[Ffmpeg-devel-irc] ffmpeg.log.20130316

[burek]

[00:00] <Zeeflo> 12000KB cache, maxrates was 6000K
[00:00] <Zeeflo> for video
[00:00] <Zeeflo> works great over flash/rtmp
[00:01] <llogan> which rtmp server?
[00:03] <dmonjo> what is better for slow internet connection vp8 ogv or mpegts?
[00:03] <dmonjo> h264
[00:04] <llogan> mpegts is a container format
[00:05] <klaxa> hmmm so is ogv, but ogv implies theora
[00:06] <Zeeflo> llogan, cloudfront
[00:07] <dmonjo> how can i use ffmpeg -f lavfi and vp8 please
[00:07] <dmonjo> cant come up wit hthe syntax
[00:07] <dmonjo> no documentation............
[00:08] <Soukyuu> hello, can someone please help me figuring out why ffmpeg won't mux my .adx file even though ffmpeg -formats lists it as muxable? I get an "input/output" error with avconv and "Truncating packet of size 36 to 19" with ffmpeg
[00:09] <Zeeflo> question: if i have a movie thats around 8000kbit in maxrate, how can I calculate how fast an internet connection needs to be to be able to have it streamed?
[00:09] <Jordan__> do i have to set yuv420p? or is that default
[00:12] <urashidmalik> hi
[00:12] <urashidmalik> i am trying to setup streaming server using ffserver
[00:12] <urashidmalik> able to get http streaming up and running using webm
[00:13] <urashidmalik> but unable to get rtsp (x264/aac working
[00:13] <urashidmalik> they only way i can stream vidio is by disabling audio
[00:13] <urashidmalik> NoAudio
[00:15] <llogan> dmonjo: https://ffmpeg.org/trac/ffmpeg/wiki/vpxEncodingGuide
[00:17] <llogan> Zeeflo: bufsize / maxrate = latency in seconds. maxrate is the minimum transfer rate (for the client) you want to support.
[00:17] <Soukyuu> fflogger, here's the paste: http://pastebin.com/1gaB9GAQ
[00:18] <Zeeflo> llogan, aha. got it!
[00:18] <dmonjo> llogan:
[00:18] <dmonjo> ffmpeg  -f lavfi -i testsrc  -c:v libvpx -b:v 1M -c:a libvorbis http://1.1.1.1:8000/event2.vp8
[00:18] <dmonjo> not working
[00:19] <llogan> Zeeflo: a typical bufsize is 2-5 seconds (depending on who you ask)
[00:19] <Zeeflo> i have this setting?
[00:20] <ubitux> 00:07:49 < dmonjo> no documentation............
[00:20] <Zeeflo> -b:v 8000k -maxrate 8000k -bufsize 16000k
[00:20] <ubitux> dmonjo: wtf.
[00:20] <Zeeflo> i was under the impression that the bufsize should be double of the maxrate ???
[00:20] <ubitux> wtf are you trying to do pushing to a http?
[00:21] <dmonjo> trying to  push from a testsrc to a webserver as vp8 stream
[00:21] <llogan> Soukyuu: does the file decode ok?
[00:21] <dmonjo> i want to push anything to see if the vp8 encoding works fine
[00:21] <Soukyuu> yes, it's playing fine in foobar or ffplay
[00:54] <Demon_Fox> Every time I try to encode an ac3 to the wav for ms
[00:54] <Demon_Fox> I get this error
[00:54] <Demon_Fox> Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height
[00:57] <Demon_Fox> http://dpaste.com/1024280/
[00:59] <mark4o> Demon_Fox: adpcm_ms does not support 5.1 channel.  Are you sure you want adpcm_ms, not just regular pcm?
[00:59] <Demon_Fox> For neroaacenc
[01:00] <mark4o> Do you want uncompressed audio?
[01:01] <Demon_Fox> I wanted to use the nero aac encoder
[01:01] <Demon_Fox> It said to use wav
[01:01] <Demon_Fox> the ms format
[01:02] <Demon_Fox> I will try just ffmpeg -i input.ac3 output.wav
[01:02] <mark4o> yes, or -acodec pcm_s16le
[01:03] <Demon_Fox> I hope it works
[01:03] <Demon_Fox> I wish it would decode flac
[01:06] <Demon_Fox> nope
[01:06] <mark4o> another ffmpeg error?
