Re: [FFmpeg-user] cannot build shared libraries on Solaris
On 12/27/2018 1:00 PM, Carl Eugen Hoyos wrote: 2018-12-24 14:20 GMT+01:00, Eric Thomas: On my Solaris machine, the library filenames do not have the version suffix. For example, "*libavcodec.so.*" I get version suffixes here: $ ls -l libavcodec/libavcodec.* lrwxrwxrwx 1 cehoyos staff 16 Dec 27 21:58 libavcodec/libavcodec.so -> libavcodec.so.58 -rwxr-xr-x 1 cehoyos staff66379668 Dec 27 21:58 libavcodec/libavcodec.so.58 I seem to recall that solaris doesn't usually set up the "symlink to the numbered lib" technique that BSD and linux use but I don't have a system to check that. Does solaris/illumos/etc use that technique? z! ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] Question about handling / unwanted changes of SAR in ffmpeg
Uwe Freese (2018-12-27): > Good. But then I totally don't understand why that 64:45 is not used and > stored at mkv container level as well. Since you have not provided the full information necessary to explain, nobody will be able to. Regards, -- Nicolas George signature.asc Description: PGP signature ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] Conversion error with aac files
2018-12-23 14:12 GMT+01:00, Mark Van Peteghem : > On 23/12/2018 12:56, Carl Eugen Hoyos wrote: >> Am 23.12.2018 um 11:10 schrieb Mark Van Peteghem >> : >>> Should I deliver some of these files as an attachment or in another way? >> Please provide one or two non-silent sample files, use a file hoster of >> your choice that does not need a login or attach them if small enough. > > I've uploaded three files. They are another example for ticket #6634. I unfortunately couldn't understand what the offending commit tried to fix, so I won't be able to help, sorry! https://trac.ffmpeg.org/ticket/6634 Carl Eugen ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] aac: Prediction is not allowed in AAC-LC.
Dear all, It seems I gave some wrong impressions. First of all thank you Carl and Mortiz for responding and sharing your inputs. I was able to see the replies on ffmpeg-user archives. As correctly pointed out by both, it doesn't need or use any remuxing as the file downloads by itself. The issue comes around half-way using mpv to view the video file. There it starts to run away and give the error. To answer the question that Moritz asked, no I do not have any specific or special instructions under .config/youtube-dl/ . In fact inside .config/ there is no youtube-dl at all. There are only couple of special instructions under .config/mpv/ but even that should have no bearing on the error I shared before. ~/.config/mpv$ cat mpv.conf alang=eng,en,english slang=en,eng, english sub-scale=1.50 save-position-on-quit -- Regards, Shirish Agarwal शिरीष अग्रवाल My quotes in this email licensed under CC 3.0 http://creativecommons.org/licenses/by-nc/3.0/ http://flossexperiences.wordpress.com EB80 462B 08E1 A0DE A73A 2C2F 9F3D C7A4 E1C4 D2D8 ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] aac: Prediction is not allowed in AAC-LC.
On Fri, Dec 28, 2018 at 01:05:15 +0530, shirish शिरीष wrote: > [ffmpeg/audio] aac: Inconsistent channel configuration. > [ffmpeg/audio] aac: get_buffer() failed > Error decoding audio. > [ffmpeg/audio] aac: Prediction is not allowed in AAC-LC. > Error decoding audio. > [ffmpeg/audio] aac: Sample rate index in program config element does > not match the sample rate index configured by the container. > [ffmpeg/audio] aac: Too large remapped id is not implemented. Update > your FFmpeg version to the newest one from Git. If the problem still > occurs, it means that your file has a feature which has not been > implemented. > [ffmpeg/audio] aac: If you want to help, upload a sample of this file > to ftp://upload.ffmpeg.org/incoming/ and contact the ffmpeg-devel > mailing list. (ffmpeg-de...@ffmpeg.org) This is youtube-dl's output. Please run youtube-dl --verbose, so that we can see the ffmpeg command line it uses. > $ youtube-dl -c -f 22 9Jomgc8Ui0E This command line doesn't even trigger use of ffmpeg here for me, because format 22 can just be downloaded (and doesn't need to go through ffmpeg). Do you have any particular youtube-dl configuration (~/.config/youtube-dl/config) which enforces reencoding or remuxing with ffmpeg? If so, why? Like Carl Eugen, I also couldn't reproduce the issue, even if manually trying to remux the downloaded file with ffmpeg. Is your youtube-dl perhaps using a different (older) ffmpeg binary? Moritz ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] down sampling
On Thu, Dec 27, 2018 at 20:29:34 +0100, Michael Koch wrote: > I think the bigger problem is "outputting through the computers > speakers". As far as I know it depends on the operating system, and > under Windows it's impossible. You can always pipe to ffplay, which plays audio also under Windows (using SDL audio). Indeed, probably a worthwhile task adding an ffmpeg "sdl audio" device. Moritz ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] Robustness on processing transport streams (TS)?
