[FFmpeg-user] How to capture an RTMP stream
I'm trying to record some of this stream: rtmp://cp258849.live.edgefcs.net/live/SonLife0616@327662 VLC will play this, so I thought it would be easy :-) ffmpeg -i "rtmp://cp258849.live.edgefcs.net/live/SonLife0616@327662" -rtmp_live live test.mp4 ffmpeg version N-73907-gc169243 Copyright (c) 2000-2015 the FFmpeg developers built with gcc 4.8 (Ubuntu 4.8.4-2ubuntu1~14.04) configuration: --enable-shared --enable-gpl --enable-nonfree --enable-libxvid --enable-libx264 --enable-libmp3lame --enable-libvorbis --enable-libfaac --enable-libopenjpeg --enable-librtmp libavutil 54. 28.100 / 54. 28.100 libavcodec 56. 50.101 / 56. 50.101 libavformat56. 40.101 / 56. 40.101 libavdevice56. 4.100 / 56. 4.100 libavfilter 5. 25.100 / 5. 25.100 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 2.101 / 1. 2.101 libpostproc53. 3.100 / 53. 3.100 Closing connection: NetStream.Play.StreamNotFound rtmp://cp258849.live.edgefcs.net/live/SonLife0616@327662: Unknown error occurred Where can I learn more about how to decipher the RTMP url into parts that ffmpeg needs? Thanks, Deron ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] How to capture an RTMP stream
On 7/25/15 6:02 AM, Moritz Barsnick wrote: On Sat, Jul 25, 2015 at 11:52:56 +, Carl Eugen Hoyos wrote: But you could test without "--enable-librtmp". Is it possible to choose different protocol handlers at runtime, as it is with decoders (analog to the choice of "mp3" vs. "mp3float" using "-c:v" as inpt option)? Or does using librtmp totally disable ffmpeg's native implementation? (Indeed, my librtmp doesn't seem to like this URL, shown by trying with rtmpdump. But I'm not the original poster. ;-)) Moritz Well, my rtmpdump _does_ work. It claims to be v2.4 and I was able to get the stream and pipe it to ffmpeg. Inelegant, but it works. Sadly, the audio/video is out of sync by about 75ms. Just enough to be annoying. What can I do to help track down the problem? Is the interest only in the native solution (try without librtmp) or ?? Side question: With the sync problem, I've used -itsoffset (and mapping of course) in the past to fix sync issues, but that obviously won't help if I don't have the streams split _before_ it gets to ffmpeg. Any solution? Thanks Deron ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] How to capture an RTMP stream
On 7/25/15 9:57 AM, Carl Eugen Hoyos wrote: Deron pagestream.org> writes: Well, my rtmpdump _does_ work. It claims to be v2.4 and I was able to get the stream and pipe it to ffmpeg. Inelegant, but it works. Sadly, the audio/video is out of sync by about 75ms. Just enough to be annoying. Does the same desync exist with FFmpeg? Carl Eugen I would have to get ffmpeg to work to begin with to test that. Should I try ffmpeg without librtmp? It could be the stream source itself for all I know, but I would still need to adjust sync someplace along the food chain if so. I was just wondering if there was some unexplored option to adjust audio sync in this case within ffmpeg. Thanks for everyone's time, Deron ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
[FFmpeg-user] Over length file
I have several files that when I try and cut out a chunk of it, I get a file that is 100+ seconds longer than I asked for. It appears to be related to the subtitle stream. If I generate the original without the subtitle stream, then I get the file segment I ask for. If it contains the subtitle, it get something 100+ seconds longer than I ask for. What could I do to correct this or determine more specifically what the problem is? Deron ffmpeg -i "/work/disk1/HD-IFYL1081.mp4" -map 0 -ss 58.908909 -t 1175.709106 -c:v libx264 -preset veryfast -qp 0 -ac 2 -ar 48000 -c:s mov_text -y "/work/disk1/HD-IFYL1081 A.mp4" ffmpeg version N-71957-g5f2b4a2 Copyright (c) 2000-2015 the FFmpeg developers built with gcc 4.8 (Ubuntu 4.8.2-19ubuntu1) configuration: --enable-shared --enable-gpl --enable-nonfree --enable-libxvid --enable-libx264 --enable-libmp3lame --enable-libvorbis --enable-libfaac --enable-libopenjpeg --enable-librtmp libavutil 54. 23.101 / 54. 23.101 libavcodec 56. 37.102 / 56. 37.102 libavformat56. 