[FFmpeg-user] Covert to TS and back
Why when I convert a video file to mpeg-ts and then convert it back to the original one there is a file size increase? Is there a way to avoid that? I want to be able to bring back the original file. Is it possible? Julien ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
[FFmpeg-user] A command for splitting a video file by file size
Hi, I want to split a file by size but starting from a specific byte or bit. For example I have a file of 6mb and I want to take the part from 4mb to the end. Is it possible to do that? Julien ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
[FFmpeg-user] Default output fps vs fps filter @ 30fps
I have an h264/vorbis video at 1k fps and want to convert it to mpeg2/ac3 at 30 fps. If I don't use the fps filter ffmpeg appears to automatically convert the output to 30 fps (which I assume is the default -r value), but the following warnings are displayed: More than 1000 frames duplicated More than 1 frames duplicated If I use the fps filter (fps=fps=30), then those warnings are not displayed and there's no indication of duplicated (or dropped) frames... although I assume there must have been. I can't see any obvious difference between the 2 output files (although I haven't watched both completely through), so I'm just curious if there IS any appreciable difference for this video. Is there any advantage (or disadvantage) to using the fps filter vs the -r option in this particular case? Here's full output for both commands: $ ffmpeg -i TEST.mkv -codec:v mpeg2video -b:v 16384k -bufsize 4096k -maxrate 3k -codec:a ac3 -b:a 448k -ar 48000 -f vob TEST-default.mpg ffmpeg version N-95488-g09581f7923 Copyright (c) 2000-2019 the FFmpeg developers built with gcc 7 (GCC) configuration: --enable-gpl --enable-nonfree --enable-libass --enable-libfdk-aac --enable-libfreetype --enable-libmp3lame --enable-libopus --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-openssl libavutil 56. 35.101 / 56. 35.101 libavcodec 58. 59.102 / 58. 59.102 libavformat58. 33.100 / 58. 33.100 libavdevice58. 9.100 / 58. 9.100 libavfilter 7. 63.100 / 7. 63.100 libswscale 5. 6.100 / 5. 6.100 libswresample 3. 6.100 / 3. 6.100 libpostproc55. 6.100 / 55. 6.100 Input #0, matroska,webm, from 'TEST.mkv': Metadata: ENCODER : Lavf57.71.100 Duration: 00:58:41.47, start: 0.00, bitrate: 2190 kb/s Stream #0:0: Video: h264 (High), yuv420p(tv, bt709, progressive), 1600x900 [SAR 1:1 DAR 16:9], 1k fps, 30 tbr, 1k tbn, 60 tbc (default) Metadata: DURATION: 00:58:41.43400 Stream #0:1: Audio: vorbis, 48000 Hz, stereo, fltp (default) Metadata: DURATION: 00:58:41.47400 Stream mapping: Stream #0:0 -> #0:0 (h264 (native) -> mpeg2video (native)) Stream #0:1 -> #0:1 (vorbis (native) -> ac3 (native)) Press [q] to stop, [?] for help Output #0, vob, to 'TEST-default.mpg': Metadata: encoder : Lavf58.33.100 Stream #0:0: Video: mpeg2video (Main), yuv420p, 1600x900 [SAR 1:1 DAR 16:9], q=2-31, 16384 kb/s, 30 fps, 90k tbn, 30 tbc (default) Metadata: DURATION: 00:58:41.43400 encoder : Lavc58.59.102 mpeg2video Side data: cpb: bitrate max/min/avg: 3000/0/16384000 buffer size: 4096000 vbv_delay: N/A Stream #0:1: Audio: ac3, 48000 Hz, stereo, fltp, 448 kb/s (default) Metadata: DURATION: 00:58:41.47400 encoder : Lavc58.59.102 ac3 More than 1000 frames duplicated 73126kB time=00:03:11.54 bitrate=3127.4kbits/s dup=985 drop=0 speed= 6.8x More than 1 frames duplicated510586kB time=00:13:46.84 bitrate=5058.7kbits/s dup=9998 drop=0 speed=6.22x frame=105644 fps=198 q=2.0 Lsize= 1967088kB time=00:58:41.50 bitrate=4576.0kbits/s dup=31190 drop=0 speed=6.58x video:1749220kB audio:192582kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 1.302177% $ ffmpeg -i TEST.mkv -codec:v mpeg2video -b:v 16384k -bufsize 4096k -maxrate 3k -codec:a ac3 -b:a 448k -ar 48000 -vf fps=fps=30 -f vob TEST-fpsfilter.mpg ffmpeg version N-95488-g09581f7923 Copyright (c) 2000-2019 the FFmpeg developers built with gcc 7 (GCC) configuration: --enable-gpl --enable-nonfree --enable-libass --enable-libfdk-aac --enable-libfreetype --enable-libmp3lame --enable-libopus --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-openssl libavutil 56. 35.101 / 56. 35.101 libavcodec 58. 59.102 / 58. 59.102 libavformat58. 33.100 / 58. 33.100 libavdevice58. 9.100 / 58. 9.100 libavfilter 7. 63.100 / 7. 63.100 libswscale 5. 6.100 / 5. 6.100 libswresample 3. 6.100 / 3. 6.100 libpostproc55. 