[00:43:57 CET] <postmodern> is it possible to join multiple mkv files together which have different aspect ratios and resolutions? [00:44:28 CET] <wiak> would think so yes [00:44:53 CET] <wiak> but dunno [01:00:53 CET] <postmodern> hmm, trying to figure out how i can tell the player that the aspect ratio changes in the middle of the mkv [01:01:26 CET] <postmodern> mpv seems to default to using the container aspect ratio, which stretches or squeezes the image [01:01:39 CET] <postmodern> first mkv is 16:9 and the others are 4:3 [01:01:45 CET] <JEEB> make sure the parameter sets are taken into mention [01:01:54 CET] <JEEB> poke #mpv for mpv specifics [02:15:15 CET] <Nanashi> Hahahahaha I found it. It contained gamma metadata which made it a little brighter than the screenshots I find online since they have it stripped. [02:15:39 CET] <Nanashi> (pains me as a man of consistency though) [03:30:15 CET] <logic_> hello [03:30:21 CET] <logic_> i need some help [03:31:27 CET] <logic_> ffmpeg -i LJ3.wmv -vcodec copy -acodec copy LJ3.mkv [03:31:43 CET] <logic_> how do i convert the audio to AAC 128kb? [03:31:50 CET] <logic_> instead of copying the audio stream [03:38:16 CET] <Nanashi> -c:a aac -b:a 128k [03:39:05 CET] <Nanashi> more information: https://trac.ffmpeg.org/wiki/Encode/AAC [05:01:54 CET] <kevmitch_> according to http://pastebin.com/Bay9wetL [05:02:03 CET] <kevmitch_> ignore that [07:05:34 CET] <Jesperhead> Hello all, trying to record my desktop with ffmpeg using the test example: ffmpeg -video_size 1366x768 -framerate 25 -f x11grab :0.0 output.mp4 only to receive a null error stating "Requested output format 'x11grab' is not a suitable output format" I appreciate any assistance while I continue googling this. [07:29:48 CET] <Jesperhead> additional note I omitted: ":0.0: Invalid argument" is one line below the earlier noted error message [07:38:15 CET] <Jesperhead> ffmpeg's suggested string was not suitable for my linux distribution [07:38:36 CET] <Jesperhead> i tried a test string from arch's wiki and all appears to work. Thank you all for.. listening? [08:17:15 CET] <LJHSLDJH1DLJH> how to know the codecs info of a .webm file under ubuntu? [08:18:07 CET] <furq> ffprobe [08:18:28 CET] <LJHSLDJH1DLJH> I can't seek (forward / backward) [08:18:50 CET] <LJHSLDJH1DLJH> tried to cut but this is the error I get pastebin.com/pK9CPaq7 [08:19:30 CET] <furq> av_interleaved_write_frame(): No space left on device [08:20:43 CET] <LJHSLDJH1DLJH> furq: yeah but it doesn't make any sense to me [08:21:01 CET] <LJHSLDJH1DLJH> what that supposed to mean and how to fix it? [08:21:23 CET] <furq> it means your disk is full [08:26:01 CET] <LJHSLDJH1DLJH> stupid me :(( [08:26:05 CET] <LJHSLDJH1DLJH> thanks furq [08:26:26 CET] <LJHSLDJH1DLJH> but still the same problem of not able to play this file properly [08:26:59 CET] <LJHSLDJH1DLJH> I actually was able to play with mplayer from command line but it crashes at certain point [08:27:11 CET] <LJHSLDJH1DLJH> then it's possible to forward or backword [12:51:16 CET] <Polochon_street> hi! I'm trying to decode songs into an array, but sometimes I have extra zeroes in it, and I don't know why... [12:51:40 CET] <Polochon_street> I'm allocating an array of size int8_t, as described here http://pastebin.com/DReryLar [12:52:30 CET] <Polochon_street> I use a while(av_read_frame(context, &avkpt)), and sometimes, it seems that I receive « extra frames » (I don't know why), and I must realloc my sample array [12:53:31 CET] <Polochon_street> I think that's where my extra zeros come from, but I don't know how to avoid it... How can I have more samples than duration*sample_rate*nb_bytes_per_samples*channels/AV_TIME_BASE ? [12:54:40 CET] <Polochon_street> (here's the full code: https://github.com/Polochon-street/bliss/blob/63bede0e4d6f0ab0caa22ab49f1f72...) [15:21:19 CET] <Polochon_street> okay, I solved my problem. Turns out getting the number of samples via a duration computation is good to approximate, but not to determine