[Ffmpeg-devel-irc] ffmpeg.log.20191231

[burek]

[00:28:20 CET] <TheWild> hello
[00:28:36 CET] <TheWild> someone knows quickly how to apply a lavfi filter?
[00:30:27 CET] <BtbN> -vf somefilter
[00:32:37 CET] <TheWild> hmm... actually it's audio. Tried -af, -a:f and -fa but didn't work. -lavfi seems to work (at least ffmpeg doesn't complain)
[00:32:58 CET] <TheWild> but it complains about -ss 31:06 -to 50:30
[00:34:26 CET] <TheWild> the file definitely has at least 55 minutes.
[00:36:17 CET] <TheWild> okay, chosen the shortened file accidentally.
[00:36:36 CET] <TheWild> Things seems to work now.
[01:36:42 CET] <rkantos> Any idea why I am getting real time buffer too full frame dropped messages when just trying to playback capture card feed with ffplay? Using OBS/VLC i get no/very little drops
[01:41:49 CET] <pink_mist> ffplay isn't really supposed to be used in normal circumstances - it's mainly a debugging tool, not a general purpose player
[01:42:08 CET] <pink_mist> if things are working in real players, I don't get why you're using it, frankly
[01:45:26 CET] <rkantos> Well.. I am experiencing the same issue with ffmpeg too. I'm trying to stream rtp_mpegts while using h264_qsv. I was trying to use ffplay for debugging this, because I was thinking that maybe the Camera that I am using is pushing too much data.. (Sony A7s)
[01:48:29 CET] <furq> rkantos: increase -rtbufsize
[01:52:42 CET] <rkantos> furq: Yeah, I have tried it, but what I see with ffplay performance wise, seems to also reflect what I get in ffmpeg. E.g. increasing rtbufsize doesn't really help and the only reason I can think is the 1080p50 yuv2 feed
[01:54:05 CET] <rkantos> I am trying to understand the differences at the moment, or why it works with obs/vlc
[02:50:44 CET] <barhom> Currently using this scale: scale=-2:'min(576,ih)' to say: "Scale to ih to 576 unless ih is <576". This still passes everything to the scaler filter.
[02:51:22 CET] <barhom> Is it possible to do an "If" check on ih and pass the scaler completely if I do not want to scale since the resolution is already correct? (and more importantly, does it save cpu cycles of so?)
[03:19:31 CET] <DHE> barhom: if scale really needs to do nothing, then it will do nothing in the pipeline. but keep in mind it may do stuff you don't realize.
[03:20:22 CET] <DHE> eg: -2 for the width, if the input had an odd width, would result in the width changing by 1 and hence resizing. the scale filter is also often used for colorspace conversions
[03:21:23 CET] <barhom> I take it from your answer that you cant really do an if-conditioned filter.
[03:22:19 CET] <barhom> I was basically hoping for something like "filter_complex=if(gt(ih,576),scale=-2:576)) Which could skip the entire filter as if I didnt even type in "scale" at all
[03:23:49 CET] <barhom> I havent had that problem with the -2, but I see your point
[03:25:28 CET] <barhom> Even though in my cases the scale filter probably does nothing since the requested scale resolution is the same as the input resolution I was hoping to be able to skip the frames even entering the scale filter. Saving minimal cpu and in the case of filtering on scale_cuda, you save a few MiB of vram
[03:30:12 CET] <DHE> the scale filter is explicitly coded to do nothing if nothing needs doing and just pass the original frame right through. so it wouldn't even copy anything
[03:33:23 CET] <barhom> Thats good, means what Im doing should be ok to continue doing. You do not see any value to be able to skip entire filters using if statements?
[03:33:46 CET] <barhom> I tried scale_cuda again with passing the same resolutions and I do see an increase of about 4MiB of vram usage when using scale_cuda and not.
[03:34:07 CET] <DHE> many filters have an "enable" option that, when evaluated to 0, do nothing. but that's more intended for editing
[15:51:36 CET] <rkantos> I get real time buffer too full even if I go to 1fps, while when having 30fps, it seems to be able to sustain about 20fps.. It makes no sense to me
[16:03:18 CET] <rkantos> Even libx264 enconding at 1fps leads to real-time buffer [Cam Link] [video input] too full or near too full
[16:56:09 CET] <grosso> hi
[16:57:19 CET] <grosso> I notice cache protocol does'nt work on Android and IOS, because ffmpeg can't open temp files on those platforms
[16:57:46 CET] <grosso> there exist a workaround to this?
[17:09:37 CET] <grosso> Trying to use cache protocol on Android and Ios, I receive this error: ff_tempfile: Cannot open temporary file
[17:09:58 CET] <grosso> there is a way to work on this?
[18:16:57 CET] <grosso> cache protocol just don't works on Android and Ios, ff_tempfile: Cannot open temporary file
[18:17:11 CET] <grosso> there is such a workaround to this?
[19:01:56 CET] <grosso> cache protocol just don't works on Android and Ios, ff_tempfile: Cannot open temporary file
[19:01:58 CET] <grosso> there is such a workaround to this?
[19:03:18 CET] <durandal_1707> do not use cache protocol
[19:08:49 CET] <grosso> I really need it
[19:09:02 CET] <durandal_1707> i doubt so
[19:10:32 CET] <grosso> well, it is intended to seek on a http stream... I can solve by reconnecting and so on, but cache protocol just does it easy and faster
[19:13:03 CET] <grosso> av_seek_frame just don't works on my stream... it tryies to reconnect under the hood, but the server don't responds as expected... I can solve it in server-side, but cache protocol is just better
[19:14:40 CET] <grosso> (sorry about bad english)
[19:16:09 CET] <grosso> maybe there is a way to specify temp dir to cache protocol, since it can't write to default temp dir on Android and IOS
[19:18:06 CET] <grosso> it's such a trivial workaround, how it's not implemented yet?
[19:22:40 CET] <grosso> https://patchwork.ffmpeg.org/patch/3211/ ?
[22:10:32 CET] <Tigzee> I'm trying to save an rtsp stream from my camera by copying (I do not wish to use cpu to re-encode).  I have it saved to mkv, the video is mostly working with the odd glitch (maybe cameras fault), but the audio isn't working at all (silence).  When using VLC to connect to the rtsp directly (stereo AAC 32000Hz) but the vlc playing back the saved mkv file says (AAC 16000Hz and does not specify stereo)
[22:14:53 CET] <Tigzee> I think ffmpeg is misreading what the input audio stream is, how do I override that maybe?
[00:00:00 CET] --- Wed Jan  1 2020