On Thu, Dec 27, 2018 at 22:21:26 +0100, Uwe Freese wrote: > -async A... simplified 1 parameter audio > timestamp matching, 0(disabled), 1(filling and trimming), >1(maximum > stretch/squeeze in samples per second) (from INT_MIN to INT_MAX) (default 0) That's the swresample option, not the ffmpeg option. "swr simple 1 parameter async, similar to ffmpegs -async" > Why does it say "float" and at the end "from INT_MIN to INT_MAX"? It's > also defined like that in the code in libswresample/options.c, also for > other parameters. *shrug* ffmpeg's "-async" option is defined differently. Moritz ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] Robustness on processing transport streams (TS)?
2018-12-27 22:21 GMT+01:00, Uwe Freese :
>> A long time back, there was ProjectX to fix transport streams, I
>> believe there is another program now (suggested by the HandBrake
>> user support) but I forgot its name.
>
> ProjectX doesn't work with HD.
HD should have no issues, h264 support is very limited iirc.
> Under Win, I used "TSDoctor", but I'm now working on
> Linux only and didn't find anything appropriate.
That's probably what I read about.
> But it seems ffmpeg is then enough for this purpose (I'm not expecting
> "very damaged" recordings).
>
>> To "insert silence", FFmpeg needs the "-async 1" option or the
>> corresponding filter chain.
>
> Very good hint, thanks!
>
> Just a small point I'm curious about. Help says:
>
> -async A... simplified 1 parameter audio
> timestamp matching, 0(disabled), 1(filling and trimming), >1(maximum
> stretch/squeeze in samples per second) (from INT_MIN to INT_MAX) (default 0)
>
> Why does it say "float" and at the end "from INT_MIN to INT_MAX"? It's
> also defined like that in the code in libswresample/options.c, also for
> other parameters.
>
> {"async", "simplified 1 parameter audio timestamp
> matching, 0(disabled), 1(filling and trimming), >1(maximum
> stretch/squeeze in samples per second)"
> ,
> OFFSET(async) , AV_OPT_TYPE_FLOAT ,{.dbl=0
> }, INT_MIN, INT_MAX , PARAM },
>
> A bug, or how does it make sense to use INT border values with a float?
I don't know but I believe this values are already too large for real
use cases.
Carl Eugen
___
ffmpeg-user mailing list
ffmpeg-user@ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-user
To unsubscribe, visit link above, or email
ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] Robustness on processing transport streams (TS)?
Hello,
If you have an input file that crashes FFmpeg, please share it!
It was only crashing Avidemux when seeking to the end. - "Sorry". ;-)
A long time back, there was ProjectX to fix transport streams, I
believe there is another program now (suggested by the HandBrake
user support) but I forgot its name.
ProjectX doesn't work with HD. Under Win, I used "TSDoctor", but I'm
now working on Linux only and didn't find anything appropriate.
But it seems ffmpeg is then enough for this purpose (I'm not expecting
"very damaged" recordings).
To "insert silence", FFmpeg needs the "-async 1" option or the
corresponding filter chain.
Very good hint, thanks!
Just a small point I'm curious about. Help says:
-async A... simplified 1 parameter audio
timestamp matching, 0(disabled), 1(filling and trimming), >1(maximum
stretch/squeeze in samples per second) (from INT_MIN to INT_MAX) (default 0)
Why does it say "float" and at the end "from INT_MIN to INT_MAX"? It's
also defined like that in the code in libswresample/options.c, also for
other parameters.
{"async" , "simplified 1 parameter audio timestamp
matching, 0(disabled), 1(filling and trimming), >1(maximum
stretch/squeeze in samples per second)"
,
OFFSET(async) , AV_OPT_TYPE_FLOAT ,{.dbl=0
}, INT_MIN, INT_MAX , PARAM },
A bug, or how does it make sense to use INT border values with a float?
Regards,
Uwe
___
ffmpeg-user mailing list
ffmpeg-user@ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-user
To unsubscribe, visit link above, or email
ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] Question about handling / unwanted changes of SAR in ffmpeg
Hello, Instead, the better solution would then be to give the SAR to whatever calculates the SAR to store at container level not as a double or int, but as a "AVRational", which contains the "num" and "den" values (in this case 64 and 45)? This is exactly what FFmpeg does (since forever). Good. But then I totally don't understand why that 64:45 is not used and stored at mkv container level as well. Regards, Uwe ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] cannot build shared libraries on Solaris
2018-12-24 14:20 GMT+01:00, Eric Thomas : > Removing the stdatomic.o file, and building with 'gmake' did > the trick for me. This may be a bug (did "make distclean" not remove the file?) but since I cannot reproduce, I'll not try to fix it. Carl Eugen ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] cannot build shared libraries on Solaris
2018-12-24 14:20 GMT+01:00, Eric Thomas : > Next question: On my Centos VM, my shared library files have the library > version as the filename suffix. > For example, "*libavcodec.so.58*". > On my Solaris machine, the library filenames do not have the version > suffix. For example, "*libavcodec.so.*" I get version suffixes here: $ ls -l libavcodec/libavcodec.* lrwxrwxrwx 1 cehoyos staff 16 Dec 27 21:58 libavcodec/libavcodec.so -> libavcodec.so.58 -rwxr-xr-x 1 cehoyos staff66379668 Dec 27 21:58 libavcodec/libavcodec.so.58 -rw-r--r-- 1 cehoyos staff 25622 Dec 27 21:58 libavcodec/libavcodec.ver -rw-r--r-- 1 cehoyos staff 85 Dec 27 21:48 libavcodec/libavcodec.version Carl Eugen ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] aac: Prediction is not allowed in AAC-LC.
