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April 2015
- 1 participants
- 60 discussions
[01:27:02 CEST] <cone-200> ffmpeg 03Andreas Cadhalpun 07master:faf9fe2c224e: alsdec: validate time diff index
[02:38:42 CEST] <cone-200> ffmpeg 03Lou Logan 07master:d1a892209803: cmdutils: indent protocols listing
[03:19:55 CEST] <cone-200> ffmpeg 03James Almer 07master:a40cee03a3be: avutil: remove pointless bmi1 define
[04:33:45 CEST] <cone-200> ffmpeg 03Andreas Cadhalpun 07release/2.5:96c1421627aa: alsdec: ensure channel reordering is reversible
[04:33:46 CEST] <cone-200> ffmpeg 03Michael Niedermayer 07release/2.5:5be683d687b6: avcodec/alsdec: Use av_mallocz_array() for chan_data to ensure the arrays never contain random data
[04:33:47 CEST] <cone-200> ffmpeg 03Andreas Cadhalpun 07release/2.5:faac8e43315d: alsdec: validate time diff index
[08:51:48 CEST] <jamrial> rcombs: wanna be among the first to know when the ticket reaches 400? :p
[08:55:14 CEST] <rcombs> jamrial: that or when something actually happens
[10:37:02 CEST] <enigma> is this the right channel if i have trouble using libav ? or is this strictly for ffmpeg development itself
[10:38:04 CEST] <enigma> I am having trouble muxing H264 bitstream and ADTS AAC streams together into FLV. If i pack H264 bitstream (source is hardware encoder) into FLV , it plays fine with proper duration and bitrate. But when I mux audio along , the bitrate and duration go all wrong and on the playback, I only hear the audio. The video gets stuck on first frame. here is my muxing code : http://pastebin.com/C6bYqtsA
[10:38:56 CEST] <enigma> ok sorry, just read the description. i
[10:39:03 CEST] <enigma> li'll stick to #ffmpeg
[12:57:57 CEST] <cone-149> ffmpeg 03James Almer 07n2.5.6:HEAD: avutil: remove pointless bmi1 define
[13:36:25 CEST] <cone-149> ffmpeg 03Michael Niedermayer 07master:3668701f9600: avformat/http: Return an error in case of prematurely ending data
[14:49:36 CEST] <cone-149> ffmpeg 03Michael Niedermayer 07master:22c0585a00d4: avformat/http: Fix 2 typos
[15:41:32 CEST] <cone-149> ffmpeg 03Michael Niedermayer 07master:9a0f60a0f89a: avcodec/mpeg4videodec: Use check_marker()
[16:45:18 CEST] <cone-149> ffmpeg 03Thomas Guillem 07release/2.4:3e1c9da38b84: matroskadec: fix crash when parsing invalid mkv
[16:45:19 CEST] <cone-149> ffmpeg 03Michael Niedermayer 07release/2.4:0b71cedfe844: Merge commit '3e1c9da38b849ce2982b516004370081fdd89ed0' into release/2.4
[17:00:54 CEST] <cone-149> ffmpeg 03Michael Niedermayer 07master:0cab0931dcf2: avformat/matroskadec: remove now duplicate doctype check
[17:02:29 CEST] <BtbN> It's quite hard to find documentation on chroma keying. No idea if i'm missing something extremely obvious about it, or if all algorithms are proprietary and hidden.
[17:06:57 CEST] <iive> what is chroma keying?
[17:14:42 CEST] <ubitux> wm4: i don't remember what test i asked for...
[17:15:20 CEST] <ubitux> wm4: about the mpv/gif (wtf are your users doing), you can trim with... trim filter in the filtergraph, i suppose
[17:15:52 CEST] <ubitux> (it will send an EOF when it reaches the end bound)
[17:16:28 CEST] <ubitux> for the starting point, you can use it but it will unfortunately not "seek" (just drop frames until it reaches the timestamp)
[17:17:06 CEST] <ubitux> for the --start we indeed have a -ss for input, and one for output
[17:17:13 CEST] <ubitux> so unless you have something equivalent..
[17:18:32 CEST] <wm4> nothing like this was ever necessary (until now=
[17:18:44 CEST] <wm4> and that user is developing a mpv frontend
[17:18:58 CEST] <wm4> so I suppose using mpv for this was an obvious thought
[17:19:03 CEST] <ubitux> i don't understand why your seek isn't exclusively on the input actually
[17:20:17 CEST] <wm4> so you accurately seek on the output
[17:20:27 CEST] <wm4> think about frame doubling deinterlacing and such
[17:21:09 CEST] <BtbN> iive, basicaly, the thing that does greenscreens.
[17:42:21 CEST] <Compn> isnt the vf framestep for that
[17:42:32 CEST] <Compn> skip 1500 frames, then start output
[17:42:40 CEST] <Compn> it would be nice if it had that feature
[17:42:47 CEST] <Compn> instead of outputting 1 frame every 1500 frames.
[17:44:32 CEST] <rcombs> Compn: use the select filter
[18:13:04 CEST] <lglinskih> fate-md5: libavutil/md5-test$(EXESUF)
[18:13:09 CEST] <lglinskih> fate-md5: CMD = run libavutil/md5-test
[18:13:30 CEST] <lglinskih> we don't need EXESUF in the second string?
[18:23:07 CEST] <jamrial> no, you can run windows executables without it
[18:23:41 CEST] <jamrial> but you need to specify it to Make when building them
[18:24:35 CEST] <wm4> yeah, on windows $(EXESUF) would expand to .exe
[18:25:00 CEST] <lglinskih> and what about other OS?
[18:25:10 CEST] <wm4> other systems don't have one
[18:25:19 CEST] <wm4> it'll expand to an empty string, so it'd do nothing
[18:26:17 CEST] <kirelagin> I bet there is an OS that has special extension for executables and wont execute them without extension ;)
[18:27:33 CEST] <kirelagin> But what matters is that ffmpeg probably isnt aiming to support them
[18:31:03 CEST] <lglinskih> is it fine if I just add $(EXESUF)?=)
[18:33:00 CEST] <iive> if binaries are not installed, you can make it to .elf :)
[18:34:44 CEST] <jamrial> lglinskih: there's no need to add it for any "CMD =" line
[20:00:03 CEST] <BtbN> RGB to YUV conversion is confusing, there are multiple formular on the net, which all claim to be exact by definition oO
[20:04:12 CEST] <BBB> BtbN: it depends on what type of yuv
[20:04:29 CEST] <BBB> BtbN: 601/709/smpte240 all have slightly different conversion formulae
[20:04:33 CEST] <BtbN> Whatever ffmpeg has in AV_PIX_FMT_YUV(A)420P
[20:04:33 CEST] <JEEBsv> in general the algo is the same, but then you have coefficients
[20:04:48 CEST] <BBB> BtbN: that depends on other descriptors in avcodeccontext
[20:05:00 CEST] <BBB> BtbN: plus the rgb it is converted into is not identical
[20:05:23 CEST] <BBB> BtbN: this is a really hairy area, you may not care enough to want to know, its like chasing your own tail for a year
[20:05:24 CEST] <BtbN> I'm in a filter, and i get an input color as user parameter, which i need as yuv
[20:05:36 CEST] <wm4> it's in AVFrame now too
[20:05:48 CEST] <wm4> AVFrame.colorspace etc.
[20:06:08 CEST] <BBB> look at AVColorSpace and related enums in libavutil/pixfmt.h
[20:06:09 CEST] <wm4> but in general, if you don't work with YUV but RGB, just make the input RGB?
[20:06:13 CEST] <BBB> they are used in AVCodecContext
[20:06:28 CEST] <BtbN> chromakey is yuv
[20:06:28 CEST] <BBB> wm4: oh cool
[20:06:58 CEST] <BtbN> The only place where i get in touch with RGB is the user parameter
[20:07:05 CEST] <BtbN> AV_OPT_TYPE_COLOR gives me rgba
[20:07:20 CEST] <wm4> maybe it would be nice if there were some utility code to return the correct matrix for a given conversion, instead of making it secret knowledge of libswscale and various standards
[20:07:24 CEST] <wm4> ah
[20:07:36 CEST] <wm4> for chomakey, wouldn't a YUV parameter make more sense?
[20:07:43 CEST] <BBB> Id just call it yuv then, yes
[20:07:44 CEST] <wm4> even if you take user convenience into account
[20:07:52 CEST] <BBB> opengl calls its buffers rgba, but its undefined what is in each channel
[20:07:56 CEST] <BBB> so people happily put yuv in it
[20:08:06 CEST] <BBB> (r=y,g=u,b=v)
[20:08:15 CEST] <BtbN> The problem with that is, that AV_OPT_TYPE_COLOR accepts strings like "green" as input
[20:08:20 CEST] <BBB> (or r=each per plane)
[20:08:21 CEST] <BtbN> which would end up as everything but green
[20:08:43 CEST] <BBB> if you want to convert in your filter, youll need the user to specify the colorspace
[20:08:48 CEST] <BBB> (AVColorspace)
[20:08:58 CEST] <BBB> and then do the conversion according to the correct coefficients
[20:09:03 CEST] <BBB> (or take it from the input yuv stream)
[20:10:29 CEST] <BtbN> Oh, there are RGB_TO_Y/U/V defines in libavutil/colorspace.h
[20:10:58 CEST] <wm4> last time I tried to use them I ended up using something else
[20:11:11 CEST] <wm4> because trying to understand these weird macros took way more effort than NIHing it
[20:11:24 CEST] <wm4> (they are absolutely horrible)
[20:12:11 CEST] <wm4> actually maybe the TO macros are ok
[20:12:24 CEST] <wm4> I tried the macros above that
[20:12:43 CEST] <wm4> which implicitly reference variables you have to declare before the macros etc.
[20:12:46 CEST] <BtbN> That filter will be horrible slow though... No way this will work in realtime.
[20:12:51 CEST] <BBB> I dont actually know which standard these macros follow
[20:13:07 CEST] <BBB> why, you need the conversion more than once per frame?
[20:13:16 CEST] <BtbN> No, the entire filter.
[20:13:18 CEST] <BBB> I mean, if the user input is rgb and you need that in yuv, its just one variable right?
[20:13:22 CEST] <BBB> so you convert it in init
[20:13:25 CEST] <BtbN> The conversion is only needed once at init
[20:13:31 CEST] <BBB> and use the converted value in your speed critical code
[20:13:33 CEST] <BBB> right
[20:13:38 CEST] <BBB> so why is it slow?
[20:13:39 CEST] <BtbN> But the entire chromakey algorithm
[20:13:45 CEST] <BBB> well thats your problem :-p
[20:13:53 CEST] <BBB> and easily simd'ed
[20:14:22 CEST] <BtbN> The algorithm i found uses floats.
[20:14:24 CEST] <BBB> (I foresee a combination of pcmpgtb)
[20:14:29 CEST] <BBB> so write it in its
[20:14:39 CEST] <BBB> ints*
[20:14:47 CEST] <BtbN> I don't think that'd work out propperly.
[20:14:50 CEST] <kierank> why not
[20:14:53 CEST] <BBB> why
[20:14:58 CEST] <kierank> the coefficients are fixed point
[20:15:05 CEST] <BBB> float is just int with an autoshifter
[20:15:16 CEST] <BBB> (sort of)
[20:21:43 CEST] <BtbN> yep, it's doing 4 fps...
[20:22:25 CEST] <BtbN> Converting these algorithms to integer math is absolutely neccesary, but i have no idea how the color distance calculation would work then.
[20:24:06 CEST] <Daemon404> why are you manually doing this
[20:24:21 CEST] <BtbN> hm?
[20:24:28 CEST] <Daemon404> swscale is a thing
[20:24:38 CEST] <BBB> Daemon404: chromakey distance != swscale
[20:24:38 CEST] <BtbN> swscale to convert an algorithm?
[20:24:53 CEST] <Daemon404> ffs i misread
[20:25:10 CEST] <Daemon404> re: chroma key, do you have places for proper blending around edges
[20:25:13 CEST] <BBB> I assume distance is expressed as 1.2*(ypicture - ykey) + 3.7*(upicture - ukey) + 0.2*(vpicture - vkey)?
[20:25:13 CEST] <Daemon404> i.e. feathering
[20:25:24 CEST] <BBB> (where I made up the coeficients)
[20:25:25 CEST] <BBB> ?
[20:25:35 CEST] <BBB> and then some abs in there
[20:25:47 CEST] <BBB> and if distance < threshold alpha=0 else alpha=1?
[20:25:58 CEST] <BBB> (or something roughly along these lines)
[20:26:41 CEST] <BtbN> It's basicaly just sqrt(diff_u² + diff_v²), with diff_u=u-key_u
[20:26:56 CEST] <BtbN> y is ignored
[20:26:59 CEST] <BtbN> for the keying
[20:26:59 CEST] <BBB> diff_u is int
[20:27:01 CEST] <BBB> ok
[20:27:04 CEST] <BBB> diff_v is also int
[20:27:06 CEST] <BBB> sqrt is int
[20:27:11 CEST] <BBB> (or can be int)
[20:27:22 CEST] <BBB> sq is obv int
[20:27:32 CEST] <BBB> so I dont see the issue
[20:27:42 CEST] <BBB> theres a lot of sqrt int algos out there
[20:28:07 CEST] <BBB> theyre not necessarily faster than float in scalar, but since theyre easier to vectorize and their precision is tunable, they tend to be faster in practice
[20:28:08 CEST] <BtbN> The values are in the range from 0-255, so with an integer sqrt it wouldn't be precise enough.
[20:28:19 CEST] <BBB> why
[20:28:33 CEST] <BBB> you can express the result as a fixed-point value
[20:28:38 CEST] <BBB> 0x100 = 1.0
[20:28:40 CEST] <BBB> 0x200 = 2.0
[20:28:43 CEST] <BBB> 0x180 = 1.5
[20:28:44 CEST] <BBB> etc.
[20:28:49 CEST] <BtbN> So basicaly just multiple up everything...
[20:28:50 CEST] <BBB> it can be as precise as you like
[20:28:56 CEST] <BBB> thats fixed point integer math, yes
[20:29:08 CEST] <BtbN> Very annoying. I want this to work with floats first.
[20:29:16 CEST] <BBB> lol ok:)
[20:29:39 CEST] <BtbN> Cause at the moment it's keying everything, so something is still wrong.
[20:29:45 CEST] <BBB> right
[20:29:46 CEST] <BBB> ok
[20:30:03 CEST] <BBB> debug it first, when youre ready for integer feel free to ask any questions here
[20:30:09 CEST] <BBB> a lot of us have experience in that area :)
[20:30:42 CEST] <BtbN> I have also no idea if the algorithm itself is ideal. There isn't much about that topic, except how to do it in various commercial softwares.
[21:24:06 CEST] <nalply> I am new here. AVInputFormat.get_device_list() is not implemented for dshow and avfoundation. I would like to rewrite avdevice so that I can list devices programmatically (i.e. not by executing `ffmpeg -list_devices true -f dshow`). What do you think? Please also see https://trac.ffmpeg.org/ticket/4486.
[21:37:48 CEST] <nevcairiel> if ffmpeg.c can list the devices, then there must be a API to do it programmatically
[21:54:50 CEST] <nalply> Sorry, no. Please read line 685 of libavdevice/avfoundation.m: https://ffmpeg.org/doxygen/trunk/avfoundation_8m_source.html#l00685
[21:56:15 CEST] <nalply> It's in the function avf_read_header(). It's invoked when the input format starts reading from the device.
[21:57:22 CEST] <nalply> It checks whether the option list_device is set then it does not invoke avdevice_list_devices() but immediately queries AVFoundation and puts the information to av_log() calls.
[22:00:48 CEST] <nalply> Similarly for dshow: https://ffmpeg.org/doxygen/trunk/dshow_8c_source.html#l01013
[22:14:42 CEST] <BtbN> Hm, it does _something_, but it keys the wrong color.
[22:17:48 CEST] <nalply> Huh?
[22:30:46 CEST] <BBB> nalply: Im fine with that being an explicit api
[22:30:59 CEST] <BBB> BtbN: well thats a bug in your code :-p
[22:33:50 CEST] <nalply> Ah, thanks. Did you write about ffvp9 last year?
[23:45:25 CEST] <cone-149> ffmpeg 03Michael Niedermayer 07master:93db2708d3b0: ffmpeg: Fix null pointer dereference in do_video_out()
[00:00:00 CEST] --- Mon Apr 20 2015
1
0
[00:42:46 CEST] <Prelude2004c> http://pastebin.ca/2976352
[00:42:56 CEST] <Prelude2004c> hey guys.. can you tell me why those errors would be coming through
[00:43:03 CEST] <Prelude2004c> i am getting the input form a local UDP port
[00:43:25 CEST] <Prelude2004c> there is no congestion.. any way to turn the packets from source to TCP with ffmpeg without transcoding?
[00:43:36 CEST] <Prelude2004c> i don't have the house power on the other server so i am trying to just forward the packets
[00:45:47 CEST] <klaxa> you can use http if you are using a single client and a single server
[00:46:31 CEST] <klaxa> Prelude2004c: https://ffmpeg.org/ffmpeg-all.html#http
[00:46:35 CEST] <klaxa> see: listen option
[00:47:47 CEST] <klaxa> or tcp: https://ffmpeg.org/ffmpeg-all.html#tcp
[01:49:42 CEST] <proshot> question, i compiled ffmpeg on a armv7 system, should i enable -mfpu=neon for neon to be detected
[02:26:33 CEST] <sine_> whats the current syntax to ignore and leave out the video container/codec
[02:26:40 CEST] <sine_> used to be -vn
[02:26:50 CEST] <c_14> still is
[02:27:07 CEST] <sine_> oh, ok.
[02:30:23 CEST] <sine_> how do i get it to use all the cores of the puc
[02:30:25 CEST] <sine_> cpu
[02:31:56 CEST] <c_14> -threads 0 or so usually
[02:36:10 CEST] <sine_> whats the default
[02:36:24 CEST] <c_14> Depends on the encoder
[02:36:24 CEST] <sine_> -threads 0 i assume means use as many as available
[02:36:57 CEST] <c_14> afaik -threads 0 means use fancy magic to try and saturate the cpu
[06:11:16 CEST] <fling> Is multiple framerates supported formultiangle?
[06:30:24 CEST] <Prelude2004c> hey guys.. looking at a ts file.. is there some way to see all the settings that constrcutred that file ? eg.. coder 1 or yadif on or off.. etc etc
[06:30:27 CEST] <Prelude2004c> is there some way to see it
[06:45:31 CEST] <jcastle> ffprobe?
[06:45:42 CEST] <jcastle> It dumps the stream info.
[07:31:28 CEST] <pjunkie> I'm trying to cut a chunk of a file while re-encoding from mpeg2 to h264, using -ss and -to, with both specified after the input filename.
[07:32:01 CEST] <pjunkie> from what I can tell from the docs and from what I can find on Google, putting -ss after the input filename instead of before (as long as you're not using copy mode) should cut exactly at that timestamp
[07:32:17 CEST] <pjunkie> but in trying it, it seems to still be cutting at the nearest previous keyframe instead
[07:33:20 CEST] <pjunkie> is it really supposed to be frame-accurate and I'm doing something wrong (in which case what info can I provide to help debug), or am I just out of luck?
[07:33:46 CEST] <pjunkie> it's version 2.6.2 self-compiled running on CentOS 7
[08:38:02 CEST] <jookiyaya> what happens if you have 5.1 surround audio on 2.0 system/speakers
[09:22:03 CEST] <rkromer> jookiyaya: My guess is that you are not going to be satisfied with a static downmix, because often this does overrule what the creator intended.
[10:27:52 CEST] <enigma> i am having trouble muxing H264 bitstream and ADTS AAC streams together into FLV
[10:28:52 CEST] <enigma> If i pack H264 bitstream (source is hardware encoder) into FLV , it plays fine with proper duration and bitrate. But when I mux audio along , the bitrate and duration go all wrong and on the playback, I only hear the audio. The video gets stuck on first frame
[10:30:58 CEST] <enigma> here is my muxing code : http://pastebin.com/C6bYqtsA
[11:46:44 CEST] <jookiyaya> i found a huge bug with .mkv
[11:46:55 CEST] <jookiyaya> who do i report it to?
[12:22:46 CEST] <bencc1> can ffmpeg application written in C mix live streams of audio/video conference where streams are added and remove dynamically?
[12:22:58 CEST] <bencc1> can it handle queues, synching and buffering for me?
[12:49:14 CEST] <tt99> anyone here who is good at programming and has knowledge of vbr mp3 headers, please msg me
[13:11:38 CEST] <seasc> tt99, you might have better chances to ask straight
[13:18:20 CEST] <tt99> well its about recording in vbr mp3 and edit afterwards to get the mp3 nicely but than the vbr mp3 header gets fucked. Instead of razorlaming mp3's (re-encode) I am looking for a tool to fix the vbr header. Its possible and I think I have the codecs. Need someone to built and check
[13:33:35 CEST] <seasc> How to fix a VBR (Variable Bit Rate)? Its been encoded to switch dynamicly the bitrates where required / possible. Its not really something you can EDIT (afaik)..
[13:37:31 CEST] <seasc> By dynamicly i mean during encoding, not afterwards, obviously. @ tt99
[13:37:53 CEST] <jookiyaya> what happens if you have 5.1 surround audio on 2.0 system/speakers
[13:39:51 CEST] <seasc> usualy, you hear only either front or backsides, or just left or just right, depends on the system / config
[13:41:02 CEST] <jcastle> https://trac.ffmpeg.org/wiki/AudioChannelManipulation#a5.1stereo
[13:41:12 CEST] <seasc> i would suggest to downcode the audio (-ac 2) but copy the videostream (-c:v copy) (remains better quality)
[14:17:37 CEST] <Guest61567> hi
[14:18:40 CEST] <Guest61567> my ffmpeg hang for [https @ 0x9d6d000] header='Server: gvs 1.0' [https @ 0x9d6d000] header='' Why?
[14:19:38 CEST] <Guest61567> it production server http://pastebin.com/UAf19thz
[14:22:14 CEST] <Guest61567> it http://pastebin.com/ddfgN7ai local host, good job. Why on development server doesn not job?