the exact number [15:23:46 CET] <grublet> Polochon_street: afaik the duration thing will always be innacurate to some degree [17:33:56 CET] <YaMoonSun> I should be able to record my screen and main audio output using ffmpeg and output as HEVC right? [17:41:41 CET] <kepstin_> YaMoonSun, in theory yeah, but you have to have a really fast computer or use really low-quality settings to get realtime hevc encoding. [17:45:57 CET] <YaMoonSun> I was using an application on ubuntu once that allowed me to record the screen and it would output it as a massive .avi, I think it was muxing the screen capture. Can I do that and then just convert to hevc later? [17:46:20 CET] <YaMoonSun> windows currently [17:47:29 CET] <furq> sure [17:48:55 CET] <kepstin_> couple options - if you've got a lot of disk space, you could use a video codec like ffvhuff; if not, try x264 with a fast profile; could also try a hardware encoder (nvenc?) and just use a really high bitrate. [17:50:02 CET] <furq> if you want a massive uncompressed screen capture then use rawvideo [17:50:45 CET] <kepstin_> rawvideo can be fun tho; your bottleneck is often the speed of your hard drive writing :) [17:51:04 CET] <YaMoonSun> If I use that to record let's say an hour, what do you think the output size will be? [17:51:34 CET] <furq> depends on the resolution [17:51:40 CET] <YaMoonSun> I've got the samsung 840 evo, speed is no problem, but size is 250gb [17:51:41 CET] <furq> unless "massive" is specific enough [17:51:53 CET] <YaMoonSun> 1080p [17:52:51 CET] <furq> what framerate [17:53:36 CET] <furq> at 60fps that'd be close to 700GB [17:53:41 CET] <furq> unless i've forgotten how to work this out [17:53:57 CET] <furq> assuming you use yuv420, rgb would be even bigger [17:54:18 CET] <YaMoonSun> well ffs [17:54:44 CET] <YaMoonSun> I'd need x4 1tb drives on raid 0 [17:54:48 CET] <furq> you could use a fast hevc profile but then there isn't much point using hevc over avc [17:55:24 CET] <YaMoonSun> Quite disappointing. [17:55:30 CET] <YaMoonSun> Thanks for the input [17:55:46 CET] <furq> there are lossless codecs if you specifically want to use hevc [17:57:23 CET] <furq> but unless you have a specific requirement to use hevc then i'd just stick with x264 [17:57:37 CET] <furq> x265 is overrated anyway [17:58:34 CET] <YaMoonSun> Not even [17:59:25 CET] <tp__> its not overrated for 4K [18:13:16 CET] <anddam> i have a 1280x720 video that I want to rescale vertically to a 480p size, I tried -vf scale=-1:480 as in https://gist.github.com/anddam/3ccf9893b7a1b2d38931 but I'm getting "Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height" [18:13:29 CET] <anddam> how can I check what the incorrect parameter is? [18:13:48 CET] <BtbN> read what it outputs [18:13:49 CET] <furq> [libx264 @ 0x7fb2a2830e00] width not divisible by 2 (853x480) [18:43:07 CET] <anddam> oh, that line I missed [18:43:51 CET] <anddam> actually I didn't _need_ 480p but the 720 point was a bit higher for an old ipad that was skipping frames so I wanted to scale it [18:47:00 CET] <anddam> I went for a close 864:486 [18:47:16 CET] <anddam> thanks [18:50:52 CET] <jackhold> hej anyone that could help me with a little problem i got trying to send udp stream and capture pictures every 30sek from a webcam?? [18:52:11 CET] <anddam> mmm "[mp4 @ 0x7fb2e5000000] Application provided invalid, non monotonically increasing dts to muxer in stream 2: 575609 >= 513977 av_interleaved_write_frame(): Invalid argument" [18:52:57 CET] <anddam> I figure that's stream in input file [18:54:25 CET] <firewated> i had a similar problem with an iPad, just re-encoding the audio to 2 channel aac and copying the video stream into an mp4 gave me smooth playback [18:54:57 CET] <YaMoonSun> -i "Input.mov" -vf scale=-2=480 -c:v libx264 -b:v 2M -c:a copy "Output.mp4" ? [18:55:42 CET] <anddam> what does -2 mean in the scale argument? [18:55:56 CET] <furq> same as -1 but mod2 [18:56:27 CET] <anddam> furq: I cannot find mod2 in the man page [18:56:51 CET] <anddam> firewated: