[Ffmpeg-devel-irc] ffmpeg.log.20161007

Sat Oct 8 03:05:01 EEST 2016

[00:49:17 CEST] <Spring> I don't understand the 'ffmpeg' sub-domain in the pastebin.com link, it just redirects to the home page, am I missing something?
[01:09:39 CEST] <llogan> Spring: what link are you referring to?
[01:12:18 CEST] <user314> How can I send a video to /dev/video0?
[01:12:21 CEST] <user314> ffmpeg -i movie.mp4 -f v4l2 /dev/video0
[01:12:27 CEST] <user314> is not correct
[01:14:10 CEST] <llogan> ffmpeg v4l2 is an input device only
[01:14:12 CEST] <Spring> llogan, the ChanServ auto channel message
[01:15:07 CEST] <llogan> Spring: ah. i don't know why that is there. seems superfluous.
[01:18:45 CEST] <llogan> user314: i was wrong. there is a v4l2 output device
[01:19:30 CEST] <llogan> lazy documentation maintainers
[01:19:47 CEST] <llogan> llogan: patch welcome
[01:20:12 CEST] <user314> I ll given the answer in a week: http://superuser.com/questions/1132310/how-can-i-send-a-video-to-my-webcam
[01:20:54 CEST] <llogan> why a week i wonder
[01:28:58 CEST] <klaxa> doesn't rawvideo need dimensions and fps? (maybe fps is optional?)
[01:29:04 CEST] <kuroro> i have 2 short clips that i have merged via iOS code that's not playing properly in Chrome (http://pastebin.com/raw/mhG2GZP2).
[01:29:28 CEST] <kuroro> i was wondering whether the file was not encoded incorrectly on the iphone side or whether its a bug in chrome
[01:29:55 CEST] <klaxa> >pipeline_error: decoder: not supported
[01:29:57 CEST] <kuroro> looking at the chrome://media-internals/ output ( which i included in the pastie), it shows "audio decoder initialization failed"
[01:30:02 CEST] <klaxa> no h264 decoder?
[01:30:12 CEST] <klaxa> ah or no aac decoder
[01:30:24 CEST] <kuroro> is there way to check whether chrome has it?
[01:30:45 CEST] <klaxa> you just did it, it doesn't have it, try playing it with something else?
[01:31:38 CEST] <kuroro> so i merged the same 2 short clips using ffmpeg instead and it plays fine (http://pastebin.com/raw/MT5ihVcP)
[01:32:28 CEST] <kuroro> the only difference between the two it seems like is ffmpeg uses (SoundHandler) and ios uses (Core Media Audio)
[01:32:31 CEST] <kuroro> but both are aac
[01:32:46 CEST] <klaxa> huh, that's weird indeed
[01:36:33 CEST] <FishPencil> Does there exist any way to "deinterlace" a progressive video that should have been originally deinterlaced? I'd imagine it could just take x pixel high horizontal bars and combine them with the next frame
[01:49:14 CEST] <DHE> you can force filters. yadif defaults to deinterlace everything
[01:49:44 CEST] <DHE> give that a spin, see how it works out for you. tweak it if needed (since it can't autodetect some settings)
[01:55:57 CEST] <FishPencil> Wow that looks to be working
[01:56:29 CEST] <FishPencil> That's pretty impressive, because my source is progressive
[02:42:35 CEST] <SchrodingersScat> ffmpeg is basically magic
[02:43:57 CEST] <realies> what would be a modernly specd headless ffmpeg transcoding/encoding server?
[02:44:13 CEST] <realies> in means of do you need cpu or gpu acceleration or both
[12:08:31 CEST] <InTheWings> Any reason, after having removed the priv from AVFrame, for not having an AV_FRAME_DATA_PRIVATE auto mapped to an AV_FRAME_DATA_PRIVATE ?
