[00:04] <DrPastah> Stream #0:0(jpn): Video: h264 (High), yuv420p, 1864x1048, SAR 1:1 DAR 233:131, 23.98 fps, 23.98 tbr, 1k tbn, 47.95 tbc (default) [00:04] <DrPastah> what does tbr, tbn and tbc mean? [00:04] <DrPastah> and SAR and DAR? [00:12] <JEEB> SAR seems to be the pixel aspect ratio set, and DAR seems to be the display aspect ratio of the end result I guess [00:45] <DrPastah> which one is bit rate? [00:45] <DrPastah> Stream #0:0(jpn): Video: h264 (High), yuv420p, 1864x1048, SAR 1:1 DAR 233:131, 23.98 fps, 23.98 tbr, 1k tbn, 47.95 tbc (default) [00:49] <DrPastah> how can i view a file's video bitrate? [01:06] <LunaVorax_win> Hello everyone! [01:07] <LunaVorax_win> I'm a longtime FFMpeg user and I recently heard about the libav fork. I would like to know what's the difference between the two project. [01:10] <LunaVorax_win> I also couldn't ignore some people telling me that they were some tensions between the two project so I precise that it's a real question. I'm not here to troll or anything. [01:16] <JEEB> Just think of it as two projects doing development work, and there being various amounts of merging between them. No real need to go further overall. [01:17] <LunaVorax_win> Hum ok JEEB thanks [01:18] <LunaVorax_win> Oh err [01:18] <LunaVorax_win> I have a technical question about lossless codecs. [01:18] <LunaVorax_win> Is it me or is the "lossless" preset of the x264 codec is just a big joke? [01:19] <JEEB> in what way? [01:19] <LunaVorax_win> The picture is all blurry and the colors spread out of the egdes [01:20] <LunaVorax_win> I've used it on a video games recording in 256colors, I had no other choice to use HuffYUV or CorePNG to have a good result [01:20] <JEEB> it is lossless compared to the 4:2:0/4:2:2/4:4:4 picture that gets input [01:20] <JEEB> (default is 4:2:0) [01:21] <JEEB> that sounds to me what happens when RGB gets converted to 4:2:0 YUV [01:21] <LunaVorax_win> So it's visually lossless or real lossless? [01:21] <LunaVorax_win> Oh [01:21] <LunaVorax_win> Maybe it's that [01:21] <JEEB> it is visually lossless if compared to the input, that is what I mean [01:22] <LunaVorax_win> Ok I see [01:22] <JEEB> also, lately there's a bgr mode in x264 as well [01:22] <LunaVorax_win> Even FFV1 messed up the colors and the black became light, you think it's the same kind of problem? [01:22] <JEEB> uhh [01:22] <JEEB> that sounds like a PC/TV range mismatch somewhere [01:22] <LunaVorax_win> Hum so I'm wrongly using the codecs you mean [01:23] <JEEB> or playing back the output incorrectly, or failing something else [01:23] <JEEB> (or ffmpeg failing something, but it's more often PEBKAC) [01:24] <LunaVorax_win> Ah I haven't checked if it was a wrong playback [01:30] <JEEBsv> anyways, from YUV 4:4:4 would be closest, and then there's the BGR mode which should be one-to-one with 8bit RGB [01:30] <JEEBsv> both supported by libx264 nowadays [01:31] <LunaVorax_win> Hum ok I'll check that [01:31] <JEEBsv> 4:2:0 YUV is what you will have to use for most output tho [01:31] <JEEBsv> as in, what you use for youtube and such [01:38] <LunaVorax_win> Hum ok, thank JEEBsv I'll definitely have to dig that, it's getting pretty complex for me. [01:39] <LunaVorax_win> To be honest, it would have been much simpler if YouTube wasn't messing already with the uploaded videos. [02:39] <lytran> Please tell me how to use segment muxer to create .ts every 10 seconds [03:03] <DrPastah> -bufsize size Set video buffer verifier buffer size (in bits). what does this buffer do exactly? Is it where it waits a certain amount before playing? [03:03] <DrPastah> The docs say this need to be set if I want to change the max bitrate [03:25] <DrPastah> ah nvm [03:35] <lytran> hi [03:35] <lytran> please let me know is there any problem with follow command [03:35] <lytran> ffmpeg -y -i udp://239.30.10.6:5106?reuse=1 -c:v libx264 -c:a libmp3lame -map 0 -f segment -container_format ts -segment_time 10 -segment_pattern %03d -segment_basename out iphonetest.m3u8 [03:35] <lytran> this command cause the error : "Could not write header for output file #0 (incorrect codec parameters ?)" [03:36] <lytran> If I remove container_format this command run [03:36] <lytran> but I cannot create ts segment [03:49] <nd456> can anyone help me to create a timelapse? (out of still images) [06:27] <MrJones_> hi [06:41] <cyrus_mc> if I want to convert a mpeg1 file to mpeg4 what command