[01:06] <Demon_Fox> nero error
[01:07] <mark4o> with the pcm_s16le wav file?
[01:08] <Demon_Fox> I think it is working if I pipe it
[01:09] <Demon_Fox> Yeah
[01:09] <Demon_Fox> It works with pipes, and can't read the file
[01:09] <Demon_Fox> Unless it is from stdin
[01:10] <Demon_Fox> Sorry, my bad I did not know this.
[01:11] <mark4o> I think it should work with files but I don't use neroaacenc
[01:11] <Demon_Fox> Well faac is just terrible
[01:12] <Demon_Fox> and celt along with opus are not supported in mkv
[01:12] <mark4o> If you want to encode aac with ffmpeg then use libfdk_aac, not libfaac
[01:12] <Demon_Fox> so I went with the best aac encoder I could find
[01:13] <mark4o> https://ffmpeg.org/trac/ffmpeg/wiki/AACEncodingGuide
[01:47] <taqattack> Can someone help me? I get no audio on livestream if I use "-bsf:a aac_adtstoasc" option (http://pastebin.com/HvH8cFLh)
[03:23] <Demon_Fox> Would I use:
[03:23] <Demon_Fox> ffmpeg -i input.vob -an -vn -scodec srt output.srt
[03:24] <Demon_Fox> to encode subtitles to the srt format?
[03:27] <Demon_Fox> I should google this
[03:49] <Demon_Fox> tried using -vn -an -codec:s:0.1 srt
[03:49] <Demon_Fox> and I got:
[03:49] <Demon_Fox> Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height
[03:49] <Demon_Fox> when taking dvdsub to srt
[03:53] <linger> hello
[03:53] <Demon_Fox> hi
[03:56] <linger> I have a question about 2-pass encoding. I am not sure which options I am supposed to use either in first pass and in the second or only in the first one?
[03:57] <Demon_Fox> There is a chance you might not need 2 pass encoding
[03:57] <Demon_Fox> Do you care if the output size is exact?
[04:00] <linger> I made the experience that witch two pass the quality is a lot better
[04:00] <linger> especially in high motion pictures
[04:02] <linger> e.g the -brd_scale and -bidir_refine option are only useable in the first pass, then the settings will be taken from the log file
[04:03] <linger> would it be the same with for example, the compare options like cmp or mbcmp?
[04:32] <Demon_Fox> linger, I was talking about variable bitrate encoding
[04:33] <Demon_Fox> If you are using x264
[04:33] <Demon_Fox> the option is -crf
[04:34] <Demon_Fox> -crf 18 is visually lossless
[04:35] <Demon_Fox> Usually takes a standard mpeg2 video from a dvd to around half its size
[04:36] <Demon_Fox> With some animations, the compression rate gets better
[04:37] <Demon_Fox> Using a decent preset like veryslow or slower
[04:38] <mudkipz> hello, I got a segmentation fault while using ffmpeg to stream.
[04:39] <Demon_Fox> We would need the output in pastebin form.
[04:39] <mudkipz> It says 1442 Segmentation fault (core dumped)
[04:39] <mudkipz> holdo n
[04:42] <mudkipz> okay, the top is the output it gave out on my command line and then after the -------contents... is the script I use.
[04:42] <mudkipz> http://pastebin.com/V1C61gF3
[04:42] <mudkipz> I removed my usernames and passwords.
[04:43] <mudkipz> most of the script is commented out there, its a very quick testy script.
[04:44] <linger> Demon_Fox, I do variable bit encoding
[04:44] <mudkipz> Im not very knowledgeable in debugging, but if it did a core dump, doesnt that mean it dumped a bunch of info into a file somewhere?
[04:44] <Demon_Fox> linger, 2 pass is fixed rate
[04:45] <Demon_Fox> usually
[04:46] <linger> allright I know what you are explaining
[04:46] <Demon_Fox> What video and audio codecs are you using?