[14:22:48 CEST] <klaxa> how long did you wait?
[14:22:48 CEST] <klaxa> not sure if -ss is passed to http as a range-request
[14:22:53 CEST] <klaxa> can you monitor network throughput? it might be downloading the file and trying to skip 2000 seconds
[14:28:44 CEST] <Guest61567> klaxa, wait slowly
[14:29:01 CEST] <Guest61567> klaxa, for local host it job good
[14:29:26 CEST] <Guest61567> On productions server doesn not job :(
[14:29:56 CEST] <klaxa> i can see that, but can you answer my question? and monitor your network throughput?
[14:30:39 CEST] <Guest61567> klaxa, how it make?
[14:30:45 CEST] <klaxa> oh wait, you are running the command on two different computers but are trying to get the same data
[14:30:48 CEST] <klaxa> now i see what you want
[14:31:02 CEST] <Guest61567> yes
[14:31:06 CEST] <Guest61567> thanks
[14:31:08 CEST] <klaxa> you can use a tool like iftop or bwm-ng or nload or just watch ifconfig
[14:31:31 CEST] <Guest61567> ifconfig i use
[14:31:51 CEST] <Guest61567> run ifconfig?
[14:31:54 CEST] <klaxa> run it once before you run the command
[14:33:13 CEST] <klaxa> note down the data received (rx) so far
[14:33:15 CEST] <klaxa> run your ffmpeg command
[14:33:16 CEST] <klaxa> and then check ifconfig again to see if it downloaded a lot of data
[14:33:17 CEST] <klaxa> it would actually be easier if you could just check the "live bandwith" with one of the three other tools i mentioned
[14:33:29 CEST] <Guest61567> klaxa, but local host job fast, very fast and speed low
[14:34:46 CEST] <klaxa> as a test, can you skip the -ss 2000 parameter?
[14:34:54 CEST] <klaxa> and maybe add -t 10 or something
[14:35:29 CEST] <Guest61567> before RX packets:1240579 errors:0 dropped:70936 overruns:0 frame:0 after RX packets:1253189 errors:0 dropped:70960 overruns:0 frame:0
[14:36:31 CEST] <klaxa> was that on your local machine or on the remote machine?
[14:36:40 CEST] <Guest61567> remote
[14:36:42 CEST] <klaxa> and did the command finish or did you abort it?
[14:37:05 CEST] <klaxa> can you try skipping -ss 2000?
[14:37:52 CEST] <klaxa> from the http header you can tell that ffmpeg is downloading the whole file
[14:38:57 CEST] <Guest61567> klaxa, it sometimes jobe
[14:38:58 CEST] <Guest61567> d
[14:39:51 CEST] <Guest61567> -ss 10 , but hang
[14:40:19 CEST] <klaxa> for how long?
[14:40:21 CEST] <klaxa> forever?
[14:40:32 CEST] <semeion> what parameter put ffplay on top of all windows in desktop? I have tried -video-on-top
[14:41:03 CEST] <klaxa> semeion: i'm pretty sure that's window-manager dependent
[14:41:45 CEST] <Guest61567> klaxa, forever
[14:42:10 CEST] <semeion> yes, i know, but ffplay donŽt make it?
[14:42:35 CEST] <klaxa> how long did your forever last? and like i said, can you maybe monitor your network throughput?
[14:42:41 CEST] <semeion> i am starting with aaplay now
[14:43:23 CEST] <semeion> he have a usefull -fs to put fullscreen
[14:43:28 CEST] <Guest61567> klaxa, still
[14:44:11 CEST] <Guest61567> klaxa, install iftop
[14:45:07 CEST] <Guest61567> klaxa, installed and runed
[14:51:10 CEST] <klaxa> does iftop display traffic when you run the ffmpeg command?
[14:53:25 CEST] <Guest61567> klaxa, show mbites
[14:53:55 CEST] <Guest61567> increasing
[14:54:03 CEST] <klaxa> sorry battery is dying
[14:54:03 CEST] <klaxa> gotta go
[14:54:56 CEST] <Guest61567> klaxa, what me make?
[14:57:07 CEST] <Guest61567> klaxa, what me make?
[15:01:26 CEST] <Guest23670> klaxa, for local host down 5mb and succed job
[15:25:30 CEST] <Guest23670> klaxa, you here?
[17:07:38 CEST] <shevy> on an .avi file I have this
[17:07:40 CEST] <shevy> Stream #0:0: Video: mpeg4 (Simple Profile) (XVID / 0x44495658)
[17:08:00 CEST] <shevy> is this now a mpeg4 file or an xvid .avi? or is that both? I am not quite familiar with the term mpeg4 there
[17:08:26 CEST] <JEEBsv> xvid is an encoder that produces video adhering to the MPEG-4 Part 2 standard
[17:08:38 CEST] <JEEBsv> MPEG-4 has like 30 parts by now
[17:09:14 CEST] <JEEBsv> so it is probably AVI if you say so container-wise, and that container contains an MPEG-4 Part 2 track with the track identifier "XVID"
[17:11:49 CEST] <shevy> hmm
[17:12:06 CEST] <JEEBsv> the container/whatever is listed before that as input format or whatever
[17:12:31 CEST] <JEEBsv> and yes, the "mpeg4" naming inside libavcodec is by now kind of vague, yes
[17:12:44 CEST] <JEEBsv> but that's the name for mpeg-4 part 2 inside lavc
[17:12:52 CEST] <JEEBsv> (because there wasn't anything else back then)
[17:13:35 CEST] <shevy> I am still a bit confused
[17:13:43 CEST] <shevy> is h264 also a mpeg4 format?
[17:14:00 CEST] <JEEBsv> yes, AVC (that's the ISO/IEC name) is mpeg-4 part 10
[17:14:05 CEST] <shevy> aaah ok
[17:14:15 CEST] <shevy> that makes a lot more sense to me then, thanks JEEB__
[17:14:21 CEST] <shevy> erm JEEBsv ... too many jeebs...
[17:14:35 CEST] <JEEBsv> part 12 is the new specification for "mp4"
[17:14:43 CEST] <JEEBsv> part 15 is how you put AVC and HEVC into part 12 :D
[17:15:07 CEST] <JEEBsv> for HEVC they actually decided to make a new group of formats
[17:15:09 CEST] <JEEBsv> MPEG-H
[17:15:34 CEST] <JEEBsv> which is kind of understandable since you had like 30 of MPEG-4 things
[17:15:35 CEST] <shevy> oh man
[17:15:39 CEST] <shevy> all those formats
[17:16:10 CEST] <shevy> yeah 30... it'll get better https://xkcd.com/927/
[17:16:51 CEST] <JEEBsv> I need to find the page that lists all of the multimedia related MPEG groups and their specs
[17:17:35 CEST] <opl> hello?
[17:18:08 CEST] <JEEBsv> https://www.itscj.ipsj.or.jp/sc29/29w42911.htm
[17:18:09 CEST] <JEEBsv> ah
[17:18:11 CEST] <JEEBsv> here we go
[17:18:50 CEST] <opl> i want to save an rtmp stream from vaughnlive to file, but the stream cuts after exactly after 8 seconds with error "rtmp server requested close"
[17:20:10 CEST] <Guest23670> hi all, help me?
[17:21:54 CEST] <Guest23670> it remote server http://pastebin.com/UAf19thz and it http://pastebin.com/ddfgN7 local server. On remote server does not job. What me make?
[17:41:50 CEST] <opl> anyone active?
[20:58:56 CEST] <Fyr> guys, what container is used for opus?
[20:59:36 CEST] <Polarwindz> I need some help please :)
[21:00:21 CEST] <Polarwindz> I was wondering how to burn in subtitles using mpv and ffmpeg
[21:02:21 CEST] <rkromer> Fyr: ogg for sure; I guess Matroska not yet
[21:02:35 CEST] <Fyr> thanks
[21:03:12 CEST] <c_14> Polarwindz: https://trac.ffmpeg.org/wiki/HowToBurnSubtitlesIntoVideo
[21:03:44 CEST] <Polarwindz> Does it keep the fonts/style of the subs?
[21:03:59 CEST] <c_14> If you use the subtitles filter with the mkv as input, yes.
[21:04:25 CEST] <Polarwindz> which is? I use vsfilter and avisynth with megui.
[21:04:31 CEST] <Polarwindz> My friends needs help so
[21:05:12 CEST] <c_14> The first one listed on that site I linked.
[21:05:35 CEST] <Polarwindz> Thank you, I heard that mpv with ffmpeg can keep the fonts/style of the subtitles
[21:07:06 CEST] <llogan> Fyr: .opus
[21:07:29 CEST] <Fyr> llogan, I couldn't find anything about this format.
[21:07:54 CEST] <Fyr> I wanted to know if it keeps pictures and id3v2 tags.
[21:40:50 CEST] <llogan> Fyr: i don't know. doesn't seem like the ffmpeg implementation does (you should doublecheck though...i was lazy)
[21:40:55 CEST] <llogan> see https://tools.ietf.org/html/draft-ietf-codec-oggopus-06
[21:41:25 CEST] <Fyr> yep
[21:41:26 CEST] <Fyr> =(
[21:41:55 CEST] <llogan> it wouldn't "keep" tags as id3v2 but ideally convert them to Vorbis comment.
[21:42:52 CEST] <llogan> but album art is currently problematic with ffmpeg (at least with things that should use vorbis comment)
[23:01:37 CEST] <t4nk028> hi
[23:05:42 CEST] <t4nk028> someone help me with ffmpeg?
[23:37:45 CEST] <sine> c_14: are you here fella
[23:38:06 CEST] <c_14> yeah, why?
[23:38:21 CEST] <Guest88130> well, your normally good for a random question or two :D
[23:40:02 CEST] <sine22> im no pro with ffmpeg but im learning bits as i need them. I have a few random clips of video of various framerates. home videos nothing special, I want them all to be 25fps. is there a way of converting them easily using the same video codec, sort of like videokeep but change framerate
[23:41:12 CEST] <c_14> You can do it with a short script in any scripting language really. Just ffprobe the input, check what codec it is and then encode to that.
[23:42:26 CEST] <sine22> ffmpeg -i clip.mp4 to check ?
[23:42:44 CEST] <sine22> i have not used ffprobe
[23:42:57 CEST] <c_14> ffprobe is usually better because you can change the output format so it's easier to parse; but that'll work as well
[23:55:25 CEST] <sine22> how do i set the fps
[23:55:44 CEST] <c_14> -vf fps=25
[23:55:51 CEST] <sine22> I am using -c:v libx264 -crf 28
[23:56:05 CEST] <sine22> I am using -c:v libx264 -crf 28 -vf fps=25
[23:56:52 CEST] <sine22> will that convert 30fps to 25fps so that it fits within the same timespace
[23:57:01 CEST] <c_14> should
[23:57:44 CEST] <sine22> ok trying that out. cheers
[00:00:00 CEST] --- Mon Apr 20 2015
1
0
[00:14:51 CEST] <cone-263> ffmpeg 03wm4 07master:e2691b809271: hls: skip to next segment if the current is unavailable
[01:15:22 CEST] <rcombs> it seems I've just implemented BFSTM support in lavf
[01:16:28 CEST] <rcombs> though it breaks on the last block
[01:18:04 CEST] <rcombs> this is some excellent Mario Kart 8 music
[01:20:22 CEST] <jamrial> haha
[01:20:44 CEST] <jamrial> i implemented a couple game audio formats some time ago, like RSD
[01:21:06 CEST] <jamrial> and so did paul
[01:21:32 CEST] <rcombs> yeah, BFSTM is basically a new header format for BRSTM
[01:21:43 CEST] <rcombs> so I just tweaked brstm.c, which is Paul's
[01:21:58 CEST] <jamrial> cool
[01:22:04 CEST] <rcombs> still have some hacks to remove
[01:22:13 CEST] <jamrial> does it use the same adpcm codecs?
[01:22:15 CEST] <rcombs> and I need to work out why decoding the last block is failing
[01:22:17 CEST] <rcombs> yup
[01:22:50 CEST] <rcombs> referenced vgstream (winamp plugin with support for the two) to work out the format differences
[01:23:49 CEST] <rcombs> the only samples I have are from Mario Kart 8, which only uses AV_CODEC_ID_ADPCM_THP
[01:24:08 CEST] <rcombs> which naturally is the codec that involves the most work
[01:25:11 CEST] <rcombs> http://puu.sh/hhCs4/c839eedb1c.png whoooo uninitialized variables
[02:56:08 CEST] <fluffrabbit|away> BBB: vpx requires MSYS. I'm not a developer anymore. Cut me some slack. I think it's probably safe to assume that the steady degradation of framerate is NOT the fault of libvpx.
[02:56:55 CEST] <jamrial> wm4: foobar2000 for example uses libavcodec's mp3 float decoder
[02:57:25 CEST] <jamrial> but they use their own demuxer for the container and not libavformat
[02:58:29 CEST] <jamrial> libavcodec for mp3, aac and vorbis. other libraries for anything else
[03:26:45 CEST] <radio_guy> Hi - anyone here who can help me with a copyright issue. May be I discovered a ffmpeg related copyright infringement.
[03:30:15 CEST] <cone-530> ffmpeg 03Michael Niedermayer 07master:80fa48a413a0: avformat/utils: Use codec_info_duration_fields in fps calculation end decission when no timestamps and thus no duration_count is available
[03:30:15 CEST] <cone-530> ffmpeg 03Michael Niedermayer 07master:edbb9b5e8fe5: avformat/utils: Add HEVC to unreliable_tb()
[04:05:08 CEST] <Prelude2004c> hey quick question guys.. i have a source multicast ( UDP ) ... i am trying to traverse from one network to another .. its a direct fiber 10 gig link.. but i am showing artifacts on the conversion to h264... so basically i ffmpeg -i < input > vcopy and acopy and -f mpegts udp://destination:xxxx
[04:05:36 CEST] <Prelude2004c> then on destination:xxx i do ffmpeg -i localhost:xxxx transcode -f HLS
[04:05:48 CEST] <Prelude2004c> how come i see artifacts every once in a while.. likely due to the UDP right?
[04:06:04 CEST] <Prelude2004c> i don't have a lot of power on the source so how do i set it to TCP so all packets always arrive
[04:20:37 CEST] <kierank> you can't multicast tcp
[04:20:45 CEST] <kierank> you need to fix your network such that it doesn't drop packets
[04:53:02 CEST] <Compn> Prelude2004c : what kind of artifacts ?
[04:53:26 CEST] <Compn> increase keyframes
[04:53:34 CEST] <Compn> then you wont need to worry about it so much :P
[04:57:52 CEST] <lordkrondor> use traceroute & mtr to figure it out heh
[06:35:50 CEST] <Timothy_Gu> How hard would it be to implement an SSIM filter like the existing PSNR one?
[16:33:09 CEST] <rkromer> Is anyone out there working on implementing the CineForm codec?
[16:36:58 CEST] <kierank> rkromer: afaik no
[16:37:08 CEST] <kierank> if a spec appeared on the mailing list I think it would help
[16:37:28 CEST] <kierank> and some test vectors
[16:39:17 CEST] <rkromer> The available specs are sadly not complete.
[16:40:20 CEST] <kierank> rkromer: really?
[16:40:24 CEST] <kierank> smpte playing games again?
[16:40:30 CEST] <rkromer> Especially not regarding the lossless flavor.
[16:41:29 CEST] <rkromer> I guess I must do some reverse engineering, as I would use it.
[16:41:49 CEST] <kierank> rkromer: do you have the specs?
[16:43:55 CEST] <rkromer> Yep, both the horribly expensive documents by SMPTE.
[16:44:47 CEST] <kierank> I am rather fed up of expensive SMPTE documents being produced that are incomplete
[16:44:53 CEST] <kierank> I need to make an angry blog post about it
[16:46:30 CEST] <rkromer> The lossless codec version interests me for archival purposes. (I am not a production or post-production person.)
[16:46:39 CEST] <kierank> I see
[16:53:29 CEST] <kierank> rkromer: can you make a list somewhere of all the deficiencies?
[16:57:56 CEST] <rkromer> I am working on it. The "standard" use is quite well documented, but the most interesting lossless flavour only... in a side note. I am caricaturing just a little.
[17:02:22 CEST] <kierank> I think vc-5 doesn't cover lossless explicitly
[17:02:58 CEST] <rkromer> In fact, I noticed its existence by doing a mistake.
[17:08:31 CEST] <rkromer> At GoPro I could not gather useful information so far. They answer to download the GoPro Studio...
[17:09:17 CEST] <kierank> yeah they won't help you
[17:12:30 CEST] <rkromer> In fact we have a free codec without a free documentation, i.e. a proprietary codec.
[17:14:10 CEST] <kierank> well it's patented too which the american archiving people don't like
[17:18:55 CEST] <rkromer> I am Swiss but I don't like neither this pseudo-free concept.
[17:19:56 CEST] <kierank> neither do I
[17:20:26 CEST] <kierank> What annoys me more is when they say the codec is standardised but the standards are not complete
[17:20:38 CEST] <kierank> and as an independent implementor you have no way of reporting bugs
[17:20:53 CEST] <kierank> At least in mpeg if you report a bug on the mailing list someone will listen
[17:20:58 CEST] <kierank> but smpte doesn't have any way of doing that
[17:22:52 CEST] <rkromer> Hmm... Should I include this as an exercise for my students? They are often much better skilled as I am!
[17:24:38 CEST] <kierank> Could write a paper about "Just because something is standardised, doesn't mean it's independently implementable"
[17:25:26 CEST] <rkromer> :-)
[17:25:58 CEST] <rkromer> I'll check how difficult it would be to implement this codec into libavcodec next semester. Thanks a lot for the chat!
[17:53:39 CEST] <cone-200> ffmpeg 03Michael Niedermayer 07master:e739cbb2bb91: avcodec/cabac: remove START/STOP_TIMER debug/benchmark code
[18:35:43 CEST] <cone-200> ffmpeg 03Mariusz SzczepaDczyk 07master:e623e8cbb033: lavf: add documentation on directory listing API
[18:41:59 CEST] <Compn> rkromer : kostya is working on cineform
[18:42:09 CEST] <Compn> theres a patch posted to libav-devel
[18:42:15 CEST] <Compn> for cfhq i think
[18:43:15 CEST] <Compn> hmm where did it go
[18:43:38 CEST] <Compn> oh no, i'm thinking of canopus hqa
[18:43:39 CEST] <Compn> my bad
[18:43:41 CEST] <Compn> :((((((((((
[18:45:42 CEST] <Compn> rkromer : but you should contact kostya , he does some RE'ing on some codecs. http://codecs.multimedia.cx/?p=419
[18:47:07 CEST] <Compn> if cineform uses wavlets, thats going to be confusing and difficult as hell to implement.
[18:47:18 CEST] <Compn> wavelet codecs are more difficult...
[18:48:45 CEST] <jamrial> i thought kostya quit developing for multimedia projects
[18:50:32 CEST] <Compn> once a reverse engineer, always a reverse engineer.
[18:50:55 CEST] <Compn> then theres maxim person who also does some codec engineering
[18:53:18 CEST] <Compn> and mike melanson does (or did) some work on writing video specs down
[18:58:30 CEST] <rkromer> Compn: Thanks a lot!
[19:01:35 CEST] <Compn> jamrial : also probably the bink devel trolling kostya helped him into doing some more work on bink :D
[19:01:49 CEST] <Compn> development by trolling
[19:20:12 CEST] <cone-200> ffmpeg 03James Almer 07master:6863249b347c: aaccoder: use put_sbits()
[19:20:13 CEST] <cone-200> ffmpeg 03James Almer 07master:b5564e041bdc: dcaenc: use put_sbits()
[19:20:14 CEST] <cone-200> ffmpeg 03James Almer 07master:979b77eec5db: dca_xll: use av_clip_intp2()
[19:20:15 CEST] <cone-200> ffmpeg 03James Almer 07master:778bac278869: adpcm: use av_clip_intp2()
[19:20:16 CEST] <cone-200> ffmpeg 03James Almer 07master:d8c52f5d04fc: proresenc_anatoliy: use put_sbits()
[19:37:00 CEST] <BtbN> Which RGB pixel format is the most common?
[19:38:42 CEST] <nevcairiel> bgr0/bgra probably, since that maps to windows rgb32 format =p
[19:39:22 CEST] <cone-200> ffmpeg 03Andreas Cadhalpun 07master:ef16501aebed: alsdec: ensure channel reordering is reversible
[19:56:11 CEST] <BtbN> What's the difference between get_video_buffer and filter_frame in an AVFilterPad? Some filters seem to implement both and the the same work in each, others only implement one of them.
[20:29:36 CEST] <wm4> just pick what seems simpler
[20:30:01 CEST] <wm4> I think there was a lot of microoptimization in lavfi that made everything more complex, though most is gone now
[20:30:08 CEST] <wm4> (like the awful buffer permissions stuff)
[20:32:46 CEST] <thardin> what I know of lavfi is that it's hard to use for simple stuff like just filtering a buffer
[20:32:50 CEST] <thardin> or has that improved?
[20:33:37 CEST] <Daemon404> no.
[20:39:23 CEST] <wm4> thardin: well, it's like a FIFO, so yes you don't just call a function which filters an AVFrame and returns it immediately
[20:39:41 CEST] <wm4> the most annoying thing about lavfi is all the boiler plate needed to set it up, though
[20:45:57 CEST] <rcombs> s/lavfi/all ffmpeg libs/
[20:53:28 CEST] <Daemon404> ^
[20:54:33 CEST] <BtbN> Hm, does YUV even carry any transparency information?
[20:54:46 CEST] <nevcairiel> there is YUVA formats
[20:55:05 CEST] <BtbN> But they don't seem to be present in ffmpeg, specialy not for the common YUV420 formats
[20:55:36 CEST] <nevcairiel> YUVA420 should exist, afaik
[20:56:14 CEST] <BtbN> AV_PIX_FMT_YUVA420P indeed
[20:56:38 CEST] <BtbN> I wonder if the transparency information in that is propperly parsed, by for example the overlay filter.
