[Ffmpeg-devel-irc] ffmpeg.log.20140918

[burek]

[00:00] <c_14> Yes, not sure if that's done automatically though...
[00:01] <quebre> ok...
[00:02] <kepstin-laptop> the -target option is already rescaling the video to the correct 720x480/5XX for the dvd; i'd expect that ffmpeg is probably guessing something reasonable for aspect by default to tell the player if stretching/squishing is needed.
[00:02] <kepstin-laptop> but -aspect could override that if something goes wrong...
[00:05] <quebre> alright
[00:05] <quebre> how can i enable CUDA for ffmpeg? this conversion takes really long time ;P
[00:11] <quebre> seems it got converted
[00:11] <quebre> is it ok that original file had 1.73gb and output is 1.02gb ?
[00:13] <kepstin-laptop> your original was a 720p h.264 video, the output is a standard definition mpeg2 video; sounds about right.
[00:13] <quebre> oki
[00:22] <quebre> is it ok that the resulotion after conversion is 1280x720 ?
[00:23] <kepstin-laptop> huh, I thought the target option would set that.... I guess I was wrong
[00:23] <quebre> ;X
[00:25] <llogan> it should scale to 720x576 or 720x480
[00:25] <kepstin-laptop> yeah, the docs say it should be scaling
[00:25] <llogan> (see ffmpeg_opt.c)
[00:25] <kepstin-laptop> the -target option does the equivalent of setting -s, which will insert a scale filter if needed
[00:25] <quebre> i did: ffmpeg cartoon.mkv -target pal-dvd dvd cartoon.mpg
[00:25] <quebre> (im using 64bit ffmpeg uner windows 7)
[00:26] <quebre> i meant ffmpeg -i (...)
[00:26] <c_14> eh
[00:26] <llogan> remover that superflous dvd
[00:26] <c_14> ^that
[00:27] <quebre> ffmpeg cartoon.mkv -target pal-dvd cartoon.mpg
[00:27] <quebre> good ?
[00:27] <llogan> sure
[00:27] <quebre> ok
[00:27] <quebre> here we go again
[00:27] <quebre> that's the only downside. speed
[00:28] <quebre> or lack of speed
[00:29] <quebre> i know ffmpeg for longer time, i guess it's best converted in the world :p
[00:29] <quebre> converter*
[00:30] <llogan> kepstin-laptop: is the k6 done yet?
[00:33] <Suchiman> quebre: just get a frickin 36 core system, ffmpeg nicely scales with CPU cores ;)
[00:33] <quebre> yeah, one day.. when i get better job :PPP
[00:34] <kepstin-laptop> llogan: still going.
[00:34] <sacarasc> You're converting to MPEG2, I wouldn't be surprised if most of the CPU is being used for decoding. :p
[00:34] <quebre> 95% cpu usage :D
[00:34] <quebre> the ammount of ram does not help much i think
[00:34] <kepstin-laptop> it's onto the filter-pixfmts tests now...
[00:35] <Suchiman> quebre: you just need a motherboard that holds two of them: http://ark.intel.com/products/81061/Intel-Xeon-Processor-E5-2699-v3-45M-Cache-2_30-GHz
[00:37] <quebre> The Xeon E5-2699 V3 is listed for 3,960.26 EUR (...)
[00:37] <quebre> right.
[00:37] <Suchiman> i wonder how much cores x264 can utilize at maximum
[00:42] <foonix> maybe video height / 48
[00:46] <quebre> ok the resolution is now 720x576
[04:18] <molikto> Hi, anyone here?
[04:18] <molikto> question: Can I get metadata for a stream? because I want to get the rotation of a video but the metadata does not appears if I use ffmetadata
[04:35] <molikto> is this channel dead?
[04:36] <EvolE_> quebre: probably all are sleeping
[04:36] <EvolE_> ...
[08:46] <anshul_mahe> how to set explicitly maximum_number_of_frames in x264
[09:04] <K4T> frame=3407399 fps= 50 q=30.0 size=37510276kB time=18:55:47.97 bitrate=4509.1kbits/s DUP=14 DROP=0
[09:04] <K4T> what is that DUP and DROP?
[09:04] <K4T> it is not always showing
[09:05] <BtbN> duplicated and dropped frames.
[09:05] <K4T> it means that 14 frames is duplicated in every second, yes?
[09:06] <sacarasc> Total, I thought.