2018-12-27 20:35 GMT+01:00, shirish शिरीष : > I downloaded a media file and got the following error - > > [ffmpeg/audio] aac: Inconsistent channel configuration. Command line and complete, uncut console output missing, I cannot reproduce. Carl Eugen ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] error with hero 7 white gopro audio stream
It seems that even though there is an error message, the audio stream is copied to the output file with no perceptible loss. I thought I had a different outcome before I posted. As far as hardware acceleration (hwaccel) is concerned, I found a couple of useful links and have successfully used my macbook air hardware with the following command. $ ffmpeg -y -threads 2 -i GH010087.MP4 -vf "lenscorrection=cx=0.5:cy=0.5:k1=-0.211:k2=0.0" -vcodec h264_videotoolbox -b:v 1K out.mp4 Adjust the bitrate (-b:v) and threads as appropriate for your desired results. Elsewhere ( https://stackoverflow.com/questions/30832248/is-there-a-way-to-remove-gopro-fisheye-using-ffmpeg) I found recommended values for GoPro correction to be k1=-0.227 and k2=-0.022. After running a test grid image through ffmpeg at different settings I settled on the settings in the example command. I found these two links helpful for figuring out a working use of hardware acceleration. https://forum.videohelp.com/threads/386398-ffmpeg-videotoolbox-options https://www.reddit.com/r/ffmpeg/comments/59qglv/your_experiences_with_h264_videotoolbox/ I need to do some more research to see if opencl can be used. When I tried, it errored out but so did my attempts with videotoolbox until I found the above links for guidance. On Wed, Dec 26, 2018 at 7:58 PM Harry Levinson wrote: > I am trying to use ffmpeg to defish my gopro videos on a mac. The lens > correction is working reasonably well though eventually I'd like to have > hardware acceleration working, but first I am having a more basic problem. > When ffmpeg gets to the audio stream it complains about the audio stream. > > Error while decoding stream #0:1: Invalid argument:00:47.31 > > I tried a few stream mapping options but no success. Below is the most > basic command and resulting output. I tried another MP4 file and it worked > fine. > > Thanks, > Harry > > > $ ffmpeg -i GH010087.MP4out.MP4 > ffmpeg version 4.1 Copyright (c) 2000-2018 the FFmpeg developers > built with Apple LLVM version 10.0.0 (clang-1000.11.45.5) > configuration: --prefix=/usr/local/Cellar/ffmpeg/4.1_1 --enable-shared > --enable-pthreads --enable-version3 --enable-hardcoded-tables > --enable-avresample --cc=clang --host-cflags= --host-ldflags= > --enable-ffplay --enable-gpl --enable-libmp3lame --enable-libopus > --enable-libsnappy --enable-libtheora --enable-libvorbis --enable-libvpx > --enable-libx264 --enable-libx265 --enable-libxvid --enable-lzma > --enable-opencl --enable-videotoolbox > libavutil 56. 22.100 / 56. 22.100 > libavcodec 58. 35.100 / 58. 35.100 > libavformat58. 20.100 / 58. 20.100 > libavdevice58. 5.100 / 58. 5.100 > libavfilter 7. 40.101 / 7. 40.101 > libavresample 4. 0. 0 / 4. 0. 0 > libswscale 5. 3.100 / 5. 3.100 > libswresample 3. 3.100 / 3. 3.100 > libpostproc55. 3.100 / 55. 3.100 > Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'GH010087.MP4': > Metadata: > major_brand : mp41 > minor_version : 538120216 > compatible_brands: mp41 > creation_time : 2018-02-08T17:33:57.00Z > firmware: H18.02.01.21.00 > Duration: 00:00:47.49, start: 0.00, bitrate: 30131 kb/s > Stream #0:0(eng): Video: h264 (High) (avc1 / 0x31637661), yuvj420p(pc, > smpte170m), 1920x1440, 30025 kb/s, 30 fps, 30 tbr, 90k tbn, 60 tbc (default) > Metadata: > creation_time : 2018-02-08T17:33:57.00Z > handler_name: GoPro AVC > encoder : GoPro AVC encoder > timecode: 00:23:00:11 > Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, > stereo, fltp, 127 kb/s (default) > Metadata: > creation_time : 2018-02-08T17:33:57.00Z > handler_name: GoPro AAC > timecode: 00:23:00:11 > Stream #0:2(eng): Data: none (tmcd / 0x64636D74) (default) > Metadata: > creation_time : 2018-02-08T17:33:57.00Z > handler_name: GoPro TCD > timecode: 00:23:00:11 > Stream #0:3(eng): Data: bin_data (gpmd / 