[20:57:22 CEST] <BtbN> looks like it does
[20:57:43 CEST] <cone-200> ffmpeg 03Michael Niedermayer 07master:7e104647a355: avcodec/alsdec: Use av_mallocz_array() for chan_data to ensure the arrays never contain random data
[21:11:05 CEST] <thardin> wm4: yeah
[21:11:42 CEST] <thardin> because what I've found is I just want a function like filter(inbuffer, outbuffer, width, height, pix_fmt, "filtername")
[21:12:14 CEST] <cone-200> ffmpeg 03Andreas Cadhalpun 07master:4c2b88678b43: alsdec: limit avctx->bits_per_raw_sample to 32
[21:16:22 CEST] <nevcairiel> unfortunately such simple filter functions dont work for quite a bunch of filters, so any generic filtering framework will need a M:N in/out FIFO-like setup
[21:18:20 CEST] <wm4> but it'd be possible to write such a function
[21:19:04 CEST] <wm4> why can't lavfi just take all the parameters from an AVFrame...
[21:21:08 CEST] <cone-200> ffmpeg 03Thomas Guillem 07master:b8d7f3186e86: matroskadec: fix crash when parsing invalid mkv
[21:21:09 CEST] <cone-200> ffmpeg 03Michael Niedermayer 07master:f98cad87590e: Merge commit 'b8d7f3186e86234f6255f5e8ee9e98573b4d9a6e'
[21:27:06 CEST] <nevcairiel> doesnt it mostly do that these days
[21:31:17 CEST] <nevcairiel> although for building the graph you need to know some information sometimes
[21:54:50 CEST] <cone-200> ffmpeg 03Paul B Mahol 07master:daf73d5d1701: avcodec/g729dec: support decoding packets with multiple frames
[21:54:51 CEST] <cone-200> ffmpeg 03Paul B Mahol 07master:df7c2b2f1b75: avformat/riff: support G729A
[22:09:31 CEST] <BtbN> Any comments on this? https://github.com/BtbN/FFmpeg/commit/8b5d20f5cdfc71749ac21c3e5a52a30e0aa92…
[22:09:42 CEST] <lglinskih> kierank, wm4: do I really need to use framecrc, or running my program and using its return code as a result of a fate-test is okay?
[22:09:43 CEST] <BtbN> vf_colorkey filter
[22:10:03 CEST] <BtbN> Propably far from optimal, but it works and should get things started
[22:10:13 CEST] <llogan> users ask for that one on occassion
[22:10:41 CEST] <BtbN> yep
[22:10:56 CEST] <BtbN> Well, they more specificly ask for chroma key, which is what lets you do greenscreens.
[22:11:03 CEST] <BtbN> This is just simple color key in RGB.
[22:11:48 CEST] <BtbN> But i think that's not too complicated to do, will look into it tomorrow.
[22:12:00 CEST] <BtbN> There are quite a few diffrent algorithms on the net.
[22:17:36 CEST] <wm4> lglinskih: fate tests can be all different
[22:17:44 CEST] <wm4> framecrc is just a helper used by some tests
[22:17:58 CEST] <wm4> what you want should work
[22:18:11 CEST] <wm4> fate just needs to be extended
[22:19:02 CEST] <wm4> see fate-run.sh
[22:19:10 CEST] <wm4> or was it fate.sh
[22:19:17 CEST] <Timothy_Gu> fate-run.sh
[22:19:34 CEST] <Timothy_Gu> fate.sh is for submitting results to fate.ffmeg.org etc.
[22:19:50 CEST] <Timothy_Gu> and yes documentation on this area is sorely missing
[22:21:16 CEST] <llogan> Timothy_Gu: perhaps you can persuade a teacher that it would be an acceptable assignment for "technical writing".
[22:21:35 CEST] <llogan> grades4code
[22:24:51 CEST] <jamrial> lots of fate tests just run a program and its return value defines if the tests passes or not
[22:25:04 CEST] <jamrial> you don't need to expand fate-run.sh for that
[22:25:29 CEST] <jamrial> see for example libavcodec.mak
[22:25:54 CEST] <jamrial> cabac-test, golomb-test, dct-test
[22:26:07 CEST] <lglinskih> wm4, Timothy_Gu: I've already read it, and want to know is it a good way, before I implement it =)
[22:28:39 CEST] <wm4> lglinskih: I think possibly the tests should be a bit more flexible (like taking input arguments and printing results, and this can be used in fate itself to test multiple interesting cases), but you probably don't have to worry about such ideas yet
[22:33:48 CEST] <llogan> damn it. forgot version.
[22:37:44 CEST] <Timothy_Gu> llogan: lol, or maybe I can just write it myself.
[22:38:41 CEST] <llogan> you have to make the system work for you
[22:43:01 CEST] <Timothy_Gu> llogan: thing is the student would probably have to take more time trying to understand the FATE system first than writing about it
[22:45:35 CEST] <llogan> Timothy_Gu: i meant that you would write it, and it can be considered homework or whatever
[22:54:45 CEST] <Timothy_Gu> llogan: oh haha i don't think my english classes have such assignments, at least not right now
[23:09:04 CEST] <cone-200> ffmpeg 03James Almer 07master:5cd216934406: wavpackenc: use put_sbits()
[23:42:52 CEST] <Timothy_Gu> llogan: 1-hour attempt: http://imgh.us/FATE.svg
[23:56:45 CEST] <llogan> Timothy_Gu: it looks nice, but it confuses me, and some of the boxes look too small for my ugly fonts: http://avmule.com/junk/fate.png
[00:00:00 CEST] --- Sun Apr 19 2015
1
0
[00:28:38 CEST] <techtopia> anyone ever downloaded netflix videos with ffmpeg?
[00:30:37 CEST] <techtopia> on other streaming video sites, i would tell ffmpeg to open the m3u8 file, and have the m3u8 file edited to point to the key locally
[00:30:53 CEST] <techtopia> with netfilx i just have the video fragments and playready drm keys
[00:31:14 CEST] <techtopia> but can't get ffmpeg to stich it back together and decrypt it
[00:39:17 CEST] <__jack__> techtopia: just x11grab hum ?
[00:42:27 CEST] <techtopia> i don't want to do a screen cap __jack__
[00:42:46 CEST] <techtopia> i have the encrypted video and i have the key so i would like to decrypt it and dump it out to an mp4
[00:43:07 CEST] <techtopia> should be higher quality than a screen cap
[00:43:12 CEST] <techtopia> if i can get it to work
[02:03:47 CEST] <sim590> how do you remove from XX:XX:XX to YY:YY:YY of an audio file ? I've tried combinations of -to and -ss, but without success.
[02:04:08 CEST] <sim590> I want to remove segments of audio.
[02:08:08 CEST] <techtopia> i wouldn't use ffmpeg for it
[02:08:22 CEST] <sim590> techtopia: what would you use ?
[02:08:37 CEST] <techtopia> is the audio alone or muxed into a video?
[02:09:02 CEST] <techtopia> i make cuts with a program called video redo
[02:09:12 CEST] <sim590> it's audio only
[02:09:15 CEST] <techtopia> then encdoe video with ffmpeg and audio with besweet
[02:09:31 CEST] <sim590> Actually, I see that I can do -f concat -i filelist.txt -c copy joinedfile.mp3
[02:09:55 CEST] <techtopia> you want to join files?
[02:10:01 CEST] <techtopia> i thought you wanted to cut
[02:10:28 CEST] <sim590> I want to do this [======music-to-keep=====\\\\\cut-here\\\\\====music-to-keep=====]
[02:10:35 CEST] <techtopia> copy /b file1.mp3 + file2.mp3 fileout.mp3
[02:11:21 CEST] <sim590> concataning direclty like that won't break any header or I don't know ?
[02:11:34 CEST] <sim590> I thought it would be better if handled by ffmpeg..
[02:11:43 CEST] <techtopia> yeah ok
[02:11:53 CEST] <techtopia> your probably right
[02:14:17 CEST] <techtopia> so somthing like this
[02:14:19 CEST] <techtopia> ffmpeg -ss 00:00:00 -i inputfile.mp3 -ss 00:03:33 -acodec copy outputfile.mp3
[02:14:46 CEST] <techtopia> first -ss is start time and needs to be specified before the -i input file
[02:15:04 CEST] <techtopia> second one is the output and needs to be specified after the -i input file
[02:19:02 CEST] <sim590> techtopia: where is your command cutting ? I'm trying to understand.
[02:21:34 CEST] <techtopia> well your telling ffmpeg to open the inputfile at 00:00:00, in this case "inputfile.mp3" and you are telling it to write it back out until 00:03:33, by just copying the source to "outputfile.mp3"
[02:21:53 CEST] <techtopia> so then you just chane your start time and end time and repeat for all your cuts
[02:30:14 CEST] <sim590> ok. I've got all my cuts. I'm trying to join them with a fade between each cuts.
[02:32:41 CEST] <techtopia> i got know clue how to do that :p
[02:34:18 CEST] <sim590> thanks. I'll just do without. I've tried something. It faded out, but didn't fade in after...
[03:02:47 CEST] <Prelude2004c> hey everyone
[03:03:00 CEST] <Prelude2004c> anyone know how to basically copy the entire transport stream and push it out udp to another destination?
[03:03:07 CEST] <Prelude2004c> i have a transport stream with 4 differnet program id's
[03:03:14 CEST] <Prelude2004c> i want to just forward that to another location to process
[03:03:28 CEST] <Prelude2004c> i tried -acopy and -vcopy and it only seems to grab the first program id
[03:03:35 CEST] <c_14> Prelude2004c: did -map 0 not work?
[03:03:40 CEST] <Prelude2004c> how does one copy them all at once
[03:03:47 CEST] <Prelude2004c> no map 0 just put everything into program ID 1
[03:04:19 CEST] <Prelude2004c> i want to keep same original structure
[03:04:48 CEST] <c_14> maybe -map p ?
[03:05:37 CEST] <c_14> There's a short section on program ids here https://ffmpeg.org/ffmpeg.html#Stream-specifiers-1
[03:05:41 CEST] <c_14> not sure if that's helpful
[03:09:29 CEST] <Prelude2004c> yup, i can select one program at a time but then i have to basically create 4 ffmpeg sessions for 4 channels
[03:09:39 CEST] <Prelude2004c> i wanted to just copy everything and forward basically the UDP content on
[03:09:46 CEST] <Prelude2004c> without changing its structure
[03:11:48 CEST] <c_14> Selecting all the programs merges them together?
[03:33:10 CEST] <radio_guy> Hi - anyone here who can help me with a copyright issue? May be I discovered a ffmpeg related copyright infringement.
[03:54:08 CEST] <Prelude2004c> hey, anyone know the best way to get the UDP packets to a differnet place.. i have a direct fiber connect from one point to another and i am doin gffmpeg -i < input > -- copy copy -f mpegts udp://another destionaton
[03:54:20 CEST] <Prelude2004c> problem is while watching the video i see artifacts sometimes as in missing stuff
[03:54:27 CEST] <Prelude2004c> not sure why.. because its UDP ?
[03:54:41 CEST] <Prelude2004c> what is the best way to solve it ?
[03:58:11 CEST] <techtopia> it couldn't write the packets fast enough
[04:01:46 CEST] <Prelude2004c> write packets fast enough?
[04:01:48 CEST] <Prelude2004c> was that for me ?
[04:04:24 CEST] <techtopia> yeah
[04:06:01 CEST] <techtopia> i do a lot of network recording of transport streams
[04:06:24 CEST] <techtopia> if you have network congestion or set a second recording going you experience what you described
[04:06:27 CEST] <techtopia> [02:54:20] <Prelude2004c> problem is while watching the video i see artifacts sometimes as in missing stuff
[04:07:54 CEST] <Prelude2004c> hum.. its a direct fiber with 10 GIG link
[04:07:59 CEST] <Prelude2004c> point A > B
[04:08:15 CEST] <Prelude2004c> there is traffic on the link of course but... how much.. well not that much
[04:08:29 CEST] <Prelude2004c> any way to fix it ? can it be tcp intead .. or something
[04:35:17 CEST] <hazardous> hi, is it possible to set a custom keep-alive in the ffmpeg requests to a rtmpt server?
[10:57:53 CEST] <__jack__> techtopia: well, drm are made to make your life bored, I don't think you'll bypass it easily; x11grab can give you the best quality, if you've powerfull enough computer (quality == raw)
[19:08:24 CEST] <crised__> Any java wrapper around thisÇ
[19:08:26 CEST] <crised__> Ç
[19:08:43 CEST] <crised__> Any java wrapper around ffmpeg?
[19:50:52 CEST] <kate_12> Hi,
[19:51:40 CEST] <kate_12> I have a problem which I am sure many of you would have faced. I am using scale filter with -1 as height but it throws me the not divisible by 2 error. T
[19:52:07 CEST] <kate_12> Here is my command: ffmpeg -i video.mp4 -vf "scale:1280:-1" out.mp4
[19:52:26 CEST] <kate_12> Is there any workaround for this?
[19:52:38 CEST] <kate_12> I read about trunc()
[19:52:47 CEST] <kate_12> but I don't where to use it.
[19:55:53 CEST] <kate_12> hi hi
[19:55:54 CEST] <kate_12> anyone?
[20:09:28 CEST] <daggs1> greetings, got a question, I'm trying to capture my screen in a c file, the fmt I'm using is PIX_FMT_BGRA, but I keep getting this error when trying to caputre: [rawvideo @ 0x24c7aa0] Invalid buffer size, packet size 1228800 < expected frame_size 8294400, what is the proper way to set the packet size do it can hold the frame size?
[21:21:12 CEST] <omegas69> My ffmpeg build keeps crashing in windows.
[21:21:21 CEST] <omegas69> Just before the process starts.
[21:21:29 CEST] <omegas69> I get the ffmpeg.exe has stopped working message.
[21:25:26 CEST] <llogan> omegas69: where did you get the build? is it a recent one, or some ancient thing?
[21:26:20 CEST] <omegas69> It is a recent build. And everything was working fine, it stopped working all of a sudden,
[21:26:35 CEST] <omegas69> I'll share the command, give me a sec.
[21:27:40 CEST] <DeadSix27> anyone an idea what: http://pastie.org/private/mjoh3zqmtaokoas8fdra
[21:27:41 CEST] <DeadSix27> means?
[21:28:04 CEST] <omegas69> ffmpeg -i 3.mp4 -i 1.mp4 -i 2.mp4 -filter_complex "[0:v]fade=t=out:st=2:d=0.5[v0];[1:v]fade=t=in:st=0:d=0.5,fade=t=out:st=2:d=0.5[v1];[2:v]fade=t=in:st=0:d=0.5,fade=t=out:st=2:d=0.5[v2];[v0][v1][v2]concat=n=3:v=1:a=0,format=yuv420p[v]" -map [v] -c:v libx264 -y out.mp4 2>log.txt
[21:28:37 CEST] <omegas69> In the console, the app crash message pops up right before the process starts
[21:28:48 CEST] <DeadSix27> http://pastie.org/private/t3ngovq4grd9fzvaelgea
[21:28:54 CEST] <omegas69> After the stream start message
[21:34:15 CEST] <llogan> omegas69: so you're saying there is no console output at all?
[21:34:46 CEST] <Prelude2004c> hey guys.. been struggling with ffmpeg quality issues for a few weeks. I am in need of some assistance. I would like to pay someone who is very familiar with video quality outputs and knows how to tune ffmpeg to reach the desired output of some live streams. Please let me know if anyone here is interested.
[21:36:13 CEST] <omegas69> Here is it:
[21:36:14 CEST] <omegas69> http://pastebin.com/nQJU465j
[21:36:18 CEST] <llogan> Prelude2004c: can you explain in more detail of what you're doing?
[21:36:36 CEST] <llogan> omegas69: your command is missing. they should be in the same paste
[21:37:32 CEST] <omegas69> there you go
[21:37:32 CEST] <omegas69> http://pastebin.com/WTuz1Ud9
[21:38:35 CEST] <llogan> omegas69: that's an old build
[21:38:57 CEST] <omegas69> Could you give me the link to the latest build?
[21:39:09 CEST] <llogan> http://ffmpeg.zeranoe.com/builds/
[21:39:12 CEST] <omegas69> What's bothering me is, this command was working fine an hour ago.
[21:39:33 CEST] <llogan> what is telling you that "ffmpeg has stopped working"?
[21:40:45 CEST] <omegas69> windows
[21:40:51 CEST] <omegas69> ffmpeg.exe has stopped working
[21:41:37 CEST] <llogan> i don't know much about windows. only think i can recommend is a recent build and do the usual Windows thing of rebooting
[21:42:11 CEST] <omegas69> yeah. thanks, llogan!
[21:42:17 CEST] <omegas69> appreciate it.
[21:42:38 CEST] <llogan> if it continues to misbehave maybe it can be narrowed down after you try that
[21:42:57 CEST] <omegas69> what should be my next step if that fails?
[21:43:12 CEST] <llogan> what has changed since an hour ago?
[21:43:27 CEST] <llogan> does any command cause the issue? certain inputs?
[21:43:48 CEST] <llogan> how do you know ffmpeg has actually "stopped"?
[21:44:02 CEST] <llogan> what if you simplify the command until you find the culprit?
[21:44:04 CEST] <omegas69> well, I scaled down these input files and then it started happening.
[21:44:18 CEST] <omegas69> it was working fine before the scaling.
[21:44:42 CEST] <llogan> does it work on another computer?
[21:45:13 CEST] <omegas69> the console has no output whatsoever, what I sent you was logged into the log.txt file using stdio stream.
[21:45:29 CEST] <omegas69> as soon as I hit enter in the cmd, it crashes and I get that message from windows.
[21:45:38 CEST] <omegas69> I do not have access to any other computer for now.
[21:46:19 CEST] <llogan> did it actually create a working output file?
[21:46:40 CEST] <omegas69> Yeah, but the size of that file was only a few bytes
[21:48:09 CEST] <llogan> i don't know...try my crappy recommendations first.
[21:49:39 CEST] <omegas69> I'll do that. Appreciate your help. I actually came here hoping to get a solution from you. Someone specifically recommended llogan at ffmpeg channel. :p
[21:49:54 CEST] <llogan> sorry to dissapoint.
[21:50:33 CEST] <petrvs> what would you do to extract dvd subs, -c:s copy foo.vob ?
[21:51:14 CEST] <omegas69> I am not disappointed. I was actually appreciating your quick response.
[21:51:15 CEST] <omegas69> Thanks.
[21:51:20 CEST] <omegas69> I am gonna go and try
[21:52:57 CEST] <petrvs> llogan: what'd you disappoint him with? :p
[21:53:16 CEST] <llogan> not having an answer. Windows.
[21:53:36 CEST] <llogan> also i have a slight hanover stupor
[21:53:37 CEST] <petrvs> what'd he want?
[21:53:41 CEST] <petrvs> :)
[21:54:10 CEST] <llogan> windows is telling him "ffmpeg has stopped working".
[21:54:39 CEST] <llogan> i recommended getting a recent build and rebooting. seems like a Windows solution to me.
[21:55:15 CEST] <petrvs> ah, yeah
[21:55:19 CEST] <petrvs> hate using windows
[21:55:21 CEST] <petrvs> so unreliable
[21:56:22 CEST] <petrvs> how would I specify a stream is the "english" audio track, or the "english" subtitle file?
[21:57:05 CEST] <petrvs> -metadata:s:s:# language=foo for subtitles?
[21:57:11 CEST] <petrvs> and for audio... -metadata:s:a?
[21:57:16 CEST] <llogan> yes, i think so
[21:57:39 CEST] <petrvs> that first s was throwing me off :) thanks
[21:57:43 CEST] <petrvs> will try
[21:57:46 CEST] <llogan> i think there is a zample in the docs
[21:58:27 CEST] <petrvs> trouble was I was only finding a sample for one or the other, and the s was confusing me, think I've decoded it in my head now
[21:58:30 CEST] <petrvs> will see in a mo
[21:58:43 CEST] <llogan> the first s is stream. otherwise it would set to global, IIRC
[22:05:49 CEST] <llogan> ffmpeg -f lavfi -i testsrc=d=5 -metadata creation_time=now time.mp4 && ffmpeg -i time.mp4
[22:06:01 CEST] <llogan> anyone else get a weird time? in my case it's creation_time : 2036-02-06 06:28:16
[22:08:31 CEST] <klaxa> i have the same timecode
[22:08:42 CEST] <klaxa> >creation_time : 2036-02-06 06:28:16
[22:08:47 CEST] <klaxa> version 2.6.1
[22:08:58 CEST] <klaxa> it's not the latest git HEAD
[22:09:08 CEST] <llogan> it must be when the world ends
[22:09:18 CEST] <petrvs> undoubtedly
[22:10:08 CEST] <klaxa> i'll try with the latest git
[22:10:19 CEST] <klaxa> will take a while to compile
[22:10:25 CEST] <c_14> I have ditto with ed2a712
[22:16:13 CEST] <llogan> appending "Z" apparently doesn't change a thing. http://ffmpeg.org/ffmpeg-utils.html#Date
[22:16:25 CEST] <llogan> <--- too lazy to look into it more
[22:18:37 CEST] <petrvs> so ffmpeg doesn't do .idx & .sub files?
[22:18:45 CEST] <petrvs> for extraction or inclusion?
[22:19:29 CEST] <petrvs> or would it take -i foo.sub and I can omit the .idx?
[22:19:42 CEST] <llogan> 2**32 secs from 1900/01/01 - 00:00:00 GMT
[22:34:00 CEST] <klaxa> my toaster finished compiling, same results with latest git
[22:34:35 CEST] <llogan> welcome to 2015.
[22:34:57 CEST] <llogan> wot CPU?
[22:35:26 CEST] <klaxa> >Intel(R) Core(TM)2 Duo CPU U9600 @ 1.60GHz
[22:35:33 CEST] <llogan> anyway, bug reported. thanks for testing y'all.
[22:35:59 CEST] <klaxa> i'm trying to get a new laptop by the end of the month
[22:36:17 CEST] <klaxa> working on hardware from 2009 just doesn't cut it anymore
[22:36:27 CEST] <llogan> what will you get? i still use an old netbook on occassion.
