> c_14 I did this command and the output has 0KB video 24169KB of audio [04:31:39 CEST] <medya> ffmpeg -i output.mp4 -vf select='gt(scene\,0.2)*gt(prev_selected_t\,294)+lt(prev_selected_t\,314)' output_fast_scene.mp4 [04:31:45 CEST] <medya> frame= 0 fps=0.0 q=0.0 Lsize= 24751kB time=02:34:40.72 bitrate= 21.8kbits/s video:0kB audio:24169kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 2.407669% [04:31:51 CEST] <medya> and when I play it has no video [04:37:07 CEST] <c_14> eh, duh [04:37:15 CEST] <c_14> I can't and with prev_selected_t if I never select anything [04:39:47 CEST] <medya> what do you mean [04:50:08 CEST] <c_14> Ok [04:50:10 CEST] <c_14> -vf select='eq(n\,0)+(gt(scene\,0.2)*gt(t\,prev_selected_t+294))+gt(t\,prev_selected_t+314)',setpts=PTS/600 [04:50:13 CEST] <c_14> should work [04:50:28 CEST] <c_14> you might have to change the 600 [04:50:37 CEST] <c_14> and you'll also need to speed up the audio [04:51:04 CEST] <c_14> or get rid of it entirely [04:53:52 CEST] <c_14> had to change a few things [05:00:37 CEST] <medya> how can I get rid of the audio entirely? [05:00:41 CEST] <c_14> -an [05:00:53 CEST] <medya> should I add -an in the end [05:01:53 CEST] <c_14> Somewhere before the output filename and after the input filename [05:11:05 CEST] <medya> c_14 the last command you gave me is producing 0 frames [05:11:07 CEST] <medya> Press [q] to stop, [?] for help frame= 0 fps=0.0 q=0.0 size= 0kB time=00:00:00.00 bitrate=N/A [05:12:32 CEST] <c_14> It's working for me, and it should definitely produce at least 1 frame [05:12:39 CEST] <c_14> What's your complete command look like? [05:12:58 CEST] <medya> ffmpeg -i output.mp4 -vf select='gt(scene\,0.2)*gt(prev_selected_t\,294)+lt(prev_selected_t\,314)' -an output_fast_scene.mp4 [05:13:36 CEST] <c_14> That's not the last one I gave you [05:13:38 CEST] <medya> oops it seems like I forgot the last part [05:13:39 CEST] <medya> sorry [05:13:40 CEST] <c_14> I gave you ffmpeg -i video -vf select='eq(n\,0)+(gt(scene\,0.2)*gt(t\,prev_selected_t+294))+gt(t\,prev_selected_t+314)',setpts=PTS/600 -an out.mkv [05:13:42 CEST] <medya> my bad [05:15:47 CEST] <medya> ok that command now has made a file with 885 bytes in it, and it is not increasing (like old ones that kept increasing) [05:16:10 CEST] <c_14> might take a while [05:16:13 CEST] <medya> and the frame status is increasing one by one very slowly [05:16:14 CEST] <medya> frame= 7 fps=0.1 q=0.0 size= 1kB time=00:00:00.00 bitrate=N/A [05:16:24 CEST] <medya> it says the bitrate is N/A [05:16:59 CEST] <c_14> hmm, time bitrate and speed should all have values [05:17:22 CEST] <medya> yeah it keeps saying 000000 and n/a [05:17:24 CEST] <medya> for those values [05:17:43 CEST] <c_14> try replacing 294 with 9 and 314 with 11 [05:17:49 CEST] <medya> ok [05:19:08 CEST] <medya> still no bitrare [05:19:39 CEST] <medya> but frames are going faster [05:19:44 CEST] <c_14> hmm [05:19:47 CEST] <c_14> try waiting for it to finish [05:19:59 CEST] <medya> it will be a long wait, it is a longer video [05:20:06 CEST] <medya> this one has a new thing called drop [05:20:12 CEST] <medya> and it keeps increasing [05:20:19 CEST] <c_14> rapidly? [05:20:22 CEST] <medya> like drop=38.. (and countruing [05:20:24 CEST] <medya> counting) [05:20:34 CEST] <medya> yeah like 3 increase per second [05:20:41 CEST] <c_14> that's a bit too fast [05:20:43 CEST] <medya> right now drop is 75 [05:20:54 CEST] <medya> now it is 85 [05:21:04 CEST] <medya> and dup is constant 0 [05:21:14 CEST] <medya> bitrate still n/a [05:21:24 CEST] <c_14> is frame= greater than drop? [05:21:33 CEST] <medya> no [05:21:34 CEST] <medya> frame is 47 [05:21:38 CEST] <medya> drop is 127 [05:21:43 CEST] <c_14> what's your current command? [05:21:57 CEST] <medya> ffmpeg -i output.mp4 -vf select='eq(n\,0)+(gt(scene\,0.2)*gt(t\,prev_selected_t+9))+gt(t\,prev_selected_t+11)',setpts=PTS/600 -an out.mkv [05:22:55 CEST] <c_14> maybe try getting rid of the setpts part [05:23:03 CEST] <c_14> to see if that changes anything [05:23:06 CEST] <medya> but it seemed it like that command hand generated a good a few seconds of time lapse after I killed it (the last one) [05:23:10 CEST] <medya> ok [05:23:25 CEST] <c_14> yeah, it should work but it also shouldn't be dropping that many frames [05:23:50 CEST] <medya> ok without setps there is no drop [05:24:12 CEST] <medya> and the output file is increasing [05:24:13 CEST] <medya> which is good [05:24:39 CEST] <medya> so 9 and 11 means.... there should be no longer than 9 or 11 seconds difference between each two frame? [05:24:52 CEST] <c_14> yes [05:25:16 CEST] <c_14> how long is the input video and how long do you want the output to be? [05:25:48 CEST] <c_14> Because you're going to need the setpts filter, but maybe 600 was too high [05:26:27 CEST] <medya> duration=18280 I want it to be like 60 seconds [05:27:05 CEST] <c_14> try adding ,setpts=PTS/300 [05:27:38 CEST] <medya> ok [05:28:36 CEST] <medya> frame= 34 fps=0.9 q=0.0 size= 1kB time=00:00:00.00 bitrate=N/A dup=0 drop=24 [05:29:03 CEST] <medya> ok now it has a bitrate [05:30:28 CEST] <medya> how can I see a the real duration of a time elapsed video, if I do the ffprobe it will tell the big number before being shortened [05:34:34 CEST] <c_14> Does the input video have subtitles? [05:34:48 CEST] <medya> no [05:34:51 CEST] <c_14> If there's any other track in the video that isn't being shortened it'll show that duration [05:35:17 CEST] <medya> oh [05:35:26 CEST] <medya> how can we short the subtitle also [05:35:35 CEST] <medya> shorten* [05:36:12 CEST] <c_14> eh, in this case it's probably better to just get rid of them. They wouldn't make sense on the shortened video anyway [05:36:40 CEST] <c_14> you can use -sn to get rid of them [05:38:02 CEST] <medya> so anything we dont want we add -n to it [05:38:05 CEST] <medya> -an -sn [05:38:08 CEST] <c_14> ye [05:38:11 CEST] <medya> what else is out there other than a and s [05:38:22 CEST] <c_14> v and d -dn -vn [05:38:29 CEST] <medya> what is d [05:38:35 CEST] <c_14> data [05:38:45 CEST] <medya> there are videos with data? [05:38:57 CEST] <medya> does it mean like metadata like author ...stuff ? [05:39:04 CEST] <c_14> data is basically anything ffmpeg can't decode [05:40:33 CEST] <medya> you are very knwoldgeable and helpful [05:40:37 CEST] <medya> do you have a github page [05:41:51 CEST] <c_14> I do, but I only really have a couple of forks of things I played around with on it [05:42:45 CEST] <medya> can I follow you there [05:43:04 CEST] <medya> btw the last time lapse command you gave me worked like a charm in the end I saw this warning in orange [05:43:05 CEST] <medya> Delayed frames seen. Reenabling low delay requires a codec flush.p=0 drop=633 [05:43:08 CEST] <medya> what does that mean [05:43:17 CEST] <c_14> sure, it's c-14 [05:43:47 CEST] <c_14> hmm [05:45:20 CEST] <c_14> Sounds like something intrenal. As long as it isn't fatal I wouldn't worry all that much about it. [05:45:52 CEST] <medya> cool [05:45:55 CEST] <medya> this is really cool [05:46:02 CEST] <medya> thank you so much again [05:46:25 CEST] <c_14> np [05:47:10 CEST] Action: c_14 needs to be heading off to bed now. I'll be back later today if you have more questions, just ask and I'll get back to you when I have time or somebody else might be able to answer them. [05:47:48 CEST] <medya> thanks I am going to bed also, so no more questions have a good night [08:00:06 CEST] <Bray90820> If my video file has multiple audio tracks ow would I choose audio track two with "-c:a copy" [08:01:25 CEST] <Bray90820> *How [08:13:07 CEST] <darkapex> Bray90820: Use -c:a:0, -c:a:1 etc to select a particular stream. (Refer to https://trac.ffmpeg.org/wiki/Map) [08:14:05 CEST] <Bray90820> Would -c:a:0 be audio stream 1 and -c:a:1 be audio stream 2? [08:14:44 CEST] <darkapex> Yes, I guess [08:15:23 CEST] <Bray90820> You guess? [08:18:56 CEST] <darkapex> Wait, my bad, what you should do is select the second stream using map [08:19:20 CEST] <darkapex> and check which is the index of the second audio stream using ffprobe [08:19:58 CEST] <darkapex> Just refer to the wiki page I linked, I need caffeine [08:20:30 CEST] <Bray90820> Alright [08:20:32 CEST] <Bray90820> Thanks [08:20:47 CEST] <darkapex> Np, sorry for the confusion, I'm sleepy af [08:21:43 CEST] <Bray90820> Same here it's ok [08:22:12 CEST] <Bray90820> And if worst comes to worst I can just try different numbers until I figure it out [09:16:06 CEST] <motbob> hello, I concatted a bunch of TS files. When I play the video, the audio is broken at the points of the joined files (there is a small, noticable glitch). But when I play the audio stream in an audio player, it sounds fine. I have no idea where to start solving this problem. [10:29:55 CEST] <maziar_> hello everyone, i want to convert mp4 to hls, can you please help me how to do it [10:44:53 CEST] <maziar_> sorry my network goes down, how can i convert mp4 to m3u8 ? can you give me a tutorial link for learning it ? [10:49:11 CEST] <spaam> maziar_: https://www.keycdn.com/support/how-to-convert-mp4-to-hls/ [10:49:54 CEST] <maziar_> spaam is there any full document about it [10:50:12 CEST] <spaam> https://www.ffmpeg.org/ffmpeg-formats.html#hls-1 [11:56:09 CEST] <roxlu> hey, where can I find if ffmpeg can read from a file which contains raw rtp data? I'm not sure where to look for in the docs [14:50:23 CEST] <ultrav1olet> How does one pass x265 params to ffmpeg? [14:50:42 CEST] <ultrav1olet> in the light of this: https://trac.ffmpeg.org/ticket/5715 [15:41:27 CEST] <Aberlin2> Hello, I hope this is the right channel for this question: how do I know if .mp4 is a good format as seen from a license perspective [15:41:33 CEST] <Aberlin2> ? [15:42:46 CEST] <Aberlin2> I just want to add it to the Wikipedia discussion, where I was looking for a hint [15:42:56 CEST] <Aberlin2> https://en.m.wikipedia.org/wiki/MPEG-4_Part_14#/talk/1 [16:04:50 CEST] <Dev^Null> I am sure i am in the wrong channel but there is no one in the other ones. I am trying to use ffmpeg to connect to a ELP 1280*720p camera I got the url from Ispy and I keep getting this error "method DESCRIBE failed: 451 ERROR [16:04:50 CEST] <Dev^Null> rtsp://admin@10.10.10.231/user=admin_password=_channel1=1_stream=1.sdp: Server returned 4XX Client Error, but not one of 40{0,1,3,4}" this is the syntax I am using ffmpeg -i rtsp://admin@10.10.10.231/user=admin_password=_channel1=1_stream=1.sdp -r 15 C:/DB_Videos/2013-04-30 17_18_34.703.mp4 can someone help me figure out what is wrong I am running ubuntu 16.04 and ffmpeg ver 3.1.1 [16:12:22 CEST] <limbo> the server's giving you an error code. [16:12:31 CEST] <limbo> no idea what the DESCRIBE metod is. [16:12:37 CEST] <limbo> *method [16:16:13 CEST] <Dev^Null> vinanao [16:16:14 CEST] <Dev^Null> asdf'lkjasdf;lkhsd [16:18:18 CEST] <Dev^Null> oops sorry [16:18:23 CEST] <Dev^Null> thanks [16:27:44 CEST] <t4nk985> hi [16:28:07 CEST] <t4nk985> iam right at the gay chat? [17:25:17 CEST] <thebombzen> does anyone know how to get -vf subtitles to work with fastseek (i.e. -ss before -i)? [17:25:50 CEST] <thebombzen> if I use ffmpeg -i input.mkv -vf subtitles=input.mkv -ss 5 then it will work correctly (i.e. render the subtitles 5 seconds in atop the video 5 seconds in) [17:26:19 CEST] <thebombzen> but if I use ffmpeg -ss 5 -i input.mkv -vf subtitles=input.mkv it will render the initial subs atop video, despite starting five seconds in [17:41:30 CEST] <kevinranks> I'm having an issue where I'm using fast seek on a remote file to grab a thumbnail at a certain point...however I am getting "Request range not satisfiable"....looking at the ffmpeg log I can see that ffmpeg does a partial content request and gets back "1349278381" as the file size....then it does a request with "Range: bytes=2054741312-" ...which makes no sense to me.... [17:42:41 CEST] <__jack__> kevinranks: you're using http/1.1 ? [17:43:05 CEST] <__jack__> (eg not http/1.0) [17:43:09 CEST] <kevinranks> correct [17:43:34 CEST] <kevinranks> "[https @ 0x1bd1fc0] header='HTTP/1.1 206 Partial Content'" [17:43:53 CEST] <__jack__> range: bytes=X- = from X to whatever you can give me, if I recall [17:44:49 CEST] <kevinranks> Right, so it's requesting "2054741312-" ...but the file size is only 1349278381 [17:47:27 CEST] <__jack__> 1349278381 is the size ffmpeg gets, or the real file size, fetched from httpd ? [17:49:04 CEST] <kevinranks> In the logs with -loglevel debug set, I see a request with: [17:49:06 CEST] <kevinranks> https @ 0x1bd1fc0] header='HTTP/1.1 206 Partial Content' [17:49:16 CEST] <kevinranks> [https @ 0x1bd1fc0] header='Accept-Ranges: bytes' [17:49:17 CEST] <kevinranks> [https @ 0x1bd1fc0] header='Content-Range: bytes 63412-1349341792/1349341793' [17:49:17 CEST] <kevinranks> [https @ 0x1bd1fc0] header='Content-Type: video/mp4' [17:49:17 CEST] <kevinranks> [https @ 0x1bd1fc0] header='Content-Length: 1349278381' [17:49:53 CEST] <kevinranks> then after that it does: [17:50:03 CEST] <kevinranks> [https @ 0x1bd1fc0] request: GET [file source] [17:50:20 CEST] <kevinranks> User-Agent: Lavf/56.25.101 [17:50:20 CEST] <kevinranks> Accept: */* [17:50:20 CEST] <kevinranks> Range: bytes=2054741312- [17:50:21 CEST] <kevinranks> Connection: close [17:50:21 CEST] <kevinranks> Host: s3-us-west-2.amazonaws.com [17:50:21 CEST] <kevinranks> [https @ 0x1bd1fc0] header='HTTP/1.1 416 Requested Range Not Satisfiable' [17:50:21 CEST] <kevinranks> [https @ 0x1bd1fc0] http_code=416 [17:50:22 CEST] <kevinranks> [https @ 0x1bd1fc0] HTTP error 416 Requested Range Not Satisfiable [17:50:56 CEST] <kevinranks> 1349341792 is the correct file size as well [17:51:14 CEST] <kevinranks> so I'm confused why it's requesting 2054741312- [18:14:55 CEST] <kevinranks> I see, it looks like the video file might be corrupt [18:21:05 CEST] <thebombzen> what does this mean: [matroska @ 0x55f9439c3de0] Using AVStream.codec to pass codec parameters to muxers is deprecated, use AVStream.codecpar instead. [18:21:08 CEST] <thebombzen> I get it all the time. [18:32:04 CEST] <maziar> what is the best way to convert a MP4 file to M3U8 different quality [18:38:08 CEST] <__jack__> maziar: you mean, a variable bitrate playlist, with multiple sub-playlist (one for "hd", one for "sd" etc) ? I don't think you can with ffmpeg [18:38:36 CEST] <maziar> __jack__ no [18:39:23 CEST] <maziar> __jack__ for example for 720p what is the best way to convert a MP4 file to M3U8 [18:39:35 CEST] <maziar> __jack__ for example for 1080p what is the best way to convert a MP4 file to M3U8 [18:39:52 CEST] <maziar> __jack__ for example for 480p what is the best way to convert a MP4 file to M3U8 [18:47:32 CEST] <thebombzen> as though asking the same question three times will get you an answer faster [18:48:27 CEST] <__jack__> maziar: for all of them : ffmpeg -i source.mp4 -map 0 -c copy output.m3u8 [18:48:41 CEST] <__jack__> the input resolution does not matter [18:56:00 CEST] <maziar> __jack__ no i have a 1080 MP4 video and i want to create a 480p m3u8 and 720p m3u8, you mean what should i do? [18:58:28 CEST] <__jack__> maziar: ffmpeg -i source.mp4 -map 0 -vf scale=720:-1 -c:a copy 720.m3u8 -map 0 -vf scale=480:-1 -c:a copy 480.m3u8 [18:58:31 CEST] <__jack__> or something like that [18:59:12 CEST] <maziar> __jack__ is there any manual or document for that? [18:59:26 CEST] <drv> m3u8 is a playlist format, not a video container... [18:59:55 CEST] <__jack__> maziar: dunno, probably [19:00:00 CEST] <__jack__> drv: sure [19:03:54 CEST] <furq> thebombzen: it means that ffmpeg.c hasn't been updated to use a new api [19:04:20 CEST] <thebombzen> furq: so it's something that someone should get around to patching but nobody has done it yet? [19:04:38 CEST] <thebombzen> cause the API you're refering to would be libavformat, not anything external [19:04:57 CEST] <furq> well really it's a situation where an api shouldn't have been marked as deprecated until the thing 99% of people use the project for stops using it [19:05:14 CEST] <furq> but your thing would also work [19:05:17 CEST] <maziar> drv i should create 4 quality in m3u8 output for adaptive bit rate , media streaming, would please help me how can i provide that quality's [19:05:56 CEST] <furq> or at the very least it shouldn't spit out a completely useless warning that most people will obviously think is something they can fix [19:13:45 CEST] <maziar> __jack__ i should create 4 quality in m3u8 output for adaptive bit rate , media streaming, would please help me how can i provide that quality's [19:35:10 CEST] <kepstin> maziar: by 'm3u8' you mean HLS streaming? [19:35:26 CEST] <maziar> kepstin yes i mean by HLS streaming [19:37:45 CEST] <kepstin> maziar: probably best to just run multiple ffmpeg commands on the same source with different scale options, and use fixed gop (disable scene detection or add min keyframe interval) so the segments match up for adaptive stream switching. [19:38:56 CEST] <maziar> kepstin, thank you so much, would you please help me about best way to convert from mp4 to hls 720 and 480 ? [19:43:01 CEST] <shincodex> hi [19:43:15 CEST] <shincodex> is your libraries of shinangans assertlevel > 0 by default in configure [19:43:25 CEST] <kepstin> maziar: I'd be able to help you if you have some specific questions about why something you've tried isn't working, or how to improve an ffmpeg command you've been working on. [19:43:29 CEST] <shincodex> cause my library that eats ffmpeg collection into one SO..... [19:43:38 CEST] <shincodex> is aborting on my throw when i have try catch handlers [19:43:52 CEST] <shincodex> so im wondering if my throw is geting stolen by ffmpeg [19:44:35 CEST] <maziar> kepstin i'm using this command to convert a 1080 MP4 file to 720 hls, is this correct command : ffmpeg -i input.mp4 -map 0 -vf scale=720:-1 -c:a copy -profile:v baseline -level 3.0 -start_number 0 -hls_time 3 -hls_list_size 0 -f hls index.m3u8 [19:45:12 CEST] <kepstin> shincodex: in general you can't expect C++ exceptions to work across C++ to C interface boundaries... [19:45:27 CEST] <maziar> kepstin please help me to improve my command [19:45:47 CEST] <kepstin> maziar: have you tried it? is there some specific issue you're having with it? [19:47:59 CEST] <kepstin> you're missing a few particular options that I'd recommend setting - in particular, I recommend always specifying the video encoder to use, you'll want to set a bitrate (and maybe vbv options depending on use case) [19:48:23 CEST] <kepstin> there's rarely a reason to use baseline encoding, very few devices require it and it's much less efficient. [19:50:37 CEST] <maziar> kepstin would you please kindly give me a sample [19:51:08 CEST] <ultrav1olet> can anyone confirm that ffmpeg doesn't pass x265 to the encoder? [19:51:19 CEST] <ultrav1olet> params [19:51:44 CEST] <ultrav1olet> if you can please chime in https://trac.ffmpeg.org/ticket/5715 [19:57:07 CEST] <maziar> kepstin would you please kindly give me a sample [19:58:47 CEST] <kepstin> ultrav1olet: i've confirmed that with the x265 version I have, using -x265-params to set crf works, but some of the other options either have no effect or i'm not sure what to look for. [19:59:06 CEST] <maziar> sry about duplicate msg its my keyboard issue [20:04:06 CEST] <kepstin> ultrav1olet: tried a few other options, looks like they all work fine. [20:04:52 CEST] <kepstin> ultrav1olet: I assume you probably are just having a syntax error or mismatched option name that's just being ignored... would have to have someone familiar with x265 options look at your command syntax [20:05:15 CEST] <shincodex> I didnt say that at all [20:05:20 CEST] <maziar> kepstin dear kepstin please kindly help me about improvement of my command [20:05:27 CEST] <shincodex> Though your right the collection of junk is in C [20:05:27 CEST] <maziar> kepstin i need your advice [20:05:32 CEST] <shincodex> C doesnt have exceptions [20:05:37 CEST] <shincodex> it just blows up instead [20:05:45 CEST] <shincodex> My problem was [20:05:48 CEST] <kepstin> ultrav1olet: in particular, the 'no-xxx' options don't work, you have to use e.g. 'xxx=0' or something, depending on the option [20:06:04 CEST] <shincodex> I was being idiot thinking throw std::string had inherited exception class [20:06:06 CEST] <shincodex> wrong [20:06:17 CEST] <shincodex> so i just blew up on uncaught exceptions instead [20:13:27 CEST] <maziar> kepstin !ping [20:14:11 CEST] <kepstin> maziar: I don't have enough time to hand-hold someone through learning basic stuff about video encoding atm :/ [20:14:44 CEST] <maziar> kepstin sir, i know that, but i need your advice [20:17:19 CEST] <shincodex> his time [20:17:23 CEST] <shincodex> has run out he died [20:22:05 CEST] <kepstin> I'm busy at work right now :) I have a limited amount of goodwill and spare time while I'm waiting for things to finish, but it's all used up now. [20:24:32 CEST] <shincodex> im at work too [20:24:48 CEST] <shincodex> working on [20:25:02 CEST] <shincodex> idk... i got 3 more projects to make 6 make files for [20:25:04 CEST] <shincodex> too much work [20:25:38 CEST] <maziar> kepstin is it possible that i send email about it to you? [20:26:03 CEST] <kepstin> maziar: nope. and I don't do any paid support [20:26:35 CEST] <maziar> kepstin please kindly let me to have your help and advice [20:28:09 CEST] <ultrav1olet> kepstin: pretty much all my options are being ignored [20:28:41 CEST] <ultrav1olet> crf can also be specified and and it has no effect on the output [20:29:00 CEST] <ultrav1olet> Maybe that's because I linked x265 statically [20:29:12 CEST] <ultrav1olet> but it's very hard to believe [20:29:29 CEST] Action: ultrav1olet has gone [20:39:24 CEST] <maziar> kepstin sir, i am here for your advice, please let me know when you are free [20:43:36 CEST] <Fyr> durandal_1707, hey, bro. [20:43:57 CEST] <Fyr> could you send your patch to the developers? [20:46:17 CEST] <maziar> Fyr i'm using this command to convert a 1080 MP4 file to 720 hls, is this correct command : ffmpeg -i input.mp4 -map 0 -vf scale=720:-1 -c:a copy -profile:v baseline -level 3.0 -start_number 0 -hls_time 3 -hls_list_size 0 -f hls index.m3u8 [20:48:09 CEST] <Fyr> looks like [20:48:23 CEST] <Fyr> -start_number 0 is not necessary. [20:49:41 CEST] <maziar> Fyr and which more? [20:50:40 CEST] <maziar> Fyr someone told me "you're missing a few particular options that I'd recommend setting - in particular, I recommend always specifying the video encoder to use, you'll want to set a bitrate (and maybe vbv options depending on use case) , there's rarely a reason to use baseline encoding, very few devices require it and it's much less efficient. " [20:51:24 CEST] <Fyr> sure [20:51:31 CEST] <Fyr> I also would use -preset [20:51:53 CEST] <shincodex> give me your money [20:52:35 CEST] <maziar> Fyr would you please kindly give me a sample ? [20:53:02 CEST] <Fyr> sorry, I've never used HLS format. [20:57:23 CEST] <maziar> Fyr thank you so much [20:57:35 CEST] <maziar> i need help :( [21:02:06 CEST] <Fyr> maziar, does this command not work as it's supposed to? [21:03:17 CEST] <maziar> Fyr this work : ffmpeg -i input.mp4 -profile:v baseline -level 3.0 -start_number 0 -hls_time 3 -hls_list_size 0 -f hls index.m3u8 ,,,,,,,,, but this does not change the quality [21:03:42 CEST] <Fyr> what kind of change do you expect? [21:04:21 CEST] <maziar> for example convert a MP4 with 1080p to a HLS with 480p [21:04:52 CEST] <Fyr> then use scale 480:-1 [21:05:13 CEST] <maziar> Fyr just this? [21:05:37 CEST] <maziar> Fyr i will be back in 5 minute i should go to ...' [21:05:37 CEST] <Fyr> -vf scale 480:-1 [21:07:54 CEST] <thebombzen_> isn't 480:-1 480 wide and whatever height? rather than 480p? [21:08:17 CEST] <Fyr> yeah, maybe [21:08:39 CEST] <thebombzen_> you'd need -1:480 for 480p [21:09:22 CEST] <thebombzen_> although ideally you'd use -2:480 to force the width to be even [21:42:51 CEST] <durandal_1707> Fyr: I'm developer [21:43:02 CEST] <Fyr> perfect! [21:43:16 CEST] <Fyr> when will you push the changes? [21:44:21 CEST] <durandal_1707> when I find way to improve it [21:51:48 CEST] <thebombzen_> how do you blockquote in TRAC? I'm trying to edit the wiki page and I don't remember how to use MediaWiki [21:53:46 CEST] <Fyr> thebombzen, [21:53:46 CEST] <Fyr> {{{ [21:53:46 CEST] <Fyr> }}} [21:53:46 CEST] <Fyr> ? [21:53:59 CEST] <thebombzen_> that does code afaik. is it different from blockquote? [21:54:03 CEST] <thebombzen_> or is it the same? [21:54:29 CEST] <Fyr> thebombzen, http://trac.ffmpeg.org/wiki/WikiFormatting [21:54:39 CEST] <Fyr> search "blockquote" [21:54:40 CEST] <thebombzen_> haha didn't realize that was an article. RTFM apparently [21:59:23 CEST] <thebombzen_> just documented force_original_aspect_ratio on https://trac.ffmpeg.org/wiki/Scaling%20(resizing)%20with%20ffmpeg [22:00:30 CEST] <thebombzen_> not sure though if what I wrote is considered "okay" so I'd appreciate it if someone could doublecheck to make sure I didn't do anything wrong with it. just added the last paragraph [22:06:03 CEST] <maziar> thebombzen_ !ping [22:06:19 CEST] <thebombzen_> yes? [22:15:01 CEST] <Ravens-S> How would one be able to mass convert an entire folder from .avi to .mp4? [22:15:16 CEST] <Fyr> with a script [22:15:38 CEST] <Fyr> either bat/bash/python/perl script [22:19:06 CEST] <Ravens-S> Looking to try and do it in a batch [22:20:17 CEST] <maziar> thebombzen_ sorry please explain it to me about "1:480 " "480" "-2:480: [22:20:43 CEST] <thebombzen_> so if you use -vf scale=-1:480 it will resize it to 480p [22:20:55 CEST] <thebombzen_> and the width will be what's needed to preserve the aspect ratio [22:21:08 CEST] <thebombzen_> if you use -vf scale=-2:480 it will do that too, but it will force the width to be an even number. [22:21:42 CEST] <thebombzen_> so if you had a 1920x1080 video and ran -vf scale=-1:480, it would resize it to 853x480, but if you did -vf scale=-2:480, it would make it 854x480 [22:21:44 CEST] <Ravens-S> I am looking to create a file that request for the input location of the files you wish to convert and the output location of where you want the files and then have FFMPEG convert the whole folder [22:22:18 CEST] <thebombzen_> even numbers are better, in that some video codecs require them, and they also work better with a yuv420p pixel format [22:23:25 CEST] <thebombzen_> Ravens-S: you could do something like this; [22:24:11 CEST] <maziar> thebombzen_ then you suggested -vf scale=-2:480 ? [22:26:30 CEST] <thebombzen_> Ravens-S: say, if you wanted to convert a folder of PNGs to JPGs, try this [22:26:55 CEST] <furq> thebombzen_: yuv420p requires an even width and height [22:27:10 CEST] <thebombzen_> furq in the past it's just whined at me [22:27:24 CEST] <thebombzen_> or silently converted. I haven't seen it now [22:27:39 CEST] <thebombzen_> Ravens-S: find FolderName/ -maxdepth 1 -type f -name '*.png' | parallel ffmpeg -i {} {.}.jpg [22:27:46 CEST] <thebombzen_> that would be an exmaple of how you coudl batch convert files [22:30:01 CEST] <furq> they said batch, so i assume that's not going to work [22:30:09 CEST] <furq> although i'm not sure why you wouldn't do that with -exec [22:35:17 CEST] <Ravens-S> It was an idea that I had and I want to see if I can make it work [22:35:41 CEST] <Ravens-S> Ill have to try some of what you are saying out [22:36:54 CEST] <Bray90820> I'm converting videos to mp4 with this command [22:36:54 CEST] <Bray90820> ffmpeg -i "video.mkv" -c:a copy "video.mp4" [22:37:04 CEST] <Bray90820> How would I choose audio track 2 instead of 1 [22:37:15 CEST] <furq> -c:a copy -map 0:v -map 0:a:2 [22:37:29 CEST] <furq> or -map 0:a:1 rather [22:37:30 CEST] <Bray90820> Thanks I'll try it [22:46:13 CEST] <Ravens-S> Bray90820 I know you can use the command of ffmpeg -i "video.mkv" -c:a copy "video.mp4" but what I am trying to do is have a batch file that ask you the location of the folder you want to convert and then have it automatically convert each file in that folder. [22:46:38 CEST] <thebombzen_> furq: parallel allows you to use the {.}.jpg notation [22:46:49 CEST] <thebombzen_> parallel also... parallizes [22:46:54 CEST] <Bray90820> Ravens-S: Did you mean to ping me? [22:47:25 CEST] <Ravens-S> Sorry i did not know that pinged you [22:47:28 CEST] <thebombzen_> so it'll run two ffmpeg's at once. this is nice for encoders/decoders that don't support threading (like afaik png and mjpeg) [22:48:01 CEST] <thebombzen_> also furq to me batch process means do a bunch of conversions at once haha. I didn't raelize he mean Windows Batch File [22:48:02 CEST] <Bray90820> It's ok [22:54:25 CEST] <Bray90820> furq: I don't think I did it right this is the code I used [22:54:27 CEST] <Bray90820> ffmpeg -i "video.mkv" -c:a copy "video.mp4" -map 0:v -map 0:a:1 [22:55:25 CEST] <Bray90820> I still took audio track 1 [22:55:36 CEST] <Bray90820> *It [23:07:16 CEST] <maziar> no matter -vf scale=-2:480 or -vf scale=-2:1080 it will resize video to same size, i mean 1080 and 480 size will be same with this command, and seems its not correct : ffmpeg -i input.mp4 -map 0 -vf scale=-1:1080 -c:a copy -profile:v baseline -level 3.0 -start_number 0 -hls_time 3 -hls_list_size 0 -f hls 1080/low_1080.m3u8 [23:18:00 CEST] Last message repeated 1 time(s). [23:18:54 CEST] <Kadigan> Hi. Can I use some part of the ffmpeg suite to grab frames from a USB framegrabber and output them directly to a preview? [23:19:04 CEST] <thebombzen_> maziar: post the exact command you're using and and output to a paste site like http://pastebin.com/, and give a link [23:19:19 CEST] <thebombzen_> the full uncut output [23:19:37 CEST] <thebombzen_> Kadigan: what do you mean by a "USB framegrabber"? [23:20:03 CEST] <Kadigan> thebombzen_: a device like, for example, a Pinnacle 510-USB, or a Logilink Grabber USB [23:20:34 CEST] <Kadigan> Something that takes Composite or S-Video w/ analog audio, and turns that into digital frames to be sucked up through dshow or something. [23:21:09 CEST] <thebombzen_> if DirectShow can read it then FFmpeg can read it too [23:21:11 CEST] <thebombzen_> see: https://trac.ffmpeg.org/wiki/DirectShow [23:21:15 CEST] <Kadigan> I think it would also be important to select the input pin, because I believe it starts in S-Video :/ [23:21:55 CEST] <thebombzen_> FFmpeg can read from DirectShow inputs. If you're trying to display it, you could display it with ffplay, or you could pipe it into another reader like MPV, ImageMagick, or IrfanView [23:22:48 CEST] <thebombzen_> You could also just dump the output to a file. [23:23:05 CEST] <maziar> thebombzen_ http://pastebin.com/p01mwrGE this is my exact command [23:23:21 CEST] <thebombzen_> >[17:19:17] <thebombzen_> the full uncut output [23:23:24 CEST] <Kadigan> thebombzen_: okay, I think I could figure out how to use ffmpeg to capture that data. How can I now put it on the screen? [23:23:44 CEST] <DHE> he just answered that [23:23:52 CEST] <Kadigan> Ah, apologies. [23:24:06 CEST] <thebombzen_> Sometimes you can just drop in ffmpeg -> ffplay and it works [23:24:18 CEST] <thebombzen_> although ffplay supports fewer commandline options than ffmpeg so you have to be careful [23:24:40 CEST] <thebombzen_> if you run into issues you could transcode it with ffmpeg to something really primitive like ffvhuff and pipe it into ffplay [23:24:56 CEST] <thebombzen_> i.e ffmpeg -i <Input> -c:v ffvhuff -f matroska - | ffplay -f matroska - [23:25:36 CEST] <thebombzen_> You might also want to consider using MPV directly because it's got Unbuffered I/O that FFmpeg doesn't. MPV reads from webcams better than FFmpeg does [23:26:06 CEST] <Kadigan> Ah, damn... Video grabbers are, in my experience, notorious for being nasty little s**ts to deal with. :/ I tried to run the command, and I got "video=Pinnacle 510-USB: Input/output error" [23:26:08 CEST] <thebombzen_> At least this is true with Video4Linux2. I haven't tested MPV on windows though so I dn't know exactly. [23:26:12 CEST] <DHE> ffmpeg isnt' really a "real-time" processes. it's more interested in getting the job done consistently [23:26:40 CEST] <maziar> thebombzen_ this is full http://pastebin.com/0Fd31PbE [23:27:19 CEST] <Kadigan> Hell, the previous one I had, which was a PCI card, basically needed two blue-screens to get the driver in correctly - for some reason it wouldn't go w/o them... If you got blue-screens installing the driver, you were going in the right direction. -_- [23:27:43 CEST] <thebombzen_> maziar: you're complaining that -1:480 and -1:1080 do the same thing [23:27:50 CEST] <thebombzen_> but you only post the output of one of those commands [23:27:59 CEST] <thebombzen_> how do you expect us to help you diagnose the issue? [23:29:16 CEST] <maziar> thebombzen_ you mean all -1:480 and -1:1080 ? you mean i should cover both of them ? [23:29:31 CEST] <thebombzen_> well if your question is "why do these two commands do the same thing" [23:29:36 CEST] <thebombzen_> then yes you should post the output of both commands [23:29:47 CEST] <thebombzen_> unless I misunderstood what you were asking [23:30:03 CEST] <maziar> thebombzen_ sorry for my missunderstanding [23:30:20 CEST] <thebombzen_> OH [23:30:35 CEST] <thebombzen_> I see. You're asking "why does -vf scale -1:1080" not change the side of the video? [23:31:12 CEST] <thebombzen_> the answer to THAT question is that -vf scale=-1:1080 resizes the video TO 1080p, not FROM 1080p. FFmepg already knows what size it starts at. [23:31:30 CEST] <thebombzen_> If you started with a video that's 1080p and you resize it to 1080p, then the size won't change. [23:31:53 CEST] <thebombzen_> In general for most intents and purposes you don't have to tell FFmpeg what the size of the input video is. That's listed in the video metadata. [23:33:16 CEST] <maziar> thebombzen_ please kindly check this and confirm is this okay or not : http://pastebin.com/QptyWx5T [23:33:17 CEST] <thebombzen_> Also, if you start with a 480p video and use -vf scale=-1:1080, it will scale it UP to 1080p, so the output will be 1080p. [23:33:45 CEST] <furq> Stream #0:0(und): Video: h264 (libx264), yuv420p, 910x480 [SAR 4096:4095 DAR 256:135], q=-1--1, 24 fps, 90k tbn, 24 tbc (default) [23:33:47 CEST] <thebombzen_> maziar: that last command looks like it works fine [23:33:48 CEST] <furq> this is 480p [23:33:50 CEST] <furq> so what's the problem [23:33:58 CEST] <thebombzen_> now I'm really confused what your'e trying to do [23:34:52 CEST] <maziar> thebombzen_ i want to convert a 1080 MP4 to 1080 HLS and 480 HLS but with this commands all size is same [23:34:58 CEST] <maziar> look at my servers vide [23:35:15 CEST] <thebombzen_> If you want to convert it to two different videos, try this: [23:35:22 CEST] <maziar> thebombzen_ this is for 1080 : 85.13.231.218/player/x1.html [23:35:56 CEST] <maziar> thebombzen_ and this is for 480 185.13.231.218/player/play.html [23:36:07 CEST] <maziar> thebombzen_ why their size is same ? [23:36:18 CEST] <thebombzen_> ffmpeg -i input.mp4 -vf scale=-1:1080 <conversion options> output_1080.hls -vf scale=-1:480 <conversion options> output_480.hls [23:36:23 CEST] <maziar> thebombzen_ this is for 1080 : http://185.13.231.218/player/x1.html [23:36:26 CEST] <thebombzen_> that will convert it to two different files at once [23:36:46 CEST] <maziar> thebombzen_ and this is for 480 http://185.13.231.218/player/play.html [23:38:06 CEST] <thebombzen_> my verdict on that is i"m not going to try to dubug you're autogenerated Flash code [23:38:11 CEST] <thebombzen_> use HTML5 <video> [23:38:47 CEST] <thebombzen_> See: http://www.w3schools.com/html/html5_video.asp [23:38:50 CEST] <maziar> thebombzen_ would you please separated this two command, for example i just want to convert a 1080 MP4 to 1080 HLS , what is the best command ? i used this : ffmpeg -i input.mp4 -map 0 -vf scale=-1:1080 -c:a copy -profile:v baseline -level 3.0 -start_number 0 -hls_time 3 -hls_list_size 0 -f hls 1080/low_1080.m3u8 [23:39:08 CEST] <thebombzen_> I literally just told you [23:39:10 CEST] <thebombzen_> but okay [23:39:24 CEST] <thebombzen_> I'll copy/paste it for your convenience [23:39:29 CEST] <thebombzen_> [17:36:16] <thebombzen_> ffmpeg -i input.mp4 -vf scale=-1:1080 <conversion options> output_1080.hls -vf scale=-1:480 <conversion options> output_480.hls [23:39:49 CEST] <thebombzen_> and oh look it appears I have to go! try scrolling up when I'm not around to help you [23:39:55 CEST] <maziar> thebombzen_ its not work [23:40:45 CEST] <Kadigan> Okay, I give up. :D I can't do it, and if I continue down this path I'll do it and it'll be 4:30 AM. Maybe. Thanks for the assist however. [23:42:11 CEST] <maziar> thebombzen when i convert a MP4 to HLS i just want to change the format not the size, why size is changed? [23:46:59 CEST] <maziar> someone please kindly help me, for example want to convert a MP4 1080 to HLS 720p, how should i do it, why there is no good sample and documents ? [23:47:40 CEST] <__jack__> maziar: didn't I answer you hours ago ? :o [23:49:10 CEST] <maziar> __jack__ its not work, when you want to convert to 1080, 720p, 480p, it will resize all of format to same size , for example check this URLs : 480 http://185.13.231.218/player/play.html ,,,,,,, 1080 : http://185.13.231.218/player/x1.html why should their size be same ? [23:49:59 CEST] <__jack__> no playable sources found [23:52:48 CEST] <maziar> __jack__ which one ? [23:53:24 CEST] <maziar> __jack__ you should use chrome [23:53:44 CEST] <__jack__> I am :) [23:53:47 CEST] <__jack__> (and both) [23:54:03 CEST] <maziar> __jack__ its work for me [23:55:40 CEST] <__jack__> maziar: that code works (480 and 720 are directories): [23:55:41 CEST] <__jack__> ffmpeg -i original.mkv -map 0:a -map 0:v -vf scale=-2:480 -c:a copy 480/play.m3u8 -map 0:a -map 0:v -vf scale=-2:720 -c:a copy 720/play.m3u8 [23:56:11 CEST] <__jack__> in 480/, you'll get a m3u8 and the related chunks, all of them being 480p [23:56:17 CEST] <__jack__> same in 720, but with 720p [23:57:35 CEST] <maziar> __jack__ just wait a minute to check [00:00:00 CEST] --- Sat Jul 16 2016