[12:09:23 CEST] <InTheWings> err AV_PKT_DATA_PRIVATE mapped to an AV_FRAME_DATA_PRIVATE
[12:30:22 CEST] <n4zarh_> any1 can help me with avcodec_fill_audio_frame function? I asked for some help here lately, but I guess I still fail hard at this
[12:33:32 CEST] <n4zarh_> what I do is: I start encoder library with 640 bytes PMC data buffer, I create frame, I set its parameters (format, channel_layout according to codec context and nb_samples = 640)
[12:33:54 CEST] <n4zarh_> i call av_frame_get_buffer(frame, 0), no errors so far
[12:35:04 CEST] <n4zarh_> next I try to fill it: avcodec_fill_audio_frame(frame, channels, sample_fmt, (uint8_t*)buffer, 640 bufferLen, 0)
[12:35:58 CEST] <n4zarh_> and I always get -22 error at this point, which AFAIR should mean "wrong parameters"
[12:36:11 CEST] <nonex86> enable ffmpeg debug outpuy
[12:36:14 CEST] <nonex86> *output
[12:36:20 CEST] <nonex86> i see the detailed error
[12:37:16 CEST] <nonex86> av_log_set_level()
[12:38:23 CEST] <nonex86> with AV_LOG_VERBOSE
[12:38:36 CEST] <nonex86> this should help :)
[12:40:19 CEST] <n4zarh_> I guess it would if I knew how to read av_log (whole ffmpeg is built with android ndk)
[12:42:10 CEST] <nonex86> well
[12:42:17 CEST] <n4zarh_> is there any way to redirect messages from av_log to another function?
[12:42:18 CEST] <nonex86> you can redirect the output :)
[12:42:21 CEST] <nonex86> interesting? :D
[12:42:26 CEST] <nonex86> sure
[12:42:45 CEST] <nonex86> av_log_set_callback() :)
[12:43:12 CEST] <nonex86> *interested
[12:44:06 CEST] <nonex86> signature should be something like void __cdecl log_callback(void *, int, const char *, va_list)
[12:55:46 CEST] <n4zarh_> couldn't use __cdecl because my functions are in c++ and compiler screams about ignoring it and having to type specifier, so I create void log_callback(void* avcl, int level, const char* fmt, va_list vl) and set it like av_log_set_callback(log_callback()) only to get an error "no matching function to call to log_callback"
[12:58:13 CEST] <jkqxz> You're calling log_callback to make its return value the argument to the function.  Remove the "()".
[12:58:42 CEST] <nonex86> yeah, pass a pointer, not function call result
[12:59:32 CEST] <n4zarh_> oh god, you're right
[13:03:42 CEST] <n4zarh_> so, I have few troubling messages: "Bitrate 7013663263447580672 is extremely low, maybe you mean 7181192518920706592k" - I set 8 as codeccontext bitrate, I guess that might be wrong
[13:08:20 CEST] <n4zarh_> "nal_unit_type: 1657579756, nal_ref_idc: 44594" - high value of the latter always ends with "unknown SEI type 1657579748" in the next line
[13:08:29 CEST] <n4zarh_> and it happens once in a while
[13:08:41 CEST] <nonex86> last message related to h264 stream
[13:08:51 CEST] <nonex86> not the audio
[13:09:13 CEST] <nonex86> check last ffmpeg output messages
[13:09:22 CEST] <nonex86> at the point you get your error
[13:09:46 CEST] <n4zarh_> maybe that would be this:
[13:09:58 CEST] <nonex86> bitrate error looks strange for me
[13:10:35 CEST] <n4zarh_> non-existing PPS 1657579628 referenced [\n] decode_slice_header error [\n] no frame! [/n] nal_unit_type: 1657579756, nal_ref_idc: 4709
[13:10:57 CEST] <nonex86> pps error is also h264 related error
[13:11:07 CEST] <nonex86> not audio ones
[13:12:01 CEST] <n4zarh_> crap, discarding any h264 and audio decoder messages might be a bit hard, let me rewrite some stuff
[13:13:00 CEST] <nonex86> usually internal messages is very clear states the problem, at the moment when you call some ffmpeg function and failed
[13:13:28 CEST] <nonex86> possible you are missed something
[13:13:48 CEST] <nonex86> or, maybe supplied something very wrong to ffmpeg?
[13:13:57 CEST] <n4zarh_> i'm just running two decoders at the same time as I try to run encoder, temporarily disabled it, let me check
[13:14:15 CEST] <nonex86> sure
[13:14:42 CEST] <nonex86> check twice your audio encoder initialization code
[13:14:50 CEST] <nonex86> and buffer allocation stuff
[13:15:14 CEST] <nonex86> check parameters
[13:15:27 CEST] <n4zarh_> only thing I get is the bitrate stuff
[13:15:57 CEST] <nonex86> which audio encoder do you use?