[04:47] <mudkipz> by the way, heres my ffmpeg configuration and version info. http://pastebin.com/pCUYg9kH
[04:47] <linger> I use the mpeg4 v codec
[04:47] <linger> and for audio mp3lame
[04:48] <Demon_Fox> linger, mpeg4 could mean part 2 or part 10
[04:49] <linger> part 2
[04:49] <Demon_Fox> linger, For any reason in particular?
[04:50] <Demon_Fox> I only ask because x264 has a preset that can compete with xvid in speed and size
[04:50] <linger> For compatibility purpose
[04:50] <Demon_Fox> Sorry then, I can't be of any further help.
[04:50] <linger> you help very well
[04:51] <Demon_Fox> If it is for embedded devices they usually have max bit rates before the decoding lags.
[04:53] <linger> the bitrates are not what is cosidering, it is the options in the passes
[04:54] <Demon_Fox> For the first pass you can usually use faster settings
[04:54] <Demon_Fox> The second pass needs the slower settings for compression
[04:55] <linger> that is what I meant
[04:55] <Demon_Fox> From what I understand
[04:56] <Demon_Fox> the first pass is more like mapping of high and low bit rate scenes to know how to drop quality to make it more unnoticeable.
[04:56] <linger> and do I have to trigger for example -bf in both passes or does ffmpeg takes the option from the first pass logfile?
[04:57] <Demon_Fox> Some/many people recommend even using the same settings for both passes.
[04:57] <relaxed> linger: both
[05:02] <linger> thank you
[05:02] <linger> that is what I needed to know
[05:29] <Demon_Fox> Anyone able to rip dvd subtitles to srt?
[06:23] <taqattack> Can someone help me? I get no audio on livestream if I use "-bsf:a aac_adtstoasc" option (http://pastebin.com/HvH8cFLh)
[09:34] <creep> hi
[09:56] <zap0> creep
[09:57] <retard> rude
[09:57] <zap0> retard
[09:57] <creep> o hai retard
[10:11] <dmonjo> there is a bug in hls
[10:12] <dmonjo> it doest stsart streamin unless i kill the ffmpehg command like 2 times
[10:12] <dmonjo> first time ts is 0
[10:12] <dmonjo> secdon time ,3u8 only created
[10:12] <dmonjo> 3times time it works ts start filling up
[10:13] <retard> riveting tale
[10:13] <retard> what's a hls
[10:13] <dmonjo> high level of seduction
[10:14] <retard> oh is this some pua shit
[10:28] <onto> Hi! I am trying to use ffmpeg to record video + audio from the webcam and stream it over RTP/UDP (locally) so that I can use opencv to process the video. I found this resource: http://ffmpeg.org/trac/ffmpeg/wiki/StreamingGuide but will it ensure that the video capture / streaming is lossless?
[10:29] <onto> is there a better way to do it?
[11:12] <dmonjo> http://current.workingdirectory.net/posts/2012/vp8-and-debian/
[11:12] <dmonjo> seems vp8 shout2send can be patched in gstreamer
[11:12] <dmonjo> anyone tried that?
[11:42] <Braden`> Hello
[11:43] <Braden`> I have a file encoded in 264, but I believe that the players are reading it as the wrong encoding.  I just want to rip the audio out.  How can I do that with ffmpeg?
[11:45] <Braden`> Anyone?
[11:46] <cbsrobot_> -vn = no video , -an = no audio
[11:50] <Braden`> err
[11:50] <Braden`> I mean, I want to save the audio and discard the video
[11:52] <Braden`> http://ideone.com/PCkjiB <-- Error I receive
[11:53] <relaxed> Your input does not contain audio.
[11:54] <relaxed> It's raw h264 video
[12:01] <Braden`> No audio at all?
[12:02] <relaxed> correct
[12:07] <Braden`> Is there a way to detect if the file has audio but is mis-encoded?
[12:11] <eladgariany> Braden, try ffprobe -show_streams -print_format json -v quiet file.ext
[12:11] <eladgariany> really helps me understand what file i'm working with
[12:13] <eladgariany> (ps. your input file does not have an audio stream = no output)
[12:13] <Braden`> http://pastebin.com/Fh4zLCyP
[12:14] <eladgariany> as I said, only 1 video stream
[12:15] <Braden`> hrm
[12:47] <xlinkz0> AVIOContext* AVFormatContext::pb
[12:47] <xlinkz0> ... what does pb come from?