0x646D7067), 30 kb/s (default) > Metadata: > creation_time : 2018-02-08T17:33:57.00Z > handler_name: GoPro MET > Stream #0:4(eng): Data: none (fdsc / 0x63736466), 10 kb/s (default) > Metadata: > creation_time : 2018-02-08T17:33:57.00Z > handler_name: GoPro SOS > File 'out.MP4' already exists. Overwrite ? [y/N] Stream mapping: > Stream #0:0 -> #0:0 (h264 (native) -> h264 (libx264)) > Stream #0:1 -> #0:1 (aac (native) -> aac (native)) > Press [q] to stop, [?] for help > [libx264 @ 0x7fa020818200] using cpu capabilities: MMX2 SSE2Fast SSSE3 > SSE4.2 AVX FMA3 BMI2 AVX2 > [libx264 @ 0x7fa020818200] profile High, level 5.0 > [libx264 @ 0x7fa020818200] 264 - core 155 r2917 0a84d98 - H.264/MPEG-4 AVC > codec - Copyleft 2003-2018 - http://www.videolan.org/x264.html - options: > cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7
Re: [FFmpeg-user] Fwd: error with hero 7 white gopro audio stream
2018-12-27 1:58 GMT+01:00, Harry Levinson : > I am trying to use ffmpeg to defish my gopro videos on a mac. The lens > correction is working reasonably well though eventually I'd like to have > hardware acceleration working, but first I am having a more basic problem. > When ffmpeg gets to the audio stream it complains about the audio stream. > > Error while decoding stream #0:1: Invalid argument:00:47.31 > > I tried a few stream mapping options but no success. Below is the most > basic command and resulting output. I tried another MP4 file and it worked > fine. > > Thanks, > Harry > > > $ ffmpeg -i GH010087.MP4out.MP4 > ffmpeg version 4.1 Copyright (c) 2000-2018 the FFmpeg developers > built with Apple LLVM version 10.0.0 (clang-1000.11.45.5) > configuration: --prefix=/usr/local/Cellar/ffmpeg/4.1_1 --enable-shared > --enable-pthreads --enable-version3 --enable-hardcoded-tables > --enable-avresample --cc=clang --host-cflags= --host-ldflags= > --enable-ffplay --enable-gpl --enable-libmp3lame --enable-libopus > --enable-libsnappy --enable-libtheora --enable-libvorbis --enable-libvpx > --enable-libx264 --enable-libx265 --enable-libxvid --enable-lzma > --enable-opencl --enable-videotoolbox > libavutil 56. 22.100 / 56. 22.100 > libavcodec 58. 35.100 / 58. 35.100 > libavformat58. 20.100 / 58. 20.100 > libavdevice58. 5.100 / 58. 5.100 > libavfilter 7. 40.101 / 7. 40.101 > libavresample 4. 0. 0 / 4. 0. 0 > libswscale 5. 3.100 / 5. 3.100 > libswresample 3. 3.100 / 3. 3.100 > libpostproc55. 3.100 / 55. 3.100 > Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'GH010087.MP4': > Metadata: > major_brand : mp41 > minor_version : 538120216 > compatible_brands: mp41 > creation_time : 2018-02-08T17:33:57.00Z > firmware: H18.02.01.21.00 > Duration: 00:00:47.49, start: 0.00, bitrate: 30131 kb/s > Stream #0:0(eng): Video: h264 (High) (avc1 / 0x31637661), yuvj420p(pc, > smpte170m), 1920x1440, 30025 kb/s, 30 fps, 30 tbr, 90k tbn, 60 tbc (default) > Metadata: > creation_time : 2018-02-08T17:33:57.00Z > handler_name: GoPro AVC > encoder : GoPro AVC encoder > timecode: 00:23:00:11 > Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, > stereo, fltp, 127 kb/s (default) > Metadata: > creation_time : 2018-02-08T17:33:57.00Z > handler_name: GoPro AAC > timecode: 00:23:00:11 > Stream #0:2(eng): Data: none (tmcd / 0x64636D74) (default) > Metadata: > creation_time : 2018-02-08T17:33:57.00Z > handler_name: GoPro TCD > timecode: 00:23:00:11 > Stream #0:3(eng): Data: bin_data (gpmd / 0x646D7067), 30 kb/s (default) > Metadata: > creation_time : 2018-02-08T17:33:57.00Z > handler_name: GoPro MET > Stream #0:4(eng): Data: none (fdsc / 0x63736466), 10 kb/s (default) > Metadata: > creation_time : 2018-02-08T17:33:57.00Z > handler_name: GoPro SOS > File 'out.MP4' already exists. Overwrite ? [y/N] Stream mapping: > Stream #0:0 -> #0:0 (h264 (native) -> h264 (libx264)) > Stream #0:1 -> #0:1 (aac (native) -> aac (native)) > Press [q] to stop, [?] for help > [libx264 @ 0x7fa020818200] using cpu capabilities: MMX2 SSE2Fast SSSE3 > SSE4.2 AVX FMA3 BMI2 AVX2 > [libx264 @ 0x7fa020818200] profile High, level 5.0 > [libx264 @ 0x7fa020818200] 264 - core 155 r2917 0a84d98 - H.264/MPEG-4 AVC > codec - Copyleft 2003-2018 - http://www.videolan.org/x264.html - options: > cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 > psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 > cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=6 > lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 > bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 > b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 > scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 > qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00 > Output #0, mp4, to 'out.MP4': > Metadata: > major_brand : mp41 > minor_version : 538120216 > compatible_brands: mp41 > firmware: H18.02.01.21.00 > encoder : Lavf58.20.100 > Stream #0:0(eng): Video: h264 (libx264) (avc1 / 0x31637661), > yuvj420p(pc), 1920x1440, q=-1--1, 0.03 fps, 15360 tbn, 30 tbc (default) > Metadata: > creation_time : 2018-02-08T17:33:57.00Z > handler_name: GoPro AVC > timecode: 00:23:00:11 > encoder : Lavc58.35.100 libx264 > Side data: > cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: -1 > Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, > stereo, fltp, 128 kb/s (default) > Metadata: > creation_time : 2018-02-08T17:33:57.00Z >
[FFmpeg-user] aac: Prediction is not allowed in AAC-LC.
Dear Friends,
Thank you for continuing to maintain this awesome free software
multimedia stack. Please CC me in case somebody responds as I am
unable to process mailing lists, just too much spam :(
I am running
$ ffmpeg
ffmpeg version 4.1-1 Copyright (c) 2000-2018 the FFmpeg developers
built with gcc 8 (Debian 8.2.0-12)
configuration: --prefix=/usr --extra-version=1 --toolchain=hardened
--libdir=/usr/lib/x86_64-linux-gnu
--incdir=/usr/include/x86_64-linux-gnu --arch=amd64 --enable-gpl
--disable-stripping --enable-avresample --disable-filter=resample
--enable-avisynth --enable-gnutls --enable-ladspa --enable-libaom
--enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca
--enable-libcdio --enable-libcodec2 --enable-libflite
--enable-libfontconfig --enable-libfreetype --enable-libfribidi
--enable-libgme --enable-libgsm --enable-libjack --enable-libmp3lame
--enable-libmysofa --enable-libopenjpeg --enable-libopenmpt
--enable-libopus --enable-libpulse --enable-librsvg
--enable-librubberband --enable-libshine --enable-libsnappy
--enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora
--enable-libtwolame --enable-libvidstab --enable-libvorbis
--enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265
--enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi
--enable-lv2 --enable-omx --enable-openal --enable-opengl
--enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883
--enable-chromaprint --enable-frei0r --enable-libx264 --enable-shared
libavutil 56. 22.100 / 56. 22.100
libavcodec 58. 35.100 / 58. 35.100
libavformat58. 20.100 / 58. 20.100
libavdevice58. 5.100 / 58. 5.100
libavfilter 7. 40.101 / 7. 40.101
libavresample 4. 0. 0 / 4. 0. 0
libswscale 5. 3.100 / 5. 3.100
libswresample 3. 3.100 / 3. 3.100
libpostproc55. 3.100 / 55. 3.100
Hyper fast Audio and Video encoder
usage: ffmpeg [options] [[infile options] -i infile]... {[outfile
options] outfile}...
Use -h to get full help or, even better, run 'man ffmpeg'
This is on a Debian testing machine.
I downloaded a media file and got the following error -
[ffmpeg/audio] aac: Inconsistent channel configuration.
[ffmpeg/audio] aac: get_buffer() failed
Error decoding audio.
[ffmpeg/audio] aac: Prediction is not allowed in AAC-LC.
Error decoding audio.
[ffmpeg/audio] aac: Sample rate index in program config element does
not match the sample rate index configured by the container.
[ffmpeg/audio] aac: Too large remapped id is not implemented. Update
your FFmpeg version to the newest one from Git. If the problem still
occurs, it means that your file has a feature which has not been
implemented.
[ffmpeg/audio] aac: If you want to help, upload a sample of this file
to ftp://upload.ffmpeg.org/incoming/ and contact the ffmpeg-devel
mailing list. (ffmpeg-de...@ffmpeg.org)
I downloaded the video from youtube and it said fine. I retried again
to verify the integrity of the file and it said it's at 100% so this
doesn't seem like a file corruption.