[13:16:05 CEST] <n4zarh_> pcm_mulaw
[13:16:06 CEST] <nonex86> AV_CODEC_*?
[13:17:28 CEST] <nonex86> one more question
[13:17:34 CEST] <nonex86> why are using 8 as bitrate?
[13:18:49 CEST] <n4zarh_> i should use 8k samplerate, 8 bitrate and 1 channel from what I saw in decoder
[13:19:00 CEST] <n4zarh_> and decoder is working fine with the same params
[13:20:50 CEST] <TwinTailed> when using libav for encoding, the last audio frame is missing
[13:21:03 CEST] <TwinTailed> ffmpeg's libav, not the external libav library
[13:21:32 CEST] <TwinTailed> i am encoding h264
[13:22:15 CEST] <nonex86> TwinTailed: if you are have missing audio frame, why are you mention h264?
[13:22:41 CEST] <TwinTailed> nonex86: just for more info
[13:22:50 CEST] <n4zarh_> h264 is video codec
[13:23:11 CEST] <nonex86> TwinTailed: do you flashed your audio decoder at the end?
[13:23:13 CEST] <TwinTailed> avc/aac
[13:23:20 CEST] <TwinTailed> nonex86: http://pastebin.com/mAbRFvH7
[13:23:25 CEST] <nonex86> TwinTailed: ffmpeg have standard flush protocol
[13:23:31 CEST] <TwinTailed> line 255
[13:23:35 CEST] <nonex86> TwinTailed: for encoders/decoders
[13:23:59 CEST] <TwinTailed> nonex86: I did flush
[13:24:12 CEST] <n4zarh_> anyway, any idea why is ffmpeg showing these crazy numbers as wrong bitrate for me?
[13:25:06 CEST] <nonex86> TwinTailed: at the line you mention i see you just allocated packets, thats all :)
[13:25:23 CEST] <TwinTailed> nonex86: line 255 and below is the transcoding process :P
[13:25:28 CEST] <TwinTailed> the whole for loop
[13:25:59 CEST] <TwinTailed> until line 385
[13:26:01 CEST] <nonex86> TwinTailed: well, sorry, i dont have time to check whole source for you :P
[13:26:42 CEST] <n4zarh_> also, it seems like the latter number is changing every time I start my encoder
[13:26:46 CEST] <TwinTailed> nonex86: its just a small code, please, no one here helped me so far with this problem
[13:27:30 CEST] <TwinTailed> i asked many times
[13:27:55 CEST] <nonex86> the idea is simple
[13:28:01 CEST] <nonex86> and looks like you do something like this
[13:28:29 CEST] <nonex86> AVPacket packet; av_init_packet(&packet);
[13:28:40 CEST] <nonex86> packet.data = nullptr; packet.size = 0;
[13:28:47 CEST] <nonex86> feed this packet to encoder
[13:29:03 CEST] <nonex86> until it return something
[13:29:08 CEST] <nonex86> that should work
[13:29:28 CEST] <nonex86> so you get your buffered AVFrame's
[13:30:28 CEST] <TwinTailed> nonex86: I know, that's what i am doing
[13:30:38 CEST] <TwinTailed> nonex86: but there is EOF at the end
[13:31:01 CEST] <nonex86>  for(;; ++j)
[13:31:06 CEST] <nonex86> nice loop btw :D:D:D
[13:31:12 CEST] <TwinTailed> :D
[13:31:21 CEST] <n4zarh_> :D
[13:31:24 CEST] <nonex86> why are you care about eof?