[22:37:35 CEST] <klaxa> i'm not entirely sure yet, but it will have to have an i7, at least 8 gb ram and an ssd
[22:46:03 CEST] <Prelude2004c> hey llogan, sorry for the delay
[22:46:15 CEST] <Prelude2004c> i am taking a live feed and transcoding it to HLS
[22:46:28 CEST] <llogan> you lost me at live and then at HLS
[22:46:37 CEST] <Prelude2004c> lol
[22:46:46 CEST] <Prelude2004c> taking an over the air feed and transcoding it into HLS segments
[22:46:52 CEST] <Prelude2004c> h264
[22:46:52 CEST] <llogan> i'm not much of a straming user
[22:46:57 CEST] <Prelude2004c> oh
[22:47:10 CEST] <Prelude2004c> ya i need someone who is good at streaming
[22:47:17 CEST] <llogan> x264 with VBV
[22:47:26 CEST] <Prelude2004c> yup.. variable
[22:47:31 CEST] <Prelude2004c> 5 - 6 Mbit/s
[22:47:51 CEST] <Prelude2004c> i can't seem to get the quality right.. i have tried soo many different things
[22:48:02 CEST] <Prelude2004c> i have a link of someone who is doing it for much less bit rate it seems
[22:48:05 CEST] <Prelude2004c> and the quality is amazing
[22:48:08 CEST] <Prelude2004c> i dont know how they do it
[22:48:20 CEST] <Prelude2004c> http://ams-lp3.9c9media.com/hls-live/livepkgr/_definst_/liveeventNoDRM/cp24…
[22:48:22 CEST] <llogan> can you ask them?
[22:48:24 CEST] <Prelude2004c> open up taht in vlc
[22:48:28 CEST] <Prelude2004c> no i dont know who put it up
[22:50:28 CEST] <llogan> lots of static stuff. do you have similar content?
[22:51:20 CEST] <Prelude2004c> yup similar
[22:53:35 CEST] <llogan> sounds like you're re-encoding a lossy input
[22:53:45 CEST] <Prelude2004c> lossy input ?
[22:53:59 CEST] <llogan> "transcoding a live feed"
[22:54:09 CEST] <Prelude2004c> yup
[22:54:20 CEST] <llogan> i'm assuming it is already somewhat crappy looking before you do anything?
[22:54:28 CEST] <Prelude2004c> no, its pretty clear
[22:54:43 CEST] <Prelude2004c> the source is the same
[23:02:58 CEST] <Prelude2004c> so what is the best way to find the right person to help ?
[23:03:22 CEST] <llogan> you can try the ffmpeg-user mailing list
[23:03:27 CEST] <llogan> or wait here
[23:04:45 CEST] <Prelude2004c> how do i get the ffmpeg-user mailng list
[23:04:49 CEST] <Prelude2004c> do you know what it is ?
[23:04:58 CEST] <llogan> http://ffmpeg.org/contact.html
[23:07:46 CEST] <seasc> Heyas, when i want to stream my screen (linux) to a friend over the internet, do i need to have librtmp enabled with ffmpeg?
[23:08:01 CEST] <llogan> ffmpeg has native rtmp support
[23:09:17 CEST] <seasc> my ffmpeg doesnt report any --enable-rtmp (or librtmp)
[23:09:42 CEST] <llogan> ffmpeg -protocols
[23:11:30 CEST] <seasc> thanks logan, so i could 'leave' this out when compiling ffmpeg and it should still support rtmp? Just to be sure.
[23:11:47 CEST] <llogan> yes
[23:11:57 CEST] Action: seasc dances
[23:12:45 CEST] <seasc> i am music now... **sing sing*
[00:00:00 CEST] --- Sun Apr 19 2015
1
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[00:36:22 CEST] <jamrial> michaelni: are you ok with applying the proper version of the dcadec patch?
[00:50:44 CEST] <michaelni> i have no opinion on the pkg config stuff, also i would avoid "proper implying the other being not so proper" wording the whole debate is heated enough already
[00:53:46 CEST] <jamrial> i said proper because it was the one oked by the author of the wrapper
[00:53:57 CEST] <jamrial> but ok, i'll keep that in mind
[00:55:27 CEST] <michaelni> thx
[01:12:47 CEST] <cone-630> ffmpeg 03James Almer 07master:8254011b2173: configure: remove old libdcadec check
[05:24:31 CEST] <cone-355> ffmpeg 03Vignesh Venkatasubramanian 07master:1c37848f9029: webmdashenc: Add better error handling
[12:56:35 CEST] <cone-263> ffmpeg 03Michael Niedermayer 07master:38f67260684a: avcodec/atrac3plusdsp: fix on stack alignment
[14:34:46 CEST] <cone-263> ffmpeg 03Michael Niedermayer 07master:8768f8f4b929: avcodec/mpegaudiodec_template: use double to build csa tables
[14:44:17 CEST] <cone-263> ffmpeg 03Michael Niedermayer 07release/2.2:1c14b09caf90: avcodec/atrac3plusdsp: fix on stack alignment
[15:06:07 CEST] <rcombs> hmm, trying to work out how best to export whether or not a TrueHD stream has an ATMOS sub stream
[15:06:36 CEST] <rcombs> could be a profile, but I don't know if that really makes sense
[15:10:14 CEST] <cone-263> ffmpeg 03Michael Niedermayer 07n2.2.15:HEAD: avcodec/mpegaudiodec_template: use double to build csa tables
[15:11:25 CEST] <rcombs> or stick it in metadata
[15:12:39 CEST] <Daemon404> rcombs, http://www.reactiongifs.com/r/btt.gif
[15:12:44 CEST] <Daemon404> since were thinking of places to stick it.
[15:12:57 CEST] Action: Daemon404 runs
[15:45:47 CEST] <cone-263> ffmpeg 03Michael Niedermayer 07release/2.5:a376ef4a17ed: avcodec/msrledec: restructure msrle_decode_pal4() based on the line number instead of the pixel pointer
[15:45:48 CEST] <cone-263> ffmpeg 03Michael Niedermayer 07release/2.5:2f67222780be: avutil/pca: Check for av_malloc* failures
[15:45:49 CEST] <cone-263> ffmpeg 03Rainer Hochecker 07release/2.5:071eb56a6a50: h264: avoid unnecessary calls to get_format
[15:45:50 CEST] <cone-263> ffmpeg 03Michael Niedermayer 07release/2.5:52ba406b94d6: avcodec/h264: Fail for invalid mixed IDR / non IDR frames in slice threading mode
[15:45:51 CEST] <cone-263> ffmpeg 03Michael Niedermayer 07release/2.5:76b289bcf2c5: avcodec/h264_refs: Do not set reference to things which dont exist
[15:45:52 CEST] <cone-263> ffmpeg 03Michael Niedermayer 07release/2.5:043cb40becb2: ffmpeg: Fix extradata allocation
[15:45:53 CEST] <cone-263> ffmpeg 03Michael Niedermayer 07release/2.5:8968de6c6193: avformat/utils: avoid discarded streams in av_find_default_stream_index()
[15:45:54 CEST] <cone-263> ffmpeg 03Michael Niedermayer 07release/2.5:48f616ceeecd: avcodec/h264: Fix race between slices where one overwrites data from the next
[15:45:55 CEST] <cone-263> ffmpeg 03Michael Niedermayer 07release/2.5:7fa861dfe06e: avcodec/h264: finish previous slices before switching to single thread mode
[15:45:56 CEST] <cone-263> ffmpeg 03Michael Niedermayer 07release/2.5:f96fdb46b773: avcodec/h264_slice: Dont reset mb_aff_frame per slice
[15:45:57 CEST] <cone-263> ffmpeg 03Michael Niedermayer 07release/2.5:fe2df122b4c3: avcodec/h264: reset the counts in the correct context
[15:45:58 CEST] <cone-263> ffmpeg 03Michael Niedermayer 07release/2.5:3e3042496120: avcodec/aacdec: Fix storing state before PCE decode
[15:45:59 CEST] <cone-263> ffmpeg 03Michael Niedermayer 07release/2.5:9333ee7c0db1: avcodec/h264: Be more tolerant to changing pps id between slices
[15:46:00 CEST] <cone-263> ffmpeg 03Michael Niedermayer 07release/2.5:6f02d93c0fbf: avcodec/h264_ps: Move truncation check from VUI to SPS
[15:46:01 CEST] <cone-263> ffmpeg 03Michael Niedermayer 07release/2.5:8c0fd44317b0: avcodec/h264: Do not fail with randomly truncated VUIs
[15:46:02 CEST] <cone-263> ffmpeg 03Michael Niedermayer 07release/2.5:74fb9df48b12: swresample: Allow reinitialization without ever setting channel counts
[15:46:03 CEST] <cone-263> ffmpeg 03Michael Niedermayer 07release/2.5:84fedd3be768: swresample: Allow reinitialization without ever setting channel layouts (cherry picked from commit 80a28c7509a11114e1aea5b208d56c6646d69c07)
[15:46:04 CEST] <cone-263> ffmpeg 03Michael Niedermayer 07release/2.5:07ea57c5bbbe: swresample: Check channel layouts and channels against each other and print human readable error messages
[15:46:05 CEST] <cone-263> ffmpeg 03Michael Niedermayer 07release/2.5:41b15817ce1e: swresample/swresample-test: Randomly wipe out channel counts
[15:46:06 CEST] <cone-263> ffmpeg 03Andreas Cadhalpun 07release/2.5:9a241d95efa1: msrledec: use signed pixel_ptr in msrle_decode_pal4
[15:46:07 CEST] <cone-263> ffmpeg 03Andreas Cadhalpun 07release/2.5:621f1a2e6364: aacdec: consistently use avctx for logging in decode_eld_specific_config
[15:46:08 CEST] <cone-263> ffmpeg 03Andreas Cadhalpun 07release/2.5:d846d3e88efd: aasc: return correct buffer size from aasc_decode_frame
[15:46:09 CEST] <cone-263> ffmpeg 03Andreas Cadhalpun 07release/2.5:1d0f9c92c550: aacpsy: avoid psy_band->threshold becoming NaN
[15:46:10 CEST] <cone-263> ffmpeg 03Andreas Cadhalpun 07release/2.5:b420354a8b66: ac3: validate end in ff_ac3_bit_alloc_calc_mask
[15:46:11 CEST] <cone-263> ffmpeg 03Michael Niedermayer 07release/2.5:0683ef4b50c4: avcodec/atrac3plusdsp: fix on stack alignment
[16:15:55 CEST] <wm4> michaelni: CBR seeking actually breaks gapless... again
[16:15:59 CEST] <wm4> probably fixable, but still
[16:16:09 CEST] <wm4> rcombs: what do you need to strip it for?
[16:16:27 CEST] <rcombs> wm4: save space; handle dumb decoders
[16:17:51 CEST] <wm4> AFAIK it's designed to be skipped by unaware decoders; ffmpeg did not do this correctly until recently though
[16:18:52 CEST] <rcombs> dingdingding
[16:19:16 CEST] <rcombs> I have to support some old ffmpegs for hysterical raisins
[16:26:03 CEST] <wm4> ah, lol
[16:26:14 CEST] <wm4> I suppose in theory a bsf could be added
[16:26:43 CEST] <saste> rcombs, any news about the segment patch?
[16:27:11 CEST] <saste> I want you to provide some info about the use case in the commit log
[16:35:54 CEST] <rcombs> saste: yes, saw, just need to get around to doing it
[16:40:36 CEST] <Daemon404> g 42
[16:48:56 CEST] <cone-263> ffmpeg 03Michael Niedermayer 07release/2.5:8a59dbbc68f7: Update or 2.5.6
[16:49:42 CEST] <rcombs> 2.5.6 \o/
[17:00:41 CEST] <wm4> michaelni: so due to codec internals, the mp3 decoder (like many other DCT based decoders, apparently) do not produce output for half a frame or so; shouldn't this silence be skipped?
[17:01:56 CEST] <kierank> wm4: that's a matter for debate iirc
[17:02:14 CEST] <kierank> and whole papers written about it
[17:03:01 CEST] <wm4> well, the gapless stuff cuts it away
[17:03:13 CEST] Action: Daemon404 has bad flashbacks to opus integration
[17:03:23 CEST] <wm4> if you start from the beginning (this appears to be explicitly designed this way in ffmpeg's mp3 code)
[17:03:41 CEST] <wm4> but if you seek in the middle, you get some silence at the start
[17:04:09 CEST] <Daemon404> i dont know why you would use lavc for music albums in the first place...
[17:04:13 CEST] <Daemon404> but thats just me
[17:04:37 CEST] <wm4> because I don't want to use hundreds of broken decoder libs?
[17:04:43 CEST] <wm4> which don't solve this anyway
[17:04:46 CEST] <Daemon404> well i wouldnt use mpv for music
[17:05:10 CEST] <wm4> most wouldn't use it for video either
[17:05:45 CEST] <Daemon404> heh
[17:07:53 CEST] <wm4> Daemon404: every audio player uses some popular mp3 lib like libmpg123 or libmad, and then hacks their own "container" handling around it
[17:07:59 CEST] <wm4> well, open sores audio player
[17:08:44 CEST] <Daemon404> meh
[17:21:57 CEST] <iive> libmad is commercially popular.
[17:47:51 CEST] <cone-263> ffmpeg 03Shivraj Patil 07master:35a7170e69b3: Makefile: Add support for MSA (MIPS-SIMD-Arch)
[17:47:52 CEST] <cone-263> ffmpeg 03Shivraj Patil 07master:4efc0e6451fa: avcodec/mips: MSA (MIPS-SIMD-Arch) optimizations for HEVC horizontal and vertical mc functions
[18:16:35 CEST] <cone-263> ffmpeg 03wm4 07master:92eef26e67ea: avformat/mp3dec: fix gapless audio when seeking in CBR mode
[18:48:11 CEST] <cone-263> ffmpeg 03wm4 07master:2a0aa318f813: fate: gapless: fix a typo in comment
[18:48:12 CEST] <cone-263> ffmpeg 03wm4 07master:49d5c24aa150: fate: gapless: test seeking to a specific position
[18:48:13 CEST] <cone-263> ffmpeg 03wm4 07master:12da8bc38dca: fate: gapless: test mp3 both with and without TOC
[19:33:34 CEST] <fluffrabbit> Is VP9 encoding speed temporal degradation a bug?
[19:36:50 CEST] <fluffrabbit> almost feels like a memory leak or something, and I know that this is not a new issue with FFMPEG, as it has been seen with other codecs
[19:49:04 CEST] <fluffrabbit> I've caught other bugs like this before so really I'd just like somebody else to test and confirm what I'm saying.
[19:50:09 CEST] <fluffrabbit> VP9 only got documentation on the FFMPEG wiki very recently so IDK if you're still considering it "beta" or whatever but it's a codec I've been watching for a long time.
[19:51:38 CEST] <fluffrabbit> Try this: ffmpeg -i in.mp4 -c:v libvpx-vp9 -pix_fmt yuv420p -crf 10 -b:v 0 out.mkv
[19:52:17 CEST] <fluffrabbit> preferably with a video > 1 minute long
[19:52:54 CEST] <fluffrabbit> watch the encoding speed drop throughout
[20:15:14 CEST] <wm4> eh, the jpeg probe check is duplicated between img2dec.c and rawdec.c
[20:15:32 CEST] <wm4> the one in rawdec.c could probably be killed completely
[20:15:58 CEST] <wm4> ubitux: for what subtitle things did you want me to add tests for?
[20:17:58 CEST] <Compn> wm4 : unless you use jpeg raw streams , jpeg webcams, etc, then sure you can remove "duplicated" code...
[20:18:05 CEST] <Compn> or you could just factor it
[20:20:08 CEST] <wm4> it looks redundant
[20:20:57 CEST] <cone-263> ffmpeg 03Diego Biurrun 07master:247aa7af7d81: avisynth: Simplify shared library name construction
[20:20:58 CEST] <cone-263> ffmpeg 03Michael Niedermayer 07master:67ac31d1bec2: Merge commit '247aa7af7d8197247c181e3fbfe8d93d75e41b29'
[20:34:49 CEST] <cone-263> ffmpeg 03Diego Biurrun 07master:1336bb06c9fb: configure: Simplify avisynth check
[20:34:50 CEST] <cone-263> ffmpeg 03Michael Niedermayer 07master:04fbab147745: Merge commit '1336bb06c9fbf9a14765e9f78616f2aad4f3a45a'
[20:45:55 CEST] <BBB> fluffrabbit: does that occur when you encode directly with vpxenc?
[20:57:37 CEST] <fluffrabbit> sorry, I'm occupied for the next hour or so
[21:37:43 CEST] <cone-263> ffmpeg 03Michael Niedermayer 07master:3acadc89f495: tests/tiny_psnr: Make the search range extend both sides from the specified shift value
[22:15:25 CEST] <fluffrabbit|away> Got a busy day today. I might test with vpxenc when I have the time.
[22:16:26 CEST] <fluffrabbit|away> Seems likely that it is an issue somewhere in ffmpeg's implementation of libvpx, but whether it's in libvpx itself has yet to be seen.
[22:17:44 CEST] <fluffrabbit|away> here I use the word "implementation" loosely as I'm not super clear on how all the little cogs fit together, the make and model of the gears and the grease in between
[00:00:00 CEST] --- Sat Apr 18 2015
1
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[00:04:44 CEST] <c_14> from the first input, yes
[00:04:50 CEST] <c_14> you can limit what it takes
[00:07:21 CEST] <Prelude2004c> but the transort stream had 5 programs
[00:07:57 CEST] <Prelude2004c> i want to relay them all > another destination without altering them
[00:08:14 CEST] <c_14> yeeeeees?
[00:08:48 CEST] <Prelude2004c> i will try -map 0 .. crashed server though so can only try tomorrow
[08:40:47 CEST] <NxtLevel> Hello all, I have a problem with capturing from IP camera RTSP stream to a jpeg image
[08:41:28 CEST] <NxtLevel> http://pastebin.com/txLJBrCG
[10:02:32 CEST] <dharriso> Hi, I have a question conserning the use of the libav* libs in an appliaction. Is this the correct IRC channel?
[10:03:08 CEST] <__jack__> sure
[10:11:43 CEST] <dharriso> So I have an MP4 file that has 5 tracks, 1 Audio, Video and 2 hints tracks. I create a variant of this file at a lower bitrate with only 2 trakcs and audio and video file, no hint tracks, this works fine, Problem is when I seek to a video keyframe, seeks request for each track, however the seek on the hint track is not what i expect
[10:12:51 CEST] <dharriso> its at a time of 3.xx seconds not at the 1.xx of the keyrame time
[10:14:16 CEST] <dharriso> not sure why this is, can you explain.
[10:14:44 CEST] <dharriso> Additionally is there a way of tell the API to ignore the hint tracks, just consume the data but dont call back to the application
[10:24:36 CEST] <jookiyaya> what is hints tracks?
[10:27:31 CEST] <dharriso> a hint track is used to include streaming information in the file for example RTP type formats
[10:27:45 CEST] <dharriso> its not needed by me but its a track in the original MP4 file
[10:49:43 CEST] <t4nkGuide> hello everyone...
[10:50:47 CEST] <t4nkGuide> im trying to make a vid from jpg files, but it give me an error [swscaler @ 0x2e87960] deprecated pixel format used, make sure you did set range correctly?? why that happening
[10:52:21 CEST] <t4nkGuide> i resize my various jpg with mogrify to same resolution.
[10:54:16 CEST] <t4nkGuide> oh yeah. ignore ratio resize.
[14:06:28 CEST] <xace> is ffmpeg able to capture the screen with libx264 using vdpau as pixel format?
[14:07:33 CEST] <iive> this doesn't make sense. ffmpeg is able to capture screen and encode it with libx264
[14:09:10 CEST] <xace> When i dont specify a -pix_fmt it prints: "No pixel format specified, yuv444p for H.264 encoding chosen. Use -pix_fmt yuv420p for compatibility with outdated media players."
[14:15:41 CEST] <retarder> hello anybody know how to use avconv to save streams to different output files simultaneously?
[14:16:40 CEST] <retarder> i need to split video and audio without discarding any, while capturing v4l2 input stream
[14:17:08 CEST] <retarder> i don't want to create a muxed container (avi, mpeg, ...)
[14:17:48 CEST] <retarder> i need to edit video and audio separately
[14:27:44 CEST] <BtbN> avconv is not ffmpeg
[14:27:46 CEST] <xace> Currently when I capture the screen, the capturing process with libx264 works fine, however later when I run the result in the mediaplayer, it complains about being unable to use hardware decoding, does anyone know of a method to capture the screen fast but which also produces a video file capable of hardwareacceleration?
[14:49:42 CEST] <retarder> ok i found out a way with -map directive
[14:50:39 CEST] <retarder> -map 0:v:0 -y /path/to/video.mpeg -map 1:a:0 -y /path/to/audio.pcm
[15:37:33 CEST] <synthecypher> How can I get ffmpeg to remove the ffmpeg2pass-0.log file after it's finished?
[16:02:55 CEST] <c_14> synthecypher: you have to remove it manually or by doing something like appending && rm ffmpeg2pass-0.log to the 2nd pass encode
[17:13:21 CEST] <Julius1241> hello all
[17:13:37 CEST] <Julius1241> any gurus with RTSP/ ipcams around
[18:10:37 CEST] <proctrap> hey there, could be an odd question, but is there an "--newline" param, so I'm not gettinga \r for the status updates ?
[18:11:06 CEST] <proctrap> (I could also use TR, but that's very cpu intense & I would like to avoid replacing \r with \n.. )
[18:18:22 CEST] <fling> What is the modern way of recoding my lossless and nearly lossless videos? I have a lot of spare cpu times and I want to get small sized files.
[18:18:40 CEST] <fling> I also need to shift audio for a bit
[18:38:13 CEST] <fling> How to create multiangle video?
[18:48:37 CEST] <fluffrabbit> what is up, my socially awkward friends?
[18:49:13 CEST] <fluffrabbit> Is this the right channel to complain?