$ youtube-dl -c -f 22 9Jomgc8Ui0E
I did run ffprobe on it when I encountered the error above to see if I
could get more idea of the nature of the issue. -
$ ffprobe Bandit\ Bashing\ –\ Kenshi\ Gameplay\ –\ Let\'s\ Play\ Part\
2-9Jomgc8Ui0E.mp4
ffprobe version 4.1-1 Copyright (c) 2007-2018 the FFmpeg developers
built with gcc 8 (Debian 8.2.0-12)
configuration: --prefix=/usr --extra-version=1 --toolchain=hardened
--libdir=/usr/lib/x86_64-linux-gnu
--incdir=/usr/include/x86_64-linux-gnu --arch=amd64 --enable-gpl
--disable-stripping --enable-avresample --disable-filter=resample
--enable-avisynth --enable-gnutls --enable-ladspa --enable-libaom
--enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca
--enable-libcdio --enable-libcodec2 --enable-libflite
--enable-libfontconfig --enable-libfreetype --enable-libfribidi
--enable-libgme --enable-libgsm --enable-libjack --enable-libmp3lame
--enable-libmysofa --enable-libopenjpeg --enable-libopenmpt
--enable-libopus --enable-libpulse --enable-librsvg
--enable-librubberband --enable-libshine --enable-libsnappy
--enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora
--enable-libtwolame --enable-libvidstab --enable-libvorbis
--enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265
--enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi
--enable-lv2 --enable-omx --enable-openal --enable-opengl
--enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883
--enable-chromaprint --enable-frei0r --enable-libx264 --enable-shared
libavutil 56. 22.100 / 56. 22.100
libavcodec 58. 35.100 / 58. 35.100
libavformat58. 20.100 / 58. 20.100
libavdevice58. 5.100 / 58. 5.100
libavfilter 7. 40.101 / 7. 40.101
libavresample 4. 0. 0 / 4. 0. 0
libswscale 5. 3.100 / 5. 3.100
libswresample 3. 3.100 / 3. 3.100
libpostproc55. 3.100 / 55. 3.100
Input #0,
Re: [FFmpeg-user] Question about handling / unwanted changes of SAR in ffmpeg
2018-12-27 10:50 GMT+01:00, Uwe Freese : > Am 26.12.18 um 15:08 schrieb Reino Wijnsma: >>> Stream #0:0: Video: h264 (High), yuv420p(progressive), 688x560 [SAR >>> 64:45 DAR 2752:1575], SAR 172:121 DAR 7396:4235, 50 fps, 50 tbr, 1k tbn, >>> 100 tbc (default) >>> >>> What does SAR (and DAR) mean in the brackets compared to the second SAR >>> 172:121, which is slightly different? >> As far as I know: >> >> 688x 64/45 = 978,49w >> 978,49 x 1575/2752 = 560h --> 978x560 @ bitstream level >> 688x 172/121 = 977,98w >> 977,98 x 4235/7396 = 560h --> 977x560 @ container level > > So when I understand correctly, it could be that somewhere in ffmpeg, > the 64/45 is rounded to 978, given to another function / class > calculating the SAR for the container level, and this calculates to > 172/121, which is best matching for 978? > Instead, the better solution would then be to give the SAR to whatever > calculates the SAR to store at container level not as a double or int, > but as a "AVRational", which contains the "num" and "den" values (in > this case 64 and 45)? This is exactly what FFmpeg does (since forever). Carl Eugen ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] down sampling
Am 27.12.2018 um 20:18 schrieb Carl Eugen Hoyos: 2018-12-27 19:01 GMT+01:00, alex jamshedi : My goal is to receive a live audio stream that is being sampled at 131,072 Hz and re-sample it at 44.1 kHz before outputting it through my computers speakers. Is this a task ffmpeg can perform? Yes, there is an output option "-ar" that accepts "44100" as argument. I think the bigger problem is "outputting through the computers speakers". As far as I know it depends on the operating system, and under Windows it's impossible. Michael ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] Robustness on processing transport streams (TS)?