[13:31:25 CEST] <nonex86> again
[13:31:25 CEST] <TwinTailed> it ends when EOF is returned
[13:31:29 CEST] <nonex86> you get your packet
[13:31:37 CEST] <TwinTailed> because without EOF I don't get all frames
[13:31:39 CEST] <nonex86> you put your packet to decoder
[13:31:41 CEST] <nonex86> decode it
[13:31:47 CEST] <nonex86> and then put frame to encoder
[13:31:50 CEST] <nonex86> fine
[13:31:54 CEST] <nonex86> you get eof
[13:32:00 CEST] <nonex86> now you need to flush your decoders
[13:32:04 CEST] <nonex86> and flush your encoders
[13:32:12 CEST] <nonex86> flush decoders if you need
[13:32:20 CEST] <nonex86> until you got all data from them
[13:32:29 CEST] <nonex86> then create this "nulldata" packet
[13:32:36 CEST] <nonex86> and feed this packet to encoder
[13:32:50 CEST] <nonex86> until you get packets from encoder :)
[13:32:52 CEST] <nonex86> thats all
[13:33:37 CEST] <nonex86> i am sorry
[13:33:45 CEST] <TwinTailed> nonex86: I already do that :(
[13:33:47 CEST] <nonex86> ah, never mind :)
[13:34:08 CEST] <TwinTailed> nonex86: when do i use avcodec_flush_buffers?
[13:34:21 CEST] <n4zarh_> hey, if I'm calling avcodec_open2(ctx, codec, NULL) and I get this error/warning/whatever at this given moment
[13:34:35 CEST] <n4zarh_> is it possible that context is not initialized properly?
[13:36:51 CEST] <nonex86> TwinTailed: documentation said:
[13:36:53 CEST] <nonex86> void avcodec_flush_buffers 	( 	AVCodecContext *  	avctx	) 	
[13:36:53 CEST] <nonex86> Reset the internal decoder state / flush internal buffers.
[13:36:53 CEST] <nonex86> Should be called e.g. when seeking or when switching to a different stream.
[13:37:08 CEST] <TwinTailed> nonex86: I mean in EOF
[13:37:29 CEST] <TwinTailed> nonex86: http://ffmpeg.org/doxygen/trunk/group__lavc__encdec.html
[13:37:39 CEST] <jkqxz> n4zarh_:  Did you allocate a clean AVCodecContext with avcodec_alloc_context3?  If you don't, all sorts of strange things will go wrong.
[13:37:55 CEST] <n4zarh_> I did
[13:38:27 CEST] <nonex86> TwinTailed: which version of ffmpeg do you use?
[13:38:33 CEST] <nonex86> TwinTailed: latest?
[13:39:06 CEST] <TwinTailed> nonex86: yes
[13:39:32 CEST] <nonex86> TwinTailed: i am still using decode_video* and all other in 3.0.0, and they are not marked as legacy :)
[13:39:40 CEST] <jkqxz> n4zarh_:  Are you sure that your headers actually match the library version you are using?  (You bitrate value above was suspiciously zero in the low half.)
[13:40:29 CEST] <n4zarh_> well, I am working on a single version from the start
[13:41:07 CEST] <n4zarh_> sometimes changing ./configure script and updating files, but never changing version
[13:42:22 CEST] <nonex86> you said its an android application?
[13:42:34 CEST] <nonex86> with some interop with ndk build ffmpeg?
[13:42:44 CEST] <jkqxz> And you don't happen to have a different version installed somewhere else on the same machine, and you are happening to use the headers from it by accident?
[13:47:44 CEST] <n4zarh_> hmm, I guess not, gonna check it out though
[13:48:15 CEST] <n4zarh_> everything my NDK uses to build app with is in jni/ffmpeg directory
[13:49:48 CEST] <n4zarh_> okay, I'm running short on time (i need to get on the bus in few minutes and I'll be back in ~1-2h
[13:50:01 CEST] <n4zarh_> so, I stopped setting bitrate, error stopped showing
[13:50:52 CEST] <n4zarh_> but I still get -22 error on filling audio frame
[13:51:00 CEST] <n4zarh_> without any information from avlog
[13:51:10 CEST] <n4zarh_> i'll be back in about an hour
[13:51:34 CEST] <nonex86> well
[13:51:41 CEST] <nonex86> from the pcm.c
[13:51:57 CEST] <nonex86> bit_rate field in context just overwrited internally
[13:52:25 CEST] <nonex86> not sure, did i pick correct file for AV_CODEC_ID_PCM_MULAW
[13:53:36 CEST] <nonex86> guess n4zarch_ just forgot to supply block_align
[13:54:10 CEST] <nonex86> that why he got random values in the debug output
[13:54:23 CEST] <nonex86> *thats
[16:54:05 CEST] <pgorley> is videotoolbox for mpeg4 broken?