[12:47] <xlinkz0> my bad didn't see the newline
[22:21] <Xgates> hi guys & gals :)
[22:21] <Mavrik> evenin
[22:22] <Xgates> what's the simplest cmd to join to avi for ffmpeg? I tried this but didn't work; http://myridia.com/dev_posts/view/197
[22:22] <Xgates> to/two avi...
[22:25] <Mavrik> Xgates, this should help you: http://ffmpeg.org/trac/ffmpeg/wiki/How%20to%20concatenate%20(join,%20merge)%20media%20files#samecodec
[22:26] <Xgates> I tried this and got permission denied;
[22:26] <Xgates> ffmpeg -i concat:sparks-plu2012-xvid.cd1.avi|sparks-plu2012-xvid.cd2.avi -f avi -acodec copy -vcodec copy output.avi
[22:26] <Xgates> -bash: ./sparks-plu2012-xvid.cd2.avi: Permission denied
[22:27] <Mavrik> you should probably use quotes around arguments that are interpreted by bash otherwise ;)
[22:27] <Xgates> ahhh just saw that :) thanks
[22:28] <Xgates> ahhh it's going now :)
[22:33] <Xgates> crap when I try to play it on my smarttv I get audio codec not supported
[22:33] <Xgates> hmm
[22:34] <relaxed> copy the video and encode the audio to mp3
[22:36] <Xgates> so just put -acodec libmp3lame
[22:37] <relaxed> ffmpeg -i input -c:v copy -c:a libmp3lame -ac 2 -b:a 192k output.avi
[22:37] <Xgates> well I'm joining with this cmd;
[22:37] <Xgates>  ffmpeg -i "concat:sparks-plu2012-xvid.cd1.avi|sparks-plu2012-xvid.cd2.avi" -f avi -acodec copy -vcodec copy output.avi
[22:38] <Xgates> so thought I'd change the -acodec copy in that cmd is all?
[22:38] <relaxed> to -acodec libmp3lame -ac 2 -ab 192k
[22:39] <Xgates>  ffmpeg -i "concat:sparks-plu2012-xvid.cd1.avi|sparks-plu2012-xvid.cd2.avi" -f avi -acodec libmp3lame -ac 2 -ab 192k -vcodec copy output.avi
[22:39] <Xgates> like that then?
[22:40] <relaxed> yes
[22:40] <Xgates> thanks relaxed
[22:41] <Xgates> oh one last one, for mp4 or mpg can I run that cmd also to join them?
[22:41] <relaxed> mpg yes, mp4 no
[22:42] <relaxed> for mp4, MP4Box -cat 1.mp4 -cat 2.mp4 -new combined.mp4
[22:42] <Xgates> ahh ok I've used MP4Box
[22:42] <Xgates> I have a little TUT on it too
[22:42] <Xgates> thanks again
[22:43] <relaxed> I usually use mkvmerge for everything but mp4, mkvmerge -o combined.mkv input1.mkv +input2.mkv +input3.mkv +input4.mkv
[22:43] <relaxed> it works with most formats
[22:44] <relaxed> then process with ffmpeg
[22:44] <Xgates> mkvmerge works on avi and mpg?
[22:44] <relaxed> yes
[22:44] <Xgates> ahh I thought it was only for mkv
[22:44] <Xgates> ahhh ok I have a simple little tut on it too
[22:44] <relaxed> ln -s mkvmerge anythingmerge
[22:44] <Xgates> I would think that mkvmerge better than concat?
[22:45] <relaxed> I believe so
[22:45] <Xgates> what do you mean by ln -s as a smylink?
[22:45] <relaxed> it was a joke
[22:45] <Xgates> oh hehe
[22:45] <Xgates> I was like what....
[22:45] <relaxed> (and an unfunny one at that)
[22:45] <relaxed> :-)
[22:47] <relaxed> burek: add an entry to the bot about concat using mkvmerge/MP4Box
[22:48] <relaxed> burek: or better yet, give me the rights and message me on how to do it.
[22:48] <Xgates> ahhh actually my tut was to extract with mkvextract and then join with MP4Box
[22:48] <Xgates>  mkvmerge -o combined.mkv input1.mkv +input2.mkv --- so this is the command and combined is the output?