6.100 / 55. 6.100 Input #0, matroska,webm, from 'TEST.mkv': Metadata: ENCODER : Lavf57.71.100 Duration: 00:58:41.47, start: 0.00, bitrate: 2190 kb/s Stream #0:0: Video: h264 (High), yuv420p(tv, bt709, progressive), 1600x900 [SAR 1:1 DAR 16:9], 1k fps, 30 tbr, 1k tbn, 60 tbc (default) Metadata: DURATION: 00:58:41.43400 Stream #0:1: Audio: vorbis, 48000 Hz, stereo, fltp (default) Metadata: DURATION: 00:58:41.47400 Stream mapping: Stream #0:0 -> #0:0 (h264 (native) -> mpeg2video (native)) Stream #0:1 -> #0:1 (vorbis (native) -> ac3 (native)) Press [q] to stop, [?] for help Output #0, vob, to 'TEST-fpsfilter.mpg': Metadata: encoder : Lavf58.33.100 Stream #0:0: Video: mpeg2video (Main), yuv420p,
Re: [FFmpeg-user] A command for splitting a video file by file size
> Am 20.10.2019 um 22:58 schrieb Julien Dotsev : > > I want to split a file by size but starting from a specific byte or bit. > For example I have a file of 6mb and I want to take the part from 4mb to > the end. Is it possible to do that? It should be possible with dd. Carl Eugen ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
[FFmpeg-user] using ffmpeg in a batch mode
My bad - sorry, I just posted about this and forgot to mention that I'm on Windows 10 and using Powershell. :) - Remember Brandolini's Law and look around. ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] Adding macrovision style copy protection to a theora video
> On Oct 19, 2019, at 2:04 PM, Marc Roos wrote: > > > Why not develop some watermarking technology? For audio you have even > some technique that can survive (some) re-encoding. And then have with > your users a license agreement. You can also just tell them you are > using this technology, while you do not ;) Speaking of licensing, where do copy protection features or producing copy protected content using GPL’ed software stand in the legalese? Just curious... I wanna say though, I feel like how copy protection mechanisms usually work is, the more invested the content producers are in implementing it, the more likely it is to be defeated. Like MPAA level studios and their distributors are really into all the DRM and copy prevention schemes, and you see, not bootlegged, but pirated versions online all the time, with all the colorimetry info, HDR metadata, I mean that 4K HDR10 Atmos .torrent is not some analog hole capture, it’s the original source with its expensive DRM chopped off! Sometimes, as long as content has the right balance of obscurity and a tricky enough copy protection to defeat, it’ll stay “protected”. So basically if your product becomes a successful commercially (a favorable situation) and the protection mechanism becomes defeated (now an eventuality) it’s like, why would you try to put the effort into something that’s only useful If your product is a failure you know? But back to the op’s question, I do know it’s possible to include program metadata that _signals_ to the decoder that copy protection should be generated in any analog signals it outputs. But it can also ignore it and not do the extra work implementing clock sensitive analog signal processing (read: mangling), and to the user, since it can play more content, it’s not hard to see how it became de facto standard practice. But a while back Vista Media Center up and refused to record some network’s shows one day (I think it was a Comcast one not sure tho), because Microsoft had implemented that particular feature, which is a real thing that nobody really did..? Especially for software? Then it turned out MS didn’t mean to do that, they just must have been really following the specs to the letter. Then the network said oh yeah that metadata flag we must have enabled it on that stream by accident sorry. But this was a rare exception. It pretty much shows almost no player respects this metadata (since other cablecard boxes had no problem) and content providers don’t even include it unless by mistake (since it known not to be very effective). If you haven’t, read up on HDCP, which I’d say is the modern, digital data analogy to MV I think, It’s history is pretty impressive for all the wrong reasons, as it was pretty much a complete failure as DRM. HDCP actually encrypts the video in transmission with respectably sized keys, but how it negotiated how devices exchanged key data was revealed to be flawed in its design. Now the master key is out in the open. That’s like, not something