[16:54:52 CEST] <pgorley> i keep getting operation not permitted
[19:17:54 CEST] <poopiefly> need to archive some video. Was asked to use HuffYUV. Should I be using ffvhuff ?
[19:19:02 CEST] <poopiefly> how compatible is ffvhuff with commercial editors? Would ffv1 be a better choice?
[19:24:16 CEST] <dgsjm> When outputting an image sequence from a video, the first three frames are always identical. What am I doing wrong? command:  ffmpeg -i input.mp4 -r 0.5 output/%03d.png
[19:29:44 CEST] <klaxa> why -r 0.5?
[19:31:36 CEST] <dgsjm> i'm trying to limit the frame rate
[19:31:37 CEST] <bornpilot> from what I understand if I downsample 4k 4:2:0 to 1080 HD I can get 4:2:2, if that is true and I have a 2k footage and I downsample the 2k 4:2:0 will it downsample as 4:2:1 ?
[19:32:15 CEST] <c_14> poopiefly: if you're using commercial editors you probably want utvideo as that's supported by most of them, otherwise I'm not sure about the support. If they use libav* for decoding it'll support them all and you can use whatever you want.
[19:32:49 CEST] <dgsjm> What I want to do is export a fixed number of frames from a video of any length, and set the file size of each exported JPG. So, regardless of the length of the video, I want 100 frames at 15k each. Having trouble finding that in the documentation.
[19:33:51 CEST] <c_14> Do the frames need to be spread out over the video?
[19:34:05 CEST] <c_14> Do you want to support videos with less than 100 frames?
[19:34:22 CEST] <dgsjm> Yes, I'd like them to be evenly spaced. No, I don't need to support videos with less than 100 frames.
[19:34:59 CEST] <iive> bornpilot: yuv420 have quarter resolutons (1/2 * 1/2) for the chroma channel. so if you downscale only the luma channel, you can get yuv444 from it.
[19:35:42 CEST] <bornpilot> live that would be from the 4k correct and then from 2k I could get 4:2:2?
[19:35:44 CEST] <iive> luma channel is gray levels, chroma is color differential.
[19:36:22 CEST] <bornpilot> I have never worked with downscaling just one chanel, interesting.
[19:36:27 CEST] <iive> you can always turn 420 into 444, it just won't have the detail. aka you won't restore lost data.
[19:36:54 CEST] <bornpilot> right I want to avoid faking 444
[19:37:37 CEST] <poopiefly> c_14 was afk. utvideo is supported by commercial. Just going to archive 420p8 colorspace video. I use to use utvideo but it tended to crash a lot. Maybe I should give it a new try?
[19:37:58 CEST] <c_14> poopiefly: libutvideo or the internal one?
[19:38:15 CEST] <poopiefly> the one that is part of ffmpeg
[19:38:55 CEST] <poopiefly> guess that is libutvideo. Not sure what I compiled in
[19:39:09 CEST] <iive> bornpilot: i'm not familiar with swscale, so I can't say if it would keep the chroma channels automatically
[19:39:28 CEST] <iive> but i't might.
[19:39:41 CEST] <bornpilot> iive so if I downsclaed just the luma channel on a 2k video could I get 442?
[19:40:02 CEST] <bornpilot> I am working with an iphone
[19:40:33 CEST] <root> please help: http://ffmpeg.gusari.org/viewtopic.php?f=11&t=3172&p=9458#p9458
[19:40:41 CEST] <iive> the standar is 444 422 420 , there are no 442 and 421 or rather... not widely used standards
[19:41:57 CEST] <bornpilot> iive okay. I am guessing then I would have to declare the pix_fmt if I am using a prores code profile 4
[19:42:06 CEST] <bornpilot> ^codec
[19:42:08 CEST] <iive> 444 have YUV at same resolution. 422 have chroma channels at half (horizontally) resolution,
[19:42:27 CEST] <iive> 420 is chroma at quarter (1/2*1/2)
[19:44:59 CEST] <poopiefly> what is the difference between 420 planer and just 420?