line would I use. I tried the following ffmpeg -i input.mpeg -vcodec mpeg4 -b 409k -r 25 -acodec mp2 -ab 192k -ar 48000 -ac 2 -async 1 -f dvd -thread 2 -y ~/test.avi [06:41] <cyrus_mc> but what came out only the audio would play [06:42] <MrJones_> is someone with some libav* knowledge here and can spot the problem in this stream initialisation? http://www.homeofjones.de/temp/ffmpeg.txt I end up with samplerate 0 and 0 channels. what do I miss to get a correct initialisation and correct samplerate/channel info? [07:14] <MrJones_> any hint appreciated :p I'm quite new to FFmpeg [09:30] <DrPastah> I'm heaving problems encoding something [09:30] <DrPastah> for some reason the audio cracks [09:30] <DrPastah> and lags [09:30] <DrPastah> what settings can i use to prevent this? [11:19] <zap0> the right ones [12:41] <osearth_> http://www.securityfocus.com/archive/1/521101/100/0/threaded [13:44] <greentemik> Hi! :) Anyone could help me please? [13:45] <greentemik> I am new user and want to compile ffmpeg with v4l support [13:47] <ubitux> should be enabled by default [13:48] <ubitux> if it doesn't support your webcam, you might want to try --enable-libv4l2 for userspace library wrapper [14:04] <jkramer> Hi [14:05] <jkramer> I'm trying to convert an MKV file to MP4 using this command: ffmpeg -i input.mkv -acodec libmp3lame -vcodec copy output.mp4 [14:05] <jkramer> But I'm getting this error: Error while opening encoder for output stream #0.1 - maybe incorrect parameters such as bit_rate, rate, width or height [14:06] <jkramer> Not sure what to do now, I think vcodec copy should just copy the video stream and not do any processing which would require those parameters [14:06] <jkramer> Ah wait, stream #0.1 is the audio stream [14:12] <darkstarbyte> There is a way to do that without reconverting the thing [14:12] <darkstarbyte> I have a script for it [14:13] <darkstarbyte> I have to go [14:13] <jkramer> Mind to share it? :) [14:13] <darkstarbyte> I can later but I have to get to work now. [14:15] <jkramer> Hmm, adding -ac 2 (found on some website) seems to work, but now I'm getting another error [14:15] <jkramer> [mp4 @ 0x86f0dc0] pts < dts in stream 0 [14:15] <jkramer> av_interleaved_write_frame(): Invalid data found when processing input [14:16] <superlinux-hp> hi. how can I concatenate two OGV files? [15:37] Last message repeated 1 time(s). [16:17] <Keyboard_Warrior> superlinux-hp, if you dont care about the ogv [16:17] <Keyboard_Warrior> mkvmerge generally has pretty great merging tools [16:17] <Keyboard_Warrior> i find that in alot of cases [16:17] <Keyboard_Warrior> using mkvmerge to append 1 file to the other [16:17] <Keyboard_Warrior> creates a perfect file with audio, subtitles, sync and everything [16:17] <Keyboard_Warrior> preserved [16:18] <Keyboard_Warrior> if you do, there are more complicated options [16:22] <superlinux-hp> OK [16:57] <anddam> hello [17:00] <anddam> I'm converting wav files to m4a using ffmpeg -i <infile> <outfile> [17:01] <anddam> I met a wav at 11 kHz and I get "Error while opening encoder for output stream #0.0 - maybe incorrect parameters such as bit_rate, rate, width or height " [17:01] <anddam> how do I specify the target sample rate? [17:02] <anddam> I checked the man and aresample= seemed the right option but I get "aresample=44100: Invalid argument " [17:13] <ubitux> anddam: -ar [17:13] <anddam> thanks, that did it [17:25] <anddam> bye [20:28] <Polarfire> hi everyone [20:28] <Polarfire> is this the right place to ask a question? [20:34] <Polarfire> I'll go ahead and ask then..... [20:34] <Polarfire> I use ffmpeg to stream to RTMP servers, and when doing a screencast my audio and video slowly get out of sync [20:35] <Polarfire> streaming a file works fine.... [20:35] <Polarfire> Is there anything I can set that will keep the two channels locked together? [20:36] <Polarfire> I am using x11grab and pulseaudio as my sources [21:05] <jfroebe> I'm writing a script that converts audiobooks (m4b) to mp3 files splitting on the chapters so I can listen to them in my car. Does anyone know if I can attach a thumbnail to the mp3 file created by ffmpeg (libmp3lame acodec) upon creation? [21:08] <jfroebe> I'm trying to keep the dependencies to a minimum if possible.. :) [23:53] <greentemik> Hi! [23:53] <greentemik> anyone could pls help me to compile ffmpeg on freebsd with v4l support? [00:00] --- Sun Jan 8 2012