[09:06] <K4T> ow, from ffmpeg start, yes?
[09:06] <sacarasc> Yes, I think so.
[09:07] <Mavrik> yes
[09:07] <Mavrik> dup= - number of frames that were duplicated to keep up with desired framerate
[09:07] <K4T> so, nothing to worry, 14 duplicated frames per 19h streaming
[09:07] <Mavrik> drop= - number of frames tropped to keep up with farmerate
[09:07] <K4T> Mavrik, thank you
[09:07] <BtbN> more like dropped because of bad network connection
[09:08] <BtbN> accidently encoding too many frames doesn't happen
[09:08] <K4T> video card is input for ffmpeg, so I thjink it is not related to network environment
[09:08] <BtbN> The input doesn't matter at all for dropping frames
[09:28] <K4T> what will happen when I run ffmpeg on machine which have 2 CPUs, each with 4 cores? All CPUs and cores (8) will be used by that ffmpeg instance?
[09:28] <sacarasc> Depending on the encoder, yes.
[09:31] <anshul_mahe> I am using libx264 when I encode the video it sets log2_max_frame_num to 4 which means maximum frames as 2^4+4 = 256, since i concat that video with other large video
[09:31] <K4T> sacarasc, I will use libc264 for video and aac for audio
[09:31] <K4T> libx264*
[09:31] <anshul_mahe> I need to set the value of log2_max_frame_num to 6 means number of frame 1024
[09:32] <sacarasc> K4T: That should use all 8 cores.
[09:33] <anshul_mahe> If I directly use x264 then there is option called --frames and how to use that option in ffmpeg, I tried -x264opts but that fails for this option
[09:47] <K4T> anshul_mahe, try -x264opts frames
[09:48] <anshul_mahe> K4T: not working
[09:48] <K4T> paste error
[09:48] <K4T> and command line
[09:51] <anshul_mahe> here is all you wanthttp://pastebin.com/NYbe9u7e
[09:51] <anshul_mahe> here is all you want http://pastebin.com/NYbe9u7e
[09:54] <K4T> what frames is supposed to do? I can not find doc
[09:56] <anshul_mahe> from x264 help       --frames <integer>      Maximum number of frames to encode
[09:57] <K4T> maybe you can use ffmpeg argument like -r or fps vilter?
[09:57] <K4T> filter*
[10:01] <K4T> anshul_mahe, pm
[10:20] <relaxed> anshul_mahe: -frames:v 1024
[10:26] <okokokno> Hi. I'm trying to use libavformat but are noticing that for many supported formats, there's a lot of missing file information
[10:27] <anshul_mahe> relaxed: not working, in actual I dont want to limit my output, I want to set log2_max_frame_num to 6 in SPS of H264
[10:28] <okokokno> for example, for mp3 files libavformat doesn't provide channel count and samplerate
[10:28] <okokokno> do i have to parse the formats myself?
[10:31] <okokokno> *The format's header info
[10:38] <myubuntu__> Can I disable faac's three frames
[10:38] <myubuntu__> Can I disable faac's three frames's buffering?
[10:57] <K4T> is RAM important for ffmpeg enconding?
[10:59] <__jack__> not really
[10:59] <K4T> I think I will have max 8 instances of FFMPEG, each of them will create 4 streams. Inputs for each instance will be video card
[10:59] <__jack__> depending on what you're encoding (source, size, format, codec etc), you 'll require a min of ram, but increase that value won't make it faster
[10:59] <K4T> 8GB will be enough or I should buy 16GB?
[11:00] <K4T> 720p quality
[11:00] <__jack__> codec ? options ? you may test
[11:01] <K4T> x264 for video and aac for audio
[11:01] <__jack__> you should test
[11:01] <__jack__> you must test ?
[11:01] <__jack__> hum
[11:02] <K4T> ok
[11:03] <__jack__> If you want a really answer: get 16GB, it's not really more expensive, and may be usefull in the future
[11:04] <K4T> yeah, 16 should be enough
[11:04] <K4T> now I have one ffmpeg with one streamd and it takes 300MB
[11:04] <K4T> I will get probably 16 streams
[13:20] <xtruder> when i sent multi bitrate streams to my rtmp server, it recognizes only one. When i do the same with FMLE it recognizes them correctly. Is this due to ffmpeg not able to align keyframes?
[13:34] <sekon> if i do ffmpeg -i /path/to/ip/file
[13:35] <sekon> I get .... Stream #0:0(eng): Audio: pcm_s16le (sowt .....