[19:45:48 CEST] <iive> 420 is usually planar, meaning that each channel is in its own array (2D)
[19:46:20 CEST] <TwinTailed> when using libav for transcoding, the last audio frame is missing (Codecs AVC/AAC), only the last audio frame
[19:47:43 CEST] <iive> yuyv uyuv are packed formats. they are 422 equivalent. e.g. it's just that all samples for a pixel are packed together. you have (y1,u,v) and (y2,u,v) where u and v are same for both luma (y) samples.
[19:50:25 CEST] <poopiefly> I really want to keep my video in 420p but need to use VirtualDubs deshaker. Think that only works with RGB24,32 ? How bad are the rounding errors etc? Looks like I got banding on my blacks now =/
[19:56:25 CEST] <furq> poopiefly: avisynth/vapoursynth have some deshake filters that work with yuv
[19:56:58 CEST] <furq> http://avisynth.nl/index.php/External_filters#Stabilization
[19:57:01 CEST] <poopiefly> I know but depan is not really all that good
[19:58:08 CEST] <c_14> dgsjm: https://pb.c-14.de/t/kng.wCJ2yr something like that should work (don't ask me why you need the decimate, I'm not quite sure myself)
[19:59:14 CEST] <c_14> furq: I based that^ on something you pasted a while back, but you didn't have the decimate. Is it supposed to work without (as in not spam the same frame about a thousand times before outputting the next)?
[19:59:44 CEST] <furq> the script i based that on has select before thumbnail
[19:59:53 CEST] <poopiefly> furq The problem with depan from mtools2 is that you can only pick between mirror, zoom, or blur for the edges. Deshaker can use temporal images and rebuild the missing border info. I use it with a slight adaptive zoom. The result is crazy nice.
[20:00:00 CEST] <furq> iirc it worked without select but i'm not totally sure
[20:00:10 CEST] <furq> i've no idea why you'd need decimate though
[20:00:27 CEST] <dgsjm> c_14: Thank you! I'll give it a try right now.
[20:00:59 CEST] <c_14> dgsjm: of course in your case you'll want jpg and the options according to that, that's just the general version
[20:05:30 CEST] <iive> poopiefly: ffmpeg supports stabilization filter through external library
[20:06:22 CEST] <furq> oh weird
[20:06:33 CEST] <furq> i was feeding thumbnail to tile, it works fine if you do that
[20:06:50 CEST] <furq> but it doesn't seem to work right with frames:v
[20:08:20 CEST] <poopiefly> iive I have tried it, its ok. What I really am looking for is that the black borders can be rebuilt with temporal frames. Not just blurry.
[20:08:24 CEST] <dgsjm> c_14: ffprobe says "Unrecognized option 'select_streams'". Maybe v 0.11.1 is too old?
[20:08:37 CEST] <furq> probably
[20:08:41 CEST] <iive> poopiefly: i see...
[20:09:22 CEST] <poopiefly> deshaker does this well. I use 150 plus and 150 minus frames.
[20:09:23 CEST] <furq> poopiefly: http://johnvansickle.com/ffmpeg/
[20:09:28 CEST] <c_14> furq: If I get rid of both the decimate and the frames:v ffmpeg outputs about a thousand copies of each frame the thumbnail filter selects
[20:09:34 CEST] <furq> yeah i just tried it
[20:09:56 CEST] <furq> it doesn't work even if frames:v is higher than the thumbnail range
[20:10:49 CEST] <furq> i don't know how that even works with tile then
[20:12:21 CEST] <c_14> furq: I found it
[20:12:28 CEST] <c_14> Adding -vsync 0 fixes it (as an output option)
[20:12:37 CEST] <c_14> It's listed under the examples for the tile filter
[20:12:59 CEST] <furq> oh neat
[20:13:45 CEST] <poopiefly> furq thx, but what am I missing that these builds might have ?
[20:14:16 CEST] <furq> well those builds aren't several years old for starters
[20:14:44 CEST] <furq> 0.11.1 is four and a half years old
[20:15:39 CEST] <c_14> dgsjm: https://pb.c-14.de/t/kng.Y42xnm updated version and you'll want something much newer (the thumbnail filter is at least 3.0.0 iirc)
[20:16:11 CEST] <furq> thumbnail is present in 2.8
[20:16:19 CEST] <furq> it is pretty recent though
[20:16:26 CEST] <dgsjm> c_14, furq: ok ill try to update. thanks.