[19:20:21 CEST] <Julius1241> guys i need some help with RSTP and ip cam
[00:00:00 CEST] --- Sat Apr 18 2015
1
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[01:57:55 CEST] <jamrial> if people ok the patch then it will be applied
[03:44:01 CEST] <cone-157> ffmpeg 03Vittorio Giovara 07master:7498f2221ef2: h264: Do not fail on unsupported timing info
[03:44:01 CEST] <cone-157> ffmpeg 03Michael Niedermayer 07master:c658269cd18c: avcodec/h264_ps: Validate num_units_in_tick/time_scale before setting them in the context
[05:13:42 CEST] <cone-157> ffmpeg 03Michael Niedermayer 07master:b50a4ca179cf: avformat/utils: Preserve packet duration when parsing is done only for headers
[13:05:48 CEST] <cone-630> ffmpeg 03Nicolas George 07master:38155865df73: lavfi/avf_showcqt: fix error code.
[13:21:00 CEST] <cone-630> ffmpeg 03Victor Anjin 07master:2db24cf746dc: ivfenc: incorrect fourcc for VP9 video
[14:58:54 CEST] <cone-630> ffmpeg 03Simon Thelen 07master:ed2a712750c3: configure: add pkg-config support for libdcadec
[15:01:13 CEST] <Daemon404> heh i figured carl would win and the retarded fallback would remain
[15:01:20 CEST] <Daemon404> as per usual
[15:01:22 CEST] <Daemon404> autism ftw
[15:14:54 CEST] <wm4> Daemon404: even though the community decided the second patch should go in
[15:15:00 CEST] <wm4> but who needs reviews
[15:20:30 CEST] <kierank> wm4: i note nicholas has not responded to you
[15:20:45 CEST] <wm4> seems so
[16:02:08 CEST] <cone-630> ffmpeg 03Andreas Cadhalpun 07master:ae6fd7300b4e: msrledec: use signed pixel_ptr in msrle_decode_pal4
[16:16:56 CEST] <cone-630> ffmpeg 03Timothy Gu 07release/2.2:67a4811c88c4: tests: Fix test name for pixfmts tests(cherry picked from commit e1ee0521a698809ed216e9e5c11bd2bbb466ed04)
[16:16:57 CEST] <cone-630> ffmpeg 03Michael Niedermayer 07release/2.2:0df90898f5a3: avcodec/h264_slice: Dont reset mb_aff_frame per slice
[16:16:58 CEST] <cone-630> ffmpeg 03Michael Niedermayer 07release/2.2:2b69da7b5ac3: avcodec/h264: reset the counts in the correct context
[16:16:59 CEST] <cone-630> ffmpeg 03Michael Niedermayer 07release/2.2:684f86391d3d: avcodec/aacdec: Fix storing state before PCE decode
[16:17:00 CEST] <cone-630> ffmpeg 03Michael Niedermayer 07release/2.2:82d3dd44aad4: avcodec/h264: Be more tolerant to changing pps id between slices
[16:17:01 CEST] <cone-630> ffmpeg 03Andreas Cadhalpun 07release/2.2:800d974cc41f: msrledec: use signed pixel_ptr in msrle_decode_pal4
[17:06:11 CEST] <Timothy_Gu> Hey guys, I see this error on http://fate.ffmpeg.org:8080/log/armv7l-panda-gcc4.4-armv5t/20150416110347/t… : "awk: line 1: syntax error at or near )"
[17:06:31 CEST] <Timothy_Gu> for atrac3p tests I added a while ago
[17:06:59 CEST] <Timothy_Gu> is there a way to have access to the box to see why that fails?
[17:07:09 CEST] <Timothy_Gu> michaelni: ^^ you own the box btw
[17:08:34 CEST] <cone-630> ffmpeg 03Michael Niedermayer 07release/2.2:ef1d4873a058: Update for 2.2.15
[17:18:39 CEST] <michaelni> Timothy_Gu, http://fate.ffmpeg.org/report.cgi?time=20150416091006&slot=armv7l-panda-gcc… says bus error, so i suspect maybe some misaligned access
[17:38:37 CEST] <Timothy_Gu> michaelni: seems like another bug in fateserver BTW
[18:17:46 CEST] <cone-630> ffmpeg 03James Almer 07master:1577b29c820a: configure: add missing dependencies for MIPS and PPC features
[19:03:39 CEST] <cone-630> ffmpeg 03Andreas Cadhalpun 07master:5b75689b987e: aacdec: consistently use avctx for logging in decode_eld_specific_config
[19:20:06 CEST] <jamrial> michaelni: Re dcadec, is the .pc file installed in your environment? the commit that added it is exactly 27 days old
[19:20:16 CEST] <jamrial> i see no reason for it to fail if it's installed
[19:22:07 CEST] <michaelni> theres no *dcadec.pc
[19:23:40 CEST] <Daemon404> your git repo is old.
[19:26:47 CEST] <jamrial> ok, should i apply the second version then?
[19:27:13 CEST] <loki_> ffmpeg can use quicksync?
[19:27:25 CEST] <loki_> is there already patch or something?
[19:28:56 CEST] <jamrial> loki_: libavcodec qsv*.c probably
[19:33:40 CEST] <cone-630> ffmpeg 03Andreas Cadhalpun 07master:0be54ad280cf: aasc: return correct buffer size from aasc_decode_frame
[20:14:35 CEST] <Timothy_Gu> are the mm* registers the same as st* but on the LSBs?
[20:15:28 CEST] <Timothy_Gu> yes
[20:36:40 CEST] <cone-630> ffmpeg 03Andreas Cadhalpun 07master:e224aa419174: aacpsy: avoid psy_band->threshold becoming NaN
[21:07:49 CEST] <cone-630> ffmpeg 03Andreas Cadhalpun 07release/2.2:3757ef017c34: aacdec: consistently use avctx for logging in decode_eld_specific_config
[21:07:50 CEST] <cone-630> ffmpeg 03Andreas Cadhalpun 07release/2.2:e863d17e6278: aasc: return correct buffer size from aasc_decode_frame
[21:07:51 CEST] <cone-630> ffmpeg 03Andreas Cadhalpun 07release/2.2:fc9514bf4dd7: aacpsy: avoid psy_band->threshold becoming NaN
[21:35:50 CEST] <cone-630> ffmpeg 03Rodger Combs 07master:55a1d75bf785: lavf/assenc: handle extra sections after Events
[21:39:55 CEST] <rcombs> heh, lavf fails to demux Matroska files containing no streams (only attachments)
[21:40:07 CEST] <rcombs> (which is weird as fuck but apparently technically valid)
[21:48:16 CEST] <wm4> rcombs: what would one use such files for?
[21:48:27 CEST] <wm4> other than the weird MPlayer feature that recently surfaced
[21:49:08 CEST] <rcombs> wm4: I'm using them mostly as a test
[21:49:15 CEST] <rcombs> see if anything fails too badly
[21:49:24 CEST] <rcombs> not sure of any real-world use-case
[21:50:01 CEST] <rcombs> but the media server branch I'm poking at with them asserts false on them, so they're certainly serving my purpose well
[23:05:09 CEST] <cone-630> ffmpeg 03Andreas Cadhalpun 07master:bc4fee7f2a51: ac3: validate end in ff_ac3_bit_alloc_calc_mask
[23:30:36 CEST] <cone-630> ffmpeg 03wm4 07master:8297d87eecbf: fate: add mp3 gapless test
[23:46:24 CEST] <cone-630> ffmpeg 03Andreas Cadhalpun 07release/2.2:a2908d49d1e1: ac3: validate end in ff_ac3_bit_alloc_calc_mask
[00:00:00 CEST] --- Fri Apr 17 2015
1
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[00:00:12 CEST] <klaxa> i'm pretty sure ffmpeg can do it too, but i don't know how from the top of my head
[00:00:44 CEST] <kyleogrg> ooh interesting
[00:01:17 CEST] <c_14> ffprobe -show_frames
[00:01:20 CEST] <c_14> check for key_frame
[00:02:57 CEST] <kyleogrg> ffprobe.exe -i VTS_01_1.VOB -show-frames
[00:03:06 CEST] <kyleogrg> Missing argument for option 'show-frames'
[00:03:29 CEST] <klaxa> it's -show_frames not -show-frames
[00:03:48 CEST] <kyleogrg> thx
[00:03:59 CEST] <kyleogrg> holy mackerel
[00:04:17 CEST] <kyleogrg> it's spitting out tons of info
[00:05:22 CEST] <c_14> yeeeaaaah; it's only really useful if you have a program to parse it.
[00:06:19 CEST] <c_14> You can use things like -select_streams v to only get video stream info though
[00:06:49 CEST] <c_14> You can also output to xml/json/csv etc
[00:07:00 CEST] <kyleogrg> so mkvtoolnix can split at idr frames?
[00:08:04 CEST] <klaxa> ffmpeg can too, not sure if it can split one file into multiple parts though
[00:08:53 CEST] <c_14> It can, but it's not trivial.
[00:08:54 CEST] <klaxa> if you invoke ffmpeg for every part you want to have split it takes some time to seek to the position you need, mkvtoolnix doesn't have that overhead
[00:09:23 CEST] <klaxa> oh? maybe it would make sense to make that easier
[00:09:29 CEST] <kyleogrg> what does mkvtoolnix do? split all the idr frames?
[00:09:46 CEST] <c_14> hmm, you could just use the segment muxer with -segment_times
[00:09:51 CEST] <c_14> that would actually not be that hard
[00:09:58 CEST] <kyleogrg> c_14: in ffmpeg?
[00:10:02 CEST] <c_14> yeyp
[00:10:21 CEST] <kyleogrg> what would the command line look like?
[00:10:33 CEST] <klaxa> mkvtoolnix is a collection of tools to create/edit/analyze matroska files, one of its capabilities is splitting a file at given timecodes/frames
[00:10:45 CEST] <kyleogrg> ok
[00:11:01 CEST] <c_14> ffmpeg -i video -segment_times 5,50,55,756,6541 -c copy out.m3u8
[00:11:13 CEST] <c_14> assuming those are the timestamps of the idr frames
[00:11:32 CEST] <kyleogrg> c_14: this would produce multiple videos?
[00:11:45 CEST] <c_14> yes
[00:12:15 CEST] <kyleogrg> so if i did this automatically for a long vob file, the command would be very very long
[00:12:18 CEST] <kyleogrg> is that ok?
[00:12:28 CEST] <c_14> yep
[00:14:09 CEST] <kyleogrg> cool. with ffprobe, how can i parse or pull out the idr frame locations?
[00:14:09 CEST] <klaxa> note: to enhance the load-balancing effect it would make more sense to make parts of similar size while still cutting at i-frames, even though adding one i-frame every once in a while doesn't drive bitrate through the roof anyway
[00:14:59 CEST] <kyleogrg> klaxa: for a DVD VOB, would i-frames more or less be distributed evenly?
[00:15:10 CEST] <klaxa> i think so
[00:15:24 CEST] <klaxa> i think the spec says an i-frame every 0.5 seconds or something
[00:16:20 CEST] <klaxa> you actually don't want an i-frame every 0.5 seconds with h265 aimed at small filesize
[00:19:00 CEST] <kyleogrg> so in my command line for converting a dvd to mkv -- i pasted the link to it earlier -- would the mkv keep all the i-frames from the vob??
[00:19:41 CEST] <c_14> depends on how you do it
[00:19:57 CEST] <klaxa> if you re-encode, that's generally not the case
[00:20:09 CEST] <kyleogrg> ok
[00:34:52 CEST] <kyleogrg> Ever heard of Ripbot264? http://forum.doom9.org/showthread.php?t=127611
[00:35:46 CEST] <kyleogrg> It can apparently do distributed encoding for x264
[00:36:20 CEST] <klaxa> i think i heard about it
[00:37:02 CEST] <klaxa> it's funny how these things get written for windows
[00:37:32 CEST] <kyleogrg> yeah
[00:37:58 CEST] <klaxa> oh yeah, i definitely remember it
[00:38:24 CEST] <klaxa> because i was thinking "cutting at 1500 frames? just like that? does that make sense?"
[00:38:44 CEST] <klaxa> but yeah, adding a few i-frames really shouldn't do much damage
[00:45:02 CEST] <pron> r
[00:47:12 CEST] <kyleogrg> back
[00:47:20 CEST] <kyleogrg> my web irc window froze
[00:48:03 CEST] <kyleogrg> That guy should make RipBot265
[00:48:31 CEST] <klaxa> >Changelog
[00:48:31 CEST] <klaxa> >v1.18.1
[00:48:31 CEST] <klaxa> >Added: Support for HEVC as input format (mkv and mp4 containers only)
[00:48:35 CEST] <klaxa> oh wait
[00:48:36 CEST] <klaxa> nvm
[00:49:03 CEST] <klaxa> actually
[00:49:06 CEST] <klaxa> >v1.18.0
[00:49:06 CEST] <klaxa> >Added: Support for x265 (DE mode works as well)
[00:49:12 CEST] <kyleogrg> what???
[00:49:25 CEST] <kyleogrg> so it would work?
[00:49:32 CEST] <klaxa> probably
[00:50:18 CEST] <kyleogrg> ok, i'll probably look at that when i have time
[02:03:08 CEST] <Guest90273> why is bluray subtitle file "SUP" so much larger in file size than dvd subtitle file "idx/vob" file
[02:12:53 CEST] <debianuser> Guest90273: maybe it contains subtitle images in higher resolution?
[02:54:57 CEST] <mtcjayne> Could someone explain to me how "Stream specifier 'main' in filtergraph description [...] matches no streams." in this paste?
[02:55:00 CEST] <mtcjayne> http://pastebin.com/tjbGETfC
[02:55:19 CEST] <mtcjayne> It seems to be that it matches painfully obviously.
[02:56:40 CEST] <kyleogrg> has anyone here had experience with ripbot264?
[02:57:03 CEST] <c_14> mtcjayne: you can't use a pad twice in one filtergraph
[02:57:12 CEST] <mtcjayne> That was a mistake on my part.
[02:57:13 CEST] <c_14> there's a filter
[02:57:41 CEST] <mtcjayne> I was trying to figure out why it was seeing a audio/video concat mismatch.
[02:57:58 CEST] <mtcjayne> http://pastebin.com/xueFGHZ9 is the correct version of the paste, with the audio/video concat mismatch.
[02:58:48 CEST] <mtcjayne> http://pastebin.com/kxNqVwjD is the error I was attempting to correct.
[02:59:11 CEST] <c_14> [pip] is video
[02:59:13 CEST] <c_14> not audio
[02:59:19 CEST] <mtcjayne> Yes.
[02:59:24 CEST] <c_14> That filtergraph doesn't have audio anywhere
[02:59:28 CEST] <c_14> change the concat to a=0
[02:59:38 CEST] <mtcjayne> Gotcha. Makes sense, thanks.
[02:59:42 CEST] <c_14> and get rid of the [a]
[03:00:57 CEST] <mtcjayne> I've really enjoyed seeing the automation that can be done with FFmpeg.
[03:01:27 CEST] <mtcjayne> Plus... for some reason... I seem to get 10x the file size with Sony Vegas Pro. Probably not 100% FFmpeg's victory there though.
[03:02:19 CEST] <c_14> What codec?
[03:07:49 CEST] <mtcjayne> 720p60 H264.
[03:08:16 CEST] <mtcjayne> I was getting >1G files from 15 minutes of footage, where the input file was about 300M.
[03:09:19 CEST] <c_14> mhm, no idea what encoder vegas uses or what settings, but x264 is pretty damn hard to beat
[03:09:34 CEST] <mtcjayne> Hmm I had some problems with my result.
[03:10:48 CEST] <mtcjayne> The audio from EndCard.mp4 wasn't included in its playback section, and I heard some footage from the primary episode footage.
[03:11:28 CEST] <c_14> That's because your command doesn't specify what audio to use.
[03:11:39 CEST] <c_14> map the audio you want from the file you want explicitly
[03:16:32 CEST] <mtcjayne> I'm not seeing how to do that. I want audio for stream 0 during its section, audio from stream 1 during its section (no audio from streams 2 or 3.) http://pastebin.com/nkmXLVMP
[03:17:01 CEST] <mtcjayne> I had that working without the "select" before
[03:17:35 CEST] <mtcjayne> Now I'm trying to give the script user control over what time periods of each file (0, 2, and 3) are used.
[03:18:43 CEST] <mtcjayne> I could simply remove the audio from streams 2 and 3 if you think that'd work.
[03:18:51 CEST] <c_14> eeeeeeeh, I'm pretty sure that concat=n=4 should be giving you an error, but w/e just add another filterchain to the graph with something like "[0:a][2:a][3:a]concat=n=3:v=0:a=1[a]" and map it
[03:19:52 CEST] <c_14> eeeeeh
[03:20:00 CEST] <mtcjayne> Yeah that's another remnant. I'll remove it here.
[03:20:09 CEST] <c_14> you might need to shorten the 0th audio stream to the same as what you put in the select
[03:20:27 CEST] <c_14> you can use aselect or something
[03:21:05 CEST] <c_14> Oh, and aren't 2+3 playing simultaneously?
[03:21:14 CEST] <c_14> You'll have to mix their audio tracks together with amix
[03:21:17 CEST] <mtcjayne> The videos are, the audio hasn't been in the past.
[03:21:34 CEST] <mtcjayne> I don't want it to be playing but I haven't actually done anything to prevent it. It's just worked out that way.
[03:21:50 CEST] <c_14> Oh, if you don't want it to be playing you can just ignore it.
[03:22:00 CEST] <c_14> just concat silence to the end of the first one.
[03:22:07 CEST] <mtcjayne> Ok
[03:22:08 CEST] <c_14> (and use -shortest)
[03:22:26 CEST] <c_14> you can generate silence with aevalsrc=0
[03:24:39 CEST] <mtcjayne> You're thinking concat <length of stream 1> seconds of silence after stream 0 in line 16? (And generate the silence the line before that?)
[03:25:53 CEST] <c_14> should work
[03:28:03 CEST] <mtcjayne> Is there an expression I can use to get the length of streams?
[03:28:49 CEST] <mtcjayne> I see that I'm using some, like "main_w" but I don't know where they're documented.
[03:29:06 CEST] <c_14> you can use ffprobe; what I would do, however, is just concat infinite amounts of silence to the end of the first audio, and then add the -shortest option. That'll make it end when the shortest stream (in this case the video) ends.
[03:29:56 CEST] <c_14> main_w etc are documented in the sections for each filter in ffmpeg-filters
[03:30:04 CEST] <c_14> https://ffmpeg.org/ffmpeg-filters.html
[03:30:26 CEST] <mtcjayne> Ah I thought they were universal or something. Thanks.
[03:38:14 CEST] <mtcjayne> This is what I've got, but I don't think I'm doing it correctly as I get the following error: "No output pad can be associated to link label 'main'." http://pastebin.com/LDzfVgQ7
[03:38:25 CEST] <mtcjayne> Oh
[03:38:38 CEST] <mtcjayne> Well first of all I've got a typo which I'll fix. Missing semicolon.
[03:39:00 CEST] <c_14> you also mispelled silence
[03:39:14 CEST] <c_14> and it should be
[03:39:15 CEST] <mtcjayne> Alright that and the missing \/n are fixed.
[03:39:21 CEST] <c_14> [main][0:a][pip][silence]
[03:39:50 CEST] <mtcjayne> Alright.
[03:39:58 CEST] <mtcjayne> I'll see what results I get here.
[03:40:40 CEST] <c_14> and I don't think the concat filter has a shortest option. What I was referring to was [main][pip]concat=n=2:v=1:a=0[v];[0:a][silence]concat=n=2:v=0:a=1[a]' -map '[v]' -map '[a]' -shortest
[03:41:08 CEST] <mtcjayne> It doesn't have one documented.
[03:47:10 CEST] <c_14> Anyway, I hope that works out. I need to go to bed now.
[03:48:44 CEST] <mtcjayne> Thanks for the help
[04:49:11 CEST] <a[0]> I'm having trouble finding documentation on this... in complex filter graphs like "... [x][1:v] paletteuse" what does [1:v] mean?
[05:34:01 CEST] <quasarlog> hi, I recently pulled from the github ffmpeg repo to update my working directory and I can't stream video using ffserver anymore.this is the command i used to send to ffserver:
[05:34:02 CEST] <quasarlog> ffmpeg -f avfoundation -i "1:" -maxrate 2500k -threads:0 4 -threads:1 4 -vcodec copyts -movflags +faststart -pix_fmt yuv420p http://localhost:8090/feed1.ffm -loglevel 48
[05:34:12 CEST] <quasarlog> i have ffserver running locally
[05:34:45 CEST] <quasarlog> with the following conf:
[05:34:46 CEST] <quasarlog> conf follows:
[05:34:46 CEST] <quasarlog> <Feed feed1.ffm>
[05:34:46 CEST] <quasarlog> File /tmp/feed1.ffm
[05:34:46 CEST] <quasarlog> FileMaxSize 5M
[05:34:46 CEST] <quasarlog> </Feed>
[05:35:06 CEST] <quasarlog> <Stream test.flv>
[05:35:06 CEST] <quasarlog> Feed feed1.ffm
[05:35:06 CEST] <quasarlog> Format flv
[05:35:06 CEST] <quasarlog> #NoVideo
[05:35:06 CEST] <quasarlog> VideoCodec flv1
[05:35:07 CEST] <quasarlog> AVOptionVideo flags +global_header
[05:35:07 CEST] <quasarlog> PixelFormat yuv420p
[05:35:08 CEST] <quasarlog> VideoFrameRate 25
[05:35:08 CEST] <quasarlog> VideoGopSize 12
[05:35:09 CEST] <quasarlog> VideoBufferSize 0
[05:35:09 CEST] <quasarlog> VideoBitRate 1872
[05:35:10 CEST] <quasarlog> NoDefaults
[05:35:10 CEST] <quasarlog> StartSendOnKey
[05:35:34 CEST] <quasarlog> VideoSize 1280x800
[05:35:35 CEST] <quasarlog> PreRoll 0
[05:35:35 CEST] <quasarlog> Noaudio
[05:35:35 CEST] <quasarlog> </Stream>
[05:37:48 CEST] <quasarlog> when i use this, the output that results is this with the recently updated and rebuilt ffmpeg:
[05:37:48 CEST] <quasarlog> Output #0, ffm, to 'http://localhost:8090/feed1.ffm':
[05:37:48 CEST] <quasarlog> Metadata:
[05:37:48 CEST] <quasarlog> creation_time : now
[05:37:48 CEST] <quasarlog> Stream #0:0, 0, 0/0: Video: flv1 (flv), 1 reference frame, yuv420p, 1280x800 (0x0), 1/25, q=1-10, 1872 kb/s, 1000k fps, 25 tbc
[05:37:48 CEST] <quasarlog> Metadata:
[05:37:49 CEST] <quasarlog> encoder : Lavc56.34.100 flv
[05:37:49 CEST] <quasarlog> Stream mapping:
[05:37:50 CEST] <quasarlog> Stream #0:0 -> #0:0 (rawvideo (native) -> flv1 (flv))
[05:37:50 CEST] <quasarlog> Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height
[05:37:51 CEST] <quasarlog> [AVIOContext @ 0x7f990b710ec0] Statistics: 0 seeks, 0 writeouts
[05:44:16 CEST] <quasarlog> http://pastebin.com/WV8D8Dgz
[06:00:40 CEST] <relaxed> quasarlog: -vcodec copyts is wrong
[06:02:48 CEST] <Guest90273> is there a difference if the source is HD or SD if output is SD?