1 0

ffmpeg-devel.log.20160714
by burek 15 Jul '16

15 Jul '16

[00:54:52 CEST] <cone-164> ffmpeg 03Rostislav Pehlivanov 07master:e24c31b65625: dirac_vlc: fix undefined shifts [00:54:53 CEST] <cone-164> ffmpeg 03Rostislav Pehlivanov 07master:9c0aba434e77: dirac_vlc: remove redundant macro declaration [00:54:54 CEST] <cone-164> ffmpeg 03Rostislav Pehlivanov 07master:b2b12b2d4ab1: diracdec: fix maximum quantization index checks [00:54:55 CEST] <cone-164> ffmpeg 03Rostislav Pehlivanov 07master:000eb01a7d14: diracdec: fix unchecked byte length [01:40:55 CEST] <atomnuker> BBB: so how did vp9 end up with just a very simple scalar quantization? [01:41:35 CEST] <jamrial> atomnuker: vc2 vsynth fate tests seem broken [01:42:19 CEST] <atomnuker> huh, tested them on mine before I pushed, let me check again [01:49:16 CEST] <atomnuker> okay, the first commit broke them, though not sure how, the residue should be zero either way [01:50:11 CEST] <BBB> atomnuker: its done in-line with actual token decoding, where its essentially negligible, but it saves a load/store pair [01:50:27 CEST] <BBB> atomnuker: if you wanted, I suppose you could do it in line in the idct but its so neglgible that nobody cares [01:50:40 CEST] <BBB> its just one op/coef in the token decoding so its really not that important [01:50:56 CEST] <jamrial> atomnuker: try multiplying instead of shifting. any compiler worth anyne's time will optimize it right [01:51:04 CEST] <BBB> for most codecs, dequant is a real process with rounding and shifting etc. (e.g. see hevc) [01:51:12 CEST] <BBB> but vp9 is just one mul op, thats all [01:51:32 CEST] <atomnuker> yeah, I thought it was kinda odd to see such a simple dequant in this day and age [01:52:06 CEST] <BBB> dunno [01:52:08 CEST] <BBB> it works for them :) [01:54:29 CEST] <TD-Linux> atomnuker, are you excited for PVQ? :) [01:55:45 CEST] <BBB> I think Im starting to see I totally misunderstood the original question [01:55:53 CEST] <BBB> you were asking about the coding technique/algorithm, not the code [01:56:04 CEST] <atomnuker> yeah [01:56:09 CEST] <BBB> yes quant can be improved quite a bit [01:56:21 CEST] <BBB> e.g. different quants per band, full quant tables [01:56:32 CEST] <BBB> esp. for larger transforms [01:57:01 CEST] <TD-Linux> AV1 has QMs as an option now [01:57:46 CEST] <TD-Linux> a lot of places in the libvpx encoder assume quantization is free, though, which makes things "interesting" [02:03:07 CEST] <cone-164> ffmpeg 03Rostislav Pehlivanov 07master:c4adcff97604: dirac_vlc: add a SET_RESIDUE macro and fix recent regression [02:08:14 CEST] <atomnuker> TD-Linux: I am, but it seems it'll take a lot more surgery to make the current codebase work well with it [02:23:15 CEST] <jamrial> atomnuker: you're back at shifting 64 bits when B is 0 [02:24:54 CEST] <jamrial> put the B ? CONVERT : 0 back in SET_RESIDUE. that should hopefully fix it [02:26:01 CEST] <atomnuker> it didn't since INIT_RESIDUE already had that when coverity did its thing [02:26:33 CEST] <atomnuker> (it's really a misdetection since it didn't take the check into account, but oh well) [02:30:11 CEST] <cone-164> ffmpeg 03Rostislav Pehlivanov 07master:f41e37b84f3d: dirac_vlc: prevent shifting a 0 by 64 bits when zeroing residue [09:20:38 CEST] <bp0> is there a way to only sync the dependencies of a certain test instead of make fate-rsync ? [09:43:04 CEST] <nevcairiel> not automatically, no [09:47:49 CEST] <cone-531> ffmpeg 03Matthieu Bouron 07master:6c8942cdfb8b: lavf/mov: fix stream extradata_size allocation [13:43:37 CEST] <kurosu> BBB, fyi, the openhevc guys probably split out the idct from residual adding, because hevc has inter-component residual prediction (...) so, basically, ease of implementation > speed [13:43:58 CEST] <BBB> thats not even part of baseline profile [13:44:24 CEST] <kurosu> ffmpeg hevc decoder supports 4:4:4, where that exists [15:40:51 CEST] <BBB> kurosu: I worry a little that our hevc decoder will perform significantly worse than commercial decoders [15:41:25 CEST] <BBB> kurosu: which maybe is a good thing given the terrible licensing situation hevcs IP holders created for themselves [15:41:32 CEST] <BBB> kurosu: but still, from a technical point of view, its not quite right [15:41:56 CEST] <BBB> does anyone want to review my 32x32 avx2 vp9 idct patch? [16:00:44 CEST] <kurosu> BBB, I don't worry much about software decoding anyway [16:01:15 CEST] <kurosu> most of the new h/w supports it [16:02:02 CEST] <nevcairiel> latest gen supposedly even supports 12-bit, but i havent seen software interfaces yet to make use of that [16:02:12 CEST] <nevcairiel> just higher chroma is left to software [16:03:48 CEST] <kurosu> actually, it's more the availability of encoders that matter [16:03:52 CEST] <kurosu> decoders are easy [16:03:57 CEST] <kurosu> good encoders are more difficult [16:04:22 CEST] <kurosu> the serving for vp9 is rather poor at the time, be it s/w or h/w [16:05:15 CEST] <nevcairiel> hw decoding for vp9 also exists, although less common then hevc [16:05:29 CEST] <kurosu> I meant encoding [16:07:33 CEST] <kurosu> as for ffmpeg hevc decoding, even if it had proper idcts, yes, there's a 10-15% performance likely missing in addition [16:08:01 CEST] <kurosu> well, not likely, provably [16:11:36 CEST] <kurosu> btw, what is netflix waiting for to switch from hevc to vp9? proper hdr support? [16:21:36 CEST] Action: ubitux going back to merges in about an hour [17:04:48 CEST] <kierank> kurosu: vp9 in mp4 support [17:04:54 CEST] <kierank> so they can do encryption [17:14:44 CEST] <BBB> kurosu: hey, vp9 has a wonderful encoder out there [17:38:06 CEST] <kurosu> kierank, ah right, if that happens - though support in ffmpeg was added recently [17:38:54 CEST] <kurosu> BBB, err yes, I was too focusing on live/well-multithreaded encoders, but I recall it was a goal of yours if your encoder got traction [17:54:42 CEST] <ubitux> http://b.pkh.me/0001-lavfi-selectivecolor-add-8bit-support.patch [17:54:44 CEST] <ubitux> ugly :( [17:54:47 CEST] <ubitux> oh well. [17:55:18 CEST] <BBB> kurosu: dont worry its ok [17:55:58 CEST] <cone-908> ffmpeg 03Michael Niedermayer 07master:de9674c5eacc: avformat/mux: Factor do_packet_auto_bsf() out [17:55:58 CEST] <cone-908> ffmpeg 03Michael Niedermayer 07master:a2c90d5f9828: avformat/mux: Apply auto bsfs in av_write_frame() too [18:23:22 CEST] <bp0> how can you look at the log output to test the result for fate? [18:26:25 CEST] <BBB> bp0: what do you mean? [18:26:44 CEST] <BBB> bp0: if you want to know what it runs, check make V=1 fate-... [18:27:37 CEST] <bp0> oh, I mean when adding a test to fate, how could you test for a line in the log as the result? [18:28:30 CEST] <BBB> like a metadata line? [18:29:07 CEST] <bp0> no, like a line that is av_log(ctx, AV_LOG_INFO, ..) ed [18:29:22 CEST] <bp0> by a filter [18:29:33 CEST] <BBB> Im not sure we have infrastructure for that [18:30:01 CEST] <BBB> you can probably just redirect output to a file and compare that file to a ref, but Im not a fate expert by any means [18:31:36 CEST] <bp0> I did make an attempt to add a function to fate_run.sh to match a line in the output, but it wouldn't let me use REF in the test definition to specify the text to match [18:32:14 CEST] <bp0> said REF needed to be a file, although elsewhere REF can be an md5 sum [18:36:19 CEST] <bp0> maybe I could not use REF and do it all as a new function in fate-fun.sh [18:54:25 CEST] <cone-908> ffmpeg 03Paul B Mahol 07master:d78fd2fa21cd: Add MagicYUV decoder [18:54:26 CEST] <cone-908> ffmpeg 03Clément BSsch 07master:c3e2e842fa91: Merge commit 'd78fd2fa21cde28465e40dd0be4446b1387d22a6' [19:00:03 CEST] <cone-908> ffmpeg 03Vittorio Giovara 07master:cc58656aca95: fate: Add tests for MagicYUV [19:00:04 CEST] <cone-908> ffmpeg 03Clément BSsch 07master:cd02df52567d: Merge commit 'cc58656aca95b5ab517989a9524b9a2b1c5653cf' [19:02:11 CEST] <cone-908> ffmpeg 03Clément BSsch 07master:9e6bb3b6eebd: fate/video: use FATE_VIDEO instead of FATE_SAMPLES_AVCONV [19:14:45 CEST] <cone-908> ffmpeg 03Clément BSsch 07master:87e9cefa9110: lavc/codec_desc: move a few codecs out of the image codecs list [19:22:01 CEST] <vade> Hi : Im curious : is it expected for an opened AVCodecContext* to change its time_base after the first call to avcodec_receive_frame wen decoding frames? [19:24:08 CEST] <nevcairiel> its possible yes [19:24:15 CEST] <nevcairiel> any info can change at any time, really [19:24:42 CEST] <vade> interesting. [19:24:47 CEST] <vade> thank you nevcairiel [19:30:33 CEST] <cone-908> ffmpeg 03Paul B Mahol 07master:470cd0c5fe63: Add TrueMotion 2.0 Real Time decoder [19:30:34 CEST] <cone-908> ffmpeg 03Clément BSsch 07master:93c13886a2d6: Merge commit '470cd0c5fe6337b6cb5276b3f84400999450fc1b' [19:30:35 CEST] <cone-908> ffmpeg 03Clément BSsch 07master:bec714525216: lavc/truemotion2rt: mark AVPacket* as const [19:36:10 CEST] <cone-908> ffmpeg 03Vittorio Giovara 07master:906ffed9b1b8: fate: Move Duck Truemotion 1 and 2 tests to vpx.mak [19:36:11 CEST] <cone-908> ffmpeg 03Clément BSsch 07master:e28d66052dd3: Merge commit '906ffed9b1b8b06979eb656989aecacb1ae75a3a' [19:38:16 CEST] <cone-908> ffmpeg 03Vittorio Giovara 07master:523c4c5b7099: fate: Add TrueMotion 2 RT tests [19:38:17 CEST] <cone-908> ffmpeg 03Clément BSsch 07master:def850676625: Merge commit '523c4c5b70994f5cd1f192b68d07cf24b292ca05' [19:38:49 CEST] <ubitux> 3rd h264 wave now. [19:43:05 CEST] <ubitux> still no one to move ffmpeg to codecpar btw? [19:43:37 CEST] <cone-908> ffmpeg 03Anton Khirnov 07master:b25cd7540e7c: h264: pass a H2645NAL to slice header decoding [19:43:38 CEST] <cone-908> ffmpeg 03Clément BSsch 07master:2c138b2c9b0e: Merge commit 'b25cd7540e7cba9868edc13817c0ce1ddef90ffc' [19:48:17 CEST] <ubitux> oooh that's going to be annoying. [19:49:06 CEST] <ubitux> guess what, it takes out stuff from the function we didn't want to move last time [19:49:08 CEST] <ubitux> :( [20:43:25 CEST] <BBB> ubitux: the downside of the whole merging business is indeed that everytime we dont maintain ourselves as a strict patchset on top of libav (which is obviously ridiculous), we have to do tons of extra work to merge non-trivial patches to divergent codebases :( [20:43:35 CEST] <BBB> ubitux: I dont think theres a good solution to that [20:43:59 CEST] <ubitux> well we didn't move it because it was breaking things only in ffmpeg in the first place [20:44:07 CEST] <ubitux> not because their change was wrong [20:58:55 CEST] <nevcairiel> (those two are not mutually exclusive) [21:09:37 CEST] <BBB> loopfilter simd is nearly 10% of runtime on this media file Im testing [21:10:23 CEST] <BBB> so thats promising [21:41:39 CEST] <cone-908> ffmpeg 03Steven Liu 07master:e29d2d9c92e1: avformat/segment: remove the check_bitstream from segment [21:55:57 CEST] <BBB> what does failed to issue emms in checkasm mean? [21:57:21 CEST] <jamrial> BBB: that's the error i mentioned the other day [21:57:34 CEST] <BBB> Im adding some new asm and Im getting that [21:57:44 CEST] <BBB> what do I do? [21:57:56 CEST] <jamrial> replace declare_func with declare_func_emms like in the other vp9 tests [21:58:11 CEST] <BtbN> My assembler knowledge is lacking, but isn't emms the instruction to reset the fp registers, and you are supposed to call if after being done with mmx instructions? [21:58:42 CEST] <jamrial> BtbN: yes [21:58:45 CEST] <BtbN> So I'd interpret it as "you are not calling emms while you should" [21:58:54 CEST] <BBB> I shouldn't [21:59:00 CEST] <BBB> I Expect the caller to do that (and it does) [21:59:25 CEST] <nevcairiel> then you should tell your test that [21:59:28 CEST] <jamrial> BtbN: sort of. it's a change to checkasm that came from a merge that's got the naming inverted i think [21:59:33 CEST] <BBB> ugh it still fails :-p [21:59:40 CEST] <BBB> (but now with a checkasm fail instead of a emms fail) [21:59:46 CEST] <BBB> so I guess the assembly is broken [21:59:51 CEST] <jamrial> BBB: declare_func_emms(AV_CPU_FLAG_MMX | AV_CPU_FLAG_MMXEXT, void, ... [22:00:00 CEST] <BBB> yup [22:00:05 CEST] <jamrial> still fails? huh [22:00:05 CEST] <BBB> ty [22:00:12 CEST] <BBB> my assembly is broken :-p [22:00:15 CEST] <jamrial> ah [22:58:31 CEST] <BBB> actually the assembly was fine, the funcptr assignment was broken (I assigned a vertical filter function to the horizontal filter index in the dsp func array) [22:58:35 CEST] <BBB> duh duh duh [23:09:50 CEST] <jkqxz> Has anyone actually tried to use qsvdec in recent times? [23:09:57 CEST] <jkqxz> In <http://git.videolan.org/?p=ffmpeg.git;a=blob;f=libavcodec/qsvdec.c;hb=HEAD#…>, how can it ever successfully decode a frame given that sync must be a null pointer when we dereference it at line 412? [23:14:00 CEST] <nevcairiel> why does it have to be null [23:14:26 CEST] <nevcairiel> its allocated there and filled by MFXVideoDECODE_DecodeFrameAsync [23:15:36 CEST] <jkqxz> Line 407 is the only way to break out of the loop, but it can only get there if it takes the else branch of the condition above (the if has a continue), where sync is unconditionally freed. [23:20:04 CEST] <michaelni> bp0, the 2 fate references are together 6mb, is that size needed ? [23:43:26 CEST] <atomnuker> >If you have a chance to put the audio in a audio editor and zoom very close in, you see that there is a clipping error, or something else. [23:43:43 CEST] <atomnuker> excellent, top tier bug reporting here [23:45:07 CEST] <atomnuker> "if you zoom into this jpeg image with a magnifying glass you can see a blocky halo around this letter here" [23:46:11 CEST] <jamrial> why is he reencoding 192k aac into 128k aac anyway? [23:47:11 CEST] <atomnuker> why is he not using opus [23:52:23 CEST] <jamrial> atomnuker: seeing he's using mp4/h264/aac, it's probably because he wants a hardware or smartphone compliant file [23:52:34 CEST] <jamrial> or simply a file that will be compatible with everything under the sun [23:53:42 CEST] <jamrial> he should stream copy that audio stream. lossy to lossy audio conversion is evil [00:00:00 CEST] --- Fri Jul 15 2016