[20:16:28 CEST] <c_14> Oh, must have gotten my timescale off by a bit then
[20:17:00 CEST] <furq> freebsd still haven't sorted out the 3.0 api changes so my nas is stuck on 2.8
[20:34:27 CEST] <teratorn> so I'm really confused why this ffmpeg command places two video output streams in to out.mp4: ffmpeg -y -i in.mp4 -filter_complex "[0:v]split=2[v1][v2];[v1]trim=start=0.5,trim=start_frame=0:end_frame=1,loop=30:size=1:start=0[out];[v2]null" -map "[out]" out.mp4
[20:34:34 CEST] <teratorn> in.mp4 just has one video stream
[20:35:12 CEST] <teratorn> I *thought* that when you used a -map that disabled the default mapping of one video/audio/subtitle (the "best" one) per output...
[20:35:18 CEST] <TwinTailed> when using libav for transcoding, the last audio frame is missing (Codecs AVC/AAC), only the last audio frame
[20:38:25 CEST] <teratorn> could anyone tell me why adding a split filter is having this effect? (two video streams going in to out.mp4, when I only expect one, since I have only one -map option)
[20:53:00 CEST] <c_14> teratorn: I think ffmpeg automatically maps unnamed outputs, in your case the output of the null filter
[20:53:03 CEST] <kepstin> teratorn: hmm, I would have expected that to give an error, unconnected pad on the filterchain.
[20:53:12 CEST] <kepstin> weird that it's putting it into the file
[20:53:35 CEST] <kepstin> i guess you only get the unconnected pad error on named pads
[20:59:15 CEST] <tua> !list
[22:11:47 CEST] <Dresk|Dev> So help me out here, we realize getting hardware decoding of H.264 is a difficult thing in ffmpeg due to all the platform differences, but AVHWAccel, what does that actually do for ffmpeg?
[23:15:15 CEST] <TwinTailed>  when using libav for transcoding, the last audio frame is missing (Codecs AVC/AAC)
[23:23:28 CEST] <Technologov> hi all !
[23:24:34 CEST] <Technologov> I have a VERY strange problem. I encoded a two videos (vcodec=libx264) that end up being playable in VLC and Windows Media Player, but not Apple iTunes/iPad.
[23:31:03 CEST] <BtbN> Embedded Devices often don't support all codec features.
[23:31:16 CEST] <furq> Technologov: pastebin the ffprobe output
[23:31:20 CEST] <BtbN> You'll have to look up what your target device supports and limit the features accordingly.
[23:32:55 CEST] <Technologov> Two of my videos have very very strange behavior:
[23:32:55 CEST] <Technologov> They can forward+rewind first few minutes of the video, and after that I can't forward/rewind/seek video (but seeking audio still works).
[23:33:06 CEST] <Technologov> iPad Pro
[23:34:59 CEST] <TwinTailed> BtbN: can you please help me with my problem :S when transcoding using LibAV to AVC/AAC the last audio frame is missing
[23:35:02 CEST] <Technologov> http://pastebin.com/Wh94xZR0
[23:35:08 CEST] <TwinTailed> I have done the EOF as u told me
[23:35:27 CEST] <BtbN> TwinTailed, does it also happen when using the ffmpeg cli tool?
[23:35:27 CEST] <Technologov> OMG : ffprobe doesn't support redirect to STDOUT. Bug ?
[23:35:38 CEST] <BtbN> Technologov, it outputs on stderr.
[23:35:56 CEST] <TwinTailed> BtbN: No it doesnt
[23:36:43 CEST] <BtbN> It's a problem in your code then.
[23:37:07 CEST] <Technologov> okay, I posted ffprobe output
[23:38:14 CEST] <Technologov> this file and another one share similar behavior: cannot rewind/seek video on Apple, beyond 1st several minutes. And no amount of encoding helps.
[23:38:31 CEST] <Technologov> (I tried different bitrates, different resolutions, ... nada)
[23:39:28 CEST] <TwinTailed> BtbN: I know it is :/ But I can't solve it, I've tried everything (Almost)
[23:40:32 CEST] <Technologov> furq, http://pastebin.com/Wh94xZR0
[00:00:00 CEST] --- Sat Oct  8 2016