[13:35] <sekon> so what does sowt mean ?
[13:36] <c_14> > 'sowt' ('twos' spelled backwards) also denotes signed linear PCM. However, 16-bit data is stored in little endian format.
[13:36] <c_14> AFAIK it's just a QuickTime identifier.
[13:36] <sekon> c_14: thanks
[13:37] <c_14> It's not like the le in pcm_s16le already stands for little endian or anything. Nah, we need more identifiers&
[13:37] <Dark-knight> kerploop
[13:39] <sekon> c_14: your help was greatly appreciated
[13:39] <sekon> not just by me .. but by others here too
[14:24] <wysaid> Anybody have the static release library of the latest ffmpeg? Please send me a link~~~~~~thanks a lot
[14:28] <c_14> wysaid: the gusari builds haven't been updated since july, just stick with the johnvansickle builds.
[14:29] <wysaid> sorry, I forgot to say that I am asking for the builds of windows platform...
[14:30] <c_14> zeranoe
[14:30] <c_14> http://ffmpeg.zeranoe.com/builds/
[14:31] <c_14> https://ffmpeg.org/download.html
[14:31] <c_14> They're all linked there.
[14:31] <c_14> For future reference.
[14:34] <wysaid> I know that, I'm using the package for Dev, but it doesn't include the dll files. The program always report an error: " missing avcodec-56.dll"
[14:34] <wysaid> The other downloads are 'exe' files
[14:35] <wysaid> Can you tell me which one is for developing? Thanks a lot~
[14:35] <wysaid> I am wasting a lot of time here
[14:37] <__jack__> that's so kind, thank you
[14:37] <c_14> If I'm reading this correctly, you need the shared and the dev package.
[14:39] <wysaid> Yes, I used to think the three downloads "static", "shared" and "dev" are for the same usage...
[14:39] <wysaid> I'm downloading the shared version...
[14:39] <wysaid> thanks~
[14:45] <wysaid> I found the DLLs! @_ at b
[14:47] <lipizzan> What would be an efficient console video player (framebuffer) to "tee" output to.
[14:48] <lipizzan> .-?
[14:48] <lipizzan> .->?
[15:00] <c_14> lipizzan: I remember mplayer having framebuffer support.
[15:05] <lipizzan> c_14:  yes, i'm testing with it. I have an old version though, and i cant re-position the video on the screen (-geometry=x:y).
[17:21] <lipizzan> when encoding dv->h.264->ts container file, is there some switch I should use to maintain audio\video sync?  video races ahead of audio on my fresh compiled mplayer when playing the ts file (linux console, X-less).
[17:25] <klaxa|work> how are you rendering the video?
[17:25] <klaxa|work> is your computer fast enough to render the video without X?
[17:26] <Mavrik> hmm
[17:26] <Mavrik> lipizzan, are you somehow clobbering the timestamps on frames?
[17:29] <lipizzan> klaxa|work: It;s <2KB/sec bitrate (-preset faster... when generated) , and cpu is < 30% during playback.
[17:29] <lipizzan> Mavrik: Not sure. How would I find out?
[17:29] <Mavrik> are you piping stuff around without containers?
[17:30] <lipizzan> The work flow is   dvgrab(dv stream)->ffmpeg(h.264->ts)->ts file.
[17:32] <lipizzan> should I be using a bitsteam filter on the audio to when creating the h.264\ts? I use one on the video.
[17:33] <lipizzan> should I be using a bitsteam filter on the audio also when creating the h.264\ts? I use one on the video.
[17:43] <excalibr> HOw do you specify milisecond with -ss and -to param?
[17:44] <Ess4_> -ss 1.23
[17:46] <Ess4_> excalibr, you can use -ss param like that : -ss 00:00:00.123
[17:48] <excalibr> thank you sir
[19:36] <tab1293> if I am transcoding a mp3 file to aac what should I use as my output format specified with the -f flag?
[19:40] <Mavrik> tab1293, mp4
[19:42] <okokokno> Hi. I've been here earlier and asked a question but later disconnected so I'll ask again, sorry.
[19:43] <tab1293> Mavrik, I am getting this error Could not write header for output file #0 (incorrect codec parameters ?)
[19:44] <tab1293> okay hold on
[19:46] <okokokno> avformat seems to, for some formats like mp3, not provide any info about the file (e.g. channels, sample rate)
[19:47] <Mavrik> okokokno, are you streaming the file?