2018-12-27 15:14 GMT+01:00, Uwe Freese : > how robust and precise especially regarding a/v sync is ffmpeg when > processing transport streams (*.ts) files from TV recordings? I have seen many samples that can be transcoded in-sync with FFmpeg. > Some programs (e.g. Avidemux on Linux) have problems (crash) If you have an input file that crashes FFmpeg, please share it! > with some *.ts files I have, and I thought it maybe is a good idea > to first convert my input to a "raw" file with: > > ffmpeg -i input.ts -map 0:v:0 -c:v huffyuv -map 0:a -c:a pcm_s16le > -sn temp.mkv I would assume that this makes possible problems with A/V sync immutable. Always use the original input file for transcoding. A long time back, there was ProjectX to fix transport streams, I believe there is another program now (suggested by the HandBrake user support) but I forgot its name. To "insert silence", FFmpeg needs the "-async 1" option or the corresponding filter chain. Carl Eugen ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] Cannot build ffmpeg 4.1 on Cygwin
2018-12-27 19:27 GMT+01:00, Eric Thomas : > libavdevice/dshow.c: In function 'dshow_cycle_devices': > libavdevice/dshow.c:264:12: error: 'VARIANT' has no member named 'vt' The line exists since 2011, so I would not assume an issue that can / should be fixed in FFmpeg, is your Cygwin old? Work-around should be "--disable-indev=dshow". Carl Eugen ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] down sampling
2018-12-27 19:01 GMT+01:00, alex jamshedi : > My goal is to receive a live audio stream that is being sampled at > 131,072 Hz and re-sample it at 44.1 kHz before outputting it > through my computers speakers. Is this a task ffmpeg can perform? Yes, there is an output option "-ar" that accepts "44100" as argument. Carl Eugen ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
[FFmpeg-user] Cannot build ffmpeg 4.1 on Cygwin
Hello, I have been trying to build a set of shared libraries in Cygwin. Here is some system info: uname –a CYGWIN_NT-6.1-WOW64 tvmf2p4-ea00063 1.7.17(0.262/5/3) 2012-10-19 14:39 i686 Cygwin gcc -v 4.5.3 posix make –-version***(no gmake)* GNU Make 3.82.90 sed –-version GNU sed 4.2.1 bash --version GNU bash 4.1.10(4)-release (i686-pc-cygwin) nasm -v version 2.10.05 Configure cmd bash ./configure –prefix=/usr/tmp/FFmpeg-4.1 -–enable-shared make CC libavdevice/alldevices.o gcc: unrecognized option '-pthread' CC libavdevice/avdevice.o gcc: unrecognized option '-pthread' libavdevice/avdevice.c: In function 'device_next': libavdevice/avdevice.c:88:13: warning: 'av_oformat_next' is deprecated (declared at ./libavformat/avformat.h:2088) libavdevice/avdevice.c:88:13: warning: 'av_iformat_next' is deprecated (declared at ./libavformat/avformat.h:2080) CC libavdevice/dshow.o gcc: unrecognized option '-pthread' libavdevice/dshow.c: In function 'dshow_cycle_devices': libavdevice/dshow.c:264:12: error: 'VARIANT' has no member named 'vt' libavdevice/dshow.c:268:46: error: 'VARIANT' has no member named 'bstrVal' libavdevice/dshow.c: In function 'dshow_show_filter_properties': libavdevice/dshow.c:516:5: warning: missing braces around initializer libavdevice/dshow.c:516:5: warning: (near initialization for 'filter_info.achName') ffbuild/common.mak:60: recipe for target 'libavdevice/dshow.o' failed make: *** [libavdevice/dshow.o] Error 1 Can anyone help solve this problem? Thanks, Eric ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
[FFmpeg-user] down sampling
Hi, Hopefully this is an appropriate question for the forums. My goal is to receive a live audio stream that is being sampled at 131,072 Hz and re-sample it at 44.1 kHz before outputting it through my computers speakers. Is this a task ffmpeg can perform? Thank you. ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
[FFmpeg-user] Robustness on processing transport streams (TS)?
Hello, how robust and precise especially regarding a/v sync is ffmpeg when processing transport streams (*.ts) files from TV recordings? Some programs (e.g. Avidemux on Linux) have problems (crash) with some *.ts files I have, and I thought it maybe is a good idea to first convert my input to a "raw" file with: ffmpeg -i input.ts -map 0:v:0 -c:v huffyuv -map 0:a -c:a pcm_s16le -sn temp.mkv and use that hopefully "corrected" version for further processing (and encoding). Question is, does ffmpeg a good job detecting gaps and errors and fill out the produces audio streams with silence etc, so they stay in sync? I mean, when I play whatever corrupt file in VLC, it does a good job in this. How is it in ffmpeg? Any other program that someone suggests to (pre)process and "correct" such ts files under Linux? Some example output I had for a TV recording (which is not from a bad reception or so - it's totally ok when playing it back): $ ffmpeg -i input.ts -map 0:v:0 -c:v huffyuv -map 0:a -c:a pcm_s16le -sn temp.mkv ffmpeg version 3.2.12-1~deb9u1 Copyright (c) 2000-2018 the FFmpeg developers built with gcc 6.3.0 (Debian 6.3.0-18+deb9u1) 20170516 configuration: --prefix=/usr --extra-version='1~deb9u1' --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-stripping --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libebur128 --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared libavutil 55. 