[06:06:28 CEST] <zumba_addict> output from HD will be a little sharper
[06:07:00 CEST] <zumba_addict> but if you are not maximizing the output, sd will be sharp
[06:08:18 CEST] <Guest90273> you think if source is HD< it will be little better?
[06:18:33 CEST] <relaxed> Guest90273: yes
[06:19:06 CEST] <Guest90273> why would that be if output is SD
[06:21:54 CEST] <relaxed> because there's more detail
[06:54:13 CEST] <Guest90273> what happens if you play 5.1 audio on 2.0 system ?
[07:05:06 CEST] <DeadSix27> depends on the system and soundcard and software ?
[07:05:11 CEST] <DeadSix27> usually gets downmixed to 2.0
[07:14:15 CEST] <urmumstty> hey guys, i have 54 frames taken with a camera rig that does a kind of matrix bullet time effect (i guess) and i was thinking i could clean it up / trick it up a bit with ffmpeg since the original technicians are not interested, basically the cameras all seem poorly calibrated so as the rotation occures it jumps around a little, also i was thinking i could use some kind of interpolation on the pts to give it an ease at the start and end
[07:14:33 CEST] <urmumstty> another idea was to do some kind of decaying motion blur so it blurs a bit at the start and end but is clear through the middle
[07:14:55 CEST] <urmumstty> i've never tried to use ffmpeg for anything like this, i've used the filter graph a fair bit, is this the right tool, should it be possible, what do i do about the jitter
[07:15:03 CEST] <urmumstty> don't think i've seen any kindo f steady cam effect
[07:15:49 CEST] <urmumstty> so i guess i'm asking: is there some kind of frame steading algorithm avaliable, can i apply an equation to the pts, can i apply a motion blur effect with its strength defined by another equation
[07:16:08 CEST] <urmumstty> would be nice to take advantage of the hardware setup and polish it up a bit rather than the current half assed output
[08:29:17 CEST] <benoliver999> I'm getting skewed screenshots when using Debian wheezy (ffmpeg from the Debian repos) - would upgrading to a more recent version fix this?
[08:36:11 CEST] <dsl420> do you use debian testing? afaik a legit ffmpeg is only in the testing reposis, debian has a fork in the reposis. they changed it in testing afaik. just grab ffmpeg from github
[08:37:03 CEST] <benoliver999> I do not, but thank you for the information. I'll give it a go.
[08:37:30 CEST] <dsl420> and if you want screenshots from an anamorphic video you would need to adjust your command as well
[08:39:54 CEST] <benoliver999> Wait that's the problem I'm having
[08:40:33 CEST] <benoliver999> but I was told somewhere (wrongly?) that a more recent version of ffmpeg solves the problem on its own
[09:44:59 CEST] <jookiyaya> what happens if you have 5.1 surround audio on 2.0 system/speakers
[10:06:21 CEST] <Anoia> depends ont he audio chipset I think
[10:27:48 CEST] <Fish-> hi
[10:28:23 CEST] <Fish-> do you know how to issue the stss box in mp4 format using the ffmpeg api?
[15:18:16 CEST] <DiegoMax> im having some issues with av_seek_frame() on h.265 files
[15:18:19 CEST] <DiegoMax> is this a known issue ?
[15:19:30 CEST] <spaam> DiegoMax: maybe you should tell us what version you are using. then maybe someone know if its a known issue or not :D
[15:19:52 CEST] <DiegoMax> latest version from the repo, pulled lastnight
[15:20:06 CEST] <DiegoMax> let me give you an exact version
[15:20:08 CEST] <DiegoMax> just a sec
[15:20:29 CEST] <DiegoMax> is there any file in the source tree with the version number or should i use the runtime version() method ?
[15:26:58 CEST] <DiegoMax> well
[15:27:00 CEST] <DiegoMax> i found version.sh
[15:27:05 CEST] <DiegoMax> and this is what it shows: N-71425-gf4f3065
[15:27:16 CEST] <DiegoMax> not sure if thats the library version or what
[15:27:38 CEST] <DiegoMax> and at runtime, avformat_version() returns 3677796
[15:27:54 CEST] <DiegoMax> the issue is that only with h.265 files, av_seek_frame() is returning -1 no matter what
[15:28:04 CEST] <DiegoMax> with any other code it works just fine
[16:12:36 CEST] <DiegoMax> i keep failing to understand the licensing model of the popular video codecs
[16:12:55 CEST] <DiegoMax> if i understand correctly, H.264 is a spec, and then we have things like x264 which are open source encoders that adhere to that spec
[16:13:30 CEST] <DiegoMax> so if i write an application that ENCODES h.264 im supposed to pay a licensing fee to that H.264 people, even if im using an open source encoder
[16:14:13 CEST] <DiegoMax> so the obvious question is, why nobody came with an open source spec ? is it something utterly complicated that no more than a few selected people can do ? or am i missing something in this story ?
[16:16:55 CEST] <nasojlsu> it is utterly complicated.
[16:17:19 CEST] <DiegoMax> but i think there are way more complicated things in this world
[16:17:22 CEST] <DiegoMax> than a video encoder
[16:17:25 CEST] <DiegoMax> that are open source and public
[16:17:42 CEST] <DiegoMax> like the linux kernel, to name one.
[16:17:42 CEST] <nasojlsu> lots of engineers come together and create something that IEEE approves and bam...
[16:18:36 CEST] <c_14> DiegoMax: there are open specs, H.264/H.265 just aren't
[16:18:53 CEST] <DiegoMax> yes my question is, why nobody ever came with an open spec ?
[16:18:58 CEST] <DiegoMax> or is there any ?
[16:19:22 CEST] <DiegoMax> what is an example of an open spec that is as good as h264/265 ?
[16:20:36 CEST] <c_14> vp9 is pretty decent (as a codec); vorbis and opus are both great
[16:20:43 CEST] <c_14> As are ogg and matroska
[16:21:21 CEST] <DiegoMax> isnt VP9 owned by Google
[16:21:22 CEST] <DiegoMax> ?
[16:21:59 CEST] <nasojlsu> yup
[16:22:02 CEST] <c_14> Developed by, not owned by
[16:22:07 CEST] <DiegoMax> i think its owned too
[16:22:13 CEST] <DiegoMax> like, they own the spec
[16:22:16 CEST] Action: DiegoMax checks
[16:22:40 CEST] <c_14> daala is the next big open video spec to keep an eye on
[16:22:42 CEST] <DiegoMax> hm interesting
[16:22:55 CEST] <DiegoMax> acoording to wikipedia: VP9 is an open and royalty free[3] video coding format being developed by Google.
[16:23:08 CEST] Action: DiegoMax looks at [3]
[16:23:34 CEST] <c_14> afaik vp9 is relesed under a BSD license
[16:23:53 CEST] <c_14> So Google might be the majority license holder, but that doesn't mean they own it.
[16:23:56 CEST] <DiegoMax> so it would be safe to assume, that the big browsers dont support vp9 natively because someone is paying them not to ?
[16:24:13 CEST] <DiegoMax> because if they did, im sure it would be widely adopted
[16:24:57 CEST] <c_14> firefox supports vp9
[16:25:01 CEST] <c_14> iirc so does chrome
[16:25:11 CEST] <DiegoMax> chrome does
[16:25:28 CEST] <DiegoMax> i dont know for firefox
[16:25:32 CEST] <c_14> No clue about IE
[16:25:35 CEST] <DiegoMax> but safari and IE im sure they dont
[16:25:50 CEST] <DiegoMax> and as far as i know android doesnt either
[16:25:58 CEST] <DiegoMax> and of course iOS doesnt either
[16:25:59 CEST] <DiegoMax> :/
[16:28:36 CEST] <c_14> Well, google did recently partner up with people for hardware-decoding support for vp9. So it shouldn't be long until at least newer devices get it.
[16:32:27 CEST] <JEEBsv> "< DiegoMax> like, they own the spec" <- are you implying that there is a spec :P
[16:32:44 CEST] <DiegoMax> and am i wrong ?
[16:32:55 CEST] <JEEBsv> there is "kind of a spec" for VP8, but with VP9 they decided that they don't need one
[16:33:15 CEST] <DiegoMax> so how they settle on the format....
[16:33:24 CEST] <JEEBsv> implementation is the definition
[16:33:42 CEST] <JEEBsv> is which is why we effectively have an implementation bug in the "format" right now
[16:33:55 CEST] <JEEBsv> because someone noticed it after the implementation was first frozen'ish
[16:33:58 CEST] <DiegoMax> i was about to say that...
[16:34:10 CEST] <DiegoMax> im by no means a vydeo codec engineer
[16:34:26 CEST] <JEEBsv> and basically they do not expect anyone else to implement it
[16:34:31 CEST] <DiegoMax> but specs are needed for much simpler things.... i cant imagine how u can build that without a spec that everyone agrees to...
[16:34:40 CEST] <JEEBsv> the guy who implemented it in libavcodec worked for G as well
[16:35:29 CEST] <DiegoMax> now let me see if i am understanding this right
[16:35:31 CEST] <JEEBsv> also G generally seems to only care that you can decode it, encoder or realtime coding was never in their plans looking at how long there was zero multithreading in the encoder
[16:35:56 CEST] <DiegoMax> suppose i write a tool that encode VP9 files
[16:35:57 CEST] <JEEBsv> now there is in-picture multithreading but that of course is at the expense of compression
[16:36:02 CEST] <DiegoMax> i then make it closed source
[16:36:07 CEST] <DiegoMax> and i sell it for $1000
[16:36:33 CEST] <stefkos> lo
[16:36:33 CEST] <DiegoMax> can anyone come after me an tell me "you have to pay a license fee" ?
[16:36:37 CEST] <JEEBsv> of course
[16:36:47 CEST] <JEEBsv> that possibility never goes away
[16:37:01 CEST] <DiegoMax> then its not "open" and "bsd licensed"
[16:37:04 CEST] <DiegoMax> as the wiki says
[16:37:10 CEST] <DiegoMax> which goes back to my original rant :p
[16:37:23 CEST] <JEEBsv> the software is open and the code is bsd licensed for libvpx, don't mix software licenses and patents etc
[16:37:30 CEST] <JEEBsv> those are two completely different problem areas
[16:38:04 CEST] <stefkos> I use configure with flags ./configure --toolchain=msvc --enable-cross-compile --arch=i686
[16:38:21 CEST] <stefkos> and then Im getting a message
[16:38:26 CEST] <stefkos> Must specify target arch and OS when cross-compiling
[16:38:29 CEST] <JEEBsv> the problem is that for AVC and HEVC you have pre-set rules and some licensing (whether or not you actually pay something depends on the licensing, for example you could distribute 100k (I think) copies of a decoder for AVC for fere per year)
[16:38:46 CEST] <JEEBsv> while vp8/9 are supposedly "free of any such stuff"
[16:39:10 CEST] <JEEBsv> in both cases someone can come to you and threaten to sue
[16:39:11 CEST] <stefkos> I want to compile Windows/VisualStudio includes and libs
[16:41:00 CEST] <JEEBsv> also AVC and HEVC are actually formats openly developed (I am registered on the jct-vc mailing list that does HEVC development), and there is a spec
[16:41:19 CEST] <JEEBsv> while G decided to do VPx development just like On² used to do it
[16:41:26 CEST] <JEEBsv> behind closed doors
[16:41:40 CEST] <DiegoMax> stefkos what is your host platform ?
[16:41:43 CEST] <JEEBsv> the only difference is that you now get the source code right away with libvpx
[16:42:15 CEST] <stefkos> DiegoMax, I must go, will back in 30 mins...
[16:42:20 CEST] <stefkos> brb
[16:42:20 CEST] <DiegoMax> ok
[16:42:33 CEST] <DiegoMax> your problem is most likely that you should set ur build to a different directory
[16:42:37 CEST] <DiegoMax> when cross compiling
[16:42:38 CEST] <JEEBsv> and I would be less negative about VPx if only either the formats were brilliant or the implementations fucking awesome
[16:43:06 CEST] <JEEBsv> right now VP8 is a limited copy of AVC and VP9 is a HEVC rip-off'ish thing
[16:43:15 CEST] <DiegoMax> yes
[16:43:27 CEST] <DiegoMax> i have been reading about some very respectable people claiming
[16:43:41 CEST] <DiegoMax> that with VP9 they just modified hevc enough to avoid patent infringement
[16:43:49 CEST] <DiegoMax> if thats true, it is really sad then
[16:44:14 CEST] <JEEBsv> well, if the result is nice then using good ideas is good
[16:44:29 CEST] <JEEBsv> anyways, off to public transport for me...
[16:44:34 CEST] <DiegoMax> lol ok
[16:44:35 CEST] <DiegoMax> cya
[19:22:17 CEST] <bencc> is ffmpeg suitable for mixing dynamic rtmp streams with audio and video?
[19:22:33 CEST] <DiegoMax> what you mean by "mixing" ?
[19:22:41 CEST] <DiegoMax> mixing as in editing live ?
[19:22:52 CEST] <bencc> it's an audio/video conference where participants may come and go
[19:23:03 CEST] <bencc> and start and stop mic and cam dynamically
[19:23:18 CEST] <DiegoMax> im not 100% sure, but i doubt it can do that out of the box
[19:23:32 CEST] <bencc> DiegoMax: not out of the box, by writing C code
[19:23:39 CEST] <DiegoMax> by writing C code, for sure
[19:23:56 CEST] <bencc> the question is, can it handle syncing of streams, mixing, adding and removing streams...
[19:24:05 CEST] <bencc> queueing...
[19:24:21 CEST] <DiegoMax> i doubt it, thats what you would have to write... for sure
[19:24:36 CEST] <DiegoMax> have you considered an streaming server instead ?
[19:25:29 CEST] <bencc> which one?
[19:25:37 CEST] <DiegoMax> as in, with ffmpeg you have the ability to do realtime rtmp, but you will still need to write the part to feed the buffers with your mixed streams and such
[19:26:02 CEST] <bencc> I was hoping ffmpeg can help me with that like gstreamer
[19:27:12 CEST] <DiegoMax> ffmpeg can build the final stream for sure
[19:27:28 CEST] <DiegoMax> what im almost 100% certain ffmpeg can NOT do, is mixing several realtime streams into one, queuing, and stuff
[19:34:59 CEST] <stefkos> I think I setup whole enviroment to compile ffmpeg for VS on windows
[19:35:14 CEST] <stefkos> now I can compile whole archive without problems
[19:35:29 CEST] <stefkos> but when I will add --enable-libtheora
[19:35:55 CEST] <stefkos> it cannot find lib. But I have it in minsys/user/lib/ (something like this) and in VC/lib/ directory
[19:36:07 CEST] <stefkos> link -nologo -libpath:ARGEADDRESSAWARE -out:./ffconf.C1yHs8yD.exe ./ffconf.QOqlO0hA.o theora_static.lib ogg_static.lib psapi.lib advapi32.lib shell32.lib
[19:36:07 CEST] <stefkos> LINK : fatal error LNK1181: cannot open input file 'theora_static.lib'
[19:37:27 CEST] <stefkos> I have this message during ./configure
[19:40:04 CEST] <stefkos> hmm maybe I made mistake, I should pass --extra-ldflags
[19:40:07 CEST] <stefkos> lets see
[19:49:18 CEST] <stefkos> nope didnt help
[21:33:24 CEST] <Jobeanie123> Hello. I just learned about ffmpeg. It's neat so far.
[23:37:08 CEST] <Prelude2004c> hey guys.. question ... i tried ffmpeg -i < input transport stream > .. -acodec copy and vcodec copy... to mpegts udp://anotherserver:port ..
[23:37:18 CEST] <Prelude2004c> some reason only one program is available on the other stream
[23:37:33 CEST] <Prelude2004c> should it not show all the program id's in the transport stream?
[23:43:31 CEST] <c_14> -map 0
[23:43:36 CEST] <c_14> (probably)
[23:58:41 CEST] <Prelude2004c> um.. if i use -map 0 it will take everyhing?
[00:00:00 CEST] --- Fri Apr 17 2015
1
0
[03:34:48 CEST] <cone-487> ffmpeg 03Michael Niedermayer 07master:fbdaebb29861: avcodec/ituh263dec: Use check_marker()
[03:58:52 CEST] <jamrial> atomnuker: in case you didn't see it https://trac.ffmpeg.org/ticket/2686#comment:374
[04:01:46 CEST] <atomnuker> Once the patch makes it through to git master I'll respond, thanks
[04:57:20 CEST] <Prelude2004c> hey guys.. tried on ffmpeg and nobody answers so i am going to try here.. using NVENC with 2 cards... only one gets used
[04:57:22 CEST] <Prelude2004c> how come ?
[04:57:37 CEST] <Prelude2004c> is there some way to say load balance NVENC through both GPUs or there is some param i can use to force it ?
[05:14:42 CEST] <Prelude2004c> can anyone help?
[05:31:33 CEST] <Prelude2004c> if nobody has the answer for NVENC .. does anyone know how to set the default GPU in linux ? in shell
[05:31:47 CEST] <Prelude2004c> if i maybe set the GPU prior to starting the ffmpeg session, maybe it will start in the correct one and i can manage that
[11:19:27 CEST] <wm4> kierank: bingo!
[11:19:44 CEST] <wm4> "If there is a point of the API that can not be accessed with the command-line tools, fix that first."
[11:26:30 CEST] <rcombs> <Timothy_Gu> lglinskih: FFmpeg developers have lives :)
[11:26:34 CEST] <rcombs> ^ > implying
[11:56:51 CEST] <cone-716> ffmpeg 03Luca Barbato 07master:0cbb1473179d: x264: Map color parameters
[11:56:52 CEST] <cone-716> ffmpeg 03Michael Niedermayer 07master:0fbe18b63eab: Merge commit '0cbb1473179d35d4eb9c9446c8ed84e197e5389b'
[12:06:23 CEST] <cone-716> ffmpeg 03Luca Barbato 07master:ec89f35e3e5e: x265: Map color parameters
[12:06:24 CEST] <cone-716> ffmpeg 03Michael Niedermayer 07master:0842df2619a0: Merge commit 'ec89f35e3e5eb9ec5221266e801efeee7ecab4df'
[13:32:43 CEST] <cone-716> ffmpeg 03Michael Niedermayer 07master:a742a0536dbb: avformat/mp3dec: Allow forcing the use of the xing TOC for CBR files
[13:32:44 CEST] <cone-716> ffmpeg 03wm4 07master:ddd429fab635: Revert "avformat/mp3dec: offset seek index to end of id3v2 tag"
[13:32:45 CEST] <cone-716> ffmpeg 03wm4 07master:3163083f7b8f: avformat/mp3dec: offset seek index to end of vbr headers
[14:03:41 CEST] <kierank> wm4: wow
[14:03:55 CEST] <kierank> lglinskih: ignore wm4 and nicolas
[14:04:07 CEST] <kierank> well ignore nicholas
[14:04:12 CEST] <lglinskih> =)))
[14:05:03 CEST] <lglinskih> kierank: I wanted comments to my patch -- now I get them=)
[14:05:15 CEST] <kierank> lglinskih: personally I would not use the test source that they recommend
[14:05:28 CEST] <kierank> I would generate a simple square wave
[14:05:34 CEST] <kierank> I want the API tests to be simple
[14:06:32 CEST] <kierank> lglinskih: but you'll need to add framecrc output so that it can be used in fate
[14:15:04 CEST] <wm4> kierank: shouldn't it use something other than framecrc?
[14:15:25 CEST] <kierank> well if it wants to go into fate, doesn't it have to?
[14:15:39 CEST] <wm4> you mean the framecrcenc pseudo-muxer, right? looks like a rather big potential error source
[14:15:48 CEST] <wm4> maybe run crc on the packet data manually
[14:16:01 CEST] <wm4> (it could be a shared utility function for all the tests)
[14:16:11 CEST] <kierank> yeah seems ok
[14:25:28 CEST] <Daemon404> kierank, for audio?
[14:25:38 CEST] <kierank> ?
[14:25:44 CEST] <Daemon404> re: fra,ecrc
[14:25:52 CEST] <kierank> for flac it's ok
[14:25:58 CEST] <kierank> which is the qualification task
[14:26:04 CEST] <Daemon404> ah
[14:27:43 CEST] <wm4> how many decoders can be made bitexact anyway?
[14:28:47 CEST] <Daemon404> wm4, id assume any except the audio codecs with no fixed point impl
[14:28:56 CEST] <Daemon404> unless there is some insane float codec
[14:28:58 CEST] <ubitux> wm4: speaking of seeking test, i wrote a "seek fuzzer" a while ago
[14:29:13 CEST] <wm4> ubitux: sounds useful
[14:29:19 CEST] <ubitux> i'll dig it out if you want when i'm back
[14:29:29 CEST] <Daemon404> does anyone run afl btw?