[19:48] <okokokno> Is there a way to get this info via the av libraries or do I have to parse the format header myself?
[19:48] <Mavrik> also, how are you reading that data?
[19:49] <okokokno> I wish to stream or otherwise read the file later, but first I wish to get its info
[19:50] <Mavrik> you're not answering my questions. HOW are you trying to read the files that you don't get that info?
[19:52] <tab1293> Mavrik here is the command and error http://pastebin.com/LrnedP9h
[19:53] <Mavrik> tab1293, you need to write mp4 / m4a to file
[19:53] <Mavrik> not pipe it somewhere.
[19:53] <Mavrik> also.
[19:53] <tab1293> Mavrik, why can't I pipe it?
[19:53] <Mavrik> [libfdk_aac @ 0x29d2b20] Note, the VBR setting is unsupported and only works with some parameter combinations
[19:53] <okokokno> I'll take a look and answer, thanks, Marvik
[19:53] <Mavrik> tab1293, because index cannot be written before file is complete.
[19:53] <Mavrik> tab1293, see, you get this error: [mp4 @ 0x29c42e0] muxer does not support non seekable output
[19:54] <tab1293> yeah
[19:54] <tab1293> so I can't transcode on the fly
[19:54] <tab1293> with aac at least
[19:54] <Mavrik> of course you can transcode on the fly.
[19:54] <Mavrik> you just can't write it to MP4/M4A container.
[19:54] <Mavrik> use something else.
[19:55] <tab1293> Mavrik, let me explain what I am trying to do
[19:56] <tab1293> I am trying to use python to transcode an mp3 to aac. So I have it set up that the mp3 file data is being passed to stdin using python's subprocess command. I would like to then take the stdout and write it to an amazon s3 bucket
[19:56] <tab1293> So I am trying to work with the raw data in python so I can upload ffmpeg's output to amazon
[19:57] <tab1293> what container supports on the fly transcoding?
[19:57] <Mavrik> you're asking the wrong question.
[19:57] <Mavrik> which containers must you support for delivery to the player?
[19:57] <Mavrik> how are you going to play stuff?
[19:57] <Mavrik> are you going to remux on delivery?
[19:57] <Mavrik> when you have those answers, then you can choose a container :)
[19:59] <tab1293> Mavrik, I was planning on using a container that is supported in most browsers hence the mp4
[19:59] <kepstin-laptop> and if the answer ends up being 'mp4', you're going to have to buffer the complete file before uploading it, for example by saving a temporary file on disk.
[20:00] <tab1293> Mavrik, does mp3 support being written to a pipe?
[20:00] <Mavrik> no idea
[20:00] <tab1293> kepstin-laptop yeah I guess I can just make a temp file
[20:00] <Mavrik> make a temp file and upload that
[20:00] <tab1293> yeah alright
[20:00] <Mavrik> browsers really don't support anything else than mp4 / mp3
[20:01] <Mavrik> so your choices are very limited in that regard
[20:01] <Mavrik> sadly people are dragging their asses as hell with HLS support which uses MPEG2-TS container
[20:01] <kepstin-laptop> mp3 in its "native" format, when not using vbr encoding, doesn't need a seek index, so you can stream it fine. If you use vbr, I think there's an optional xing header to add seeking, I'm not sure if it can be written at the end of the file or needs space reserved at the start.
[20:02] <tab1293> Mavrik, I was actually thinking about trying to write a hls client in javascript. It seems like it could be done using aurora.js
[20:02] <Mavrik> yuck :)
[20:02] <Mavrik> we just used flash
[20:02] <Mavrik> way more supported, way better experience for users :)
[20:03] <tab1293> Isn't flash a nono nowadays?
[20:35] <Mavrik> tab1293, why would it be?
[20:35] <Mavrik> mobile platforms support HLS natively
[20:35] <Mavrik> desktop people can use Flash because it still isn't as shitty as 5000 lines of JS decoder :)
[20:36] <tab1293> Mavrik, mobile browsers support HLS natively?
[20:36] <tab1293> how would you embed the HLS stream for mobile?
[20:37] <Mavrik> ?
[20:37] <Mavrik> you just point the video tag to m3u8
[20:39] <tab1293> Mavrik, hmm okay I didn't know that. So are there any open source flash hls players?