even fee-paying content participants with executed licensing contracts had access to. To top it all off, it's the frequent cause of compatible sinks and sources not working claiming the other device is incompliant. I’ve been rambling for a while now, but actually I think yeah it’s probably possible to insert metadata that signals copy protection using ffmpeg into a program stream. What I’m pretty sure of is that it can’t generate one on its own. Something else to consider, for this to work, the media decoder in the console(?) needs to be able to see that and insert the vblank pulses. And whatever media your game is distributed on is only compatible with a device that implements the copy protection mechanism. Basically you need control of the whole ecosystem, something like the NES comes to mind, what exactly is your distribution method? > Is there a way to get a video output by ffmpeg to send an extra vblank with > random pixels that a TV would ignoreicense agreement. You can also just tell > them you are > using this technolbut would mess up a video recorded on > a vcr? In other words, what is between the “video output” and “ffmpeg”? Obviously it’s not ffmpeg -> video output, or is it some hardware device that actually makes this possible? Ted ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] Split a video file in two by size and merge itback
Yes, the ts file works exactly as you said. It can be split and every part is readable. But As I said I want to be able to keep the original file. So if it is mp4 for example and I converted to mpeg-ts, when I reconvert it back there is a file size increase. Is there a way to avoid that? Julien На чт, 17.10.2019 г. в 13:26 ч. Julien Dotsev написа: > Hey Moritz, > That's great advice! > I will definitely try it. > > Thank you very much! > Julien > > > On Thu, 17 Oct 2019, 12:14 Moritz Barsnick, wrote: > >> On Thu, Oct 17, 2019 at 03:44:49 +0200, Julien Dotsev wrote: >> > Actually every bit. I want to be able to bring back the original file >> > unchanged up to every bit. But every chunk playable. >> >> If you want not only the content (i.e. the stream), but also everything >> else including the container to be identical, I believe the best >> (perhaps only) choice is to use MPEG-TS as a container. >> >> In theory, you should be able to split MPEG-TS files at arbitrary >> bytes. I have bad experiences with that, but it works great at packet >> boarders, meaning your segments need to be exact multiples of 188 >> bytes. This works great for all my players (ffplay, mplayer, mpv, VLC, >> and my Set Top Box), and the binary splits can of course just be >> reassembled to create the original file. >> >> I often cut my files on Linux with dd: >> >> $ dd if=input.ts bs=188 skip=10325 count=11877 of=output.ts >> >> but for equally sized files, you would use "split", e.g.: >> >> $ split --bytes=$((188 * 11877)) -d input.ts output.ts. >> >> Cheers, >> Moritz >> ___ >> ffmpeg-user mailing list >> ffmpeg-user@ffmpeg.org >> https://ffmpeg.org/mailman/listinfo/ffmpeg-user >> >> To unsubscribe, visit link above, or email >> ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe". > > ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] Covert to TS and back
> Am 20.10.2019 um 23:36 schrieb Julien Dotsev : > > Why when I convert a video file to mpeg-ts and then convert it back to the > original one there is a file size increase? Is there a way to avoid that? Command line and complete, uncut console output missing. > I want to be able to bring back the original file. Is it possible? In general this is not possible but see above. Carl Eugen ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] A command for splitting a video file by file size
What is dd? On Mon, 21 Oct 2019, 00:00 Carl Eugen Hoyos, wrote: > > > > Am 20.10.2019 um 22:58 schrieb Julien Dotsev : > > > > I want to split a file by size but starting from a specific byte or bit. > > For example I have a file of 6mb and I want to take the part from 4mb to > > the end. Is it possible to do that? > > It should be possible with dd. > > Carl Eugen > ___ > ffmpeg-user mailing list > ffmpeg-user@ffmpeg.org > https://ffmpeg.org/mailman/listinfo/ffmpeg-user > > To unsubscribe, visit link above, or email > ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe". ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
[FFmpeg-user] How to use ffmpeg in batch mode?