[14:29:52 CEST] <ubitux> that would be nice
[14:29:55 CEST] <wm4> ubitux: lglinskih should be interested in it (at least if the gsoc application or whatever it was succeeds)
[14:31:29 CEST] <ubitux> wm4: yeah but i'm really short in time, so in one week or so, maybe
[14:33:51 CEST] <ubitux> and yeah afl looks nice; i think we could follow the logic of what that guy did with the sql thing
[14:34:17 CEST] <ubitux> use afl on every fate test basically
[14:34:21 CEST] <ubitux> that could be a nice project
[14:34:29 CEST] <ubitux> anyway, going away
[14:34:36 CEST] <wm4> what's afl?
[14:43:21 CEST] <kierank> a fuzzer
[14:44:44 CEST] <Daemon404> american fuzzy lop
[14:44:47 CEST] <Daemon404> stupid name
[14:44:49 CEST] <Daemon404> good fuzzer
[14:45:46 CEST] <wm4> sounds pretty interesting
[15:01:59 CEST] <lglinskih> kierank: ok, I don't understand exactly how to do it (I don't know much about fate), but I'll read and ask you later)
[15:14:15 CEST] <kierank> fate basically is a script that compares the crc of the decoded file to a reference
[15:14:24 CEST] <kierank> you should have a look at the fate instructions on ffmpeg.org
[15:17:53 CEST] <wm4> michaelni: I might try to add a test for gapless - can the sample file I posted be added to fate, or is it too big?
[16:45:42 CEST] <michaelni> wm4, 250kb AG.mp3 ? that would be ok, in which directory should i put it ?
[16:46:45 CEST] <wm4> ok... no need to do add it yet, I first have to write a patch
[17:56:41 CEST] <wm4> is there a test yet which includes AVFrame or AVPacket side data?
[17:57:29 CEST] <michaelni> i think so but i dont remember which
[18:00:03 CEST] <Prelude2004c> good day everyone. Tried yesterday but got no responses. Can someone point me in the right direction please. I am using NVENC to encode and i ahve 2 cards in my system. I don't know the flag to have the system load balance between both cards as right now it is only using 1 GPU. Can anyone help?
[18:00:31 CEST] <wm4> Prelude2004c: enter a support contract with nvidia
[18:00:53 CEST] <nevcairiel> the nvenc encoder has a "gpu" option which lets you select which gpu is being used
[18:00:58 CEST] <nevcairiel> 0, 1, 2 etc
[18:01:02 CEST] <nevcairiel> thats all we can do
[18:01:06 CEST] <nevcairiel> otherwise, ask nvidia
[18:01:18 CEST] <Prelude2004c> really? its some param ? i tried looking everywhere on google. nobody posts it as an example
[18:01:31 CEST] <Prelude2004c> i hope its something simple like a param that i set
[18:01:35 CEST] <nevcairiel> you should really read the help output from ffmpeg
[18:01:35 CEST] <Prelude2004c> but not sure where to set it
[18:02:58 CEST] <Prelude2004c> i looked at all the ffmpeg doc, nvidia nvenc stuff.. nothing does it say to select GPU do this.
[18:03:08 CEST] <Timothy_Gu> Prelude2004c: yes it's -gpu
[18:03:34 CEST] <kepstin-laptop> Prelude2004c: you set it in the same way as any other encoder avoption.
[18:04:22 CEST] <Prelude2004c> really?
[18:04:26 CEST] <Prelude2004c> hum.. i will try thank you
[18:04:54 CEST] <Timothy_Gu> Prelude2004c: by default it uses the gpu with the most CUDA computing unit
[18:06:25 CEST] <Prelude2004c> you are the best
[18:06:27 CEST] <Prelude2004c> thank you
[18:06:36 CEST] <Prelude2004c> ya but cards are identical
[18:06:45 CEST] <Prelude2004c> one has like 90% load and the other is 0
[18:06:51 CEST] <Prelude2004c> setting up -GPU worked
[18:06:52 CEST] <Prelude2004c> :)
[18:06:56 CEST] <Prelude2004c> so much thank you's
[18:07:09 CEST] <Prelude2004c> i was at it for a few hours .. couldn't find anything on it
[18:19:37 CEST] <philipl> You cannot split an encoding job across GPUs. That's not how the hardware works. What you're attempting to do cannot be done.
[18:19:46 CEST] <philipl> It's not anything like multi-threaded CPU encoding.
[18:20:03 CEST] <philipl> The hardware has decided silicon for encoding with a fixed pipeline.
[18:39:02 CEST] <Timothy_Gu> Prelude2004c: yeah ffmpeg doesn't have load balancing..
[18:39:30 CEST] <Timothy_Gu> philipl: no he's trying to have multiple ffmpeg instances using different cards
[18:46:48 CEST] <philipl> Timothy_Gu: ah.
[19:23:49 CEST] <jc0n> hey if #ffmpeg-user is empty and I pop in here to ask a quick question about ffserver are you guys gonna kill me?
[19:25:32 CEST] <lordkrondor> #ffmpeg seems like the place to ask questions
[19:25:39 CEST] <jc0n> alright thank you
[19:56:50 CEST] <cone-716> ffmpeg 03Michael Niedermayer 07master:574978d2bdbc: avcodec/intelh263dec: Use check_marker()
[19:56:51 CEST] <cone-716> ffmpeg 03Nedeljko Babic 07master:f4ccf3890285: libavutil/softfloat: Change order of fields in SoftFloat structure.
[20:15:46 CEST] <wm4> so why does "ffmpeg -ss 0 -i some.mp3" not seek to 0, but ts=353604?
[20:16:14 CEST] <wm4> it seems to include the start time (who knows why the heck gapless was designed to move the start time from 0 to a later timestamp)
[20:16:59 CEST] <wm4> and it seems to do so unconditionally
[20:25:56 CEST] <wm4> is this acceptable for a fate test? http://sprunge.us/IOSI
[21:03:57 CEST] <cone-716> ffmpeg 03Rostislav Pehlivanov 07master:c5d4f87e8111: aaccoder: Implement Perceptual Noise Substitution for AAC
[00:00:00 CEST] --- Thu Apr 16 2015
1
0
[00:01:34 CEST] <kyleogrg> How could using yadif speed up the video? It shouldn't change the number of frames or the framerate right?
[00:02:02 CEST] <lordkrondor> just my experience where the video ended before audio
[00:02:07 CEST] <lordkrondor> I didn't try to figure it out
[00:02:16 CEST] <kyleogrg> hmm ok
[00:11:03 CEST] <kyleogrg> -vf "fps=30000/1001,fieldmatch,yadif=deint=interlaced,decimate"
[00:11:23 CEST] <kyleogrg> What does "yadif=deint=interlaced" mean?
[00:11:39 CEST] <c_14> Only deinterlace frames that are marked as interlaced
[00:11:42 CEST] <c_14> iirc
[00:12:26 CEST] <kyleogrg> ok. and why is the fps part necessary?
[00:12:29 CEST] <kepstin-laptop> that looks like it should in theory give you a vfr video with detelecined and deinterlaced sections as appropriate
[00:13:17 CEST] <kepstin-laptop> kyleogrg: dunno, but it might be required on dvd-video sources, which switch between 30 and 24 fps.
[00:13:44 CEST] <kepstin-laptop> (ideally, you'd be applying softpulldown telecine to the video rather than duplicating frames, imo)
[00:13:54 CEST] <kyleogrg> kepstin-laptop: all right. well, what i have is an edited dvd of a 24p film.
[00:14:21 CEST] <kyleogrg> so there was (i think) a telecined dvd which somebody edited *again*
[00:14:46 CEST] <kyleogrg> and made an edited dvd version. so it's still telecined, but with no consistent pattern.
[00:14:51 CEST] <kyleogrg> now I want to detelecine
[00:14:57 CEST] <kepstin-laptop> it's very common for animated stuff to have additional edits being made after the telecine was done :/
[00:15:08 CEST] <kyleogrg> and this isn't animation...
[00:16:01 CEST] <kyleogrg> So I'm looking for a smart way to detelecine, then deinterlace the remaining interlaced frames...
[00:16:50 CEST] <kepstin-laptop> detelecine then deinterlace remaining frames, you mean?
[00:17:17 CEST] <kyleogrg> yes, i guess. but obviously to only deinterlace frames which need it
[00:19:38 CEST] <kepstin-laptop> kyleogrg: then the filterchain you wrote there is probably one of the better ways to do it in ffmpeg
[00:20:27 CEST] <kepstin-laptop> kyleogrg: might also be worth considering 'pullup,idet,yadif=deint=interlaced'
[00:21:00 CEST] <kepstin-laptop> pullup uses future context, so it's sometimes better in wonky edited content than the fieldmatch filter
[00:22:05 CEST] <kepstin-laptop> hmm. probably want to throw a dejudder on the end of that too, if you're using pullup
[00:22:35 CEST] <kyleogrg> hmm i need to think for a sec
[00:23:10 CEST] <kepstin-laptop> hmm. I should try "pullup,idet,yadif=deint=interlaced,dejudder" on some of my problematic samples
[00:23:44 CEST] <kepstin-laptop> I honestly kept a copy of mencoder around just so I could use its 'filmdint' filter before ffmpeg grew additional detelecining stuff...
[00:25:36 CEST] <kyleogrg> so what is dejudder?
[00:26:25 CEST] <kyleogrg> i can google
[00:26:43 CEST] <kepstin-laptop> kyleogrg: filter that fixes the timestamps of detelecined edited video to be constantish, rather than do something like 1, 2, 3, missing, 5,
[00:26:53 CEST] <kyleogrg> "May change the recorded frame rate of the container. Aside from that change, this filter will not affect constant frame rate video. "
[00:27:08 CEST] <kyleogrg> Can this mess with the pace or framerate of the movie?
[00:28:22 CEST] <kepstin-laptop> it fixes "judder", which is when you see e.g. a smooth horizontal pan and it looks like it moves smoothly for a few frames then jumps, moves smoothly for a few frames, jums, repeated.
[00:28:55 CEST] <kepstin-laptop> which can happen with some detelecine filters, particularly on edited video
[00:28:59 CEST] <kyleogrg> how does it know when to fix this, though?
[00:29:52 CEST] <kepstin-laptop> kyleogrg: it basically looks for a sequence of frames where e.g. one in five frames has double the pts different from the others, then spreads those out evenly.
[00:30:06 CEST] <kyleogrg> ok. whats pts?
[00:30:23 CEST] <kepstin-laptop> frame play timestamp
[00:30:41 CEST] <kyleogrg> hmm. ok, i'll try this!
[00:31:00 CEST] <kepstin-laptop> I'm heading home now, i should try some of these filters on a few of my problematic sources
[00:31:05 CEST] <kyleogrg> with that sequence you gave (pullup,idet,yadif=deint=interlaced,dejudder), should I customize this at all?
[00:31:14 CEST] <kyleogrg> any special settings for yadif, for instance?
[00:31:15 CEST] Action: kepstin-laptop has some fun anime dvds with horribly broken telecine patterns.
[00:31:20 CEST] <kyleogrg> haha
[00:31:39 CEST] <kepstin-laptop> kyleogrg: if you like; but make sure yadif is running in a one-frame-in,one-frame-out mode.
[00:32:02 CEST] <kepstin-laptop> you can play around with the thresholds and edge exclusion in pullup, and the idet thresholds too
[00:32:09 CEST] <kyleogrg> ok. just one more question: how do i put this in your sequence?
[00:32:19 CEST] <kyleogrg> like yadif=3:0:0 or something
[00:32:33 CEST] <kyleogrg> yadif=3:0:0=deint....?
[00:32:42 CEST] <kepstin-laptop> kyleogrg: you start by reading the 'ffmpeg-filters' document to see what the syntax for ffmpeg filter chains is ;)
[00:32:48 CEST] Action: kepstin-laptop is off now.
[00:32:53 CEST] <kyleogrg> ok, thanks!
[00:33:52 CEST] <kepstin-laptop> but you can't mix the two parameter formats, you can either use "yadif=foo=10:bar=20" or "yadif=10:20"
[00:34:03 CEST] <kepstin-laptop> well, you can, but it's confusing.
[00:34:07 CEST] <kepstin-laptop> so I don't do that ;)
[02:29:36 CEST] <FedoraUser> I am trying to open analog tv with ffmplay under linux, but I can't seem to find enough information on how to parse options (such as input and frequency)
[02:30:47 CEST] <FedoraUser> 'ffplay -f v4l2 -standard PAL-BG /dev/video0' produces black picture
[02:31:58 CEST] <FedoraUser> is there a documentation that I'm missing that someone can point me to?
[02:41:28 CEST] <Prelude2004c> hey guys.. anyone now how nvenc will select a GPU to use ? i have 2 ( k4200 ) cards and its only using the one card not both
[02:41:33 CEST] <Prelude2004c> anyone know why?
[03:02:36 CEST] <FedoraUser> I'll be lurking here for a while, because I'm really stuck, so if you see my question later on don't hesitate to answer :)
[03:32:02 CEST] <kyleogrg> how can i set a volume level for all of the audio channels?
[03:32:22 CEST] <kyleogrg> it's a 5.1 surround sound video, and i need to be able to change the volume or silence it
[03:32:42 CEST] <kyleogrg> not individual tracks -- i mean the all the audio tracks
[03:35:06 CEST] <kyleogrg> never mind -- i think i got it
[04:23:57 CEST] <Prelude2004c> hey guys.. anyone now how nvenc will select a GPU to use ? i have 2 ( k4200 ) cards and its only using the one card not both
[05:42:06 CEST] <Prelude2004c> can anyone help with nvenc ?
[08:06:42 CEST] <gridtank> Tying to set Scan type to Interlaced. using recommended -flags +ildct in ffmpeg. The source is Upperfield and the target is DNxHD MXF. But media info reports back Scan type: Progressive and so does th Avid Mediacomposer. What am i doing wrong. SAMPLE: ./ffmpeg -i OLLETEST_MANSZELM_ANIMATION.mov -timecode 00:10:00:00 -threads 16 -map 0:0 -map 0:1 -map 0:1 -map_channel 0.1.0:0.1 -map_channel 0.1.1:0.2 -vcodec dnxhd -
[08:06:42 CEST] <gridtank> b 120M -acodec pcm_s24le -ar 48000 -flags +ildct -y test.mxf
[08:07:43 CEST] <gridtank> -b:v 120M offcourse
[09:38:40 CEST] <GuiToris> is there anybody here who knows how to use the deshake with ffmpeg?
[12:43:09 CEST] <d3fault_> Say you have a folder of N audio and video(only) files. The audio files came from one device and the video files from another. If the timestamps of the files are accurate (pointing to when the media file BEGAN), this tool I wrote can stitch togetether all of the intersecting timeapspans of audio/video. It can do multiple audio files per video (inserting silence as needed), and there's even a mode to interactively calculate +/- audio delays (with 15
[12:43:20 CEST] <d3fault_> second previews). It obviously uses ffmpeg. https://github.com/d3faultdotxbe/autobackup/tree/master/Projects/Directorie…
[12:48:05 CEST] <sleepypikachu> I'm trying to look up what I want to do but having a little bit of trouble expressing it. I have a directory containing images and a wav audio file, I want to create a video which cross fades between the images in a loop for as long as the wav file is (with the wav file as the background audio)
[12:50:57 CEST] <c_14> Possible, but very fiddly with the commandline tools.
[12:51:52 CEST] <sleepypikachu> c_14: Something else I considered was just calculating the length of time each image would need to be on screen for to fill the whole wav file
[12:52:25 CEST] <c_14> Both versions are possible, but would need the support of a scripting language.
[12:52:33 CEST] <c_14> Well.
[12:52:43 CEST] <c_14> If you only want to do it once you can also do the calculations manually.
[12:53:04 CEST] <sleepypikachu> Sadly no, I'm doing it more than once
[12:53:21 CEST] <c_14> Then you'll need a scripting language of some sort.
[12:53:21 CEST] <sleepypikachu> Guess I could just use python?
[12:53:42 CEST] <sleepypikachu> either with an explicit exec or some kind of binding
[12:53:53 CEST] <c_14> Popen is usually the method of choice
[12:54:23 CEST] <c_14> Which version would you prefer (each image has constant length, or only one loop)?
[12:54:50 CEST] <sleepypikachu> Probably constant length, if the wav is long and the number of images is low it would look strange
[12:55:26 CEST] <c_14> Right. So first you'll want to create a video clip that represents "single-loop"
[12:56:30 CEST] <sybariten> hey, can ffmpeg be run with a sort of short script as input as opposed to command line parameters, and could it make things clearer? :)
[12:57:29 CEST] <sleepypikachu> sybariten: You could create a script which has access to a sub set of the options and then calls with the command line parameters for you
[12:57:47 CEST] <sleepypikachu> sybariten: Or you can write a program which binds to the library directly using libav but I've done that and it's horrible.
[12:58:39 CEST] <c_14> With 2 images the command would look kinda like: ffmpeg -loop 1 -i image1.png -loop 1 -i image2.png -filter_complex '[0]trim=duration={time},fade=in=0:{fade_in_time},fade=out:{first frame to start fading out}:{number of frames to fade out}[v1];[1]{as_above}[v2];[v1][v2]concat=n=2:v=1:a=0[v]' -map '[v]' outvideo
[12:59:38 CEST] <c_14> Then you need to find the length of the video; concat that video with itself duration/video_length times (rounded up) with -shortest as an output option and add the wav audio.
[13:00:18 CEST] <sleepypikachu> -shortest?
[13:00:36 CEST] <c_14> Stops producing output when the shortest input (in this case the wav) ends.
[13:01:03 CEST] <sleepypikachu> ah. gotcha
[13:04:47 CEST] <sybariten> sleepypikachu: thanks for the insigt. Bummer that it's not a more established method
[15:26:33 CEST] <Popara> Hello
[15:26:49 CEST] <Popara> I'm trying to make FFmpeg to keep original timestamps and not to correct anything
[15:27:09 CEST] <Popara> I'm using this: ffmpeg -async -1 -vsync 0 -copyts -start_at_zero -i "stream" ...
[15:27:31 CEST] <Popara> Is that good? And also is there anything needed to put after the INPUT? or is ok as it is?
[17:47:53 CEST] <Popara> anyone?
[17:50:24 CEST] <capradmar> I handle timestamp like this: ffmpeg -i input.m4v -metadata timecode=10:00:00:00 output.m4v
[17:59:47 CEST] <Prelude2004c> good day everyone. Tried yesterday but got no responses. Can someone point me in the right direction please. I am using NVENC to encode and i ahve 2 cards in my system. I don't know the flag to have the system load balance between both cards as right now it is only using 1 GPU
[17:59:51 CEST] <Prelude2004c> can anyone help?
[18:16:10 CEST] <kyleogrg> I am experimenting with libx265, placebo. (I know it's super slow.) I am coverting DVD VOBs (originally 24p movies, now 30i) to 24p detelecined MKV.
[18:16:33 CEST] <kyleogrg> My question is, is there a film "tune" setting for libx265?
[18:17:43 CEST] <kyleogrg> Also, I am using crf 23. What other settings would (theoretically, not practically) improve the output quality (not size per se)?
[18:19:06 CEST] <RobotsOnDrugs> if you want quality nad don't care about size, there's a lossless option
[18:19:13 CEST] <RobotsOnDrugs> and*
[18:19:46 CEST] <kepstin-laptop> kyleogrg: all the options that really matter in regards to quality should be covered by the -preset
[18:20:10 CEST] <kepstin-laptop> other than that, it's just picking the quality vs. file size tradeoff, which you do with crf.
[18:20:15 CEST] <kyleogrg> okay, but i was reading about x264 (maybe that's a different animal) and folks were recommending a tune setting as well
[18:20:49 CEST] <kepstin-laptop> kyleogrg: x265 doesn't have particularly great psy optimizations yet, so afaik it doesn't have as many tune-type options as x264.
[18:21:15 CEST] <kyleogrg> ok, thanks
[18:21:38 CEST] <kepstin-laptop> (they are completely different projects, of course; the x265 people just picked the name for marketting reasons ;)
[18:23:28 CEST] <kyleogrg> ahhh yes
[18:23:45 CEST] <kyleogrg> well, there is apparently a tune grain setting, but not tune film.
[18:24:02 CEST] <kyleogrg> "--tune grain tries to improve the retention of film grain in the reconstructed output. It helps rate distortion optimizations select modes which preserve high frequency noise:"
[18:24:26 CEST] <kyleogrg> Could this be good if my source is a DVD?
[18:24:52 CEST] <kepstin-laptop> only if the source has heavy film grain that you want to preserve
[18:25:10 CEST] <kyleogrg> it is film
[18:25:19 CEST] <kyleogrg> or it was before it was telecined
[18:25:38 CEST] <kepstin-laptop> sue, but a) does it have heavy film grain in the video, and b) do you want to preserve this, rather than smooth it out?
[18:25:40 CEST] <kyleogrg> but you're saying it's not important unless it has unusually heavy grain?
[18:25:42 CEST] <kepstin-laptop> sure*
[18:25:50 CEST] <RobotsOnDrugs> DVDs usually mess up grain anyway
[18:26:11 CEST] <kepstin-laptop> yeah, dvd is pretty low res for any sort of decent looking film grain
[18:26:21 CEST] <RobotsOnDrugs> MPEG2, especially at DVD standards, is not efficient
[18:26:31 CEST] <kyleogrg> yes
[18:26:34 CEST] <kyleogrg> i was thinking that
[18:27:23 CEST] <kyleogrg> so is that a good reason to preserve whatever grain is left?
[18:27:23 CEST] <RobotsOnDrugs> i personally treat it as noise
[18:27:41 CEST] <RobotsOnDrugs> it's a personal preference
[18:27:59 CEST] <RobotsOnDrugs> watch the video and decide for yourself
[18:28:22 CEST] <kepstin-laptop> the main reason to preserve film grain (particularly "artificial" grain on high-def sources) is that it can mask banding in some cases.