[20:39] <Mavrik> I think JWPlayer has HLS module
[20:39] <tab1293> but you developed your own player?
[20:40] <okokokno> Hi Mavrik. I'm opening the mp3 file via avformat_open_input, providing a path. Is this what you meant by how I'm reading the data?
[20:40] <okokokno> (Sorry it took so long to answer)
[20:41] <Mavrik> okokokno, yes, and you're getting the data form AVStream right?
[20:41] <Mavrik> tab1293, honestly I do backend work so I don't know which option the frontend guys took at the end :)
[20:41] <Mavrik> I just deliver video ;)
[20:42] <okokokno> I'm not getting any data before I'm trying to get the format's info
[20:42] <okokokno> E.g. via <codec context>->channels, which is 0
[20:43] <Mavrik> uhm
[20:43] <okokokno> I need the codec's/format's info long before I want to start reading the actual audio samples
[20:44] <okokokno> Do I need to do something different?
[20:45] <Mavrik> okokokno, you need to call avformat_find_stream_info on the codec context
[20:45] <Mavrik> to probe the input and get all that data populated
[20:45] <Mavrik> that will read your input up to probesize / analyzeduration and extract data
[20:46] <okokokno> Ah
[20:46] <okokokno> I'll try that, thanks
[20:48] <EvolE> tab1293: for hls you can use http://www.flashls.org/ with videojs for example (or with any osmf player)
[20:49] <tab1293> Mavrik, do you know if you use ffmpeg to create a HLS live stream with the +live flag do the segments get automatically deleted?
[20:49] <Mavrik> I doubt it
[20:49] <okokokno> Thanks, it worked!
[20:49] <tab1293> EvolE, awesome thanks
[20:50] <tab1293> Mavrik, so what does the +live flag change?
[20:50] <Mavrik> hmm, I don't even see the option here: https://www.ffmpeg.org/doxygen/trunk/hlsenc_8c_source.html
[20:51] <Mavrik> ah
[20:51] <Mavrik> 00573     { "live",              "enable live-friendly list generation (useful for HLS)", 0, AV_OPT_TYPE_CONST, {.i64 = SEGMENT_LIST_FLAG_LIVE }, INT_MIN, INT_MAX,    E, "list_flags"},
[20:51] <tab1293> Mavrik, https://www.ffmpeg.org/ffmpeg-formats.html#segment_002c-stream_005fsegment_002c-ssegment under options
[20:51] <Mavrik> https://www.ffmpeg.org/doxygen/trunk/segment_8c-source.html
[20:52] <Mavrik> it seems to just add this to m3u8: 00203                         "#EXT-X-TARGETDURATION:%"PRId64"\n", seg->time / 1000000);
[20:53] <tab1293> hmm okay
[20:54] <tab1293> do you think adaptive bitrate streaming is useful when just streaming audio? I am trying to decide if my app should stream with hls or just transcode to m4a and serve them statically
[20:55] <Mavrik> I doubt it's useful
[20:55] <Mavrik> I even doubt the audio does enough bitrate difference for players to even switch
[20:55] <tab1293> Both formats are supported by the popular browsers (hls with flash) and mobile so I am trying to decide if hls will give better performance
[20:56] <tab1293> or any benefit over using m4as
[20:57] <tab1293> so you think not?
[21:34] <EvolE> tab1293: using hls should be ok if you plan to live stream and have a lot of concurrent listeners, because it's easy to cache on the server or cdn. not sure how it works with audio, but i suppose it should be ok
[21:44] <Suchiman> is it just me or does ffmpeg rarely match the timestamps other players report? e.g. when i cut something with the timestamps from the players, the result will not be exactly what i've targeted
[21:45] <jehar_work> Are you cutting h264 videos? There's the keyframes to keep in mind whenever you're slicing without doing a re-encode.
[21:46] <Suchiman> jehar_work: the difference is up to 30 seconds so i don't believe it's just about keyframes, is it? i also tried moving -ss before and after the Input file
[21:46] <Suchiman> but in this case it's slicing a vob file
[21:46] <Suchiman> so mpeg2
[21:46] <jehar_work> Hmm, that'd be outside my head then.
[21:50] <kepstin-laptop> vob files are mpeg-ps; a quick search reveals that at least some players (vlc and maybe gstreamer stuff?) have had issues with timestamps in those. Fixed by using the avformat (ffmpeg) demuxer.