I have been trying to convert a bunch of mkv files to mp4 using ffmpeg in batch mode via a .bat file but I can't get it right. I've followed a number of suggested commands from various hits I've got on google but none of them work. Don't need anything fancy. Just a basic convert from mkv to mp4 if anyone can help? - Remember Brandolini's Law and look around. ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] Default output fps vs fps filter @ 30fps
> Am 20.10.2019 um 23:05 schrieb Hans Carlson : > > Is there any advantage (or disadvantage) to using the fps filter vs the -r > option in this particular case? The main difference between the fps filter and the constant frame rate output that can be forced by the command line utility is that the filter is less verbose. The algorithms are not equally greedy but it is likely that this makes little or no difference for your use case. Carl Eugen ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] 453 Not Enough Bandwidth Error when streaming from IP camera
Am Mo., 21. Okt. 2019 um 01:03 Uhr schrieb Michael Shaffer
:
>
> I have a Sanyo IP camera that I'm streaming to Youtube via rtsp and ffmpeg.
> The camera is supposed to be capable of 1080p but I can only get it to work
> in 720p mode. When I try 1080p mode ffmpeg gives the following error below.
> If anyone has any ideas I'd really appreciate it.
>
> user@user-computer:~$ ffmpeg -rtsp_transport tcp -y -rtbufsize 200M -i
> "rtsp://admin:password@192.168.1.103:554/VideoInput/1/h264/1" -c:a copy
> -c:v copy -f flv 1080test.mp4
>
> ffmpeg version N-50401-gdd01947397-static https://johnvansickle.com/ffmpeg/
> Copyright (c) 2000-2019 the FFmpeg developers
> built with gcc 6.3.0 (Debian 6.3.0-18+deb9u1) 20170516
> configuration: --enable-gpl --enable-version3 --enable-static
> --disable-debug --disable-ffplay --disable-indev=sndio
> --disable-outdev=sndio --cc=gcc-6 --enable-fontconfig --enable-frei0r
> --enable-gnutls --enable-gmp --enable-libgme --enable-gray
> --enable-libfribidi --enable-libass --enable-libfreetype
> --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb
> --enable-libopenjpeg --enable-librubberband --enable-libsoxr
> --enable-libspeex --enable-libsrt --enable-libvorbis --enable-libopus
> --enable-libtheora --enable-libvidstab --enable-libvo-amrwbenc
> --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265
> --enable-libxml2 --enable-libxvid --enable-libzimg
> libavutil 56. 35.101 / 56. 35.101
> libavcodec 58. 59.102 / 58. 59.102
> libavformat58. 33.100 / 58. 33.100
> libavdevice58. 9.100 / 58. 9.100
> libavfilter 7. 62.101 / 7. 62.101
> libswscale 5. 6.100 / 5. 6.100
> libswresample 3. 6.100 / 3. 6.100
> libpostproc55. 6.100 / 55. 6.100
> [rtsp @ 0xbafb380] method SETUP failed: 453 Not Enough Bandwidth
> rtsp://admin:password@192.168.1.103:554/VideoInput/1/h264/1: Server
> returned 4XX Client Error, but not one of 40{0,1,3,4}
I may be missing something but this is an error that your camera
returns (FFmpeg only shows the error to you).
Carl Eugen
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[FFmpeg-user] 453 Not Enough Bandwidth Error when streaming from IP camera
I have a Sanyo IP camera that I'm streaming to Youtube via rtsp and ffmpeg.
The camera is supposed to be capable of 1080p but I can only get it to work
in 720p mode. When I try 1080p mode ffmpeg gives the following error below.
If anyone has any ideas I'd really appreciate it.