1 0

ffmpeg.log.20160714
by burek 15 Jul '16

15 Jul '16

[00:50:04 CEST] <fastpenguin> Hey, I can't figure out how to properly install ffmpeg on OS X. I downloaded the zip file here: http://www.ffmpegmac.net/ and I know it's not installed, but when I click on the ffmpeg it just opens the command line. [01:28:03 CEST] <mccc> Hi, I'm trying to cross compile and install ffmpeg with alsa and fdk-aac plugins for Raspberry Pi. [01:29:01 CEST] <mccc> I'm a bit new but I've been experimenting with both static and shared builds. [01:29:54 CEST] <mccc> I'm having trouble compiling alsa as a shared library right now. What I'm doing is first building on my development environment (Debian) using cross compiling tools, and then copying to the Raspberry Pi to install. [01:31:21 CEST] <mccc> Here are the commands I'm running to compile on each platform: [01:31:22 CEST] <mccc> http://pastebin.com/RPSx96Mm [01:31:49 CEST] <mccc> Right now on the Raspberry Pi, after running sudo make install, I get the following error: [01:31:53 CEST] <mccc> control/.libs/libcontrol.a: member control/.libs/libcontrol.a(cards.o) in archive is not an object [01:32:34 CEST] <mccc> Can someone point me in the right direction to diagnose this, or confirm that I'm going about cross compiling and installing the right way? Thank you. [02:02:07 CEST] <nd> Hello, i wonder if ffserver is still worth using. I heared ppl talking about it beeing no longer supported. Anybody got sources/information for me? [02:05:19 CEST] <c_14> https://ffmpeg.org/index.html#news [02:06:09 CEST] <nd> ok now i feel stupid... thanks a lot [02:15:26 CEST] <mccc> Hi, am I cross-compiling alsa for ffmpeg the right way? ./configure, make on my dev box, copy sources to Raspberry Pi, ./configure, sudo make install on the Raspberry Pi? [06:08:03 CEST] <FFM> Hello [06:08:29 CEST] <FFM> Anybody knows how encoding to LPCM on the ffmpeg? [06:23:05 CEST] <SpaceManiac> Hi, I noticed that ffmpeg supports .xm format, but not FMOD's .oxm extension; does anyone know if there is an IP encumbrance or would a patch to add support be accepted? [07:14:13 CEST] <mccc> Hi, am I cross-compiling alsa for ffmpeg the right way? ./configure, make on my dev box, copy sources to Raspberry Pi, ./configure, sudo make install on the Raspberry Pi? [09:20:19 CEST] <bp0> is there a way to only sync the dependencies of a certain test instead of make fate-rsync ? [09:20:32 CEST] <bp0> sorry, should be in devel [10:56:44 CEST] <debianuser> mccc: if your raspberry pi includes build toolchain you can just build it there, without any crosscompiling, just copy ffmpeg sources to rpi and `./configure <your options> && make && sudo make install` there. [13:01:04 CEST] <DelphiWorld> 'lo all [13:01:18 CEST] <DelphiWorld> anyone yet have any sucsess on using qsv h.264 encoder? [13:03:18 CEST] <BtbN> On Windows it should just work if you select it. [13:03:30 CEST] <BtbN> On Linux, I'd recommend forgetting its entire existence. [13:11:19 CEST] <DelphiWorld> why? [13:28:46 CEST] <DelphiWorld> so i shouldn't dream for qsv on linux? [14:11:26 CEST] <friki2015> what means "drop" in this context? "-timecode hh:mm:ssSEPff: Specify Timecode for writing. SEP is : for non drop timecode and ; (or .) for drop." [15:18:55 CEST] <md2k> Hi All [15:19:12 CEST] <md2k> cn anyone help me with -stream_loop option? [15:20:26 CEST] <md2k> no matter what i do when last clip from list file (using concat) played and it is time to read 1st file again i have error: Error while filtering: Illegal seek [15:24:38 CEST] <md2k> version of ffmpeg N-80901-gfebc862 (7:3.1.1~trusty) [15:38:03 CEST] <Conder> hi, whats command to set video title in mkv? parameter -metadata title="xx" set name of whole mkv, i want just title of video stream [15:51:22 CEST] <limbo> it is title, but it's the chapter title. [17:04:55 CEST] <friki2015> -metadata:s:v:0 title="my first video" [17:05:39 CEST] <friki2015> Conder: i think it should work -metadata:s:v:0 title="my first video" [17:07:13 CEST] <Conder> i already edited it with mkvtoolnix, but anyway thanks, i'll use it next time [18:31:11 CEST] <md2k> Hi guys, so anyone know if `-stream_loop` can work with `-f concat` functionality in ffmpeg? because i always getting `Illegal Seek` error [19:07:47 CEST] <vade> is it expected for a CodecContext to change its time_base after the first call to avcodec_receive_frame ? [19:40:04 CEST] Action: DelphiWorld is back [19:40:09 CEST] <DelphiWorld> still fighting intel QSV... [19:40:39 CEST] <DelphiWorld> someone got intel qsv encoding working on linux? [19:41:03 CEST] <furq> isn't it recommended to use vaapi for that on *nix now [19:41:34 CEST] <DelphiWorld> furq: i'm confused a lot with vaapi... and for a live stream isn't qsv faster ? [19:41:45 CEST] <furq> no idea, i don't have an intel cpu that supports it [19:42:11 CEST] <DelphiWorld> my ffmpeg is allready libva enabled, but it look like the cmdline is much complicated [19:42:56 CEST] <furq> https://ffmpeg.org/pipermail/ffmpeg-user/2016-May/032153.html [19:43:19 CEST] <DelphiWorld> i am still wondring how did some people got it to work [19:44:51 CEST] <DelphiWorld> furq, i read that. and yay they sayd it's not pur hardware transcoding [19:49:14 CEST] <kepstin> DelphiWorld: just use vaapi. It uses the same hardware encoder block as the libmfx stuff, and is a lot easier to get working [19:49:48 CEST] <DelphiWorld> kepstin: but please please please. what's WRONG with libmfx? [19:50:34 CEST] <kepstin> (partially) closed source, have to agree to an intel eula, latest versions drop support for older chips, and it's difficult to actually get working. [19:50:57 CEST] <DelphiWorld> crap..., and intel documenting how to do it in ffmpeg! [19:54:08 CEST] <DelphiWorld> kepstin: that's my current lock: [19:54:10 CEST] <DelphiWorld> ffmpeg -vaapi_device /dev/dri/renderD128 -hwaccel vaapi -hwaccel_output_format vaapi -i B4.mp4 -vf 'format=nv12|vaapi,hwupload,scale_vaapi=w=1280:h=720:format=yuv420p,hwdownload' -c:v libx264 -acodec copy out.mp4 [19:54:29 CEST] <DelphiWorld> libva error: /opt/intel/mediasdk/lib64/iHD_drv_video.so init failed [19:54:48 CEST] <kepstin> ... why is libva trying to use mediasdk stuff? [19:55:08 CEST] <DelphiWorld> i have no idea, cause i used intel sdk to try to get qsv [19:55:10 CEST] <kepstin> you should have the open-source driver stuff installed, it doesn't need mediasdk at all [19:55:26 CEST] <DelphiWorld> ok, where to get it? [19:55:37 CEST] <kepstin> i guess you have a partially installed/broken mediasdk that's interfering with the open source drivers :/ [19:55:50 CEST] <DelphiWorld> i dont even know how should i remove the existing, afraid to lock my system [19:58:04 CEST] <furq> you ought to be able to remove anything from /opt without breaking your system [19:58:15 CEST] <furq> emphasis on "ought to" because people can't be trusted [19:59:33 CEST] <DelphiWorld> furq: should i remove the intel libva library? [19:59:59 CEST] <furq> no idea [20:00:01 CEST] <kepstin> if vaapi is loading a driver that's trying to load the mediasdk library, it's probably installed some stuff outside of /opt [20:00:43 CEST] Action: DelphiWorld is lost, so. [20:00:54 CEST] <kepstin> DelphiWorld: what does 'ls /usr/lib/dri' show? (probably best to pastebin it) [20:01:34 CEST] <DelphiWorld> there's no such /usr/lib/dri, kepstin [20:02:08 CEST] <furq> what distro [20:02:12 CEST] <kepstin> hmm, could be system specific. iirc on debian it might be /usr/<arch string>/lib/dri? [20:02:16 CEST] <DelphiWorld> ubuntu server [20:02:20 CEST] <furq> /usr/lib/yourarch/dri [20:03:17 CEST] <furq> presumably /usr/lib/x86_64-linux-gnu/dri [20:03:44 CEST] <DelphiWorld> i am pbing it [20:08:02 CEST] Action: DelphiWorld is locked with DNS issue... ;) [20:10:42 CEST] <DelphiWorld> kepstin: will be back, restarting my dumb router [20:15:46 CEST] <shincodex> find? [20:16:16 CEST] <shincodex> maybe he link with it [20:26:21 CEST] <JEEB> /win32 [20:32:56 CEST] <DelphiWorld> furq, kepstin: i'm back [20:33:06 CEST] <DelphiWorld> pastebined: http://paste.ubuntu.com/19393656 [20:39:32 CEST] <DelphiWorld> fucking i7 not pretty usefull without QSV... crapy, intel! [20:39:32 CEST] <DelphiWorld> model name : Intel(R) Core(TM) i7-6700 CPU @ 3.40GHz [20:40:13 CEST] <shincodex> amd is shit [20:40:19 CEST] <shincodex> they just blew up tons of poor nerds machines [20:40:22 CEST] <Fyr> ((= [20:40:23 CEST] <shincodex> they robbed me of stocks [20:40:27 CEST] <shincodex> You better love your intel [20:40:35 CEST] <Fyr> NVIDIA rules. [20:40:37 CEST] <shincodex> 0_0 [20:40:47 CEST] <shincodex> class action coming soon [20:40:49 CEST] <DelphiWorld> Fyr: +++ [20:41:05 CEST] <DelphiWorld> intel making me crazy with there media sdk [20:41:24 CEST] <DelphiWorld> is gpu transcoding anyy better thant that QSV shit? [20:41:24 CEST] <shincodex> windows media? [20:41:43 CEST] <DelphiWorld> shincodex: no, ffmpeg with Intel suposed QuickSync [20:43:27 CEST] <shincodex> is that supposed to be something that helps sync audio, ass, video together [20:43:53 CEST] <furq> what [20:44:14 CEST] <DelphiWorld> check it furq, http://paste.ubuntu.com/19393656 [20:44:23 CEST] <furq> like i said, i've never used vaapi [20:44:35 CEST] <DelphiWorld> ok, i'll wait for kepstin so [20:47:03 CEST] <furq> https://packages.debian.org/jessie/i965-va-driver [20:47:06 CEST] <furq> i assume you need that [20:47:56 CEST] <DelphiWorld> furq: sadly allready there! [20:49:27 CEST] <furq> http://packages.ubuntu.com/trusty/amd64/i965-va-driver/filelist [20:49:31 CEST] <furq> i don't see that .so in your paste [21:07:27 CEST] <DelphiWorld> how to get rid of that shitty intel sdk... [21:17:56 CEST] <kepstin> hmm, that's weird. you should have a file 'i965_drv_video.so' if the vaapi driver is installed correctly [21:18:17 CEST] <kepstin> i'm guessing that the intel mfx stuff must be loaded via the vdpau compatibility driver :/ [21:20:44 CEST] <kepstin> given that the command you gave as an example was using qsv to decode then x264 to encode, i'd expect a mild, but not particularly impressive speed increase by moving the decode off the cpu... [21:20:52 CEST] <kepstin> eh, he's gone again [21:28:01 CEST] <furq> DelphiWorld: [21:28:02 CEST] <furq> 20:17:56 ( kepstin) hmm, that's weird. you should have a file 'i965_drv_video.so' if the vaapi driver is installed correctly [21:28:05 CEST] <furq> 20:18:17 ( kepstin) i'm guessing that the intel mfx stuff must be loaded via the vdpau compatibility driver :/ [21:28:40 CEST] <DelphiWorld> thank furq [21:28:45 CEST] <DelphiWorld> kepstin: how to use that vdpauN [21:28:48 CEST] <DelphiWorld> vdpau ? [21:28:59 CEST] <furq> do you even have that driver installed [21:29:08 CEST] <DelphiWorld> i'll look [21:29:09 CEST] <DelphiWorld> hold on [21:29:15 CEST] <furq> it wasn't there in your pastesdfsd [21:29:28 CEST] <furq> how hard is it to stay on irc [21:32:13 CEST] <DelphiWorld> vdpau allready installed..., ding [23:34:38 CEST] <mccc> @debianuser Thank you. I want to build on another VM for faster builds. I did get this working (my notes are here: http://pastebin.com/JpvK0EJP) but I still suspect I should be creating a deb package rather than running make and make install on separate machines. Is that true? [23:36:57 CEST] <furq> just copy the ffmpeg binary to /usr/local/bin on the rpi [23:37:40 CEST] <furq> you should also be able to just use libasound2-dev:armhf on the host machine [23:38:02 CEST] <furq> https://wiki.debian.org/Multiarch [23:38:31 CEST] <mccc> How about alsa? When I tried compiling alsa statically in to ffmpeg, it was missing alsa.conf. I am just copying the ffmpeg bin directory over. [23:39:52 CEST] <furq> nvm i misread that [23:40:48 CEST] <furq> afaik if you install libasound2-dev:armel and libasound2-data:armel on both boxes then it should work without having to compile alsa [23:40:55 CEST] <furq> that's the whole point of multiarch [23:42:43 CEST] <mccc> Okay, thank you, I'll try that now. [00:00:00 CEST] --- Fri Jul 15 2016