32.100 / 56. 32.100 libavdevice56. 4.100 / 56. 4.100 libavfilter 5. 16.101 / 5. 16.101 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 1.100 / 1. 1.100 libpostproc53. 3.100 / 53. 3.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/work/disk1/HD-IFYL1081.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 encoder : Lavf56.32.100 Duration: 00:29:31.80, start: 0.021333, bitrate: 64906 kb/s Stream #0:0(und): Video: h264 (High 4:4:4 Predictive) (avc1 / 0x31637661), yuv422p, 1280x720 [SAR 1:1 DAR 16:9], 64775 kb/s, 59.94 fps, 59.94 tbr, 11988 tbn, 119.88 tbc (default) Metadata: handler_name: VideoHandler Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 124 kb/s (default) Metadata: handler_name: SoundHandler Stream #0:2(und): Subtitle: mov_text (tx3g / 0x67337874), 1280x720, 0 kb/s (default) Metadata: handler_name: SubtitleHandler No pixel format specified, yuv422p for H.264 encoding chosen. Use -pix_fmt yuv420p for compatibility with outdated media players. [libx264 @ 0xa85e40] using SAR=1/1 [libx264 @ 0xa85e40] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX [libx264 @ 0xa85e40] profile High 4:4:4 Predictive, level 3.2, 4:2:2 8-bit [libx264 @ 0xa85e40] 264 - core 146 r2538 121396c - H.264/MPEG-4 AVC codec - Copyleft 2003-2015 - http://www.videolan.org/x264.html - options: cabac=1 ref=1 deblock=1:0:0 analyse=0x1:0x111 me=hex subme=2 psy=0 mixed_ref=0 me_range=16 chroma_me=1 trellis=0 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=0 chroma_qp_offset=0 threads=48 lookahead_threads=5 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=0 weightp=1 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc=cqp mbtree=0 qp=0 Output #0, mp4, to '/work/disk1/HD-IFYL1081 A.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 encoder : Lavf56.32.100 Stream #0:0(und): Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv422p, 1280x720 [SAR 1:1 DAR 16:9], q=-1--1, 59.94 fps, 11988 tbn, 59.94 tbc (default) Metadata: handler_name: VideoHandler encoder : Lavc56.37.102 libx264 Stream #0:1(und): Audio: aac (libfaac) ([64][0][0][0] / 0x0040), 48000 Hz, stereo, s16, 128 kb/s (default) Metadata: handler_name: SoundHandler encoder : Lavc56.37.102 libfaac Stream #0:2(und): Subtitle: mov_text ([8][0][0][0] / 0x0008), 1280x720 (default) Metadata: handler_name: SubtitleHandler encoder : Lavc56.37.102 mov_text Stream mapping: Stream #0:0 -> #0:0 (h264 (native) -> h264 (libx264)) Stream #0:1 -> #0:1 (aac (native) -> aac (libfaac)) Stream #0:2 -> #0:2 (mov_text (native) -> mov_text (native)) Press [q] to stop, [?] for help frame=70473 fps=258 q=-1.0 Lsize=10888940kB time=00:19:35.72 bitrate=75869.9kbits/s video:10868826kB audio:18364kB subtitle:22kB other streams:0kB global headers:0kB muxing overhead: 0.015869% [libx264 @ 0xa85e40] frame I:356 Avg QP: 0.00 size:368826 [libx264 @ 0xa85e40] frame P:70117 Avg QP: 0.00 size:156857 [libx264 @ 0xa85e40] mb I I16..4: 74.5% 0.0% 25.5% [libx264 @ 0xa85e40] mb P I16..4: 23.7% 0.0% 1.0% P16..4: 21.4% 3.1% 2.7% 0.0% 0.0%skip:48.1% [libx264 @ 0xa85e40] 8x8 transform intra:0.0% inter:0.0% [libx264 @ 0xa85e40] coded y,uvDC,uvAC intra: 77.0% 73.5% 73.4% inter: 33.8% 32.3% 32.3% [libx264 @ 0xa85e40] i16 v,h,dc,p: 65% 34% 0% 0% [libx264 @ 0xa85e40] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 39% 36% 3% 3% 6% 4% 4% 3% 2% [libx264 @ 0xa85e40] i8c dc,h,v,p: 27% 30% 44% 0% [libx264 @ 0xa85e40] Weighted P-Frames: Y:0.5% UV:0.4% [libx264 @ 0xa85e40] kb/s:75729.76 and now you can
Re: [FFmpeg-user] Over length file
On 9/2/15 8:44 PM, Deron wrote: I have several files that when I try and cut out a chunk of it, I get a file that is 100+ seconds longer than I asked for. It appears to be related to the subtitle stream. If I generate the original without the subtitle stream, then I get the file segment I ask for. If it contains the subtitle, it get something 100+ seconds longer than I ask for. What could I do to correct this or determine more specifically what the problem is? Deron The problem boiled down to a subtitle entries that held for over 2 minutes. I corrected the original srts and all is well(?) Should ffmpeg have cut in the middle of the subtitle? Should I create a bug entry? Deron ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