[21:50] <kepstin-laptop> so, make of that what you will :/
[21:59] <Suchiman> kepstin-laptop: well right now i try to repair a broken DVD Video, 2 titles won't read by MakeMKV but Play fine in VLC. copying them with mplayer (dvdnav just stops with error, dvd works) will break the titles even more. But i managed to put all vobs together using ffmpeg -i "concat:..." -c copy huge.vob (which plays fine) and now i'm trying to slice it by
[21:59] <Suchiman> adding the title lengths 1-3 reported by mplayer to get the start Position of title 4 and then stream copy that Segment with fflags +genpts and -ss ... -t ... result.mkv which produces a playable mkv again.
[21:59] <Suchiman> all fun stuff lol
[22:02] <okokokno> I see AVCodecContext::block_align tells me the size of each packet read from the file but not decoded yet
[22:03] <okokokno> Is there a similar value that tells me the decoded size of each packet, if known?
[22:03] <okokokno> (For codecs where this is constant)
[22:59] <active8> is there a syntax error here? ffmpeg -f alsa -ac 2 -i pulse -f x11grab -r 30 -s 1920x1080 -i :0.0 -a:c pcm_s16le -v:c libx264 -preset ultrafast -crf 0 -threads 0 09-14-2014-cast.mkv
[22:59] <active8> Unrecognized option 'a:c'. Error splitting the argument list: Option not found
[23:00] <lipizzan> active8: think that should be -c:a
[23:00] <active8> yeah - copy pasted from article, duh!
[23:00] <lipizzan> -c:a   aka, codec:audio
[23:02] <active8> right. now this: Invalid loglevel "libx264". $ffbin -f alsa -ac 2 -i pulse -f x11grab -r 30 -s 1920x1080 -i :0.0 -c:a pcm_s16le -v:c libx264 -preset ultrafast -crf 0 -threads 0 09-14-2014-cast.mkv Invalid loglevel "libx264". Possible levels are numbers or:
[23:02] <active8> i tried it without the preset, too. I might not have the preset, but even without it, same error
[23:03] <active8> $ffbin is a shortcut to a recent build
[23:03] <active8> = ffmpeg
[23:06] <Suchiman> active8: -c:v not v:c
[23:07] <Suchiman> c:v = codec:vide
[23:07] <active8> oh my. I must really need a shot to the head. lemme try that.
[23:09] <active8> that's better. sorry, but thanks.
[23:09] <active8> why, when there's no audio playing do I get: Non-monotonous DTS in output stream 0:1  ?
[23:09] <active8> I sometinmes get that even with audio playing
[23:28] <jehar_work> I'm doing some initial tests with x265 - it looks like I'm getting the same filesize with 1 pass as I am with 2 - is there any implication that I can get everything done with 1?
[23:34] <kepstin-laptop> jehar_work: nah, that just means that their 1-pass bitrate algorithm is really strong. You should still prefer 2-pass, because it uses knowledge of future frames to create a file that's roughly constant quality
[23:35] <kepstin-laptop> rather than dropping quality in hard bits and wasting file size on easy bits
[23:36] <llogan> jehar_work: use -crf, one pass. or use -b:v with two passes if you are targeting a specific output file size.
[23:36] <Suchiman> *thought that crf is better than two pass and two pass should only be used if e.g. constant file size is desired
[23:36] <jehar_work> gotcha
[23:37] <kepstin-laptop> with x264, crf and 2-pass are exactly equivalent, given the same output file size. I don't know if that holds for x265
[23:37] <Suchiman> what x_X
[23:37] <llogan> i read 26*4*
[23:37] <llogan> damned similar names...
[23:37] <jehar_work> Me neither - I'm just scoping out the current time/filesize footprint currently.
[23:38] <kepstin-laptop> x264's 2-pass algorithm works by doing one pass, using the results to calculate a crf value, and using that for the second pass. I have no idea what x265 does for that :/
[23:38] <jehar_work> Fortunately I can likely go "there's no huge gain at the moment, let's check again in 2 months" and get on with my day :D
[23:39] <kepstin-laptop> well, to see the difference between 1- and 2- pass, you'd have to watch the output and see how the quality is
[23:39] <kepstin-laptop> both should be able to get very close to the target bitrate in a good codec.
[23:40] <jehar_work> Yeah, I've been having trouble checking the visual quality with the vlc nightly - is there a decoder that's working well?
[00:00] --- Fri Sep 19 2014