1 0

ffmpeg-devel.log.20160713
by burek 14 Jul '16

14 Jul '16

[00:00:01 CEST] <BBB> I think I asked that before [00:00:28 CEST] <jamrial> also, websites like vlc's defaulting to 32 bits builds and requiring the user to click and browse through four pages to get a 64 bits build doesn't help either [00:01:00 CEST] <nevcairiel> i work for a company making a commercial media management solution .. ie. a glorified player with a file database [00:02:21 CEST] <nevcairiel> and there is of course my open-source plugin based on avformat and avcodec which integrates both into the DirectShow framework, used by a large variety of windows players [00:02:54 CEST] <BBB> I sometimes forget that windows still exists [00:02:57 CEST] <BBB> these blissful moments [00:03:16 CEST] <llogan> then it pops up back into your life like a lip herpy [00:03:41 CEST] <nevcairiel> i've been pushing for work to at least start offering a 64-bit version with a big notice that plugins wouldnt work, but its complicated [00:08:01 CEST] <BBB> llogan: I just close my eyes and pretend it didnt happen :-p [00:13:25 CEST] <friki2015> (offtopic?) the seek time on a video depends on the length of the jump, isn't it? [00:14:06 CEST] <nevcairiel> generally, no [00:14:58 CEST] <friki2015> so, i'll redo my yesterday tests [00:15:00 CEST] <nevcairiel> most seekable formats would have an index for direct seeking, or at least directly readable timestamps for a binary search [00:15:21 CEST] <nevcairiel> some formats are not really designed for seeking, so they may behave differently [00:16:22 CEST] <friki2015> by default a ffmpeg encoded video should bee "direct seekable"? (i dont' remenber exaclti... h264+aac+(flv/ts) ) [00:16:37 CEST] <nevcairiel> it depends on the format [00:16:45 CEST] <nevcairiel> both flv and ts are not very good formats for file storage [00:17:02 CEST] <nevcairiel> ts is for streaming/broadcasting, and flv is just generally bad :p [00:17:05 CEST] <nevcairiel> use mp4 or mkv [00:17:37 CEST] <friki2015> thanks [00:19:19 CEST] <friki2015> i plan to seek very long videos (max 1 week) [00:22:36 CEST] <llogan> atomnuker: pushed the ffserver news [00:22:51 CEST] <atomnuker> llogan: saw, thanks [00:59:50 CEST] <BBB> friki2015: the seek time (for exact seeking) generally depends on the distance to the preceeding i frame. if seeking to a i frame, the seek time should be near-instant unless something else buggers you (player overhead, playback decoded frame cache, network latency for non local files, etc.) [01:00:31 CEST] <BBB> friki2015: for some file formats without indices (seekcache in matroska parleance), seeking is much more complex [02:57:30 CEST] <friki2015> BBB: thanks, i'm working to build a 1 week video in order to test disctinct file formats. i really don't care about the fileformat if it can seek fast on very big videos [03:03:26 CEST] <Compn> ehe [03:03:34 CEST] <Compn> i bet there are a lot of time limits hidden in various demuxers [03:03:52 CEST] <Compn> but people have been streaming for weeks so maybe not [06:05:25 CEST] <bp0> have you heard of slow tv in norway? [06:05:39 CEST] <bp0> friki2015 [06:05:45 CEST] <bp0> sorry that was 3 hours ago [06:06:25 CEST] <bp0> anyway, it is hours long or days long video of something uncut, like a train journey or a cruise, or burning firewood [06:06:41 CEST] <bp0> broadcast on public tv [08:35:06 CEST] <cone-134> ffmpeg 03Vignesh Venkatasubramanian 07master:acca56d9629b: libvpx: Enable vp9 alpha encoding [13:17:45 CEST] <robUx4> nevcairiel: have you tried VP9 Profile 1 (10 bits) with the DXVA2 Profile 0 decoder ? [13:17:54 CEST] <robUx4> and VP8 ? [13:19:14 CEST] <nevcairiel> the decoders support exactly the profile they advertise, trying to decode anything else is futile [13:21:02 CEST] <robUx4> ok [14:08:38 CEST] <friki2015> bp0: never heard about it... in spain when we need to take a snap we just watch 'Le Tour' ;) [14:09:34 CEST] <friki2015> my 1 week records will be some think similar to a security cam record [14:40:21 CEST] <mateo`_> michaelni: i've updated my libav-merge branch (https://github.com/mbouron/FFmpeg/commit/2cd0ad839d2ea4c77ee21cf8ca89e61e8f…) the first crash is fixed, the regression while playing http://samples.ffmpeg.org/ffmpeg-bugs/trac/ticket1918/aark15sd_9A62E2FA.mp4 not yet, because some stsc entries (1503-1800~)from this file seems corrupt and the merge adds a check on those values (they must not be greater [14:40:23 CEST] <mateo`_> than sc->stsd_count) [14:42:14 CEST] <mateo`_> should I drop compatibility with this file *or* should i drop the check and improve/harden the new code that rely on sc->stsc[index].id to deduct the stsd id [14:47:50 CEST] <mateo`_> second proposition sounds better (imho) [15:00:31 CEST] <cone-567> ffmpeg 03Martin Vignali 07master:f2b08a07028e: libavcodec/exr : cosmetics, rename variable in b44_uncompress func [15:01:02 CEST] <michaelni> mateo`_, i agree 2nd sounds better [15:05:25 CEST] <mateo`_> should the fix be separate commit ? [15:05:52 CEST] <nevcairiel> ideally master should never be broken [15:06:39 CEST] <mateo`_> ok, i'll document the changes i made in the commit message then [15:13:48 CEST] <michaelni> mateo`_, theres a crash remaining, ill sent you the sample to repro [15:14:18 CEST] <mateo`_> thanks [16:15:13 CEST] <mateo`_> michaelni: i've updated the branch [16:38:45 CEST] <ubitux> i'll start back the merging effort in the next days [16:41:37 CEST] <mateo`_> michaelni: regarding the sample you provided, it seems there is regression between 3.0.2 et current master, ie: the sample does not play as it did before [16:42:32 CEST] <michaelni> mateo`_, which sample ? [16:42:55 CEST] <mateo`_> the one you sent to me [16:43:22 CEST] <mateo`_> the private one to be more precise [16:44:09 CEST] <michaelni> it doesnt need to play, its not a valid h264 stream [16:44:33 CEST] <mateo`_> it did before [16:45:12 CEST] <mateo`_> anyway, i'll push the merge commit in a few minutes [16:46:25 CEST] <michaelni> i see what you mean about the sample [16:55:02 CEST] <cone-567> ffmpeg 03Vittorio Giovara 07master:76729970049f: mov: Implement support for multiple sample description tables [16:55:03 CEST] <cone-567> ffmpeg 03Matthieu Bouron 07master:3c058f570128: Merge commit '76729970049fe95659346503f7401a5d869f9959' [17:06:44 CEST] <cone-567> ffmpeg 03Vittorio Giovara 07master:846a3e78a535: mov: Support prores with multiple stsd [17:06:45 CEST] <cone-567> ffmpeg 03Matthieu Bouron 07master:354336490da0: Merge commit '846a3e78a535f05ee61bb23a160f3378f041f751' [17:14:15 CEST] <cone-567> ffmpeg 03Matthieu Bouron 07master:495a40cecb4d: tests/checkasm: reduce cosmetic diff with libav [17:23:14 CEST] <cone-567> ffmpeg 03Martin Storsjö 07master:105998fb5ca3: checkasm: Add tests for h264 idct [17:23:15 CEST] <cone-567> ffmpeg 03Matthieu Bouron 07master:a91c330a29f3: Merge commit '105998fb5ca3c343f5c8cb39ce3197f87a5e4d36' [17:26:20 CEST] <mateo`_> michaelni: does our ffv1 decoder support transparency ? [17:28:49 CEST] <mateo`_> I'm looking at the next merge (ec9f044 ffv1: Error out on unsupported format) [17:29:13 CEST] <mateo`_> sorry the question would be more like does it support transparency with format other than yuv [17:30:29 CEST] <michaelni> i see AV_PIX_FMT_RGB32 in there [17:30:38 CEST] <michaelni> thats with alpha [17:32:35 CEST] <mateo`_> mmm, I wonder if this commit should be a noop [17:32:57 CEST] <michaelni> probably [17:34:17 CEST] <michaelni> is this supposed to be a reimplementation of 878c3a36451eaf1ae3ec3d8eab0af11dab0a7695 ? [17:35:26 CEST] <jamrial> mateo`_: can i push something or will it force you to locally rebase your work? [17:36:44 CEST] <mateo`_> jamrial: you can push [17:36:48 CEST] <mateo`_> thanks for askin ;) [17:36:50 CEST] <mateo`_> +g [17:37:42 CEST] <cone-567> ffmpeg 03James Almer 07master:fde9c5e42430: fate: fix fate-vp8 dependencies [17:37:55 CEST] <jamrial> mateo`_: no prob [17:41:19 CEST] <cone-567> ffmpeg 03Jerome Martinez 07master:ec9f04423b82: ffv1: Error out on unsupported format [17:41:20 CEST] <cone-567> ffmpeg 03Matthieu Bouron 07master:57fa9608e419: Merge commit 'ec9f04423b82afa323e90f5b2c677be74302c1fd' [17:55:15 CEST] <BBB> vp9_inv_dct_dct_32x32_add_8_avx: 2690.9 [17:55:15 CEST] <BBB> vp9_inv_dct_dct_32x32_add_8_avx2: 1572.9 [17:55:23 CEST] <BBB> sweet [17:55:41 CEST] <BBB> so thats 1.7x [17:56:17 CEST] <BBB> (for the full idct; I didnt do sub-idcts yet) [17:58:19 CEST] <mateo`_> durandal_1707: http://0x5c.me/diff.html should i take all their changes ? [17:58:58 CEST] <mateo`_> durandal_1707: nevermind, this diff is not relevant [18:01:45 CEST] <mateo`_> durandal_1707: i've updated the diff comparing their commit and the current magicyuv.c code in master [18:03:15 CEST] <ubitux> #include "../libavutil/pixdesc.h" you don't want the "../" here [18:03:23 CEST] <ubitux> and you want to keep the AV_QSORT() [18:03:31 CEST] <ubitux> (and copyrights) [18:03:45 CEST] <ubitux> otherwise i think you can take everything to reduce diffs [18:03:58 CEST] <durandal_1707> Yes [18:06:29 CEST] <mateo`_> I'll stop with the merges for today [18:30:56 CEST] <BBB> I also figured out why my bench output didnt work for sub-idcts [18:55:26 CEST] <jamrial> BBB: nice! [18:56:57 CEST] <jamrial> it's amazing how many of these functions get a massive boost even with the simplest simd implementation [18:57:09 CEST] <jamrial> from 14k to 2.6k with only sse2, for example [18:58:22 CEST] <jamrial> bah, technically not simplest. the missing instructions in sse2 make it more complex to write :p [18:58:32 CEST] <jamrial> least efficient, then [19:41:20 CEST] <BBB> the C code is, admittedly, incredibly badly written [19:42:33 CEST] <BBB> you can see how ssse3 is hugely helpful compared to sse2, though [19:42:38 CEST] <BBB> which is not surprising [19:42:41 CEST] <BBB> pmulhrsw FTW [19:46:25 CEST] <atomnuker> yeah, but that can easily overflow [19:49:09 CEST] <jamrial> BBB: except in 8_32 it seems. your test shows it's slower than sse2 somehow [19:49:43 CEST] <BBB> oh I Think I know now [19:49:52 CEST] <BBB> the subidct idea is not implemented for sse2? [19:49:58 CEST] <BBB> it probably should be, but I didnt bother [19:50:33 CEST] <BBB> it could shortcut the first idct_1d at least [20:15:31 CEST] <BBB> jamrial: and that then explains why sse2 would be faster for the full one, since its already loaded in cache whereas the ssse3 one isn't [20:32:04 CEST] <cone-567> ffmpeg 03Michael Niedermayer 07master:e879819e7b27: avfilter/vf_uspp: Check for encoding failure [22:29:14 CEST] <bp0> michaelni, do you have concerns about the other hdcd patches? [22:47:22 CEST] <michaelni> bp0, no but do you have testcases for the patches ? [22:48:09 CEST] <bp0> yes, I do [22:48:15 CEST] <michaelni> fate test(s) with small samples would also be usefull [22:50:15 CEST] <michaelni> ideally each bugfix / feature would be covered by a fate test, one fate test could cover many features / bugfixes though [22:51:21 CEST] <michaelni> bp0, can you post patches that add (a) fate test(s) for the patches ? [22:51:58 CEST] <bp0> yeah, I think I could, ... I'll have to look into creating fate tests [22:52:03 CEST] <bp0> I've not done that yet [22:52:40 CEST] <bp0> I have an example false positive for the patch regarding that, and an example with many detectable errors [22:52:49 CEST] <bp0> I suppose they will have to be cut down to :30 or so [22:52:52 CEST] <bp0> right? [22:52:58 CEST] <bp0> for copyright reasons [22:53:43 CEST] <michaelni> smaller is better for diskspace & network bandwidth reasons [22:53:54 CEST] <bp0> for the patch regarding the change in call to hdcd_update_info, maybe a test with known counts? [22:53:59 CEST] <bp0> what would be good for that [22:55:34 CEST] <michaelni> hmm, if nothing in the sample/timestamp/metadata/... output of the filter changes then testing it is harder [22:56:44 CEST] <bp0> oh you are saying the result of the test must be detectable in the output stream, not in the log [22:57:44 CEST] <michaelni> the log could be tested for too in principle [22:59:29 CEST] <bp0> I've personally tested these patches on about 520 CDs, finding about 30 HDCD-encoded among them [22:59:53 CEST] <bp0> so I have examples of false positives, and those with errors [23:00:04 CEST] <bp0> but I know it should be possible to make a test that can be used by others [23:00:50 CEST] <bp0> and I will try [23:01:14 CEST] <michaelni> ok, if you need help, ask [23:01:19 CEST] <bp0> thank you [00:00:00 CEST] --- Thu Jul 14 2016