user@user-computer:~$ ffmpeg -rtsp_transport tcp -y -rtbufsize 200M -i
"rtsp://admin:password@192.168.1.103:554/VideoInput/1/h264/1" -c:a copy
-c:v copy -f flv 1080test.mp4
ffmpeg version N-50401-gdd01947397-static https://johnvansickle.com/ffmpeg/
Copyright (c) 2000-2019 the FFmpeg developers
built with gcc 6.3.0 (Debian 6.3.0-18+deb9u1) 20170516
configuration: --enable-gpl --enable-version3 --enable-static
--disable-debug --disable-ffplay --disable-indev=sndio
--disable-outdev=sndio --cc=gcc-6 --enable-fontconfig --enable-frei0r
--enable-gnutls --enable-gmp --enable-libgme --enable-gray
--enable-libfribidi --enable-libass --enable-libfreetype
--enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb
--enable-libopenjpeg --enable-librubberband --enable-libsoxr
--enable-libspeex --enable-libsrt --enable-libvorbis --enable-libopus
--enable-libtheora --enable-libvidstab --enable-libvo-amrwbenc
--enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265
--enable-libxml2 --enable-libxvid --enable-libzimg
libavutil 56. 35.101 / 56. 35.101
libavcodec 58. 59.102 / 58. 59.102
libavformat58. 33.100 / 58. 33.100
libavdevice58. 9.100 / 58. 9.100
libavfilter 7. 62.101 / 7. 62.101
libswscale 5. 6.100 / 5. 6.100
libswresample 3. 6.100 / 3. 6.100
libpostproc55. 6.100 / 55. 6.100
[rtsp @ 0xbafb380] method SETUP failed: 453 Not Enough Bandwidth
rtsp://admin:password@192.168.1.103:554/VideoInput/1/h264/1: Server
returned 4XX Client Error, but not one of 40{0,1,3,4}
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Re: [FFmpeg-user] Default output fps vs fps filter @ 30fps
On Sun, 20 Oct 2019, Carl Eugen Hoyos wrote: Am 20.10.2019 um 23:05 schrieb Hans Carlson : Is there any advantage (or disadvantage) to using the fps filter vs the -r option in this particular case? The main difference between the fps filter and the constant frame rate output that can be forced by the command line utility is that the filter is less verbose. The algorithms are not equally greedy but it is likely that this makes little or no difference for your use case. Ah, so the fps filter *could* display the same/similar kind of "frames duplicated" messages, it just doesn't. Thanks Carl, that's kind of what I suspected. PS. I do realize there may be instances (eg. different codecs and/or increasing fps) where the filter vs -r option could make more of a difference. But in this instance, as you say, probably nothing noticeable. ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
[FFmpeg-user] how to copy timed metadata from flv to mpegts
Hi, I have a flv file, which has timed metadata tags. how to copy timed metadata as id3 metadata from flv to mpegts? Thanks! Regards Andrew ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] HLS conversion fails with "Cannot use rename on non file protocol..." error
Le 20/10/2019 à 11:27, Dennis Mungai a écrit : > On Sun, 20 Oct 2019 at 11:14, Hoggins! wrote: >> Le 20/10/2019 à 00:33, Dennis Mungai a écrit : >>> On Sat, 19 Oct 2019 at 20:21, Hoggins! wrote: Hello there, First post on this list. I'm trying to convert a file to HLS without reencoding, and I'm stuck with an error I quite don't understand. Here's my command: ffmpeg -i rockinspace-okoyome.mp4 -c:v copy -flags +cgop -g 50 -hls_time 1 -hls_playlist_type vod rockinspace-okoyome.m3u8 So it starts normally and creates a bunch of .ts files as expected, but at file no 1764, it stops with an error I cannot relate to: [hls @ 0x55657b07ec00] Opening 'rockinspace-okoyome1761.ts' for writing [hls @ 0x55657b07ec00] Opening 'rockinspace-okoyome1762.ts' for writing [hls @ 0x55657b07ec00] Opening 'rockinspace-okoyome1763.ts' for writing [hls @ 0x55657b07ec00] Opening 'rockinspace-okoyome1764.ts' for writing [hls @ 0x55657b07ec00] Cannot use rename on non file protocol, this may lead to races and temporary partial files frame=69230 fps=436 q=-1.0 Lsize=N/A time=00:46:07.60 bitrate=N/A speed=17.4x video:1352238kB audio:42374kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown [aac @ 0x55657b19b280] Qavg: 174.694 Source and destination files are on a local filesystem, so I quite don't understand why I would get this error. But considering it always happens at this particular file, I'm guessing there might be something odd with the source file. But what? Source file is available here if you have a chance to test it: https://radiom.fr/media-rockinspace-okoyome.mp4 Note that the same thing happens with another file but on another segment: [hls @ 0x557732d14140] Opening 'rockinspace-enola2178.ts' for writing [hls @ 0x557732d14140] Opening 'rockinspace-enola2179.ts' for writingd=15.4x [hls @ 0x557732d14140] Opening 'rockinspace-enola2180.ts' for writing [hls @ 0x557732d14140] Opening 'rockinspace-enola2181.ts' for