[18:28:26 CEST] <kepstin-laptop> imo
[18:28:33 CEST] <RobotsOnDrugs> heh
[18:28:40 CEST] <kepstin-laptop> other than that, it's just personal taste
[18:28:50 CEST] <kyleogrg> ok
[18:29:00 CEST] <kepstin-laptop> and if you really like grain, but want efficient encoding, you can always use a video filter after decoding to add fake grain :)
[18:29:08 CEST] <RobotsOnDrugs> are you sure that the banding isn't caused by trying to filter out the grain?
[18:29:41 CEST] <RobotsOnDrugs> (or by smoothing it out in the lossy encoding process)
[18:30:06 CEST] <kepstin-laptop> the banding is caused by the limited bit depth of the colour; the grain sort of acts like dithering.
[18:30:33 CEST] <kyleogrg> ok
[18:30:41 CEST] <kyleogrg> if i want to use tune grai, how do i?
[18:30:43 CEST] <RobotsOnDrugs> depends on the source i guess
[18:31:02 CEST] <kyleogrg> how do i do it with -x265-params?
[18:31:26 CEST] <kepstin-laptop> kyleogrg: don't need to, x265 takes a -tune avoption
[18:31:42 CEST] <kyleogrg> ok
[18:33:12 CEST] <kepstin-laptop> I'm kind of surprised that there's no avoption for crf.
[18:34:25 CEST] <kyleogrg> after cropping the vob, the resolution is 707x461. Then I scale it to 706x460. I then set aspect ratio to 1.85. is this good/safe in your opinion?
[18:35:22 CEST] <kepstin-laptop> kyleogrg: probably better to round the crop to a multiple of 8 or 16, then don't scale.
[18:35:23 CEST] <Mavrik> might wanna crop it instead of scaling it
[18:36:02 CEST] <kepstin-laptop> most sd content you'll want to crop to a width of 704, and high varies based on content :/
[18:36:15 CEST] <kyleogrg> argh, but then i have to either lose video information or have a black border somewhere :)
[18:36:43 CEST] <kepstin-laptop> kyleogrg: yeah, well, someone watching on a television has overscan anyways, so you're already ahead :)
[18:36:50 CEST] <kyleogrg> yeah haha
[18:37:04 CEST] <kepstin-laptop> you're basically cropping off bits that aren't intended to be seen.
[18:37:35 CEST] <kepstin-laptop> you shouldn't have to manually set aspect ratio at any time; ffmpeg will propagate that through if the source is correct.
[18:38:23 CEST] <kyleogrg> that's another thing
[18:38:31 CEST] <kyleogrg> 1.85:1 is very close to 16:9
[18:38:51 CEST] <kepstin-laptop> dvd video isn't exactly 16:9 to start with, because of all the overscan and stuff.
[18:39:12 CEST] <kepstin-laptop> you just want to make sure the pixel aspect ratio is correct, then the video is whatever size it is.
[18:39:12 CEST] <kyleogrg> This film was 1.85 according it imdb
[18:40:09 CEST] <kyleogrg> well, the par is not square pixels, but when i set aspect ratio to 1.85 it looks ok
[18:40:11 CEST] <kepstin-laptop> that might explain why you're having to crop off the top and bottom, that's not needed on most content
[18:40:42 CEST] <kepstin-laptop> you shouldn't have to set the aspect ratio, ffmpeg will keep the par throughout and it'll just work.
[18:41:05 CEST] <kepstin-laptop> is this ntsc or pal source?
[18:41:06 CEST] <kyleogrg> but vlc doesn't display it correctly unless i do so
[18:41:23 CEST] <kepstin-laptop> hmm, they you might need to throw a setsar in the filter chain
[18:41:26 CEST] <kyleogrg> it's concatenated VOB files
[18:41:42 CEST] <kepstin-laptop> yes, but is it ntsc or pal video standard?
[18:41:53 CEST] <kyleogrg> ntsc
[18:42:38 CEST] <kepstin-laptop> right, throw a "setsar=40:33" at the start of your filter chain, and it should be good
[18:42:57 CEST] <kepstin-laptop> but that really should work automatically, if the vob files are correct :/
[18:43:04 CEST] <kyleogrg> http://pastebin.com/XaqimjzN
[18:43:22 CEST] <kyleogrg> ok.
[18:43:41 CEST] <kyleogrg> i don't know *exactly* what you mean. can you edit my paste?
[18:44:12 CEST] <kyleogrg> also, the "-tune grain" gives me an error: "Invalid preset or tune."
[18:44:30 CEST] <kepstin-laptop> kyleogrg: you take the filter "setsar=40:33" and put it at the start of your filter chain in the -vf option
[18:44:35 CEST] <kepstin-laptop> kyleogrg: don't bother with the tune.
[18:44:56 CEST] <kepstin-laptop> (also, drop the -aspect option if you add the setsar filter)
[18:44:56 CEST] <kyleogrg> and what is setsar?
[18:45:02 CEST] <kepstin-laptop> it sets the sample aspect ratio
[18:45:19 CEST] <kepstin-laptop> 40:33 is the nominal shape of a pixel in ntsc anamorphic widescreen video
[18:46:07 CEST] <kyleogrg> ok i'll try this
[18:46:16 CEST] Action: kepstin-laptop is off for lunch, back in a while.
[18:46:30 CEST] <kyleogrg> thanks, bye
[19:26:34 CEST] <jc0n> a quick ffserver question, I am useing ffmpeg to configure a ton of settings that ffserver also configures, should I copy all of the settings from ffmpeg to ffserver?
[19:26:56 CEST] <jc0n> things like audio rate, video and autio bitrate, etc
[19:44:15 CEST] <sfan5> jc0n: iirc ffmpeg automatically uses all the settings set in the ffserver configuration when you stream to ffserver
[19:44:40 CEST] <jc0n> sfan5: shouldn't it work when I use NoDefaults though?
[19:45:00 CEST] <jc0n> also it shows messages like Setting default value for audio bit rate when I already have it set
[19:45:09 CEST] <sfan5> shouldn't what work?
[19:45:14 CEST] <jc0n> ffserver
[19:45:41 CEST] <sfan5> ffserver wants to have configuration which is why it complains when you use NoDefaults and don't set the needed values
[19:46:26 CEST] <jc0n> oh do those get overridden by the ffmpeg command?
[19:46:35 CEST] <sfan5> not sure
[19:47:01 CEST] <c_14> ffserver ignores settings from ffmpeg (unless you add the override switch)
[19:47:07 CEST] <sfan5> you have NoDefaults in your ffserver configuration and pass a/v settings to ffmpeg?
[19:47:23 CEST] <jc0n> from ffmpeg
[19:48:06 CEST] <jc0n> ah the override switch
[19:48:09 CEST] <jc0n> perfect
[19:49:21 CEST] <jc0n> one last thing, I'm trying to add a Launch command to the Feed section of my conf and I have like 40 options to pass
[19:49:32 CEST] <jc0n> newlines break it
[19:49:57 CEST] <jc0n> is there some xml tag to escape the newlines?
[19:50:17 CEST] <jc0n> or should I flatten the whole thing into a huge line
[19:51:00 CEST] <jc0n> assuming the conf is aprsed as xml that is idk what the formatting is
[19:51:04 CEST] <sfan5> i would just do the latter
[19:51:15 CEST] <jc0n> alrighty then 0_0
[19:54:47 CEST] <jc0n> one last thing would you use http or rtp to stream locally
[19:57:07 CEST] <sfan5> i heard rtp performs better for streaming than http
[19:57:16 CEST] <sfan5> but that could be entirely wrong
[19:57:22 CEST] <sfan5> I'd suggest you to try what works best for you
[19:57:26 CEST] <kepstin-laptop> well, if you need low latency, rtp will normally be better
[19:57:28 CEST] <jc0n> alright thanks
[19:57:33 CEST] <kepstin-laptop> http might be easier to set up.
[19:57:42 CEST] <kepstin-laptop> depends.
[19:57:54 CEST] <jc0n> local network
[20:13:24 CEST] <Jackster22> Hi all! I need to copy a RTMP stream without bitrate loss
[20:13:32 CEST] <Jackster22> I have tried -copy but it changes things
[20:14:57 CEST] <kepstin-laptop> Jackster22: you mean -c copy?
[20:20:24 CEST] <Jackster22> yea something like that
[20:22:25 CEST] <kepstin-laptop> well, it can't be something like that, it has to be exactly that...
[20:22:41 CEST] <kepstin-laptop> (or, you know, -codec copy or one of the aliases, or with stream selection).
[20:25:00 CEST] <Jackster22> I don' have tjhe code right this min
[20:26:12 CEST] <kepstin-laptop> well, to dump the audio/video from an rtmp stream into a file, it should be as simple as "ffmpeg -i rtmp://example.com/whatever -c copy outputfile.mkv"
[20:26:39 CEST] <Jackster22> I am doing RTMP > RTMP
[20:26:50 CEST] <Jackster22> And that always results i n a lower quality stream
[20:27:40 CEST] <kepstin-laptop> how do you mean lower quality? is it getting corrupted?
[20:28:03 CEST] <Jackster22> It goes from 1mbps to 300k with -c copy
[20:28:13 CEST] <kepstin-laptop> ffmpeg doesn't reencode anything if you use -c copy, so the output quality is exactly the same.
[20:28:19 CEST] <Jackster22> It was a while sionce I have done it, just about to try again and probably will be the same
[20:28:22 CEST] <Jackster22> Ill try
[20:29:03 CEST] <kepstin-laptop> (make sure you actually look at the streams to compare; it's quite possible that a nominally "1mbit" source video actually has lower bitrate in some sections...)
[20:35:20 CEST] <pmlamotte> hello, I have an mp4 file that encounters an error when played through ie11 on windows 7 (but oddly enough, not windows 8 with the same windows media player and IE version). I ran it through ffmpeg and came across an error message and was curious what it meant.
[20:35:27 CEST] <pmlamotte> pastbin here: http://pastebin.com/6H2L9YJU
[20:35:46 CEST] <pmlamotte> relevant error is at line 95, the "non monotonically increasing" error
[20:41:20 CEST] <Jackster22> So I tried copy http://pastebin.com/K4K68Kyr
[20:41:52 CEST] <Jackster22> That was "ffmpeg -i rtmp://server/app/stream -c copy -f flv "rtmp://server/app/key""
[20:43:01 CEST] <Jackster22> not had that before
[20:43:45 CEST] <kepstin-laptop> not really enough context there for those messages to be meaningful.
[20:51:12 CEST] <jc0n> hey what stream format should I choose?
[20:51:14 CEST] <jc0n> avi?
[20:51:19 CEST] <jc0n> for audio and video
[21:04:04 CEST] <kepstin-laptop> jc0n: depends, what software do you want to be able to play it?
[21:04:50 CEST] <kepstin-laptop> i'd suggest avoiding avi for newer video codecs, it handles those poorly. mkv is a generally ok popular modern format.
[21:05:29 CEST] <jc0n> kepstin-laptop: for the stream format in ffserver
[21:06:31 CEST] <kepstin-laptop> jc0n: doing an http stream?
[21:06:40 CEST] <kepstin-laptop> what software are you using as the player?
[21:06:52 CEST] <jc0n> mpv
[21:07:07 CEST] <jc0n> yeah http or rtsp im experimenting
[21:07:18 CEST] <jc0n> looks like avi is the best video option but cant be sure
[21:07:43 CEST] <kepstin-laptop> i'd use mkv, probably for that case, it supports streaming fine.
[21:07:51 CEST] <jc0n> is that an option?
[21:08:05 CEST] <jc0n> it gives a list of formats in the example.conf
[21:08:13 CEST] <jc0n> that isn't one of them 0_0
[21:08:55 CEST] <kepstin-laptop> i wouldn't suggest using avi, but it might work if you don't need seeking, i dunno.
[21:09:24 CEST] <jc0n> if mkv is an option ill use it let me test it
[21:18:38 CEST] <DeadSix27> is there a way to make ffmpeg use a soundfont when converting midis?
[21:19:48 CEST] <kepstin-laptop> i'm not sure what you mean... ffmpeg can't play midi afaik...
[21:20:14 CEST] <DeadSix27> huh
[21:20:35 CEST] <DeadSix27> ffmpeg.exe -i "test.mid" -c:a flac test.flac
[21:20:36 CEST] <DeadSix27> worked tho.
[21:21:16 CEST] <DeadSix27> guess it uses timidiy
[21:21:29 CEST] <DeadSix27> as i get (on a windows machine) load_pat > can not open /usr/local/share/timidity/timidity.cfg, use environment variable MMPAT_PATH_TO_CFG for the directory
[21:23:40 CEST] <kepstin-laptop> that's pretty strange, standard ffmpeg builds don't do midi at all.
[21:23:57 CEST] <DeadSix27> its zeranoe's build
[21:24:07 CEST] <DeadSix27> maybe its compiled different, idk
[21:24:52 CEST] <kepstin-laptop> DeadSix27: what does the configure line on ffmpeg -version say?
[21:26:04 CEST] <kepstin-laptop> the zeranoe website doesn't list timidity as one of the bundled packages or anything.
[21:26:11 CEST] <DeadSix27> kepstin-laptop: http://pastebin.com/YKc2PJTn
[21:26:40 CEST] <kepstin-laptop> hmm.
[21:26:47 CEST] <kepstin-laptop> i wonder if it's something in libgme
[21:27:05 CEST] <DeadSix27> well, i aint mad
[21:27:23 CEST] <DeadSix27> would be nice to define soundfonts though
[21:27:55 CEST] <kepstin-laptop> you could just use e.g. timidity++ or fluidsynth or something directly to render the midi to a wave file with options of your choice.
[21:29:49 CEST] <DeadSix27> lets see
[21:34:57 CEST] <jc0n> woohoo its working
[21:35:06 CEST] <jc0n> ffserver -formats helped, format is 'matroska'
[21:35:20 CEST] <jc0n> last step is to get this playing on my rpi
[21:35:37 CEST] <jc0n> and voila drm free netflix streaming with pipelight
[22:15:14 CEST] <kyleogrg> hey
[22:15:48 CEST] <kyleogrg> I'm converting VOB files to MKV to watch on a laptop. Should I change the color space?
[22:18:40 CEST] <kepstin-laptop> probably not
[22:19:47 CEST] <kyleogrg> it's yuv420
[22:20:07 CEST] <kyleogrg> is that sort computer rgb?
[22:20:18 CEST] <kyleogrg> sort of
[22:20:29 CEST] <kepstin-laptop> no, but yuv420 is what the codecs can efficiently compress
[22:20:46 CEST] <kepstin-laptop> it's converted back to rgb in the player after decoding, usually on the graphics card.
[22:20:47 CEST] <kyleogrg> ok, so it's very common
[22:21:05 CEST] <kyleogrg> so VLC is converting it to computer RGB?
[22:21:32 CEST] <kepstin-laptop> either vlc is doing it, or vlc is sending the yuv420 stream to the gfc card and telling the gfx card to do it.
[22:22:02 CEST] <kyleogrg> cool, thanks
[22:24:11 CEST] <BtbN> You should be able to convert vob to mkv without re-encoding. Will still be mpeg2 then.
[22:24:22 CEST] <BtbN> And propably quite big, but without any quality loss.
[22:25:16 CEST] <kepstin-laptop> BtbN, kyleogrg is doing detelecineing, so it has to be reencoded anyways.
[23:10:06 CEST] <kyleogrg> Yeah, and I've been testing libx265, and I'm impressed by how SLOW it is.
[23:10:14 CEST] <kyleogrg> I'm using placebo.
[23:10:17 CEST] <kyleogrg> It's SO SLOW
[23:11:14 CEST] <kyleogrg> But can deliver impressive results at smaller file sizes than x264
[23:18:59 CEST] <klaxa> it's next gen, so obviously it's slower but better
[23:19:45 CEST] <kyleogrg> yes
[23:19:58 CEST] <kyleogrg> are there any tricks to make ffmpeg faster?
[23:20:23 CEST] <c_14> Don't use placebo?
[23:20:27 CEST] <kyleogrg> is it using all the cores in my processer?
[23:20:38 CEST] <kyleogrg> I mean without changing the encoding
[23:20:43 CEST] <c_14> Depends on the filters you're using.
[23:20:49 CEST] <c_14> You can always optimize the codepaths.
[23:20:59 CEST] <kyleogrg> you mean optimize my command line/
[23:21:00 CEST] <kyleogrg> ?
[23:21:53 CEST] <c_14> I meant the source code. If I can't see your command line I don't know if you could make it better.
[23:24:16 CEST] <kyleogrg> Here's my command line: http://pastebin.com/sXMSrgTA
[23:24:47 CEST] <kyleogrg> Idk if you see anything that can be optimized?
[23:25:18 CEST] <c_14> mhm, iirc scale doesn't thread very well. Should still be faster than the encoding though so unlikely to be the limiting factor
[23:25:53 CEST] <c_14> x265 is probably stealing all the cpu time
[23:26:09 CEST] <kyleogrg> yup
[23:26:16 CEST] <kyleogrg> super duper slow
[23:26:30 CEST] <kyleogrg> it's been going like 15 hours and has encoded about 8 minutes
[23:27:23 CEST] <kyleogrg> it's just a test anyway. this probably can't go any faster with my settings.
[23:27:36 CEST] <kyleogrg> Can ffmpeg use server farms?
[23:27:41 CEST] <kyleogrg> i mean render farms
[23:28:08 CEST] <c_14> You can cut the video up into tiny chunks, encode each of the chunks, then concat them together.
[23:28:34 CEST] <kyleogrg> oh? can you pretty much concat any codec like that?
[23:28:39 CEST] <kyleogrg> without problems
[23:29:30 CEST] <kyleogrg> i've wondered about that
[23:30:28 CEST] <c_14> Most.
[23:31:40 CEST] <kyleogrg> i wonder if there's any relatively easy way to use ffmpeg that way
[23:32:03 CEST] <kyleogrg> if there's a gui that will take an input video and split it up among many computers
[23:32:55 CEST] <c_14> https://github.com/klaxa/Distributed-encoding
[23:32:58 CEST] <c_14> ^may or may not work
[23:33:01 CEST] <c_14> It used to work.
[23:33:04 CEST] <klaxa> huh what?
[23:33:05 CEST] <klaxa> lol
[23:33:06 CEST] <klaxa> maybe
[23:33:15 CEST] <c_14> You can look at it for ideas though.
[23:33:19 CEST] <klaxa> i should probably do some stuff with it
[23:33:22 CEST] <c_14> Not sure if there's a gui for something like taht.
[23:33:36 CEST] <klaxa> not yet
[23:33:41 CEST] <c_14> *that
[23:33:43 CEST] <kyleogrg> THE klaxa?
[23:33:54 CEST] <klaxa> it's basically map-reduce with a server-client model
[23:34:11 CEST] <kyleogrg> what's map-reduce
[23:34:15 CEST] <klaxa> by now i would probably rewrite it from scratch
[23:34:27 CEST] <klaxa> https://en.wikipedia.org/wiki/MapReduce
[23:34:49 CEST] <kyleogrg> that would be cool if you rewrote it
[23:34:54 CEST] <kyleogrg> i'd test it for you
[23:35:56 CEST] <kyleogrg> so it would split a video, distribute it among the clients, and stitch it together?
[23:38:21 CEST] <klaxa> yeah pretty much
[23:39:04 CEST] <kyleogrg> Hmm, so if I wrote something like this
[23:39:28 CEST] <kyleogrg> on the client end, i could just have ffmpeg encode the vob chunk given to it
[23:39:47 CEST] <kyleogrg> as a x265 stream file
[23:40:16 CEST] <kyleogrg> the server would encode the audio, stitch the video chunks, and mux it
[23:40:37 CEST] <klaxa> no reason to not let the client encode audio too
[23:40:46 CEST] <klaxa> the server can be a client too connecting to itself
[23:41:39 CEST] <kyleogrg> yeah
[23:43:20 CEST] <kyleogrg> I guess the source video needs to be copied to all the clients?
[23:45:20 CEST] <c_14> only the parts that the client will be encoding
[23:45:50 CEST] <kyleogrg> but would the server simply break the file
[23:46:20 CEST] <kyleogrg> if the source video is a vob, will the server break the vob into binary chunks?
[23:48:16 CEST] <kyleogrg> because i wonder if ffmpeg would ever have trouble with a broken chunk of a video file
[23:49:48 CEST] <c_14> It should be fine so long as you split at i-frame boundaries
[23:50:10 CEST] <kyleogrg> ahh, so this isn't going to be simple for me
[23:50:26 CEST] <kyleogrg> because i don't know how to do that
[23:50:37 CEST] <klaxa> check the code i wrote, maybe there are hints
[23:50:46 CEST] <klaxa> i don't remember what i did, but i split at i-frames
[23:51:03 CEST] <kyleogrg> ok, and i don't even really know what i-frames are
[23:51:10 CEST] <c_14> klaxa: you used mkvinfo/mkvmerge so that wouldn't work in the general non-mkv case
[23:51:14 CEST] <kyleogrg> which codecs have them?
[23:51:15 CEST] <klaxa> oh yeah
[23:51:31 CEST] <klaxa> but you can put basically anything into mkv
[23:52:13 CEST] <klaxa> kyleogrg: basically all codecs if i'm not horribly mistaken
[23:52:35 CEST] <kyleogrg> haha
[23:52:37 CEST] <klaxa> an i-frame (or in this case more specifically an IDR-frame) is a frame at which a decoder can start decoding
[23:52:53 CEST] <klaxa> it's a full frame and no frame at a later point references a frame before the i-frame
[23:53:10 CEST] <kyleogrg> hmm, but the decoder wants to decode all frames right?
[23:53:35 CEST] <kyleogrg> ok, so it's like a chapter marker?
[23:58:54 CEST] <kyleogrg> ok, googled it
[23:59:16 CEST] <kyleogrg> but i need to know technically how to find idr frames in a mpeg2 stream
[23:59:32 CEST] <klaxa> https://en.wikipedia.org/wiki/Group_of_pictures
[23:59:58 CEST] <klaxa> well i went the way of muxing it to mkv and then using mkvtoolnix because it can output them in a simple format
[00:00:00 CEST] --- Thu Apr 16 2015
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