1 0

ffmpeg.log.20160713
by burek 14 Jul '16

14 Jul '16

[00:01:03 CEST] <Bray90820> jkqxz: How would I check to see what my mp4 files are currently because I thought the default was h.264 [00:01:38 CEST] <jkqxz> ffprobe will tell you. [00:03:15 CEST] <Bray90820> jk looks like h.264 to me [00:03:17 CEST] <Bray90820> Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 720x480 [SAR 8:9 DAR 4:3], 799 kb/s, 29.97 fps, 29.97 tbr, 30k tbn, 59.94 tbc (default) [00:03:23 CEST] <Bray90820> jkqxz: [00:03:45 CEST] Action: Bray90820 can't remember to hit tab [00:05:34 CEST] <jkqxz> With the options and quality you wanted? If that copied, it's probably fine. Otherwise you should play some back to make sure it's sensible. [00:07:05 CEST] <Bray90820> I already know the quality is fine. without pausing it and looking at them side by side i can't really tell a difference and it takes up a third of the space and is iPad compatible so I think i'm all good [00:09:03 CEST] <Bray90820> jkqxz: Unless there is something else you think I should know [00:11:38 CEST] <jkqxz> No, you should be good with that. The output working is the thing that matters. [00:12:43 CEST] <Bray90820> jkqxz: Thanks [01:40:31 CEST] <Jnorthrup> does anyone have a working nnedi weights file for zaranoe? [01:57:20 CEST] <c_14> https://github.com/dubhater/vapoursynth-nnedi3/blob/master/src/nnedi3_weigh… [01:57:23 CEST] <c_14> It's listed in the docs https://ffmpeg.org/ffmpeg-filters.html#nnedi [02:01:54 CEST] <pfelt1> is hevc the only option for ffmpeg for h.265/hevc decoding? [02:04:39 CEST] <Jnorthrup> c_14: for the zaranoe 32 bit build it says unexpected size of file nnedi3_weights.txt [02:05:07 CEST] <Jnorthrup> s/txt/bin/ [02:10:16 CEST] <BNS> Hello. I have just downloaded ffmpeg 3.1.1 [02:10:37 CEST] <BNS> I see in the release notes that OpenEXR support has been expanded [02:11:28 CEST] <BNS> Are there are command line options to deal with multiple channels embedded in the EXR file? [02:57:55 CEST] <c_14> Jnorthrup: the same file should work across all OS 32/64 bit etc [02:59:51 CEST] <c_14> BNS: ffmpeg -h decoder=exr, don't think so though [05:12:40 CEST] <zhrts> hello, I am trying to use a udp multicast h264 stream as an input to ffmpeg. VLC plays it perfectly, but ffplay/ffprobe/mpv all fail with inconsistent errors when I try to open it with them. Is there a way I can find the exact codec settings VLC is using and use them with ffmpeg? [07:38:23 CEST] <strongcoffee> "Even though AAC should generally give better quality than MP3 at the same bitrates, libmp3lame should give better quality output than libfaac." [07:38:44 CEST] <Spring> at the same settings? [07:39:14 CEST] <Spring> so the libfaac encoder is shit? or what :p [07:44:10 CEST] <ritsuka> yes, faac is not a good encoder [09:06:08 CEST] <bp0> Spring, there are many better aac encoders than faac. [09:06:09 CEST] <bp0> http://wiki.hydrogenaud.io/index.php?title=AAC_encoders [09:06:46 CEST] <bp0> if fdk aac is available, you can use that with ffmpeg [09:07:02 CEST] <JEEB> heck, even the internal one is now better than faac [09:07:12 CEST] <JEEB> after atomnuker's improvements [09:07:50 CEST] <bp0> faac is old. the developer went to work at Nero years ago, and Nero AAC is its successor [09:07:57 CEST] <bp0> but now, even Nero is old [09:08:51 CEST] <bp0> ffmpeg native aac is much improved [09:13:06 CEST] <bp0> idk about that that "libvorbis >= libfdk_aac" bit in https://trac.ffmpeg.org/wiki/Encode/HighQualityAudio [09:17:08 CEST] <furq> seems about right to me [09:17:38 CEST] <furq> fdk is decent but there are better aac encoders [09:18:23 CEST] <bp0> by HA testing, only apple's [09:18:37 CEST] <furq> i expect fhg is better as well [09:21:01 CEST] <bp0> I don't know that the two fraunhofer encoders have been tested against each other [09:21:10 CEST] <furq> they haven't to my knowledge [09:21:27 CEST] <furq> but it's probably a safe bet that the closed source floating point encoder is better than the open source fixed point one [09:23:21 CEST] <bp0> is there a way to get a recent version of it? has there been any development since last winamp? [09:23:50 CEST] <bp0> fdk is actively developed, but you may very easily be right [09:23:59 CEST] <furq> i've never used fhg [09:24:15 CEST] <furq> i only ever use aac in mp4 video [09:24:30 CEST] <bp0> ah [09:27:12 CEST] <bp0> I've used the fhg wrapper cli utility, in wine, just to try it out [09:27:44 CEST] <bp0> if I'm going to that trouble I would use apple's instead [09:28:12 CEST] <bp0> i'm pleased that fdk exists, and find it appealing in different ways [09:29:38 CEST] <bp0> I have an archive of FLACed CD's, and then a mirror of all of that as aac that gets synced to devices [09:40:58 CEST] <bp0> i've been playing with the new hdcd filter in ffmpeg, and i realize that fdk will only accept 16-bit input [09:41:11 CEST] <bp0> so I'll have to change to something else to take advantage of that anyway [09:44:09 CEST] <Spring> bp0, cool so the native aac is better [11:41:38 CEST] <k_sze[work]> What does ffmpeg do about corrupted P-frames or B-frames? [11:43:52 CEST] <k_sze[work]> I have these .nut files with corrupted frames in them. When I decode them using ffmpeg, I see that the frame time skips when a corrupted frame occurs. [11:44:35 CEST] <k_sze[work]> I'm not entirely sure if ffmpeg actually skipped over the corrupted frame (which means that the frame time is from the frame right after the corrupted frame) [12:09:50 CEST] <termos> I upgraded libffmpeg and now the AVCodecContext::bit_rate is always 0 for the incoming stream, before this was set to the correct avg value of the input stream. Something deprecated? [13:15:47 CEST] <termos> aha it's been replaced by AVStream::codecpar, no more AVCodecContext in AVStream [13:37:25 CEST] <termos> I still get 0 bitrate when accessing the bit_rate in the new AVStream::codecpar [14:34:21 CEST] <gener1c> hey , i am trying to open an rtp stream with ffmpeg/ffmplay and it keeps on telling me the packet is too short and that codec info couldnt be fetched [14:34:59 CEST] <gener1c> what do i need to do to get more information? vlc autodetected the codecs and information from the rtp multicast stream [14:37:14 CEST] <gener1c> http://paste.ubuntu.com/19268632/ [14:41:43 CEST] <gener1c> looks like its supposed to be mpegts over rtp but i want to analyze the stream to be sure but i have no clue how, ill google it [14:42:41 CEST] <pgorley> have you tried setting -loglevel? [14:46:33 CEST] <eaglex> I'm streaming a rtsp stream from a "DS-7216HGHI-SH / A" to youtube live; all fine and dandy until there is a lot of movement in front of the camera, then I start getting these: http://pastebin.com/znC9V2Fz and then the stream stops -- I'm pretty sure it's the shitty firmware in the Hikvision, but is there anything I can do to work-around this issue? [15:17:04 CEST] <amanda_> Hi! I only get 50fps with ffmpeg -f x11grab at 2560x1440, is this due to slow cpu or something (i5 2500k, 3.3ghz) [15:18:14 CEST] <c_14> hard to say without more information [15:18:19 CEST] <c_14> What's your full commandline? [15:19:39 CEST] <BtbN> The framebuffer has to be pulled back from your GPU, which is usualy one of the slowest things you can ask from them. [15:19:56 CEST] <amanda_> ffmpeg -f x11grab -s 2560x1440 -r 60 -i :0.0 -f alsa -ac 2 -i hw:0 -f flv -ac 2 -ar 48000 -c:v nvenc -g 120 -keyint_min 60 -b:v 4000k -minrate 4000 k -maxrate 4000k -pix_fmt yuv420p -s 2560x1440 -tune film -acodec aac -threads 4 -strict normal -bufsize 4000k test.mkv [15:21:35 CEST] <amanda_> in 1920x1080 it works great, is this due to bad cpu speed? i can oc to 4.6ghz [15:21:49 CEST] <amanda_> (ffmpeg 3.1) [15:21:50 CEST] <furq> do you get 50fps with ffmpeg -f x11grab -s 2560x1440 -r 60 -i :0.0 -c:v rawvideo -f null /dev/null [15:22:09 CEST] <amanda_> no i dont, i get stable 60fps then [15:22:17 CEST] <furq> presumably it's nvenc's fault then [15:22:30 CEST] <BtbN> Is this on a Kepler card? [15:22:35 CEST] <amanda_> it's a gtx 970 [15:22:58 CEST] <debianuser> amanda_: I _guess_ it's due to slow yuv420p conversion. Test it: `ffmpeg -f x11grab -s 2560x1440 -r 60 -i :0.0 -vf format=yuv420p -t 30 -f null -` [15:23:01 CEST] <c_14> don't use -r, use -framerate [15:23:25 CEST] <amanda_> debianuser: that gives me 45fps [15:23:28 CEST] <BtbN> Maxwell should be easily fast enough for 1440p at 60fps [15:24:14 CEST] <BtbN> also, why do you write flv to a .mkv file? [15:24:17 CEST] <debianuser> amanda_: ffmpeg fault then? ;) [15:24:29 CEST] <furq> i assume -f flv is because this is going out over rtmp [15:24:37 CEST] <furq> or will be at some point [15:24:50 CEST] <amanda_> point is to stream to twitch :-) [15:25:10 CEST] <BtbN> your bitrate is too high then, and your keyint too low. [15:25:21 CEST] <amanda_> what should keyint be then? [15:25:30 CEST] <BtbN> ah, nvm. 60 fps, not 30. [15:25:56 CEST] <BtbN> anyway, 1440p, at that low bitrate, at 60fps, encoded with nvenc... poor viewers. [15:26:01 CEST] <furq> 4mbit seems pretty reasonable for 1440p30 [15:26:10 CEST] <furq> and by reasonable i mean not too high, not good [15:26:15 CEST] <amanda_> :() [15:26:23 CEST] <furq> especially with nvenc [15:26:36 CEST] <amanda_> so i end up with 30fps? [15:26:52 CEST] <furq> depends how compressible the source is [15:26:53 CEST] Action: debianuser had a patch for faster yuv420p conversion somewhere... [15:26:55 CEST] <amanda_> sure.. i can stream rawvideo to twitch, but i think they get angry of 1gbit/s [15:27:09 CEST] <furq> zscale should do faster format conversion [15:27:12 CEST] <BtbN> but yeah, x11grab gives you rgb I'd guess. nvenc needs yuv, ideally nv12. And that conversion has to be done on the CPU [15:27:37 CEST] <amanda_> BtbN: so overclocking to 4.6ghz would fix it? [15:27:49 CEST] <BtbN> is your CPU maxed out? [15:27:50 CEST] <furq> if it's going to twitch then it'll presumably need to be yuv420p [15:28:14 CEST] <BtbN> well, it has to be yuv420 [15:28:18 CEST] <furq> and -vf format is singlethreaded iirc so it'd just need to be one core that's maxed [15:28:19 CEST] <BtbN> the format nvenc gets it in isn't part of h264 [15:28:28 CEST] <BtbN> It prefers nv12 [15:28:44 CEST] <amanda_> only one core maxes out [15:28:47 CEST] <amanda_> for x11grab [15:28:49 CEST] <furq> ^ [15:28:55 CEST] <amanda_> rest is offload on nvenc [15:29:14 CEST] <amanda_> so if i do 4.6ghz, will it work? cpu only gets 60c with 4.6ghz [15:29:45 CEST] <BtbN> Twitch already hates you if you go beyond 3.5Mbit/s [15:29:59 CEST] <amanda_> oh ok [15:30:02 CEST] <furq> amanda_: does your build have zscale [15:30:08 CEST] <furq> that should do the format conversion faster [15:30:10 CEST] <amanda_> furq: how do i know? [15:30:18 CEST] <BtbN> In theory, the format conversion could be done via CUDA. [15:30:25 CEST] <BtbN> But someone would have to write a filter for that. [15:30:27 CEST] <furq> -vf zscale,format=yuv420p [15:30:34 CEST] <furq> if that works then you've got zscale [15:30:41 CEST] <furq> or you could look through your configure line but that takes longer [15:31:13 CEST] <amanda_> where in cmdline should -vf zscale,format=yuv420p be included? [15:31:26 CEST] <furq> replace -pix_fmt yuv420p with that [15:31:51 CEST] <furq> also i'm pretty sure -threads 4 and -tune film have no effect with nvenc [15:31:54 CEST] <amanda_> [AVFilterGraph @ 0x12db660] No such filter: 'zscale' [15:31:54 CEST] <debianuser> amanda_: A few years ago I wrote this to get 60fps conversion: http://pastebin.com/QAwE7Tvs It's basically a copypaste from the ff_rgb24toyv12_c() function above with offsets adjusted. I don't think it was ever included in ffmpeg. But if you don't mind building ffmpeg yourself you can try it. :) [15:32:09 CEST] <furq> if you're going to build ffmpeg yourself then just build it with zscale [15:32:19 CEST] <amanda_> where do i get zscale? [15:32:50 CEST] <furq> https://github.com/sekrit-twc/zimg/ [15:33:10 CEST] <furq> be sure to compile it with --enable-x86simd because it's disabled by default for some insane reason [15:33:15 CEST] <furq> or it was last time i tried [15:34:27 CEST] Action: amanda_ builds zimg [15:35:25 CEST] <amanda_> --enable-libzimg enable z.lib, needed for zscale filter [no] [15:35:30 CEST] <amanda_> that's what's needed for ffmpeg? [15:35:34 CEST] <furq> right [15:36:38 CEST] <amanda_> Unknown option "--enable-x86simd". [15:36:40 CEST] <amanda_> ? [15:36:42 CEST] <amanda_> :) [15:36:44 CEST] <amanda_> do i need that [15:36:56 CEST] <furq> that's in zscale's configure [15:37:03 CEST] <furq> if it's not there any more then maybe it's the default now [15:42:02 CEST] <amanda_> Is there any documentation regarding zimg? ffmpeg complains about ERROR: zimg not found using pkg-config [15:42:40 CEST] <furq> tail -50 config.log [15:43:49 CEST] <amanda_> furq: in ~/ffmpeg-3.1/config.log or ~/zimg/config.log? [15:43:56 CEST] <furq> ffmpeg [15:47:04 CEST] <amanda_> pkg-config --exists --print-errors zimg -> package zimg was not found in the pkg-config search path \n Perhaps you should add the directory containing `zimg.pc' \n to the PKG_CONFIG_PATH environment variable \n No package 'zimg' found [15:47:18 CEST] <furq> did you install zimg [15:47:27 CEST] <amanda_> i did, it gave me a .a file [15:47:58 CEST] <furq> make install should have installed a .pc file to your default pkg-config dir [15:48:34 CEST] <amanda_> oh thanks for help [15:49:29 CEST] <amanda_> ok [15:50:41 CEST] <amanda_> furq: the .pc file was placed in /usr/lib/pkgconfig, copied to /usr/lib64/pkgconfig/, seems to work. *compiles ffmpeg-3.1* [15:56:27 CEST] <amanda_> [Parsed_zscale_0 @ 0x26fe700] code 3074: no path between colorspaces/A speed= 0x [15:56:31 CEST] <amanda_> ? [15:56:31 CEST] <amanda_> :D [15:56:56 CEST] <shincodex> ffmpeg actually uses pkg-config correctly? [15:56:57 CEST] <shincodex> meh [15:57:04 CEST] <amanda_> it did, because it compiled :P [15:58:17 CEST] <amanda_> : ] [15:58:21 CEST] <amanda_> thanks for your help! [15:58:23 CEST] <shincodex> yasm compile? [15:58:27 CEST] <shincodex> or debug [15:59:43 CEST] <amanda_> shincodex: how do i know? [15:59:59 CEST] <shincodex> in the slew of text mostly full of CC [16:00:03 CEST] <shincodex> youll see YASM appear [16:00:13 CEST] <shincodex> or open aterminal and type yasm.exe [16:00:19 CEST] <shincodex> err sorry wrong os [16:00:21 CEST] <shincodex> type yasm [16:00:27 CEST] <amanda_> yeah i have yasm [16:01:07 CEST] <shincodex> finally [16:01:15 CEST] <shincodex> you have --disable-debug in your configure [16:01:25 CEST] <shincodex> and dont have rtlo turned on [16:02:13 CEST] <amanda_> what's rtlo? [16:02:24 CEST] <shincodex> run time link optimizations. [16:02:36 CEST] <shincodex> Youll know its on cause it will take like > 10 minutes to link lol [16:03:08 CEST] <xx_pk_xx> Hey guys. I'm doing a little project, and I am trying to encode DVB subtitles. I'm trying to use png images as input format for DVB subtitles. However, maybe I'm doing something wrong - I'm not sure if I'm changing AVFrame to AVSubtitle corectly, because output is never right. Would anyone at least point me into right direction? I could use ASS subtitles as input as well, but from some sources it seems like it's impossible. [16:03:30 CEST] <amanda_> shincodex: so i should do ./configure (100 options) --disable-debug ? [16:04:04 CEST] <amanda_> shincodex: how do i enable rtlo? [16:04:23 CEST] <amanda_> ./configure --help |grep rtlo <- doesnt show anything [16:05:58 CEST] <shincodex> the tests for yasm assembled files run faster than rtlo. [16:06:12 CEST] <shincodex> Yes 100 options --disable-debug as your last statement if you like [16:06:40 CEST] <shincodex> but if you plan to debug ffmpeg stuff like gdb or other tool youll not want that statement [16:07:07 CEST] <amanda_> i will try oc to 4.6ghz [16:07:09 CEST] <amanda_> maybe it works :) [16:07:22 CEST] Action: amanda_ recompiles ffmpeg with --disable-debug [16:07:59 CEST] <debianuser> amanda_: If you won't succeed with zimg/zscale - you can try my patch too. :) Just apply the patch, build as usual and test if fps is any different. :) [16:08:18 CEST] <amanda_> debianuser: zimg works, but i get errors in ffmpeg [16:08:40 CEST] <amanda_> ffmpeg -f x11grab -s 2560x1440 -framerate 60 -i :0.0 -f alsa -ac 2 -i hw:0 -f flv -ac 2 -ar 48000 -c:v nvenc -g 120 -keyint_min 60 -b:v 4000k -minrate 4000k -maxrate 4000k -vf zscale,format=yuv420p -s 2560x1440 -tune film -acodec aac -threads 4 -strict normal -bufsize 4000k test.mkv [16:08:51 CEST] <amanda_> [Parsed_zscale_0 @ 0x1d0b560] code 3074: no path between colorspaces [16:10:36 CEST] <amanda_> brb [16:10:39 CEST] <amanda_> reboot and oc to 4.6ghz [16:10:43 CEST] Action: amanda_ crosses her fingers [16:30:50 CEST] <Kadigan> Hey. Quick question -- I have a 2048x1536 video which contains only 1504x1504 material (from upper left corner). The rest is black. So, naturally, I attempted to "-vf 'crop=1504:1504:0:0'" to get the desired material. [16:31:04 CEST] <Kadigan> However, for some reason (encoding to mp4) this gives me a 3008x1504 video. [16:31:18 CEST] <Kadigan> What am I doing wrong? [16:47:04 CEST] <Kadigan> http://pastie.org/private/b8mfgtq9lylpgecvafrzdw here You go [16:55:02 CEST] <c_14> that should work [16:55:06 CEST] <c_14> It lists the correct size in the output [16:55:14 CEST] <c_14> What are you using to check the dimensions of the output? [17:05:04 CEST] <Kadigan> QuickTime Player, OS X Preview and 360fly Director, which all show wrong sizes and 360fly incorrectly warps the video (it's supposed to be a 360 dome) [17:05:28 CEST] <c_14> try checking with ffprobe [17:07:22 CEST] <Kadigan> http://pastie.org/private/ul9wc0j5rttrtnyhbejahg there. [17:08:21 CEST] <c_14> The size is correct there [17:08:21 CEST] <Kadigan> ffprobe claims it's 1504x1504, but SAR/DAR show 2:1, with a tbc of 60000/1001 [17:08:31 CEST] <Kadigan> which is weird. [17:08:47 CEST] <Kadigan> (it's supposed to be 30000/1001, 1:1 as in square pixels) [17:08:52 CEST] <c_14> the sar/dar is weird [17:09:05 CEST] <c_14> try adding ,setsar=1 after the crop in the conversion command [17:09:08 CEST] <Kadigan> I think I know what's causing this. Can You tell me off the top of Your head [17:09:13 CEST] <Kadigan> ah, yeah, that. Will do. [17:10:02 CEST] <Kadigan> I believe Final Cut Pro X is doing something weird to SAR at export of 2K. [17:10:10 CEST] <Kadigan> Okay, now it's 1504x1504 as it should be, [17:10:17 CEST] <Kadigan> but the original material is squished to half width. [17:10:26 CEST] <Kadigan> (ie. left half content, right half empty) [17:10:35 CEST] <Kadigan> I think I need to tell it to treat it differently at input. [17:10:50 CEST] <c_14> try setsar then crop [17:10:54 CEST] <c_14> maybe also setsar,setdar,crop [17:11:14 CEST] <Kadigan> Trying now. [17:11:54 CEST] <Kadigan> Now I get 1128x1504 :D [17:12:01 CEST] <Kadigan> (SAR and DAR show up as 3:4) [17:12:27 CEST] <lilibox> hi [17:13:51 CEST] <amanda_> OC TO 4.6ghz -> stable 60 fps [17:13:56 CEST] <amanda_> stupid mother board [17:13:56 CEST] <amanda_> :D [17:14:45 CEST] <Kadigan> Okay, I can't continue dealing with this right now. I'll come back tomorrow morning. c_14 - thanks very much for Your help. [17:15:12 CEST] <lilibox> i have got such crazy problem, i am dealing with bunch of 23.97fps footages [17:16:01 CEST] <lilibox> when i put them to adobe premiere main project is also 23.97fps Export media option panel sets export frame rate to 24fps which is imho weird [17:16:27 CEST] <lilibox> but i can chose 23.97fps also [17:17:34 CEST] <lilibox> is possible to tell ffmpeg to every parameter specific frame rate, i mean if i can ad -r 23.97 for input video and audio and also for exporting format [17:17:51 CEST] <lilibox> for output rate to be sure [17:18:10 CEST] <lilibox> thank you for any hints [17:40:35 CEST] <pfelt> afternoon all. i'm trying to decode a 4k multicast stream and the best i can get is about .88x (80% of realtime). i wonder if there is *anything* i can do to get that faster? Here's the relevant information: http://pastebin.com/XFq3FUqh [17:40:52 CEST] <pfelt> (this is git head on fedora 23) [17:48:23 CEST] <furq> pfelt: not unless you have a gtx 950 or better [17:53:11 CEST] <jkqxz> That's bounded by the CPU but not actually using all of it because there isn't enough parallelism in the decoder, presumably? You might do better with a smaller but faster CPU. [17:54:15 CEST] <pfelt> i hoped you guys weren't going to say that [17:55:16 CEST] <pfelt> that's a huge bummer [17:55:53 CEST] <pfelt> are there any tools that could do it? perhaps gstreamer with libdbe (or whatever their hevc library was) [17:57:25 CEST] <jkqxz> How much of the available CPU time are you actually using? [18:00:15 CEST] <pfelt> jkqxz: http://pastebin.com/Y2ZHEUQm [18:00:16 CEST] <jkqxz> (You might want to try setting an explicit -threads value lower than the default it comes up with (which is presumably 40).) [18:00:28 CEST] <pfelt> *lower* than 40? [18:00:44 CEST] <pfelt> i'm actually surprised that i'm able to use this much processor [18:01:28 CEST] <pfelt> i also tried libx265 to see if that would do any better and it's about the same in terms of the .88x speed [18:02:14 CEST] <pfelt> or wait, no. that was zimg when i was still trying to scale [18:02:20 CEST] <pfelt> libx265 is only encode. sorry :( [18:03:10 CEST] <jkqxz> Yeah, start at 1 or 2 and increase it until you get no gain. [18:04:23 CEST] <pfelt> so, why would lowering the number of threads increase performance? [18:04:56 CEST] <pfelt> one thread gets me about .33x [18:05:05 CEST] <pfelt> (oh, but maybe that's your point.) [18:06:24 CEST] <jkqxz> Having too many threads gains nothing if you don't have work for them to do, and can make things slower because of resource contention. [18:07:32 CEST] <pfelt> frame= 3222 fps= 61 q=-0.0 size=N/A time=00:00:55.15 bitrate=N/A speed=1.04x [18:07:39 CEST] <pfelt> that's at 25 threads [18:07:52 CEST] <pfelt> i wonder if i can do the scaling now [18:08:25 CEST] <pfelt> heh. nope. .7 [18:09:13 CEST] <pfelt> ooh. although i could potentially do that in another ffmpeg process [18:28:40 CEST] <Dat> how can I convert mkv's to mp4s? [18:47:09 CEST] <Dat> think i got it [18:48:27 CEST] <kbarry> I thought i read somewhere that it is possible to set a delay for when an input file will "begin" to be played. [18:48:39 CEST] <kbarry> I have a library of 100+ wave files (spoken numbers), and would like to have each file begin playing 1 second apart, so that the resultant stream is "one, two, three..." with 1 second between the beginning of each number tract. [18:48:47 CEST] <Threads> how can i get ffmpeg to stop adding the DURATION tag to video/audio [18:54:11 CEST] <thebombzen> kbarry: the WAV file format doesn't support pregaps. You could try something silly like this: [18:54:36 CEST] <thebombzen> for i in list_of_wav_files; do ffplay $i; sleep 1; done [18:56:49 CEST] <kbarry> thebombzen: That would be silly. [18:57:03 CEST] <kbarry> I am not sure what a pregap is, but I am refering to: [18:58:06 CEST] <kbarry> ffmpeg -i "one.wav" -i "two.wav" ...... -delayplaybackstart 0:1 -delayplaybackstart 1:2 [18:58:58 CEST] <thebombzen> a "pregap" is a tag that says "wait this long before playing this track" [18:59:05 CEST] <thebombzen> the WAV file format doesn't support that [18:59:58 CEST] <thebombzen> you could manually append one second of silence to each file, with ffmpeg -i sample.wav -lavfi 'anullsrc=r=48000:cl=stereo [a1], [a1] atrim=0:1 [a2], [0:a] [a2] concat=n=2:v=0:a=1 [a]' -map '[a]' sample_with_extra_second.wav [19:08:43 CEST] <soulshock> ffmpeg -y -i videofile.mov -pix_fmt yuv420p -filter_complex "[0:v]yadif=0:-1:0,split=3[in1][in2][in3];[in1]scale=1280:720[out1];[in2]scale=854:480[out2];[in3]scale=640:360[out3]" -map [out1] -an -c:v libx264 -b:v 4000k -map [out1] videofile_4000k.mp4 -an -c:v libx264 -b:v 3000k videofile_3000k.mp4 -map [out2] -an -c:v libx264 -b:v 2000k videofile_854_2000k.mp4 -an -c:v libx264 -b:v 1500k videofile_854_1500k.mp4 -map [out3] -an -c: [19:08:43 CEST] <soulshock> gives me [19:08:43 CEST] <soulshock> Output with label 'out1' does not exist in any defined filter graph, or was already used elsewhere. [19:08:52 CEST] <soulshock> isnt it possible to reuse the same output for several bitrates? [19:10:06 CEST] <furq> you've got two -map [out1] [19:10:19 CEST] <furq> also -pix_fmt yuv420p needs to be specified for each output [19:10:36 CEST] <furq> or add format=yuv420p before split [19:10:51 CEST] <soulshock> ahh format=yuv420p that's how it works in filter_complex [19:11:18 CEST] <soulshock> but can I reuse out1? [19:11:30 CEST] <furq> -map [19:11:36 CEST] <furq> -map [out1] -an -c:v libx264 -b:v 4000k -map [out1] videofile_4000k.mp4 [19:11:41 CEST] <furq> get rid of one of those maps [19:13:04 CEST] <furq> i'm not sure if you can reuse it for two different outputs but evidently you can't for the same output [19:13:53 CEST] <furq> you could just use split again [19:15:37 CEST] <soulshock> splitting the output of the scale? [19:21:12 CEST] <soulshock> ah yes that does work [19:21:13 CEST] <soulshock> cool [19:21:26 CEST] <soulshock> I'm daily impressed by the flexibility and power of ffmpeg [19:22:31 CEST] <kbarry> It looks like there is a switch "itsoffset" but it may only apply if the source if video. [19:24:13 CEST] <soulshock> this works: -i videofile.mov -pix_fmt yuv420p -filter_complex "[0:v]yadif=0:-1:0,split=3[in1][in2][in3];[in1]scale=1280:720,split=2[out1_1][out1_2];[in2]scale=854:480,split=2[out2_1][out2_2];[in3]scale=640:360,split=2[out3_1][out3_2]" -map [out1_1] -an -c:v libx264 -b:v 4000k videofile_4000k.mp4 -map [out1_2] -an -c:v libx264 -b:v 3000k videofile_3000k.mp4 -map [out2_1] -an -c:v libx264 -b:v 2000k videofile_854_2000k.mp4 -map [out [19:51:01 CEST] <Threads> dont matter manged to do it [20:05:56 CEST] <anddam> is there a rule of thumb for bitrate for acceptable quality in spoken audio, e.g. a talk-only show like a podcast? [20:06:04 CEST] <anddam> in AAC [20:08:30 CEST] <Threads> anddam 128 maybe or 200 max ? [20:09:33 CEST] <anddam> oh wow, that's what I use for music [20:09:40 CEST] <anddam> I expected way les [20:09:43 CEST] <anddam> less [20:10:00 CEST] <anddam> I'll make a few test and see [20:11:26 CEST] <Kadigan_KSB> Hi. I was in here earlier asking about why ffmpeg has issues with me cropping out a part of the video frame and re-encoding. Seems the original .mov I'm trying to process (2k, 2048x1536) has SAR and DAR set to 2:1. I'm currently doing `ffmpeg -i file -vf "crop=1504:1504:0:0" -vframes 1 -y ./test.png` trying to debug it. [20:12:06 CEST] <kepstin> Kadigan_KSB: what's the difference between what you're expecting, and what you're getting? [20:12:11 CEST] <Kadigan_KSB> Originally the frame came out 3008x1504 instead of 1504x1504. We tried using 'setsar=1' which made it 1504x1504, but the video is now squished. [20:12:35 CEST] <anddam> Threads: ah found this perceptual quality comparison http://i231.photobucket.com/albums/ee166/img_f/figure_1.png [20:12:36 CEST] <kepstin> not sure the sar will be respected properly by the png encoder or viewer [20:12:37 CEST] <Kadigan_KSB> kepstin: I was expecting a video filling the frame (it's a video of res 2048x1536 with 1504x1504 in there, from 0,0, rest is black) [20:12:57 CEST] <Kadigan_KSB> kepstin: I'm getting the same results as I got from encoding mp4, so I assume it's accurate. [20:13:08 CEST] <furq> anddam: there aren't usually useful rules of thumb for bitrate [20:13:15 CEST] <furq> but for voice i'd have thought 32k he-aac would be fine [20:13:26 CEST] <furq> you'll need a build with fdk for he-aac [20:13:37 CEST] <kepstin> Kadigan_KSB: you can't have both SAR and DAR set to 2:1 on a 2048x1536 video, that doesn't make sense [20:13:42 CEST] <kepstin> Kadigan_KSB: what's the actual SAR? [20:13:48 CEST] <furq> also opus is optimised for that use case but i assume you can't use that [20:14:30 CEST] <Kadigan_KSB> kepstin: interesting, on Mac it said sar 2:1 dar 2:1 with tbc 60000/1001, [20:14:35 CEST] <Kadigan_KSB> on windows I get "Stream #0:0(und): Video: prores (apcn / 0x6E637061), yuv422p10le, 2048x1536, 201508 kb/s, SAR 2:1 DAR 8:3, 29.97 fps, 29.97 tbr, 30k tbn, 30k tbc (default)" [20:14:35 CEST] <kepstin> Kadigan_KSB: keep in mind that the crop filter preserves SAR. [20:14:47 CEST] <kepstin> no, that says DAR 8:3 :) [20:14:51 CEST] <kepstin> huh, weird [20:14:56 CEST] <Jnorthrup> do i need amerge to transcode 4 pcm tracks from one asset into 1 4channel pcm track in the output asset? [20:15:23 CEST] <kepstin> Kadigan_KSB: the only way for sar and dar to both be 2:1 is if the input video has same number of pixels height and width [20:15:55 CEST] <furq> Jnorthrup: i think you can use -map_channel [20:16:02 CEST] <Kadigan_KSB> I don't quite remember what SAR and DAR meant. I remember pixel aspect ratio. Screen Aspect Ratio? And the other one? [20:16:04 CEST] <furq> there's no reason not to use amerge if you're transcoding though [20:16:15 CEST] <kepstin> SAR is sample aspect ratio, basically the same as pixel aspect ratio [20:16:30 CEST] <Kadigan_KSB> Ah, so pixel and frame AR basically. [20:16:34 CEST] <Kadigan_KSB> Or so? [20:16:48 CEST] <anddam> furq: that's https://trac.ffmpeg.org/wiki/Encode/AAC#fdk_aac right? [20:16:54 CEST] <kepstin> Kadigan_KSB: if you take your 2048x1536 SAR 2:1 DAR 8:3 video and crop to 1504x1504 then you get a 1504x1504 SAR 2:1 DAR 2:1 video. [20:16:54 CEST] <furq> yes [20:17:06 CEST] <Kadigan_KSB> In any case, I need to get that square into a square :) [20:17:22 CEST] <kepstin> Kadigan_KSB: sounds like you need to scale the video :) [20:17:28 CEST] <Kadigan_KSB> f... [20:17:34 CEST] <furq> -c:a libfdk_aac -profile:a aac_he_v2 -b:a 32k [20:17:49 CEST] <Kadigan_KSB> The whole point of me rendering it in 2k was so that I would avoid scaling (and therefore degrading) the picture. It's already shitty as-is. [20:18:13 CEST] <kepstin> well, you're using 2:1 sar pixels rather than square pixels. if you want square pixel output, you need to scale. [20:18:35 CEST] <Kadigan_KSB> *sigh* [20:18:40 CEST] <kepstin> if you want to squish the video to half the width, then you could do it with a 'setsar' filter instead. [20:18:44 CEST] <furq> can't you just use -vf setsar [20:18:45 CEST] <furq> yeah [20:19:22 CEST] <Kadigan_KSB> kepstin: I'm already having it squished (the original 1504x1504 is now taking up 752x1504) [20:19:27 CEST] <kepstin> crop=1504:1504,setsar=1:1 will give you a 1504x1504 SAR 1:1 DAR 1:1 video that's squished horizontally compared to the original [20:19:34 CEST] <furq> https://ffmpeg.org/ffmpeg-filters.html#setdar_002c-setsar [20:19:35 CEST] <Kadigan_KSB> I need it to expand to full frame. [20:19:41 CEST] <Kadigan_KSB> so what, setsar=1:2? [20:20:01 CEST] <furq> 1:1 sounds right [20:20:51 CEST] <Kadigan_KSB> Well, using setsar=1:2 gave me 752x1504 at least, so that's something [20:21:05 CEST] <Kadigan_KSB> (instead of 1504x1504 with left half used, right half black) [20:21:09 CEST] <kepstin> setsar doesn't change the pixels in the video, just the display size [20:22:15 CEST] <kepstin> so a 1504x1504 video with SAR=1:2 will be scaled during playback to display as if it was a 1504x3008 video (depending on the player) [20:22:50 CEST] <Kadigan_KSB> Meh, so I need to output from another setting altogether. [20:23:51 CEST] <kepstin> basically, if your original video has the correct sar (i.e. a circle displayed on screen is a circle, not an oval), then ffmpeg's "crop" filter will make sure there's no deformations [20:24:08 CEST] <Jnorthrup> C:\Users\Jnorthrup.WOHLER\Desktop\ffmpeg-20160712-df1dc52-win32-static [20:24:10 CEST] <Jnorthrup> >ffmpeg -i SpongeBob_TEST_Paramount.mov -map 0 -c copy -c:v prores_ks -profile:v [20:24:11 CEST] <kepstin> if your original video does not have correct sar, you can use the 'setsar' filter to fix that [20:24:11 CEST] <Jnorthrup> 3 -qscale:v 2 -c:a pcm_s16le -filter_complex "[0:1][0:2][0:3][0:4]amerge" -y [20:24:13 CEST] <Jnorthrup> SBTPFA2.mov [20:24:14 CEST] <Jnorthrup> [AVFilterGraph @ 0477e2c0] Too many inputs specified for the "amerge" filter. [20:24:16 CEST] <Jnorthrup> Error initializing complex filters. [20:24:17 CEST] <Kadigan_KSB> My original file has correct ratios, so things must happen in Final Cut Pro X, because the resultant .mov is borked. [20:24:17 CEST] <Jnorthrup> Invalid argument [20:25:15 CEST] <Kadigan_KSB> At least now I know what I'm dealing with. This is probably something that some course in 2k video covers, and it's probably an "industry standard" or something. Anyway, thanks for the assist! [20:25:32 CEST] <furq> commercial video tool in bad at video shocker [20:25:35 CEST] <kepstin> Jnorthrup: please use a pastebin site next time, but in the mean time, you need to read the docs for the amerge filter: https://www.ffmpeg.org/ffmpeg-filters.html#amerge-1 (hint: it has a parameter that you need to set) [20:26:09 CEST] <Kadigan_KSB> furq: I'm kind of used to having things set behind the scenes, but this was a first I actually saw non-square pixels in output. Like I said, it's probably an "industry" 2K thing. [20:26:35 CEST] <Kadigan_KSB> (ie. I output rect pixels, I process rect pixels, I master BR rect pixels, something like that) [20:26:36 CEST] <kepstin> Kadigan_KSB: it sounds like your original source is a movie encoded for theatrical display? IIRC, some of the 2k formats for that were anamorphic [20:27:02 CEST] <Kadigan_KSB> kepstin: no, my original source is 1:1, it's my NLE that makes it 2:1. [20:27:07 CEST] <Kadigan_KSB> (apparently) [20:27:34 CEST] <Jnorthrup> kepstin thanks that's remarkably similar to a dozen prior attempts based on other docs [20:27:36 CEST] <Kadigan_KSB> The only beef I can have is that I wasn't told this would happen. (which is probably something I'm expected to know if I'm dealing with 2k video) [20:29:14 CEST] <Kadigan_KSB> I assume it may have something to do with stereoscopic 2k video. [20:29:33 CEST] <Kadigan_KSB> In any case, I've a few other formats to try out tomorrow. Thanks! [20:30:40 CEST] <Jnorthrup> now i have 4 channel audio occupying ouptut stream 0. [20:31:02 CEST] <kepstin> Jnorthrup: yep, that's what you told the filter to do. [20:35:34 CEST] <Jnorthrup> kepstin, for the life of me im not seeinf a google result that says reorder ffmpeg output tracks with filter_complex. any hints? [20:36:17 CEST] <kepstin> Jnorthrup: that's because you don't do it with filter complex. [20:36:21 CEST] <kepstin> what's your actual goal? [20:36:30 CEST] <kepstin> then we can work forwards to the correct solution :) [20:37:04 CEST] <kepstin> what do you currently have as a source? what would you like to have as an output? [20:37:16 CEST] <Jnorthrup> converting prores to prores with deinterlace and pcm channels merged 4 into 1, and video t5rack before audio track for the sake of tradition [20:37:33 CEST] <furq> -filter_complex "[0:1][0:2][0:3][0:4]amerge=inputs=4[out]" -map 0:v -map [out] [20:38:04 CEST] <kepstin> Jnorthrup: what exactly do you mean by "pcm channels merged 4 into 1" [20:38:08 CEST] <Jnorthrup> technically prores uses metadata in track 0 [20:38:20 CEST] <kepstin> would you like a mono audio track with the sound from 4 input channels? [20:38:29 CEST] <kepstin> or a single 4-channel audio track, or ... ? [20:38:34 CEST] <Jnorthrup> no 4 channels from 4 pcm sreams [20:39:42 CEST] <kepstin> Jnorthrup: right, then your filter_complex is fine and you just need to add some -map stuff like furq said to set the stream order correctly [20:41:01 CEST] <Jnorthrup> i have that syntax without the [out] but im almost groking it [20:41:32 CEST] <kepstin> in order to use -map on the output of a filter graph, you need to give a label to the output stream. That's what that does. [20:48:20 CEST] <Jnorthrup> furk your addition to the working amerge created 1 pcm per track in addition to the amerge. im using -c copy prior [20:48:53 CEST] <Jnorthrup> changing to -c:d copy does not fix the amerge chaos [20:52:08 CEST] <kepstin> if you're using '-map', only the streams explicitly indicated with the -map will be included in the output file [21:08:20 CEST] <Jnorthrup> i got it from your example. -map v:? -map :s? -map [ao] -map :d? [21:21:36 CEST] <Conder> hello, i have 1920x804 video and i want resize it to 720p. what exactly resolution i should use? [21:23:04 CEST] <thebombzen_> Conder: you can use -vf scale=-1:720, which will resize the height to 720 and the width to whatever you use to keep the aspect ratio [21:23:37 CEST] <thebombzen_> although, 1920x804 is like a widescreen 1920x1080, so you might want to resize it to use -vf scale=1280:-1, which will make the width 1280 and the height will be less than 720 [21:24:18 CEST] <Conder> yea, second option is better. thx [21:24:58 CEST] <thebombzen_> some video formats don't like odd numbers, so you can use 1280:-2 if that's a problem [21:25:36 CEST] <Conder> and one more question, if input have 2 audio streams and i use -c:a copy, it will copy both? [21:29:14 CEST] <furq> Conder: you'd need -map 0:a as well [21:29:20 CEST] <furq> or just -map 0 [21:30:35 CEST] <Conder> alright, ty [21:35:27 CEST] <kepstin> Conder: note that 720p is nominally exactly 1280x720; so if you use scale=1280:-1, you might also have to pad the video to add black bars top and bottom if you need full 720p size [21:36:16 CEST] <Conder> yes i know [21:51:08 CEST] <Conder> hm, am i doing it correct? :D reencode video and copy all audio streams [21:51:10 CEST] <Conder> ffmpeg -i input.mkv -map 0 -vf scale=1280:-1 -c:v libx264 -crf 20 -preset medium output.mkv [21:52:20 CEST] <kepstin> if you want to copy audio streams, add a -c:a copy [21:52:31 CEST] <kepstin> otherwise it'll re-encode them (probably to vorbis, in an mkv file) [21:52:55 CEST] <Conder> but then it copy only 1 audio stream, i want copy 2 [21:54:45 CEST] <c_14> -map 0:a [21:55:15 CEST] <c_14> though you already have -map 0, so it should work [21:59:31 CEST] <Conder> used this [21:59:33 CEST] <Conder> ffmpeg -i input.mkv -map 0:a -vf scale=1280:-1 -c:v libx264 -crf 20 -preset medium output.mkv [21:59:41 CEST] <c_14> that won't have video [21:59:42 CEST] <Conder> and output has only audio streams reencoded to vorbis [21:59:54 CEST] <c_14> -map 0 -c:a copy [22:01:10 CEST] <Conder> ffmpeg -i input.mkv -map 0 -c:a copy -vf scale=1280:-1 -c:v libx264 -crf 20 -preset medium output.mkv [22:01:17 CEST] <c_14> yes [22:01:17 CEST] <Conder> right? [22:01:21 CEST] <Conder> finally, thanks :D [22:01:40 CEST] <c_14> you might want -2 instead of -1 though in the scale [22:01:47 CEST] <c_14> just in case you want to use that on different input [22:02:01 CEST] <c_14> (assuming your video is yuv420p) [22:02:10 CEST] <c_14> That'll make sure the height is always divisible by 2 [22:02:39 CEST] <thebombzen_> also you might also want to set the audio bitrate [22:02:57 CEST] <thebombzen_> the default for mkv is vorbis. you might want to use -c:a opus or set -b:a somehow [22:03:06 CEST] <c_14> thebombzen_: he's using -c:a copy [22:03:18 CEST] <thebombzen_> ah my bad [22:26:33 CEST] <Bray90820> Is there a way to concatenate file with ffmpeg? [22:28:04 CEST] <__jack__> yep, check out the doc [22:28:16 CEST] <__jack__> like that one: https://trac.ffmpeg.org/wiki/Concatenate [00:00:00 CEST] --- Thu Jul 14 2016