writing [hls @ 0x557732d14140] Cannot use rename on non file protocol, this may lead to races and temporary partial files frame=64024 fps=385 q=-1.0 Lsize=N/A time=00:43:03.63 bitrate=N/A speed=15.5x video:1250493kB audio:40609kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown [aac @ 0x55772e883cc0] Qavg: 161.026 (source file is https://radiom.fr/media-rockinspace-enola.mp4) Also note that I was previously using the following command, but I quit because it resulted in audio and video being out of sync: ffmpeg -i rockinspace-okoyome.mp4 -c:v -c:v libx264 -crf 17 -preset slow -flags +cgop -g 50 -hls_time 1 -hls_playlist_type vod rockinspace-okoyome.m3u8 Any idea? Should I try to reencode the source file to "normalize" it? Thanks! Hoggins! >>> Hello Hoggins, >>> >>> Please post the full console output from FFmpeg, such that we can see >>> the build version, etc. >>> On my end, using the latest ffmpeg git tip, I'm able to run your >>> command with no issues whatsoever. >>> >>> Note that only git tip is supported here. >> I will try to compile something using the latest git tip to see where I >> can get. >> Do you mean that you successfully ran my command along with the provided >> source file on your end? >> >> Anyway, here is the requested console output using the second example >> provided (https://radiom.fr/media-rockinspace-enola.mp4): >> >> ffmpeg -i ../../rockinspace-enola.mp4 -c:v copy -flags +cgop -g 50 >> -hls_time 1 -hls_playlist_type vod rockinspace-enola.m3u8 >> ffmpeg version 4.1.4 Copyright (c) 2000-2019 the FFmpeg developers >> built with gcc 9 (GCC) >> configuration: --prefix=/usr --bindir=/usr/bin >> --datadir=/usr/share/ffmpeg --docdir=/usr/share/doc/ffmpeg >> --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 >> --mandir=/usr/share/man --arch=x86_64 --optflags='-O2 -g -pipe -Wall >> -Werror=format-security -Wp,-D_FORTIFY_SOURCE=2 >> -Wp,-D_GLIBCXX_ASSERTIONS -fexceptions -fstack-protector-strong >> -grecord-gcc-switches -specs=/usr/lib/rpm/redhat/redhat-hardened-cc1 >> -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -m64 -mtune=generic >> -fasynchronous-unwind-tables -fstack-clash-protection >> -fcf-protection' --extra-ldflags='-Wl,-z,relro -Wl,--as-needed >> -Wl,-z,now -specs=/usr/lib/rpm/redhat/redhat-hardened-ld ' >> --extra-cflags=' ' --enable-libopencore-amrnb >> --enable-libopencore-amrwb --enable-libvo-amrwbenc --enable-version3 >> --enable-bzlib --disable-crystalhd --enable-fontconfig >> --enable-frei0r --enable-gcrypt --enable-gnutls --enable-ladspa >>
Re: [FFmpeg-user] Writing PGM files
On 10/20/19, Michael Koch wrote: > Hello, > > FFmpeg can read PGM (Portable Gray Map) files that are either ASCII > coded (beginning with "P2") or binary coded (beginning with "P5"). When > writing a PGM file, the default format is binary. Is it also possible to > write an ASCII coded PGM file? Nope. > > Thanks, > Michael > > ___ > ffmpeg-user mailing list > ffmpeg-user@ffmpeg.org > https://ffmpeg.org/mailman/listinfo/ffmpeg-user > > To unsubscribe, visit link above, or email > ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe". ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
[FFmpeg-user] File metadata in m4a causing player problem?
I have a set of m4a audio files which I can play but on which the progress bar does not work. If I select anywhere along the progress bar playing starts again from the beginning. The only difference I can see when using ffprobe to examine the metadata is that files which have a creation_time set have a non functioning progress bar problem. Files which do NOT have the Stream metadata :creation_time, play correctly. The metadata is shown:- Stream #0:0(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 319 kb/s (default) Metadata: creation_time : 1970-01-01T00:00:00.00Z handler_name: SoundHandler This creation_time seems to me like a default not the actual time when file was created and may be the cause of the player problem. What I wish to do is remove the tag: creation_time : 1970-01-01T00:00:00.00Z I confess using ffmpeg to achieve this, if indeed this is the correct tool, is well beyond my ability and I am seeking help here. I appreciate this still may not fix my problem but I must try and eliminate possible causes and this is one possible fix. Can anybody help please. Budgie ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
[FFmpeg-user] Writing PGM files
Hello, FFmpeg can read PGM (Portable Gray Map) files that are either ASCII coded (beginning with "P2") or binary coded (beginning with "P5"). When writing a PGM file, the default format is binary. Is it also possible to write an ASCII coded PGM file? Thanks, Michael ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