34.101 / 55. 34.101 libavcodec 57. 64.101 / 57. 64.101 libavformat 57. 56.101 / 57. 56.101 libavdevice 57. 1.100 / 57. 1.100 libavfilter 6. 65.100 / 6. 65.100 libavresample 3. 1. 0 / 3. 1. 0 libswscale 4. 2.100 / 4. 2.100 libswresample 2. 3.100 / 2. 3.100 libpostproc 54. 1.100 / 54. 1.100 [h264 @ 0x558e3216ee00] SPS unavailable in decode_picture_timing [h264 @ 0x558e3216ee00] non-existing PPS 0 referenced [h264 @ 0x558e3216ee00] SPS unavailable in decode_picture_timing [h264 @ 0x558e3216ee00] non-existing PPS 0 referenced [h264 @ 0x558e3216ee00] decode_slice_header error [h264 @ 0x558e3216ee00] no frame! [h264 @ 0x558e3216ee00] SPS unavailable in decode_picture_timing [h264 @ 0x558e3216ee00] non-existing PPS 0 referenced [h264 @ 0x558e3216ee00] SPS unavailable in decode_picture_timing [h264 @ 0x558e3216ee00] non-existing PPS 0 referenced [h264 @ 0x558e3216ee00] decode_slice_header error [h264 @ 0x558e3216ee00] no frame! [h264 @ 0x558e3216ee00] SPS unavailable in decode_picture_timing [h264 @ 0x558e3216ee00] non-existing PPS 0 referenced [h264 @ 0x558e3216ee00] SPS unavailable in decode_picture_timing [h264 @ 0x558e3216ee00] non-existing PPS 0 referenced [h264 @ 0x558e3216ee00] decode_slice_header error [h264 @ 0x558e3216ee00] no frame! [h264 @ 0x558e3216ee00] SPS unavailable in decode_picture_timing [h264 @ 0x558e3216ee00] non-existing PPS 0 referenced [h264 @ 0x558e3216ee00] SPS unavailable in decode_picture_timing [h264 @ 0x558e3216ee00] non-existing PPS 0 referenced [h264 @ 0x558e3216ee00] decode_slice_header error [h264 @ 0x558e3216ee00] no frame! [h264 @ 0x558e3216ee00] SPS unavailable in decode_picture_timing [h264 @ 0x558e3216ee00] non-existing PPS 0 referenced [h264 @ 0x558e3216ee00] SPS unavailable in decode_picture_timing [h264 @ 0x558e3216ee00] non-existing PPS 0 referenced [h264 @ 0x558e3216ee00] decode_slice_header error [h264 @ 0x558e3216ee00] no frame! [h264 @ 0x558e3216ee00] SPS unavailable in decode_picture_timing [h264 @ 0x558e3216ee00] non-existing PPS 0 referenced [h264 @ 0x558e3216ee00] SPS unavailable in decode_picture_timing [h264 @ 0x558e3216ee00] non-existing PPS 0 referenced [h264 @ 0x558e3216ee00] decode_slice_header error [h264 @ 0x558e3216ee00] no frame! [h264 @ 0x558e3216ee00] SPS unavailable in decode_picture_timing [h264 @ 0x558e3216ee00] non-existing PPS 0 referenced [h264 @ 0x558e3216ee00] SPS unavailable in decode_picture_timing [h264 @ 0x558e3216ee00] non-existing PPS 0 referenced [h264 @ 0x558e3216ee00] decode_slice_header error [h264 @ 0x558e3216ee00] no frame! [h264 @ 0x558e3216ee00] SPS unavailable in decode_picture_timing [h264 @ 0x558e3216ee00] non-existing PPS 0 referenced
[FFmpeg-user] Sync subtitles with HLS VOD
Hi, i'm trying to sync webVTT subtitles with HLS video. my process: 1. segment the mp4 to 10sec segments 2. repeat step 1 for multiple renditions * these steps are done by using the script found here: https://gist.github.com/mrbar42/ae111731906f958b396f30906004b3fa 3. segment the webVTT file and update the master playlist with the subtitles info. * using this command: ffmpeg -i SOURCE -f segment -segment_time 10 -segment_format webvtt -scodec copy -segment_list PLAYLISTNAME-sub.m3u8 -segment_list_type m3u8 SEGMENTNAME-%d.vtt problem: subtitles are not synced with video/audio cause: i noticed that my VOD script changes the video start time and my first segment of the video gets pkt_pts = 133200 , whereas in the original video it starts with 0. i'm not sure if a.i should force my script to set the initial ts to 0 - was not able to accomplish that b. add an offset to my VTT files. - seems really strange that i would have to ffprobe each segmented video for pkt_pts value and then edit my VTT files accordingly. what am i missing? is there an easier way to achieve this? Thanks. ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] Question about handling / unwanted changes of SAR in ffmpeg
Am 26.12.18 um 15:08 schrieb Reino Wijnsma: Stream #0:0: Video: h264 (High), yuv420p(progressive), 688x560 [SAR 64:45 DAR 2752:1575], SAR 172:121 DAR 7396:4235, 50 fps, 50 tbr, 1k tbn, 100 tbc (default) What does SAR (and DAR) mean in the brackets compared to the second SAR 172:121, which is slightly different? As far as I know: 688x 64/45 = 978,49w 978,49 x 1575/2752 = 560h --> 978x560 @ bitstream level 688x 172/121 = 977,98w 977,98 x 4235/7396 = 560h --> 977x560 @ container level So when I understand correctly, it could be that somewhere in ffmpeg, the 64/45 is rounded to 978, given to another function / class calculating the SAR for the container level, and this calculates to 172/121, which is best matching for 978? Instead, the better solution would then be to give the SAR to whatever calculates the SAR to store at container level not as a double or int, but as a "AVRational", which contains the "num" and "den" values (in this case 64 and 45)? Could this be done, or is there a reason not to do that? Regards, Uwe ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