1 0

ffmpeg-devel.log.20160712
by burek 13 Jul '16

13 Jul '16

[00:45:38 CEST] <cone-706> ffmpeg 03Rostislav Pehlivanov 07master:80721cc1ff1f: diracdsp: add dequantization SIMD [00:45:39 CEST] <cone-706> ffmpeg 03Rostislav Pehlivanov 07master:bd61f3c6bfb8: diracdsp: add SIMD for the 10 bit version of put_signed_rect_clamped [00:45:40 CEST] <cone-706> ffmpeg 03Rostislav Pehlivanov 07master:17caae72025f: diracdec: simplify golomb parsing and dequantization [00:45:41 CEST] <cone-706> ffmpeg 03Rostislav Pehlivanov 07master:b9c6c5f4539d: diracdec: decode HQ profile slices in rows [00:45:42 CEST] <cone-706> ffmpeg 03Rostislav Pehlivanov 07master:09d89d940635: diractab: expose the maximum quantization index as a macro [00:45:43 CEST] <cone-706> ffmpeg 03Rostislav Pehlivanov 07master:c43485f70765: diracdec: rewrite HQ slice decoding [00:45:44 CEST] <cone-706> ffmpeg 03Rostislav Pehlivanov 07master:0eb0f93109aa: diracdec: implement a LUT-based Golomb code parser [00:45:45 CEST] <cone-706> ffmpeg 03Rostislav Pehlivanov 07master:dcad4677d637: diracdec: do not allocate and free slice parameters every frame [00:45:46 CEST] <cone-706> ffmpeg 03Rostislav Pehlivanov 07master:209456292309: diracdec: do not memset the entire coefficient buffer for HQ pictures [00:55:07 CEST] <llogan> anyone going to VDD? [00:56:31 CEST] <llogan> i've never been to VDD...or Deutschland [01:04:21 CEST] <llogan> 4500mi/7300km away. $1300-1600 USD for the ticket... [01:38:28 CEST] <Compn> llogan : you going this year ? [01:38:52 CEST] <Compn> llogan : btw are you anywhere near fairbanks ? [01:39:49 CEST] <Compn> llogan : pretty sure kierank is going. j-b is going. BBB i think is going... [01:40:00 CEST] <BBB> Im going yes [01:40:09 CEST] <Compn> i might be going [01:40:11 CEST] <cone-706> ffmpeg 03Rostislav Pehlivanov 07master:df1dc52195a5: diracdsp_init: add missing ARCH_X86_64 check [01:40:48 CEST] <Compn> carl went last year. [01:40:53 CEST] <Compn> dont think i've seen iive yet [01:43:00 CEST] <llogan> Compn: i'm considering it, but i'll be travelling already for a while before VDD. I'm in Juneau. [01:44:51 CEST] <llogan> ...also i don't want to miss out on coho fishing this year [01:44:58 CEST] <Compn> i was in alaska once [01:45:03 CEST] <Compn> felt a lot like the midwest :D [01:50:57 CEST] <llogan> but the midwest is so flat [01:51:57 CEST] <rcombs> it's got huge& tracts of land! [03:53:43 CEST] <Timothy_Gu> nevcairiel: yeah I'm gonna try but I'm in China right now and the Internet sucks here (no surprise) [10:47:26 CEST] <mateo`_> michaelni: I can't reproduce the ffprobe segfault you got, 1918 does not fail (except the decoding errors near the end of file, which are already here with master) [10:52:48 CEST] <mateo`_> anyway multiple_stsd.mp4 play better with the merge commit but the audio is out of sync [10:58:07 CEST] <michaelni> 1918 just gives here: [mov,mp4,m4a,3gp,3g2,mj2 @ 0x2f194e0] error reading header [10:59:47 CEST] <michaelni> ffprobe doesnt segfault under valgrind here, but it shows some anomalies [11:01:01 CEST] <michaelni> mateo`_, http://pastebin.com/cVhC14xR [11:01:03 CEST] <mateo`_> I also have some issues with multiple_stsd.mp4 I didn't have on my other machine yesterday [11:01:44 CEST] <mateo`_> I've run ffprobe under valgrind too, but no invalid read :( [11:03:54 CEST] <michaelni> for ticket 1918 this fails ./ffmpeg_g -i http://samples.ffmpeg.org/ffmpeg-bugs/trac/ticket1918/aark15sd_9A62E2FA.mp4 -f null - [11:08:30 CEST] <michaelni> mateo, you are on 5d3f3a421d70890eff73434b93cb55c233731bd1 ? [11:09:43 CEST] <mateo`_> michaelni: i've clean/reset my repo and i'm currently rebuilding [11:11:53 CEST] <mateo`_> michaelni: sorry for the noise, got the sames issues you have reported [11:12:23 CEST] <michaelni> if you want the exact same test2.mov i used: http://www.ffmpeg.org/~michael/test2.mov [11:13:36 CEST] <mateo`_> and libav just segfaults the same way [11:13:53 CEST] <mateo`_> i'll fix the crashes and let you know [11:46:11 CEST] <cone-432> ffmpeg 03Rostislav Pehlivanov 07master:a337cb736142: diracdec: fix #coeffs -> byte conversion [11:52:59 CEST] <robUx4> nevcairiel: do you know if HEVC Main10 hardware decoder in libavcodec is supposed to work ? [11:53:18 CEST] <nevcairiel> of course it works [11:53:40 CEST] <robUx4> I only get AV_PIX_FMT_YUV420P10LE as possible output for the sample I use [11:54:13 CEST] <robUx4> do you have a working sample ? [12:02:03 CEST] <nevcairiel> any hevc main10 sample works for me [12:02:33 CEST] <robUx4> you get something else than AV_PIX_FMT_YUV420P10LE ? [12:02:46 CEST] <robUx4> even for 10 bits H264 I only get that [12:03:04 CEST] <robUx4> so I can't even try to use the hardware decoder [12:03:05 CEST] <nevcairiel> 10-bit h264 is not hardware accelerated [12:03:18 CEST] <nevcairiel> no consumer hardware supports that [12:03:56 CEST] <nevcairiel> hevc main10 is also only supported with dxva2 at this time, nothing else [12:04:03 CEST] <nevcairiel> (ie. windows only) [12:06:17 CEST] <robUx4> yes I'm trying on windows [12:06:36 CEST] <robUx4> (d3d11va but it doesn't propose dxva2_vld anyway) [12:06:55 CEST] <robUx4> here's a Main10 sample where it only proposes software decoding https://drive.google.com/open?id=0ByOSGN3npCKxQkpzYXlibkRuNTA [12:08:12 CEST] <nevcairiel> works for me [12:14:34 CEST] <robUx4> ok, I'll dig in the code to find out why it doesn't for me [12:17:13 CEST] <nevcairiel> ffmpeg -hwaccel dxva2 -threads 1 -i D:\video-x265-10bit-420.mp4 -f null - [12:17:23 CEST] <nevcairiel> uses hwaccel just fine on your sample [12:25:49 CEST] <robUx4> never mind, I was using the other libavcodec [12:25:58 CEST] <robUx4> I see there's more stuff in ffmpeg [13:33:48 CEST] <cone-432> ffmpeg 03Michael Niedermayer 07master:9157ac2f9c86: avcodec/dirac_vlc: Fix mixed declaration and statements [13:56:44 CEST] <Chloe> what would an appropriate AVERROR be for corrupted memory? [14:01:10 CEST] <kierank> corrupted memory? [14:01:13 CEST] <BtbN> Uhm, what? How do you know the memory is corrupted, and why don't prevent it in the first place? [14:04:41 CEST] <Chloe> An external function which will fail if the file in memory is corrupted [14:04:57 CEST] <BtbN> And how did it get corrupted? [14:05:21 CEST] <kierank> AV_INVALIDDATA or whatever it's called [14:11:13 CEST] <Chloe> BtbN: because of a corrupted file at the start [14:13:00 CEST] <BtbN> So not memory corruption, just a broken file. [14:13:22 CEST] <BtbN> Invalid Data seems appropiate [14:13:34 CEST] <Chloe> ok, thanks [16:26:14 CEST] <cone-432> ffmpeg 03James Almer 07master:4acdbb1c6c19: avformat/oggenc: always use the time base stored in the theora header [18:08:09 CEST] <KGB> [13FFV1] 15michaelni pushed 1 new commit to 06master: 02https://git.io/vKWLc [18:08:09 CEST] <KGB> 13FFV1/06master 1441cc0f2 15Michael Niedermayer: FFV1: Add security considerations... [18:48:53 CEST] <bp0> if asked to split a patch into two separate patches, can I resubmit them to ffmpeg-devel in a single email with two attachements, or should I use two emails with one attachment each? [18:51:17 CEST] <JEEB> you should use send-email preferably and have each patch in its own message with [PATCH] => [PATCH v2] [18:51:39 CEST] <JEEB> https://git-scm.com/docs/git-send-email [18:53:34 CEST] <JEEB> after making the two patches the two "topmost" commits in your repo, you can do `mkdir -p patchset && git format-patch HEAD~2 -o patchset` and then edit the .patch files to have [PATCH v2] instead of [PATCH]. then you can use git send-email according to the documentation specifying the directory (you can test it with --dry-run) [18:54:16 CEST] <bp0> If using send-email instead of format-patch, how do you comment to the list without it being in the commit message? [18:54:44 CEST] <nevcairiel> you put your comments below the --- line [18:55:21 CEST] <bp0> alright, thanks [18:55:29 CEST] <JEEB> --compose can also be used to create an introductory/cover message [18:55:43 CEST] <JEEB> which will be [PATCH 0/2] [18:55:59 CEST] <JEEB> anyways, more than one way to skin a cat [18:56:35 CEST] <JEEB> also one thing I notice is that with send-email you generally want to disable CC'ing [18:57:24 CEST] <JEEB> --suppress-cc=all is what I generally use (and set in my gitconfig) [19:32:54 CEST] <cone-476> ffmpeg 03Michael Niedermayer 07master:2408f92678f8: avcodec/dirac_vlc: Fix avutil.h include [21:08:17 CEST] <llogan> yet another user help question sent to ffmpeg-user-admin. they never tell me why they do that when i ask. [21:10:02 CEST] <llogan> i'm not exactly sure where they find it either since I don't know why they would click "ffmpeg-user list run by michael, babtiste, and lou" [21:55:13 CEST] <BBB> jamrial: dont you think we should allow ext lib tests in fate? [21:55:34 CEST] <nevcairiel> its not a terrible idea, but the majority of stations wont test them [21:55:48 CEST] <jamrial> not for a library like vpx. there's no way to do version checks and such [21:56:05 CEST] <jamrial> components that depend on zlib sure, since that's been the same since the dawn of time [21:57:39 CEST] <jamrial> for that matter, that makes me wonder if the minimum version we support for libvpx can decode vp8a and vp9a, or if this patch will need some extra checks [22:13:19 CEST] <BBB> jamrial: I think thats worth asking :) [22:15:19 CEST] <jamrial> or could be a good excuse to bump the minimum required version and get rid of all the ifdeffery :p [22:17:26 CEST] <BBB> or that [22:17:29 CEST] <BBB> many options... [22:17:48 CEST] <BBB> someone could also make the internal vp8/9 decoders support alpha [22:17:51 CEST] <BBB> *hint* *hint* [22:18:03 CEST] <BBB> I can mentor that if theres some outreachy student up for that [22:52:00 CEST] <nevcairiel> doesnt that literally require firing up a second decoder with a one-plane bitstream? the vpx patch makes it look like that [22:53:29 CEST] <BBB> yes [22:53:42 CEST] <BBB> but the decoder can be fired up internally so its hidden from the outside [22:53:52 CEST] <BBB> it looks fairly hacky but whatever Im not in charge of bitstream design over there [22:54:01 CEST] <BBB> ( :-p ) [22:54:21 CEST] <nevcairiel> its not even a bitstream design thing, its like the webm guys solved it without the vp8/9 people wanting to cooperate =p [22:54:45 CEST] <nevcairiel> you dont want to give us a true alpha mode? we'll encode two bitstreams, ha! [22:54:56 CEST] <BBB> dunno, not involved in the discussions [22:55:00 CEST] <BBB> I think they did this in vp6 also [22:55:08 CEST] <BBB> (from what I remember) [22:58:01 CEST] <rcombs> ProRes 4444 it is, then [23:06:15 CEST] <kierank> nevcairiel: dct on alpha is pointless [23:06:24 CEST] <kierank> the whole point of alpha is to have nice sharp edges [23:27:34 CEST] <cone-476> ffmpeg 03Burt P 07master:0d8caeb41e50: af_hdcd: integrate() renamed hdcd_integrate() to be consistent with the other function names [23:27:35 CEST] <cone-476> ffmpeg 03Burt P 07master:7af44ce2f7c2: af_hdcd: don't log full HDCD stats if HDCD was not detected [23:39:37 CEST] <jamrial> speaking of vp8, just sent a patch to add vp8 muxing support for ogg [23:39:41 CEST] <jamrial> for all you ogg lovers :p [23:41:08 CEST] <TD-Linux> did we ever write a spec for that even [23:41:43 CEST] <jamrial> there is one [23:41:49 CEST] <BBB> omg ;( [23:41:52 CEST] <BBB> really [23:41:57 CEST] <jamrial> https://people.freedesktop.org/~slomo/ogg-vp8/ogg-vp8.pdf [23:41:58 CEST] <BBB> can we all please agree that ogg should just die [23:42:13 CEST] <BBB> ogg is one of these horrible ideas that will just not die [23:42:16 CEST] <BBB> I dont understand why [23:42:30 CEST] <BBB> they keep trying to patchwork around its brokenness even though there is matroska [23:42:37 CEST] <BBB> just use matroska and let ogg die [23:42:40 CEST] <BBB> pretty please? [23:42:43 CEST] <jamrial> BBB: xiph keeps it alive with opus and now daala [23:43:36 CEST] <BBB> so sad [23:44:01 CEST] <atomnuker> btw, since no one's saying a thing about the ffserver news, maybe llogan can push it? [23:44:53 CEST] <BBB> jamrial: I also realize something else thats rather painful about avx2 (although its not strictly avx2s fault) [23:45:21 CEST] <jamrial> what? [23:45:35 CEST] <BBB> jamrial: most butterfly idct implementations that I know of do the outermost interleave pass at the end, which means that you load pairs of n, and num-1-n coefficients into memory for a sumsub [23:45:37 CEST] <BBB> e.g. 0 and 31 [23:45:41 CEST] <BBB> and then 1 and 30 [23:45:41 CEST] <BBB> etc. [23:46:05 CEST] <BBB> but the subsequent transpose needs to have pairs of 16 and you have only 16 regs so you end up doing 16 full emmory loads and stores [23:46:32 CEST] <BBB> for sse2 this is hidden because you have 16 regs but only need 8 for the transpose, so you can transpose the first and last 8 in the 16 regs without stores/loads [23:46:40 CEST] <BBB> and then the middle 2 pairs of 8 [23:46:46 CEST] <BBB> for avx2 this is not possible [23:46:47 CEST] <jamrial> BBB: avx512 would solve that with its 32 simd regs [23:46:55 CEST] <BBB> yeah when is that coming out again? [23:46:57 CEST] <jamrial> but yeah, avx2 not so much [23:47:06 CEST] <jamrial> soon(tm) [23:47:07 CEST] <BBB> I think releasing avx2 w/o 32 regs was kind of a mistake maybe [23:47:17 CEST] <BBB> also doesnt avx512 introduct 64byte zmm regs? [23:47:21 CEST] <BBB> or is that the one after? [23:47:24 CEST] <jamrial> yeah [23:47:30 CEST] <BBB> I want 64 zmm regs [23:47:33 CEST] <BBB> 32 ymm regs [23:47:34 CEST] <BBB> 16 xmm regs [23:47:44 CEST] <BBB> [etc] [23:48:24 CEST] <BBB> 16 ymm regs is slightly painful [23:48:46 CEST] <jamrial> well, with avx512 you'll get 32 zmm, 32 ymm (mapping to low bits of zmm regs) and 32 xmm (same), just like avx does with ymm/xmm, so that's something [23:48:52 CEST] <nevcairiel> it seems odd that the size and the count of regs would double, might as well quadruple the size of the step before it and use the same hardware :D [23:49:11 CEST] <nevcairiel> ymm uses silly lane stuff anyway, so why not expose those as 32 xmm regs [23:51:47 CEST] <BBB> 32 xmm is kind of much [23:51:50 CEST] <BBB> I mean sure why not [23:51:55 CEST] <BBB> but Im not sure anyone will ever need them [23:51:56 CEST] <BBB> ? [23:52:32 CEST] <nevcairiel> why do you need 32 ymm but not 32 xmm [23:53:21 CEST] <nevcairiel> couldnt you implement the same things that need 32 ymm regs in sse? [23:53:35 CEST] <nevcairiel> say, like a 32x32 idct [23:54:13 CEST] <BBB> nevcairiel: well, sure [23:54:28 CEST] <BBB> nevcairiel: but that would only work on machines that have zmm regs [23:54:42 CEST] <BBB> nevcairiel: so then you wouldnt need the avx2 32reg or sse2 32reg version on that machine anyway [23:54:56 CEST] <BBB> nevcairiel: do the deployment potential of a function like that seems fairly minor [23:54:59 CEST] <nevcairiel> well yes, but maybe you dont need 512 bit registers [23:55:07 CEST] <BBB> for 32x32? yes you do [23:55:15 CEST] <BBB> 32 words = 64 byte = 512 bits [23:55:39 CEST] <BBB> likewise, 16x16 = 16 words = 32 byte = 256 bits = ymm regs to do it stackless/loopless [23:55:51 CEST] <BBB> so the 16x16 is in fact stack- and loopless on avx2 [23:56:01 CEST] <BBB> but the 32x32 avx2 will be stack/loop based [23:56:56 CEST] <BBB> (i.e. you do a 16x32 idct_1d and transpose results to stack (2x) and then another 16x32 idct_1d 2x in the other direction, which you residual_add to the prediction [23:57:11 CEST] <nevcairiel> i'm really more disappointed that 32-bit only has 8 xmm regs, did they just plan badly or did the instruction coding not allow for more [23:57:29 CEST] <BBB> who cares about 32bit :-p [23:57:33 CEST] <jamrial> seriously, avx512 with 8 zmm regs will be hilarious [23:58:08 CEST] <BBB> does anyone here write avx2 that works on 32bit? [23:58:12 CEST] <BBB> all my avx2 is 64bit only [23:58:18 CEST] <jamrial> i did [23:58:20 CEST] <BBB> Im too lazy to care about 32bit at this point [23:58:21 CEST] <nevcairiel> i also wonder why they didnt fix it in vex coding or somethng [23:58:27 CEST] <nevcairiel> must be some deep technical reason why it just doesnt work [23:58:31 CEST] <jamrial> but because it was relatively easy [23:59:01 CEST] <jamrial> i had to work around the fact i had 7 gprs more often than the fact i had 8 simd regs [23:59:35 CEST] <BBB> oh right theres that also [23:59:38 CEST] <nevcairiel> BBB: unfortunately many of my users still use 32-bit, mostly for reasons of the host app that uses avcodec is bound by legacy reasons, like a plugin landscape and shit like that [23:59:54 CEST] <BBB> nevcairiel: so what kind of work do you do? :-p [00:00:00 CEST] --- Wed Jul 13 2016