[22:49] <relaxed> correct
[22:49] <Xgates> sweet thanks I'll give that a go next
[22:53] <Demon_Fox> Is there anything that produces smaller lossless audio output than flac?
[22:54] <BotoX> Hello, I'm running into problems with ffmpeg. I try to stream video via ffserver which is running on a remote server. ffmpeg just sigsev's when running this: ffmpeg http://my-server.tld:8090/feed1.ffm
[22:54] <BotoX> this happens with ffmpeg 1:1.1.3 and the git version
[22:57] <BotoX> okay interesting, with a different ffserver version it seems to work but it gives me a error message about missing audio, let me just fix that.
[22:59] <relaxed> Demon_Fox: I think wavpack may
[22:59] <relaxed> but I doubt it's by very much
[23:00] Action: relaxed thinks JEEB knows
[23:00] <Demon_Fox> thanks
[23:07] <Xgates> relaxed:  this cmd, the first cd part has audio but at the second part that is joined there's no audio ; ffmpeg -i "concat:sparks-plu2012-xvid.cd1.avi|sparks-plu2012-xvid.cd2.avi" -f avi -acodec libmp3lame -ac 2 -ab 192k -vcodec copy output.avi
[23:07] <Xgates> errrrrr
[23:07] Action: Xgates bangs head
[23:09] <Xgates> ok I'm use to channels that up to 4 lines is cool
[23:09] <Xgates> ok...
[23:09] <LithosLaptop> Xgates: bit rates, channels, samples rates etc the same?
[23:10] <llogan> the concat protocol does not work for all inputs. just ones that support file level concatenation (mpeg-1/2, dv, etc)
[23:10] <Xgates> don't really know didn't make it
[23:10] <LithosLaptop> oh
[23:10] <Xgates> does ffmpeg have a cmd that shows this?
[23:11] <llogan> ffmpeg -i input1 -i input2
[23:11] <Xgates> hmm maybe mkvmerge is simpler for joining?
[23:11] <Xgates> thanks
[23:11] <llogan> try the concat demuxer
[23:12] <llogan> http://www.ffmpeg.org/ffmpeg-formats.html#concat
[23:12] <llogan> http://www.ffmpeg.org/ffmpeg-filters.html#join
[23:12] <llogan> https://ffmpeg.org/trac/ffmpeg/wiki/How%20to%20concatenate%20%28join%2C%20merge%29%20media%20files
[23:12] <Xgates> ok the info on both files is the same
[23:12] <llogan> that 2nd url is wrong....i meant http://ffmpeg.org/faq.html#How-can-I-join-video-files_003f
[23:15] <f0x> Hi there, was wondering if anyone could give me some feedback on my current RTMP - > .m3u8 command; I'm running on windows so hence the manual path to the preset. http://pastebin.com/xp2hF1Rx
[23:15] <Xgates> thanks I looked at it before, couldn't understand then what is the concat demuxer cmds I wanted to use...
[23:15] <f0x> Works fine when playing it on VLC but on my phone it lags sometimes (Iphone 4S) and was wondering if my settings could be the cause.
[23:15] <Xgates> ok so converting it to mpg first?
[23:16] <Xgates> ok I'll try the ones below where it says; Additionally, you can use the concat protocol instead of cat or copy
[23:17] <llogan> f0x: the complete console output is missing
[23:18] <f0x> what would llogan , what would you like me to include, there isnt any error anything, its works well; was just wondering if the settings seem correct for phone viewing and that im not doing something that would affect playback.
[23:18] <Xgates> ok I'm trying this, then I get this fail msg; http://pastebin.ca/2333887
[23:25] <llogan> f0x: nevermind then.
[23:48] <Demon_Fox> What does it mean when mpeg outputs this error?:
[23:49] <Demon_Fox> Application provided invalid, non monotonically increasing dts to muxer in stream 0:?
[23:59] <f0x> Demon_Fox - there is no error; perhaps I wasnt clear at first. The command runs fine, the stream is segmented as I would expect. Play back is also 100%. My only issue is playback on my phone (local network) sometimes it hangs for a few seconds and was wondering if it has anything to do with the transcoding settings im using.
[00:00] --- Sun Mar 17 2013