I'll try, why 2 channels? [18:57:14 CET] <anddam> I got a lot of [18:57:16 CET] <anddam> [mp4 @ 0x7f9343820c00] Non-monotonous DTS in output stream 0:1; previous: 13412354, current: 13412344; changing to 13412355. This may result in incorrect timestamps in the output file. [18:57:45 CET] <anddam> this is the audio stream, I worked around the previous error on stream 2, that was subtitle, by just stripping it out [18:58:52 CET] <YaMoonSun> Sounds like it's being a dingleberry. Are you using the latest build? [18:59:06 CET] <firewated> I can't remember if there was a specific reason, does the iPad have more than 2 speakers? [18:59:40 CET] <firewated> Also, does the mp4 container event support DTS? [18:59:46 CET] <firewated> even* [19:02:15 CET] <jackhold> Why do this just overwrite the output and not auto incrementing?? ffmpeg -v info -r 25 -s 1280x720 -f video4linux2 -vcodec mjpeg -i /dev/video1 -f image2 -r 1/30 ~/indput/out%20d.png -vcodec copy -f mjpeg udp://test.eu:33333 [19:05:37 CET] <anddam> YaMoonSun: nope [19:05:54 CET] <anddam> firewated: I think the iPad has just the one speaker [19:06:09 CET] <firewated> So why do you want DTS audio [19:10:06 CET] <anddam> I don't, I don't even know what that is [19:10:37 CET] <anddam> I'm just copying the audio stream to avoid recoding [19:13:18 CET] <firewated> try this: ffmpeg -i in.mkv -c:s mov_text -ac 2 -b:a 320K -c:v copy out.mp4 [19:18:26 CET] <eksrow> Hi, i'm using the overlay filter to slide an image across a video(with overlay=x=-'(t*20) - 50') and it's working great. but the animation looks 30fps ish while the framerate of the video itself is 60. is it possible to get this animation to 60? [19:44:28 CET] <kepstin> eksrow: that should be recalculated every frame. Perhaps make sure both inputs to the overlay filter are the same framerate? [20:09:39 CET] <eksrow> kepstin: good to know it's possible, i'l keep trying. each input has -r 60, that should be correct right? [20:12:01 CET] <anddam> firewated: lots of those "Non-monotonous DTS in output stream 0:1; " then again the error "Application provided invalid, non monotonically increasing dts to muxer in stream 2: 575609 >= 513977 av_interleaved_write_frame(): Invalid argument" [20:13:19 CET] <firewated> can you post the full output from ffmpeg, i don't see how DTS is getting to the output, it should be getting re-encoded [20:21:08 CET] <anddam> sure, give one moment [20:23:48 CET] <anddam> it's _very_ long [20:23:59 CET] <anddam> I figure you just want to see the start and the end [20:24:16 CET] <firewated> sure, in a gist [20:28:53 CET] <anddam> https://gist.github.com/anddam/3ccf9893b7a1b2d38931 updated [20:29:47 CET] <anddam> notice the massive "[mp4 @ 0x7fb0bb01fa00]" starting line cut at L105 [20:30:58 CET] <firewated> hmm your input audio is aac, not sure why there's references to DTS at all [20:33:34 CET] <furq> firewated: decoding timestamp [20:34:36 CET] <firewated> oh, right [20:34:51 CET] <firewated> ffmpeg version is recent too, not sure [20:35:03 CET] <furq> anddam: try -fix_sub_duration as an input option [20:39:42 CET] <firewated> and you probably don't need to re-encode the audio either as it's already aac [20:45:02 CET] <anddam> "ffmpeg -y -fix_sub_duration -i in.mkv -c:s mov_text -c:a copy -c:v copy out.mp4" results in the same error [20:45:10 CET] <anddam> [mp4 @ 0x7f9159802600] Application provided invalid, non monotonically increasing dts to muxer in stream 2: 573373 >= 513977 [20:45:18 CET] <anddam> av_interleaved_write_frame(): Invalid argument [20:45:58 CET] <anddam> I'm halfway scaling the video anyway [20:46:49 CET] <firewated> do you also still get the timestamp out of range messages? [20:49:58 CET] <anddam> not with the rescale, I dropped the sub stream, copied the audio and rescaled video using libx264, [20:51:36 CET] <firewated> may be because of the sub stream [20:55:30 CET] <jookiyaya> anybody here? mirc [21:10:34 CET] <anddam> what does that even mean? [00:00:00 CET] --- Mon Feb 29 2016