1 0

ffmpeg.log.20160712
by burek 13 Jul '16

13 Jul '16

[04:25:16 CEST] <Bray90820> "pts has no value" what does that warning mean [11:51:37 CEST] <pskosinski> Guys, I'm trying to record desktop with recordmydesktop, it's using ffmpeg. I have a problem with parts of my wallpaper getting visible sometimes, like that: https://imgur.com/0XT7jnf [11:52:41 CEST] <pskosinski> I'm getting many artefacts if anything on my screen changes position, when I scroll webpage or even move mouse pointer [12:20:44 CEST] <lilis> Hi guys, I want to combine two webcam streams and output it as another webcam stream. Is it possible to do? [12:21:07 CEST] <lilis> I want to do something like 360 video using ffmpeg [12:23:22 CEST] <Fyr> guys, I have a very blurry text in a movie. how do I make it sharpened? [12:24:15 CEST] <Fyr> I have a screenshot of a video that I'm trying to magnify and recognize label written on the wall. [13:02:49 CEST] <lilis> I figured out how to place two streams onto a single output file, but I can't seem to get it to output to /dev/video2 (through v4l2loopback) [13:02:55 CEST] <lilis> Does anybody know why? [13:05:31 CEST] <lilis> This is one of the errors: [v4l2 @ 0x562187227920] Unknown V4L2 pixel format equivalent for yuva420p [13:05:58 CEST] <lilis> Actual command is [13:06:00 CEST] <lilis> ffmpeg -re -i /dev/video0 -i /dev/video1 -filter_complex "nullsrc=size=640x480 [base]; [0:v] setpts=PTS-STARTPTS, scale=320x240 [upperleft]; [1:v] setpts=PTS-STARTPTS, scale=320x240 [upperright]; [base][upperleft] overlay=shortest=1 [tmp1]; [tmp1][upperright] overlay=shortest=1:x=320" -f v4l2 /dev/video2 [13:19:14 CEST] <whald> lilis, it seems you pipe is generating frames w/ an alpha channel, which is v4l sink cannot accept. the acceptable pix fmts are listed here: https://www.ffmpeg.org/doxygen/3.0/v4l2-common_8c_source.html#l00064 [13:20:16 CEST] <whald> lilis, i'm not too familiar with the ffmpeg command line so i'm afraid i don't know where, but probably you'll need to throw in a -pix_fmt=<...> somewhere. [13:21:41 CEST] <lilis> whald, thanks! I'll try that out [13:22:27 CEST] <furq> add ;format=yuv420 to the end of filter_complex [13:22:32 CEST] <furq> yuv420p [13:23:09 CEST] <furq> ,format=yuv420p [13:23:11 CEST] <furq> i got there in the end [13:23:54 CEST] <lilis> D: [13:24:00 CEST] <lilis> It worked! [13:24:05 CEST] <lilis> Thanks [13:27:03 CEST] <lilis> Is there a way to reduce lag between the streams? [13:27:31 CEST] <lilis> It _is_ different webcams, so I might be asking too much [13:28:10 CEST] <whald> lilis, are you talking about some offset between the different inputs or total input->output latency? [13:28:55 CEST] <lilis> Different inputs I believe [13:29:14 CEST] <lilis> It shows in the combined output [13:29:39 CEST] <lilis> brb [14:14:21 CEST] <hdare> is there any web player supporting the stream from ffmeg trough icecast ? [14:14:46 CEST] <hdare> i mean video stream [14:17:19 CEST] <BtbN> Nothing that's not flash, and even flash is quite restricted. [14:17:42 CEST] <BtbN> Use HLS or DASH for streaming to a Browser. [14:18:57 CEST] <furq> hdare: you should just be able to point an html5 video tag at it [14:19:09 CEST] <furq> whether or not it'll work well is another question [14:27:04 CEST] <__jack__> hls on browser, hum [14:27:21 CEST] <__jack__> won't work, unless you're using flash, or unless "browser" = "safari" [14:27:31 CEST] <BtbN> hls.js works fine in every browser. [14:27:41 CEST] <BtbN> And HLS is less insane than DASH [14:29:01 CEST] <ritsuka> hls works on edge too, and some version of the android browser [14:29:15 CEST] <__jack__> consumes a lot a CPU, no ? [14:29:21 CEST] <BtbN> It works in every Browser that supports MSE, so, all of them. [14:29:35 CEST] <__jack__> I just tested http://dailymotion.github.io/hls.js/demo/, doh my poor PC [14:29:56 CEST] <BtbN> So your PC doesn't support hardware decoding? [14:30:27 CEST] <BtbN> Firefox plays this while using 6% CPU [14:30:35 CEST] <BtbN> while it's using 5% when doing nothing... [14:30:52 CEST] <__jack__> dunno, you may be right [14:31:06 CEST] <__jack__> it worth looking at that code, thanks for the tip [14:35:08 CEST] <hdare> with video.js and medialement the playback of hls works well and with clappr too [14:35:52 CEST] <hdare> but how to stream in hLS mode ? [14:36:11 CEST] <hdare> trough icecast [14:36:28 CEST] <BtbN> I don't think icecast supports HLS. [14:36:49 CEST] <hdare> it s what i think too [14:37:54 CEST] <hdare> html5 accept only audio stream but no video [14:38:23 CEST] <BtbN> Browsers support only mp4 for video [14:38:28 CEST] <BtbN> And mp4 can't be streamed. [14:38:37 CEST] <BtbN> So it's cut into short segments. [14:40:05 CEST] <hdare> it s must be streamed to an rtmp server to redirect stream to hls format [14:42:52 CEST] <viric> does firefox have hardware decoding? [14:43:06 CEST] <BtbN> for h264 and stuff, yes [14:43:11 CEST] <BtbN> Not for VP9 i think [14:43:12 CEST] <furq> hdare: ffmpeg has an hls muxer [14:43:14 CEST] <viric> I thought it did not [14:43:26 CEST] <viric> (on GNU/Linux) [14:43:32 CEST] <furq> if you don't need to push to a remote server then you can just serve the fragments with any httpd [14:43:43 CEST] <BtbN> viric, check about:support for hardware decoder status. [14:43:58 CEST] <BtbN> for me it says "Hardware H264 Decoding: Yes; Using D3D11 API" [14:44:10 CEST] <furq> and yeah hls.js doesn't use much cpu in my experience [14:44:17 CEST] <viric> Supports Hardware H264 Decoding No [14:44:24 CEST] <viric> BtbN: that's Windows [14:44:37 CEST] <BtbN> I'm quite sure it supports hw accell on Linux as well [14:44:43 CEST] <BtbN> they recently switched to ffmpeg for that iirc [14:44:48 CEST] <furq> No; D3D9 blacklisted with DLL atiumd64.dll (9.14.10.1128) [14:44:49 CEST] <furq> huh [14:44:54 CEST] <furq> you'd think i'd have noticed [14:45:17 CEST] <BtbN> And before that it was done via gstreamer [14:45:33 CEST] <BtbN> So if you install the neccesary gstreamer plugins, it can use whatever hwaccel gst supports. [14:46:34 CEST] <viric> BtbN: hm. I remember a bugzilla ticket about not being available at all. [14:49:11 CEST] <viric> I see it builds something from firefox-47.0.1/media/ffvpx/libavcodec/x86/... [14:53:36 CEST] <viric> h264 being inside 'ffvpx' sounds strange [14:55:48 CEST] <BtbN> they only use lavc, and only for its VP8/9 decoder [14:56:15 CEST] <viric> BtbN: https://hg.mozilla.org/mozilla-central/file/tip/media/ffvpx/README_MOZILLA [14:56:25 CEST] <viric> --disable-vaapi --disable-vdpau [14:57:59 CEST] <viric> vp8 and vp9, ok [14:58:09 CEST] <viric> BtbN: and where is the h264 part? [14:58:25 CEST] <DHE> aside, isn't --disable-everything redundant with all the other --disable-* commands? [14:58:35 CEST] <BtbN> it should be [14:59:22 CEST] <furq> other way round, surely [15:02:43 CEST] <viric> BtbN: what about that Cisco openh264? [15:02:56 CEST] <DHE> isn't that baseline only? [15:03:02 CEST] <BtbN> It's an encoder, for baseline only. [15:03:05 CEST] <viric> ahh. [15:03:10 CEST] <DHE> oh, I thought it was a decoder as well [15:03:19 CEST] <furq> it does decoding as well [15:03:21 CEST] <furq> but presumably only baseline [15:03:24 CEST] <viric> then where is the decoder source of h264 in the firefox tree?I Can't find it [15:03:35 CEST] <BtbN> viric, firefox does not contain any h264 [15:03:39 CEST] <viric> mh [15:03:42 CEST] <viric> how does it decode it? [15:03:46 CEST] <BtbN> that would require them to pay patent feees [15:03:47 CEST] <BtbN> it doesn't [15:03:54 CEST] <viric> how does it play h264? [15:03:54 CEST] <BtbN> it relies on external stuff doing it [15:04:01 CEST] <viric> like what? You said that it uses ffmpeg [15:04:06 CEST] <BtbN> It does [15:04:12 CEST] <BtbN> Installed via your package manager. [15:04:21 CEST] <viric> auhm [15:04:27 CEST] <BtbN> And on windows it uses the Microsoft Media APIs [15:04:45 CEST] <BtbN> Might also still use gst, no idea how far their migration got. [15:05:01 CEST] <viric> I fail to find it in the recipe [15:05:59 CEST] <BtbN> I remember some Firefox developer asking in ffmpeg-devel. They seem to have come up with a complex runtime-dynamic-loader for libav*, that detect the version and uses the correct ABI at runtime. [15:06:14 CEST] <viric> ahhh [15:06:22 CEST] <viric> and how does it check for hardware acceleration? [15:06:29 CEST] <BtbN> No idea [15:14:57 CEST] <viric> I can't find it [15:18:38 CEST] <viric> BtbN: how would my system ffmpeg decide to decode accelerated or not accelerated? [15:19:05 CEST] <BtbN> The library user has to decide that. ffmpeg hwaccels don't just work. [15:19:13 CEST] <viric> Aha [15:19:15 CEST] <BtbN> It needs quite a bit of work from the user. [15:19:29 CEST] <viric> then it does not do anything I'd say [15:20:03 CEST] <viric> https://hg.mozilla.org/mozilla-central/file/tip/dom/media/platforms/ffmpeg [15:23:43 CEST] <viric> it sets up decoder threads... https://hg.mozilla.org/mozilla-central/file/tip/dom/media/platforms/ffmpeg/… [15:23:46 CEST] <viric> no accel [15:24:26 CEST] <viric> BtbN: how is it accelerated on windows?Doesn't it use ffmpeg also? [15:24:45 CEST] <BtbN> No, it uses the Windows API for decoding and playing stuff [15:24:55 CEST] <viric> the WMF thing [15:25:09 CEST] <viric> that *yes* has things about hw acceleration. [15:25:48 CEST] <viric> also android, also omx (rpi), .. not ffmpeg, there. [15:28:03 CEST] <BtbN> It will use whatever is available on the platform it's running on. [15:28:13 CEST] <BtbN> Linux doesn't have a native API, so it uses ffmpeg or gst there. [15:34:02 CEST] <viric> BtbN: but it could set up hwaccel in ffmpeg [15:34:26 CEST] <viric> in any case, it matches what I knew about it for more than one year: no support at all for hw acceleration in GNU/Linux [15:42:33 CEST] <DHE> in theory you can use vdpau under linux, sure. but there's usually a trick to playback, like requiring the matching vdpau video-output driver for rendering. it may not be easy, or possible, to get the raw decoded video frames out [15:42:51 CEST] <DHE> similar for other hwaccel methods. which is why it doesn't "just work" like the software decoders [15:45:44 CEST] <BtbN> well, you can always just pull the frames back to system memory and continue as usual [15:45:54 CEST] <BtbN> Just like ffmpeg.c does it [15:46:11 CEST] <pgorley> doesn't that hunder performance though? [15:46:35 CEST] <pgorley> *hinder, sorry [15:46:47 CEST] <BtbN> sure, but still faster than decoding in software. [15:47:02 CEST] <pgorley> true [17:18:50 CEST] Action: Fyr slaps durandal_1707 around a bit with a large trout [17:59:48 CEST] <durandal_1707> Fyr: ? [18:00:28 CEST] <Fyr> you wrote about ffmpeg-mt a few months ago. have it been merged with the master branch? [18:01:15 CEST] <durandal_1707> you mean mt audio encoding? [18:01:27 CEST] <Fyr> yes [18:04:19 CEST] <k_sze> Is there a streaming format that can readily embed a real wall clock timestamp with each frame, as metadata? [18:06:28 CEST] <Fyr> I would dig into TS. [18:07:47 CEST] <k_sze> I see that ffmpeg has a use_wallclock_as_timestamps option, and the nut container format also has a -syncpoints flag. [18:07:55 CEST] <k_sze> I wonder how usable those are. [18:08:32 CEST] <k_sze> I would be somewhat surprised if all container formats allow wallclock timestamps. [18:08:32 CEST] <Fyr> durandal_1707, still, the 16-core Intel Xeon cluster converts FLAC as fast as my Banana Pi. [18:08:48 CEST] <Fyr> that thing makes me sad. [18:09:17 CEST] <k_sze> FLAC encoding doesn't multithread readily, IIRC. [18:10:04 CEST] <Fyr> k_sze, wrong, a file can be divided into pieces and then merged into one. [18:10:12 CEST] <Fyr> each piece can be converted with FFMPEG. [18:10:20 CEST] <Fyr> a simple bash wrapper can do that. [18:10:49 CEST] <k_sze> And have you looked at your CPU usage? [18:10:56 CEST] <k_sze> Maybe the CPUs are not the bottleneck. [18:10:58 CEST] <Fyr> unfortunately, the developers don't want the users to be happy. [18:11:29 CEST] <k_sze> Maybe it's the I/O, if you don't have fast enough storage to saturate 16 cores. [18:12:26 CEST] <Fyr> 16-CORE INTEL XEON CLUSTER DOESN'T HAVE FAST ENOUGH STORAGE? [18:12:36 CEST] <Fyr> how about my Banana Pi? [18:12:53 CEST] <k_sze> Is your Xeon cluster running on SSD? [18:13:29 CEST] <Fyr> it has an SSD, but for users it uses an HDD. [18:13:50 CEST] <k_sze> And your audio file is on the spinning HDD. [18:14:02 CEST] <Fyr> I/O has never been a problem on the cluster. [18:14:07 CEST] <k_sze> measure [18:14:15 CEST] <Fyr> but, the Banana Pi seems to have one. [18:14:21 CEST] <catchin> is this ffmpeg at work? https://www.instagram.com/p/BHwDgD0AU4E/ [18:20:03 CEST] <durandal_1707> Fyr: I wrote patch ... [18:21:08 CEST] <kepstin> as far as mpeg-ts goes, the timestamp format only holds ~26.5 hours iirc; they're designed to be able to hold a 24-hour realtime timestamp, wrapping each day. If you want to also store the date, it would have to be in metadata - which i'm sure there's a way to do, I just don't know it :) [18:22:56 CEST] <k_sze> Fyr: how's the build quality of the banana pi? What model do you have btw. [18:23:07 CEST] <Fyr> Banana Pro [18:23:18 CEST] <k_sze> I have a Raspberry Pi 3 and an incoming Raspberry Pi Zero. [18:23:19 CEST] <Fyr> I'm pretty much satisfied with the purchase. [18:23:42 CEST] <k_sze> Maybe I'll try the Banana Pi sometime later. [18:23:55 CEST] <Fyr> if you try to use it hard, you should buy something more reliable such as Intel NUC, ZBOX or whatever else. [18:24:09 CEST] <k_sze> Fyr: can't see a Banana Pro listed here: http://www.banana-pi.com/eindex.asp [18:24:16 CEST] <k_sze> "hard"? [18:24:26 CEST] <Fyr> Banana Pi may be hot under full load of the CPU. [18:24:31 CEST] <k_sze> ah [18:24:47 CEST] <Fyr> k_sze, https://en.wikipedia.org/wiki/Banana_Pi#Banana_Pi_Pro [18:26:57 CEST] <k_sze> ah [18:35:09 CEST] <friki2015> Some days ago i asked here for help about rtmp ingest to Akamai. Finaly i got it working using ffmpeg rtmp implementation (non librtmp). ffmpeg ... -f flv "rtmp://USER:PASSWD@p.epCPCODE.i.akamaientrypoint.net/EntryPoint/STREAMNAME_ANGLE_BITRATE@STREAMID" [18:36:16 CEST] <friki2015> just take care about avoid open new connection simultaneously (akamai reject logins in this situation, as they have internaly documented) [18:52:49 CEST] <kbarry> I'm trying to grab only the audio from a single programs in an HLS stream. I'm a little confused with the syntax. I believe I need to be using -map, ie, 0:p:2 should refer to the streams from the program "1", but I am unable to select just the audio stream from this particular program. [18:56:24 CEST] <furq> Fyr: you could use flaccl if your encoding box has a decent gpu [18:56:30 CEST] <__jack__> p ? use -map 0:a:0 [18:56:32 CEST] <furq> it's .net though which sucks [18:56:44 CEST] <__jack__> (as in : first input, first audio stream) [18:56:53 CEST] <Fyr> furq, flacCL doesn't have descent compression. [18:57:08 CEST] <Fyr> however, it's 60x times faster, than the cluster. [18:57:20 CEST] <furq> er [18:57:25 CEST] <furq> neither of those tally with my experience [18:57:38 CEST] <furq> although i haven't used it on a decent gpu [18:58:31 CEST] <furq> last time i tried it was roughly on a par with flac -8 and ~3x as fast [18:59:26 CEST] <Fyr> FFMPEG has the best FLAC compression with -multi-dim on. [18:59:35 CEST] <DHE> kbarry: 0:a means all audio streams (there should only be one). you can also use ffprobe to report stats on the stream [19:00:04 CEST] <kbarry> DHE: Well, ther eis one other problem. [19:00:31 CEST] <kbarry> This is a TV signal. I'm trying to strip out the audio only, but it appears that the signal configurations is changing drastically when commercials appear [19:01:12 CEST] <kbarry> http://pastebin.com/S0eqJWN3 during News Broadcast [19:03:06 CEST] <DHE> oh... yeah that's always a pain when the PMT changes [19:10:04 CEST] <friki2015> kbarry_: -map p:<program_id>:<audio_stream_index> or just -map p:5 -vn [19:11:35 CEST] <kbarry_> well, the stream configuration keeps changing, [19:12:37 CEST] <kbarry_> between "commercial" and "non-commercial" periods. [19:13:08 CEST] <friki2015> i never faced this problem, sorry [19:16:01 CEST] <friki2015> could be you can fix it after recording all audios (detect commercial timeranges and remove them -ss start -to firstcommercial) [19:48:14 CEST] <maziar> hi i want to know everything about converting MP4 to m3u8 and other formats, what should i read? i need something like book, pdf and etc. [19:52:53 CEST] <Fyr> maziar, some short manuals on "how to convert MP4 into MP3" or "how to remux an MP4". [19:53:03 CEST] <Fyr> there are plenty of them. [19:53:46 CEST] <maziar> Fyr, i need a perfect document perfect and full [19:53:49 CEST] <Jnorthrup> what is the mp4 encoding mojo to get "422 hq" ? [19:54:34 CEST] <Fyr> maziar, https://wiki.archlinux.org/index.php/FFmpeg [19:55:27 CEST] <Fyr> that one is good too: [19:55:28 CEST] <Fyr> http://trac.ffmpeg.org/ [19:55:41 CEST] <maziar> fry how about this : https://www.ffmpeg.org/documentation.html [19:56:02 CEST] <maziar> Fyr, which part should i read fot that ? [19:56:05 CEST] <furq> Jnorthrup: isn't that prores [19:56:27 CEST] <furq> maziar: https://www.ffmpeg.org/ffmpeg-formats.html#hls-1 [20:08:16 CEST] <Fyr> maziar, every. [20:08:43 CEST] <maziar> Fyr, sorry, every what ? [20:08:51 CEST] <Fyr> every part [20:09:02 CEST] <Fyr> you needed full manual. [20:09:56 CEST] <Fyr> those ones cover conversion broadly. [20:10:17 CEST] <furq> is there anyone in the entire world who's read all of the ffmpeg docs [20:10:37 CEST] <furq> not including the libav* docs [20:14:37 CEST] <maziar> Fyr, for full manual, what should i do ? [20:15:54 CEST] <llogan> man ffmpeg-full [20:16:12 CEST] <llogan> *man ffmpeg-all [20:19:08 CEST] <wiak> the windows compile guide for av1 is a whreck so i made a better one https://nwgat.ninja/compile-av1-using-msys2-mingw/ [20:19:09 CEST] <wiak> ^^ [21:47:11 CEST] <Mista-D> "-xerror" exits when using "-f concat" with in/out timestamps, at "[ac3 @ ...] frame sync error". Anyway to set concat to not break AC3 frame sync or force xerror to ignore the AC3 error please? [21:51:20 CEST] <Mista-D> http://pastebin.ca/3657801 [23:31:50 CEST] <Bray90820> Can someone tell me what this error means? [23:31:50 CEST] <Bray90820> http://pastebin.com/raw/qnGDmMk4 [23:36:32 CEST] <jkqxz> "codec not currently supported in container"? It means that the codec is not currently supported in the container. [23:36:49 CEST] <Bray90820> No shit [23:36:51 CEST] <jkqxz> (Try writing to a different container instead (Matroska, say), or just transcode.) [23:37:24 CEST] <Bray90820> so something other than MP4? [23:37:38 CEST] <jkqxz> Yeah. [23:39:05 CEST] <Bray90820> I just transcoded it [23:39:15 CEST] <friki2015> is there a recommended way to seek to the end of a "growing" video file? somethink like -ss "-00:00:20" [23:39:35 CEST] <friki2015> of course i can get the duration and seek acoddingy [23:41:58 CEST] <jkqxz> Bray90820: To that? Some random MPEG-4-alike is not really a sensible thing to use nowadays - you're probably better off transcoding again to something more modern, such as H.264 or VP9. [23:45:56 CEST] <Bray90820> jkqxz: I just used -c:a instead of -c [23:48:39 CEST] <jkqxz> And now it's transcoding the video to whatever the default video codec is, using some default settings you probably don't want, and that works in mp4. [23:49:58 CEST] <jkqxz> (While copying the audio stream directly.) [23:52:53 CEST] <Danimal> I have a thing that's streaming mpegts out to a multicast address. Everything is fine. it seems that once I have 6 of them sending to different multicast addresses on the same network, the audio sync gets off by about a second(a lot). [23:53:48 CEST] <Danimal> this seems very weird to me. Anyone have an idea what might be going on? [23:56:00 CEST] <Bray90820> jzHow would I transcode it to h.264 then [23:56:05 CEST] <Bray90820> jkqxz: [23:57:14 CEST] <Bray90820> Or copy it [23:57:54 CEST] <pfelt> afternoon all. how does the native hevc performance compare with libx265 ? [23:58:11 CEST] <pfelt> (at least as i understand it, that's an apples-to-apples comparison) [23:59:46 CEST] <jkqxz> Bray90820: Transcode would be something like "ffmpeg -i input.avi -c:a copy -c:v libx264 -crf 20 output.mp4" (with a suitable value for 20). For copy, you need to pick a different container, so "ffmpeg -i input.avi -c copy output.mkv" or similar. [00:00:00 CEST] --- Wed Jul 13 2016

1 0

ffmpeg-devel.log.20160711
by burek 12 Jul '16

12 Jul '16

[01:25:58 CEST] <cone-304> ffmpeg 03Marton Balint 07master:593987810ef8: avutil/parseutils: dont assume standard time when parsing a timestamp [08:35:01 CEST] <bms20> Hi. I'm trying to work out how to obtain an AVCodecContext* from an AVStream*, without using the deprecated codec pointer in AVStream. Is there a way to do this in a non-deprecated form? [08:35:37 CEST] <andrey_turkin> you don't. Instead you create new AVCodecContext and populate it from AVStream [08:36:26 CEST] <andrey_turkin> side remark - doc/examples really ought to be updated... [08:39:55 CEST] <bms20> Ok - I am trying to forward port code from ffmpeg 2.6; the code does not create an AVCodecContext from what I can see. With older ffmpeg(s) were the stream's codec fields automatically populated in avformat_open_jnput? [08:41:37 CEST] <andrey_turkin> right. Now you are not supposed to use that codec context. Instead after open_input and after find_stream_info you should create new codec contexts for streams you are interested in, and populate then via avcodec_parameters_to_context [08:42:31 CEST] <andrey_turkin> so basically instead of getting stream->codec you just create new one using avcodec_alloc_context3 and fill it in using avcodec_parameters_to_context [08:42:53 CEST] <bms20> Ok - that sounds very sensible. In which case I'll hold the contexts myself. [08:43:11 CEST] <andrey_turkin> right, and it is decoupled from format context [08:43:26 CEST] <bms20> and use codecpar for obvious things like width/height, etc. [08:43:32 CEST] <bms20> Sounds like a sensible decoupling. [08:44:39 CEST] Action: andrey_turkin is waiting for someone to usher this conversation to #ffmpeg [08:52:53 CEST] <bms20> Andrey - thanks for your help; will crash and try this tomorrow. [10:20:58 CEST] <KGB> [13FFV1] 15michaelni pushed 1 new commit to 06master: 02https://git.io/vKZWf [10:20:58 CEST] <KGB> 13FFV1/06master 146c03d12 15Michael Niedermayer: remove trailing whitespaces [10:24:26 CEST] <KGB> [13FFV1] 15michaelni pushed 1 new commit to 06master: 02https://git.io/vKZWZ [10:24:26 CEST] <KGB> 13FFV1/06master 145abe932 15Michael Niedermayer: Favor https URLs where the servers support them... [12:57:02 CEST] <cone-190> ffmpeg 03Steven Liu 07master:b45a976ed4d4: add split_by_time flag for support splite mpegts segment at non-keyframe [13:15:03 CEST] <cone-190> ffmpeg 03Michael Niedermayer 07master:37c658b56b4c: doc/muxers: Fix Thisimproves typo [13:15:04 CEST] <cone-190> ffmpeg 03Michael Niedermayer 07master:92139c1bd190: avcodec/mediacodecdec_h264: Fix occured typo [15:04:48 CEST] <cone-190> ffmpeg 03Michael Niedermayer 07master:9585c50942a4: avcodec/iff: remove useless () [15:04:49 CEST] <cone-190> ffmpeg 03Michael Niedermayer 07master:a2550e7d2e4c: avcodec/iff: Check that there are enough bytes in dgb before reading [15:33:06 CEST] <cone-190> ffmpeg 03Matthieu Bouron 07release/3.1:1410732621dd: lavc/mediacodecdec_h264: properly convert extradata to annex-b [16:15:40 CEST] <cone-190> ffmpeg 03Ronald S. Bultje 07master:f0a2b6249bb2: vp9: add 16x16 idct avx2 (8-bit). [16:15:41 CEST] <cone-190> ffmpeg 03Ronald S. Bultje 07master:70d418c7e6af: Revert "PPC64: Add versions of functions in libswscale/input.c optimized for POWER8 VSX SIMD." [16:52:58 CEST] <ubitux> vp9_inv_dct_dct_16x16_add_8_8 improvements are kinda impressive [16:53:06 CEST] <ubitux> > 23x faster [16:53:16 CEST] <ubitux> with avx2 [16:53:20 CEST] <ubitux> i like this [16:53:27 CEST] <ubitux> much better than a 0.91x :D [16:54:29 CEST] <nevcairiel> the C code looks phenomenally slow though, thats kind of insane improvements [16:54:45 CEST] <BBB> maybe the c code was compiled with O0, I Dont really know [16:54:52 CEST] <BBB> the point was more the avx vs. avx2 comparison [16:55:03 CEST] <BBB> that should hold OK regardless of C compiler flags [16:56:17 CEST] <ubitux> 1.5x for avx vs avx2 is nice too [17:03:08 CEST] <BBB> yeah the avx2 speed improv is quite good [17:03:14 CEST] <BBB> Ill try 32x32 when I find time also [17:07:03 CEST] <BBB> if anyone feels like redoing the loopfilter, I can explain how the 2x merging in the caller (vp9.c) works, we need to change that to 4x merging to get it to work for avx2/8bit [17:07:14 CEST] <BBB> (i.e. its not exactly trivial) [17:39:07 CEST] <jamrial> BBB: would have been 2x if you didn't have to deal with the crosslane crap [17:39:28 CEST] <BBB> the crosslane crap only affects the store operations though [17:39:35 CEST] <BBB> (and the transpose of course) [17:39:58 CEST] <jamrial> that's the speed up i saw in most avx2 functions compared to ssse3/avx where i didn't use any perm or insert/extract [17:42:01 CEST] <jamrial> like in ME functions [17:43:42 CEST] <jamrial> nevcairiel: would be interesting to rebench the c version now that we disabled tree vectorization [17:46:36 CEST] <outcast> hi! is this the right place to ask for help with a C# question using ffmpeg? [17:47:16 CEST] <BBB> outcast: no [17:47:25 CEST] <BBB> outcast: this channel is only for development of ffmpeg, not with ffmpeg [17:47:49 CEST] <outcast> k is there a channel for that [17:48:11 CEST] <BBB> jamrial: interesting& I guess idct is somewhat special because as register size increases, the size of things like transpose increases accordingly which naturally dilutes the 2x optimization ideal [17:48:33 CEST] <BBB> but the MC results are promising and its what Id expect for the horizontal loopfilter, I guess [17:48:38 CEST] <BBB> (the ones which dont need to transpose0 [17:48:43 CEST] <RiCON> outcast: it's in the topic (#ffmpeg) [17:49:40 CEST] <outcast> ok thanks i was over there just not found what im looking for in the documents in the topic there yet [17:49:47 CEST] <outcast> i'll head back over there [18:22:55 CEST] <cone-190> ffmpeg 03James Almer 07master:f60b54902f45: avformat/oggenc: make flac the default for oga muxer [20:50:18 CEST] Action: mateo`_ working on the next libav merge [21:01:26 CEST] <JEEB> mateo`_: did you still want that 10bit AVC sample btw? [21:01:38 CEST] <JEEB> to see how your things revert from hwdec to swdec [21:03:03 CEST] <mateo`_> JEEB: the decoder should properly fail at init time right now (if it was not the case before) [21:03:12 CEST] <JEEB> oh, cool [21:03:21 CEST] <JEEB> because before it went all the way to decoding [21:03:39 CEST] <JEEB> I will reinstall my android development VM as I somehow borked its / :D [21:03:58 CEST] <mateo`_> I added some profile checks in e452abc5c2441356d1aa22ffcd8a9183aa1fac33 [21:04:15 CEST] <JEEB> coolio [21:07:45 CEST] <mateo`_> JEEB: anyway, if you still can share this sample, I'm happy to do some testing [21:09:06 CEST] <JEEB> mateo`_: the oldest one being http://cccp-project.net/beta/test_files/H.264-10bit-sample-railgun.mkv :) (my apologies for the animated content) [21:10:52 CEST] <JEEB> but I guess this is better than superheroes in spandex ;) [21:10:53 CEST] <mateo`_> no worries, I tend to consume a lot of anime content [21:13:30 CEST] <mateo`_> "any" content is fine except that bunny thing [21:14:49 CEST] <JEEB> tell me about it... [21:22:50 CEST] Action: mateo`_ crying watching big buck bunny for the 1000000 time [21:23:49 CEST] <JEEB> at this point if someone asks for a sample for something I push out tears of steel [21:23:59 CEST] <JEEB> it's a) shorter and b) less BBB [21:24:23 CEST] <nevcairiel> dont hate on BBB, what did he ever do to you [21:24:46 CEST] <JEEB> being overused is the only sin it did [21:24:51 CEST] Action: JEEB 29 [21:24:54 CEST] <JEEB> welp [21:25:12 CEST] <BBB> can we please rename that sample to something else? [21:25:31 CEST] <BBB> like PBR (phat buck rabbit) [21:25:41 CEST] <JEEB> :D [21:27:20 CEST] <ubitux> and what do we do about beaglebone black? [21:29:59 CEST] <mateo`_> make it decode PBR, and grab a hammer [21:33:26 CEST] <mateo`_> michaelni: are there other samples than this one ffmpeg-bugs/trac/ticket3962/multiple_stsd.mp4 to test multiple stsd ? [21:40:07 CEST] <llogan> PBR is a shitty beer. [21:40:25 CEST] <llogan> hipster juice [22:35:49 CEST] <michaelni> mateo`_, the samples in the fate-quickdraw and fate-gsm-toast might contain multiple stsd [22:37:33 CEST] <mateo`_> michaelni: thanks, btw the merge (https://github.com/mbouron/FFmpeg/commit/5d3f3a421d70890eff73434b93cb55c233…) passes fate and handle ffmpeg-bugs/trac/ticket3962/multiple_stsd.mp4, i'll do more testing tomorrow morning [22:38:41 CEST] <michaelni> tickets/4400/cartest_supers.mov might also be multiple stsd [23:14:54 CEST] <michaelni> mateo`_, the commit breaks ffmpeg-bugs/trac/ticket1918/aark15sd_9A62E2FA.mp4 [23:19:53 CEST] <michaelni> mateo`_, ./ffmpeg -i matrixbench_mpeg2.mpg -t 1 -frag_duration 200k test2.mov ; ./ffprobe test2.mov -show_packets -print_format compact [23:19:58 CEST] <michaelni> ffprobe segfaults [23:21:18 CEST] <michaelni> tickets//2991/wwwq_cut.mp4 fails very similar to ticket1918 [23:23:01 CEST] <cone-706> ffmpeg 03Carl Eugen Hoyos 07master:244d22452c7d: lavc/Makefile: Fix standalone compilation of the lame encoder. [23:34:45 CEST] <llogan> Compn: i'll be gone from july 13 - august 2. could you tend to ML then? mostly from july 25-aug 2 as I not have internet/cell range then. [23:40:18 CEST] <flux> $9 [23:40:34 CEST] <flux> (oops) [00:00:00 CEST] --- Tue Jul 12 2016

1 0

ffmpeg.log.20160711
by burek 12 Jul '16

12 Jul '16

[00:01:33 CEST] <DHE> as two files, or in a format that doesn't care about that [00:14:01 CEST] <jaggz> dhe, I'd like the final to be in .mp4 [00:14:24 CEST] <jaggz> I have a screen capture, and audio recording -- the audio is over an hour, the screencapture failed because my screensaver turned on at 10 minutes [00:14:55 CEST] <jaggz> the screencap contains audio too, but I'm wondering if I can chop out the video from the 11th minute on to save filespace in the final vide [00:17:44 CEST] <friki2015> Hi, i've encoded a video at 25fps. ffmpeg says: "25 fps, 25 tbr, 90k tbn, 50 tbc". Using some players (mplayer) it plays at 2x speed (50fps) but ffplay works fine [00:18:09 CEST] <JEEB> try mpv and latest vlc rather than mplayer [00:20:49 CEST] <friki2015> the video works fine with both players. May be a mplayer bug or a encoding/packaging problem? [00:21:14 CEST] <JEEB> most probably mplayer just being mplayer [00:21:46 CEST] <friki2015> :-) [00:22:03 CEST] <JEEB> mpv and vlc are probably the two things you should generally test things with. mpv is the least retarded fork of mplayer that uses FFmpeg's libraries, and VLC generally uses something els. [00:22:15 CEST] <JEEB> so you can thus test two ways of reading the files in many cases. [00:22:50 CEST] <JEEB> http://lists.mplayerhq.hu/pipermail/mplayer-dev-eng/2015-September/073131.h… [00:23:03 CEST] <JEEB> this was a good post on original mplayer's mailing list from last year [00:23:06 CEST] <friki2015> thanks for the tip. i was thinking that tbc=50 was the problem [00:24:50 CEST] <friki2015> "MPlayer is still my favorite video player, and I really would hate to see it go down the drain." hehehe love this final words at the post [00:31:52 CEST] <JEEB> friki2015: also since container timestamps generally win over anything in the actual stream, for matroska you can use mkvinfo and for mp4/mov type things you can use L-SMASH's boxdumper [00:32:07 CEST] <JEEB> if you want to be ever stare at the timestamps saved in the actual container [00:32:18 CEST] <JEEB> *if you ever want to stare... [01:13:33 CEST] <haasn> furq: I remember wanting to show you screenshots of my wow interface. I managed to recover some from chat logs: https://i.imgur.com/PBeX3PJ.jpg https://i.imgur.com/0L24roU.jpg [04:56:46 CEST] <Blakes5> Hello. I'm trying to setup a ffserver for a http stream. I have what looks like a working conf file and ffmpeg appears to be encoding the stream but I can't seem to access it using VLC. Help? [05:00:20 CEST] <Blakes5> ffserver.conf is here http://pastebin.com/zJ0P32YK [05:00:53 CEST] <Blakes5> This is the ffmpeg command ffmpeg -r 30 -s 752x416 -f video4linux2 -i /dev/video0 http://localhost:8090/office.ffm [05:02:50 CEST] <Blakes5> http://localhost:8090/office.ffm is the address I'm using to try to access it using VLC. [05:04:01 CEST] <Blakes5> I can see get requests. [05:04:38 CEST] <Blakes5> [GET] "/office.ffm HTTP/1.1" 404 158 I'm guessing the 404 means ffserver can't find the stream? [05:05:25 CEST] <Blakes5> I don't understand. [08:47:19 CEST] <antiPoP> hi, I'm trying to run ffmpeg and I get "ffmpeg: error while loading shared libraries: libx264.so.148: cannot open shared object file: No such file or directory" [10:39:03 CEST] <asio> I've got 2 movies: movie1.mp4 and movie2.mp4, that I want merged together but one was filmed during sunshine and other when cloudy, is it possible to adjust them by colorbalance/brightness to so one matches the other? [11:26:56 CEST] <sybariten> oh hai [11:27:17 CEST] <sybariten> could someone please explain to me the concept of separate input and output frame rates when makeing videos from sequences of still images [11:31:58 CEST] <spirou> hi [11:32:33 CEST] <spirou> from still images, there would be no input frame rate would it? [11:33:26 CEST] <furq> the input framerate is how long you want each image to appear for [11:33:38 CEST] <furq> the output framerate is what you think it is [11:34:08 CEST] <spirou> but I assume you can tell ffmpeg about what rate those images come "these are 50 frame/s" and then tell that you want output frame "25 frame/s" and so it will skip all even frames :D [11:34:20 CEST] <furq> you sure can [11:35:50 CEST] <furq> it's more likely that you'd want something like -framerate 1 -i %03d.png -r 25 [11:36:10 CEST] <furq> which will duplicate frames [11:57:31 CEST] <sybariten> furq: Hm. I'm sorry but could you explain this again, butt differenlty [11:57:37 CEST] <sybariten> eh, BUT not butt [12:20:53 CEST] <sybariten> it seems like they would control the same thing? [12:34:07 CEST] <spirou> yeah you could se out framerate so each in-frame will be used multiple times in a video yeah [12:34:23 CEST] <spirou> *set [12:36:07 CEST] <spirou> if you want a 25fps video but don't really have that many pictures [13:48:47 CEST] <Blakes5> I'm trying to setup a ffserver stream. It looks like it's working but I can't access the stream. Can someone help me pick through this? ffserver.conf http://pastebin.com/zJ0P32YK [13:49:03 CEST] <Blakes5> This is the ffmpeg command ffmpeg -r 30 -s 752x416 -f video4linux2 -i /dev/video0 http://localhost:8090/office.ffm [13:49:14 CEST] <Blakes5> http://localhost:8090/office.ffm is the address I'm using to try to access it using VLC. [15:21:08 CEST] <istee> i am trying to convert a h264 file from a dvr cctv camera. it works fine until about five minutes into the video where the console is flooded with messages like [15:21:11 CEST] <istee> [h264 @ 0x174df60] Cannot use next picture in error concealment [15:21:14 CEST] <istee> [h264 @ 0x174df60] concealing 396 DC, 396 AC, 396 MV errors in P frame [15:22:00 CEST] <istee> is there any way to get it to drop frames with these errors? it plays fine with mpv if you skip through it but it ends up freezing and then being just a gray color while converting with ffmpeg [15:27:10 CEST] <bencoh> the error_concealment flag iirc? [15:27:13 CEST] <bencoh> istee: ^ [15:29:12 CEST] <istee> i couldnt find any relevant flag @ http://ffmpeg.org/ffmpeg.html [15:29:40 CEST] <istee> i found references to -ec/-er but they're deprecated? [15:30:30 CEST] <istee> url was supposed to be http://ffmpeg.org/ffmpeg-all.html , oops [15:31:44 CEST] <istee> nevermind, i put it after the input and it gave me a vague splitting error. -ec before works [15:32:22 CEST] <bencoh> input-related options should always sit before input :) [15:38:01 CEST] <istee> "-ec 0" just kept stuttering between the frame in a loop, not moving forward :( [16:13:54 CEST] <istee> update: i tried several players and all of them do the same thing when playing the damaged section, requiring stopping said player and reloading. but it works fine when you seek past it [16:14:35 CEST] <ChocolateArmpits> Did w3fdif get updated? Previously I remember it was slower than yadif, but now on an old two core system it's pushing over 300fps at complex while yadif manages only 220fps at send_field_nospatial [16:14:39 CEST] <istee> however ffmpeg with both -ss after and before input causes the error [16:17:15 CEST] <ChocolateArmpits> bwdif pushes around 260fps and looks to have better output than yadif or w3fdif [16:17:56 CEST] <istee> https://paste.debian.net/hidden/77c0c9d0/ log while trying to play said section with mpv [16:21:43 CEST] <ChocolateArmpits> istee: Are you looking for any solution ? [16:23:03 CEST] <istee> i want to be able to transcode whats recoverable from this stream to a non-corrupt DVD [16:24:08 CEST] <ChocolateArmpits> istee: Ok, pipe the bytes of the file to ffmpeg [16:25:07 CEST] <istee> https://carl.moe/bad.mp4 i trimmed the section it chokes on 5:20 to be exact w/ -vcodec copy [16:27:41 CEST] <istee> i can brute force recover by seeking before input and doing -vcodec copy for these sections to get them away but i would like a less hassle solution [17:34:09 CEST] <aarrgg> Good morning :) I am struggling with FFmpeg, I have a persistent error -->Unknown input format: 'alsa' [17:34:09 CEST] <aarrgg> ...and my extensive searches are not shedding light on my problem [17:36:14 CEST] <aarrgg> help anyone? [17:36:44 CEST] <outcast> hi is there a page with code snippets? [17:36:59 CEST] <outcast> looking for a sample of how to do a progress bar in c# [17:38:19 CEST] <DHE> aarrgg: need to compile with ALSA support. you may need to install a package like alsa-dev for your distribution [17:39:30 CEST] <aarrgg> E: Unable to locate package alsa-dev [17:41:31 CEST] <aarrgg> DHE , i'm using Jessie on a Raspberry Pi, I have not seen a specific example of compiling ffmpeg with ALSA support [17:42:46 CEST] <DHE> officially you use ./configure --enable-indev=alsa (and similar for outdev) but I'm guessing you lack the dev library. [17:44:21 CEST] <aarrgg> apt-get install alsa-dev is what I have tried, and thanks for the config line [17:48:45 CEST] <debianuser> aarrgg: apt-get install libasound2-dev [17:49:19 CEST] <debianuser> "libasound" is alsa :) [17:49:54 CEST] <aarrgg> ill try a recompile with your suggestion, thanks. [17:50:48 CEST] <aarrgg> debianuser: thanks, tried and already the newest version :) [17:51:03 CEST] <outcast> is the forum in the topic the best place to ask a c# question [17:55:30 CEST] <bencoh> I dunno if you'll find any C#/friendly dev (I'm not) but I guess you could always ask (provided it's ffmpeg-related ;) [17:57:32 CEST] <aarrgg> its nearly 2AM in Australia, so i'm compiling and off to bed, thanks for your help, love linux but its documentation leaves a bit to be desired [17:58:10 CEST] <furq> are you compiling ffmpeg on an rpi [17:58:29 CEST] <aarrgg> yes on a Rpi [17:58:33 CEST] <furq> that sounds like a miserable experience [17:58:43 CEST] <viric> furq: much less noisy than what you are used to [17:58:48 CEST] <viric> ;) [17:59:06 CEST] <furq> i've got an rpi with ffmpeg on it [17:59:10 CEST] <furq> but i compiled it on a real computer [17:59:14 CEST] <aarrgg> not too bad, 15min? and not often [17:59:49 CEST] <viric> with a good scheduler, you just queue the task with nice, and at some point it's done [17:59:51 CEST] <furq> are you disabling everything or something [18:00:06 CEST] <furq> that sentence is confusing [18:00:30 CEST] <furq> pretend i didn't say "or something" [18:00:40 CEST] <outcast> sorry phone rang [18:01:16 CEST] <outcast> looking for way to use -progress i think it is to make a percentage bar [18:01:35 CEST] <aarrgg> furq: ffmpeg has not been recognizing alsa....OR...you could recommend a better way to make video with sound :) [18:01:59 CEST] <furq> there is no better way than to recompile ffmpeg [18:02:11 CEST] <furq> but you can cross-compile it on a faster cpu [18:02:34 CEST] <aarrgg> thanks, at least I'm not barking up the wrong tree :) [18:02:43 CEST] <viric> can ffmpeg do something to record sound/video in sync, if it captures from independent devices? webcam, soundcard, ... [18:02:52 CEST] <furq> it's slightly complicated if you need external dependencies but i assume debian multiarch handles that ok [18:03:07 CEST] <furq> i didn't need to deal with it because i just wanted to see how bad omx's h264 encoder actually is [18:03:10 CEST] <furq> answer: it's bad [18:03:17 CEST] <viric> I have a sound recorder that deviates ~1s per hour [18:03:42 CEST] <aarrgg> ....I'm in enough trouble as it is [18:04:00 CEST] <viric> furq: ah I've never seen it. Worse than x264 ultrafast? [18:04:28 CEST] <furq> i've not done a side by side comparison [18:04:36 CEST] <furq> i'd guess yes though [18:06:08 CEST] <furq> you can't really expect much from a £30 computer though [18:06:33 CEST] <furq> or £9 if the pi zero has the same gpu [18:21:49 CEST] <aarrgg> ...having some joy, thanks again, off to bed! [19:37:59 CEST] <blue_misfit> hey guys - anyone familiar with HLS? I'm struggling to package a stream with 23.98 video while respecting #EXT-X-TARGETDURATION > #EXTINF. If I have 4 second chunks for example #EXTINF is usually 4.004 - presumably due to 23.98 vs 24. How can I fix? #EXT-X-TARGETDURATION is integer only [19:41:23 CEST] <Bray90820> pts has no value is what does that error mean? [19:41:28 CEST] <Bray90820> or warring either [19:41:34 CEST] <Bray90820> *Rather [19:44:39 CEST] <JEEB> blue_misfit: that's a bug in the muxer then [19:44:58 CEST] <JEEB> post an issue on the tracker pointing towards https://tools.ietf.org/html/draft-pantos-http-live-streaming-19#section-4.3… [20:24:55 CEST] <wiak> vp9 at faster speed and higher bitrate or slower speed at lower bitrate? [20:25:34 CEST] <wiak> -deadline realtime -speed 8 -b:v 6000K [20:25:56 CEST] <wiak> for live streaming [20:39:25 CEST] <ChocolateArmpits> wiak: Depends how many concurrent streams you need and how many you can sustain [20:40:08 CEST] <wiak> nah with vp9 the issue lies with the slow encoder [20:40:33 CEST] <wiak> encoding to 720p is like 36fps and the source is 30fps [20:40:48 CEST] <wiak> vp9 has awful multicoresupport [20:41:00 CEST] <ChocolateArmpits> If it's stable 36fps then I don't see no problem in livestreaming [20:41:08 CEST] <wiak> true [20:43:25 CEST] <ChocolateArmpits> Google/Youtube parallelizes their encoding by splitting the file into parts with each encoded separately. So for them multithreading isn't that vital. [20:43:53 CEST] <ChocolateArmpits> I guess they might do the same for livestreams [20:44:00 CEST] <ChocolateArmpits> or maybe not [20:44:16 CEST] <ChocolateArmpits> nah scratch that, doesn't make sense for livestreams [20:44:25 CEST] <furq> do they actually do that or is that just conjecture [20:45:44 CEST] <ChocolateArmpits> furq: they do http://youtube-eng.blogspot.com/2016/05/machine-learning-for-video-transcod… [20:45:48 CEST] <ChocolateArmpits> "The only way we can keep up with that rate of ingest and quickly show you your transcoded video in YouTube is to break each file in pieces called chunks, and process these in parallel. Every chunk is processed independently and simultaneously by CPUs in our Google cloud infrastructure" [20:46:33 CEST] <ChocolateArmpits> The post is all about how they try to improve quality of these chunks [20:47:55 CEST] <furq> At YouTube we care about the quality of the pixels we deliver to our users. [20:48:00 CEST] <furq> handcrafted artisanal pixels [20:49:05 CEST] <furq> i like how it says that's the only way they can keep up [20:49:17 CEST] <furq> i mean they could make their encoder have less shit multithreading [20:55:12 CEST] <SuperSQL> Hello. I have a problem I can't figure it out with ffmpeg. Recently I decided to digitize old video camera cassette tapes using PCI video card and ffmpeg to record. Problem that I have is that some tapes are abit damaged and it looks like some frames are being dropped (ffmpeg doesn't report that). Problem now is it looks like when I try to capture using ffmpeg those dropped frames get replaced by old frame, so when playing finished video [20:55:12 CEST] <SuperSQL> you'd see single frames jumping back in time. [20:55:50 CEST] <SuperSQL> I used this command ffmpeg -channel 1 -s 920x576 -input_format yuyv422 -thread_queue_size 10000k -i /dev/video0 -f alsa -ac 1 -thread_queue_size 2000k -i hw:0 -vcodec copy -acodec copy "/home/user/Desktop/012.mkv" [20:57:15 CEST] <ChocolateArmpits> SuperSQL: Are you sure it's ffmpeg duplicating frames and not the pci card ? [20:57:17 CEST] <SuperSQL> I tried many things with it, it always ends up the same. There is nothing wrong with the video card since I tried using other video programs such as motv and it worked perfectly, it just dropped those frames [20:57:32 CEST] <ChocolateArmpits> oh ok [20:57:33 CEST] <obelisk29> Is there any way to encode h.265 files with the hvc1 atom rather then the default hevc? Premiere encodes using hvc1 and I tried just changing the tag -tag:v hvc1 but premiere won't import it. [20:59:07 CEST] <SuperSQL> It's really strange, like if the frame buffer overflowed and went to the oldest one in memory [20:59:33 CEST] <SuperSQL> no warnings or errors [21:00:00 CEST] <wiak> ChocolateArmpits http://nwgat.ninja:8000/live.webm [21:00:01 CEST] <wiak> ^^ [21:00:22 CEST] <ChocolateArmpits> wiak: huh ? [21:01:03 CEST] <wiak> 1080 > 720, 36fps with speed 8, deadline realtime [21:01:13 CEST] <wiak> 4mbps [21:01:35 CEST] <ChocolateArmpits> wiak: What's the CPU usage ? [21:02:16 CEST] <wiak> 20.9% [21:02:22 CEST] <wiak> windows 10, FX-8350 [21:03:47 CEST] <nick0> Is this the right place to ask about libav issues? [21:03:55 CEST] <wiak> #libav? hehe [21:03:59 CEST] <JEEB> if you mean libav* libraries, yes [21:04:06 CEST] <JEEB> Libav the fork project, no [21:04:07 CEST] <nick0> err yeah, haha [21:04:10 CEST] <nick0> the libraries [21:04:20 CEST] <wiak> wait what there is a libav fork? [21:04:29 CEST] <furq> it was really nice of libav to pick such an unambiguous name [21:04:42 CEST] <wiak> fufork? [21:04:43 CEST] <wiak> :P [21:04:56 CEST] <wiak> mey the forking commence [21:05:59 CEST] <JEEB> wiak: more like very early 2011 and FFmpeg merges most things done there [21:06:18 CEST] <wiak> https://gist.githubusercontent.com/nwgat/1cb7e8e69d0b1fbc6a57cb7631a807c5/r… [21:06:46 CEST] <wiak> vp9 is a pain, its much nicer to use vp8, that one can maintain like 50fps [21:07:24 CEST] <nick0> Okay, so basically I'm having an issue with the resampler: if I don't flush it I only get 5 seconds of my original input, but if I do, I get my entire input and an extra minute of it played backwards. [21:07:35 CEST] <nick0> My code is here: http://pastebin.com/u7ThNPXq [00:00:00 CEST] --- Tue Jul 12 2016

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