[FFmpeg-user] Decklink CC Output
Has anyone looked into adding closed caption (CEA-708) output to the Decklink driver? I have most of the documentation necessary, but am missing the SMPTE 333 (SMPTE 333M-1999) documentation needed. Otherwise, has anyone captured VANC data to figure out how 708 gets put into the VANC data for output? Thanks, Deron ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] Decklink input buffer overrun
On 9/23/15 1:47 PM, Flávio Pontes wrote: Hi, I need to stream the 2 inputs from a DeckLink Duo card in FHD. So far I could compile ffmpeg with decklink support and capture one input at a time without problems. When I try to capture from the 2 inputs I start getting the following message repeatedly on both terminals: [decklink @ 0x32287e0] Decklink input buffer overrun! I use these commands in the terminals: /opt/ffmpeg_build/bin/ffmpeg -f decklink -re -i 'DeckLink SDI (1)@7' -g 90 -s hd1080 -profile:v baseline -pix_fmt yuv420p -preset veryfast -c:v libx264 -b:v 2000k -bufsize 1400k -minrate 2000k -maxrate 2000k -c:a libfdk_aac -b:a 96k -threads 4 -f flv rtmp://10.12.20.62/myapp/tvines /opt/ffmpeg_build/bin/ffmpeg -f decklink -re -i 'DeckLink SDI (2)@7' -g 90 -s hd1080 -profile:v baseline -pix_fmt yuv420p -preset veryfast -c:v libx264 -b:v 2000k -bufsize 1400k -minrate 2000k -maxrate 2000k -c:a libfdk_aac -b:a 96k -threads 4 -f flv rtmp://10.12.20.62/myapp/tvescola I tried many combinations but couldn't get it to work. I wonder if the input buffer is shared between the 2 inputs. Is there any way to set an input buffer size independently for each command? Thx in advance! Flávio. My first thought was you are simply not keeping up. Are you positive your computer can process these two streams at the same time? I've not tried to capture from a Duo, but I've captured two streams simultaneously from a Quad using other software. Since you are running two independent copies of ffmpeg, I don't think there is a shared internal buffer to increase. Hopefully you will get some more useful advice from others. Deron ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] Export Subtitles live from HD-SDI
On 10/7/15 7:05 AM, florian.he...@rbb-online.de wrote: Hello everyone, i'm working on a project where we want to livestream one TV-Program with MPEG-DASH. There for we need seperate subtitles. The encoder gets the signal over HD-SDI wich supports embedded subtitles in Teletext with SMPTE 2031. Our capturecard will be a Blackmagic Decklink Duo wich supports VANC-Data. I know that ffmpeg can extract closed captions from files. But what about Livesignals like from SDI-Signals? Thank you, Florian I am currently trying to get the decklink card to output CEA-708 closed captioning, so I can tell you a little about what I know so far. The current driver does not support capture or playback of VANC/closed captioning. The capture does support 10bit as I recall, which is the first thing you would need just to be able to capture the VANC. I don't know anything about SMPTE 2031, but at the very least you would need to ask for the correct DID/SDID for SMPTE 2013 and then place that data in a Teletext stream. That would require a patch to the current ffmpeg. When/if I get this output to work, I would certainly be willing to try my hand at creating a patch for this. I wish someone out there had a decklink card that could capture live tv in North America. Having some real world data to look at would make creating the CEA 708 output easier... Deron ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] Export Subtitles live from HD-SDI
On 10/7/15 1:40 PM, Florian Hesse wrote: Are you sure its a problem by the driver? The Blackmagic productpage says VANC is supported and as far as i know they developed the drivers? There just seems to be a problem with some lines below the SDI switching points https://forum.blackmagicdesign.com/viewtopic.php?f=12&t=32029 Do you know if it works with dectek-cards and drivers? I'm not sure if you are asking me or someone else, but I'll do my best to answer. VANC is just a means by which Ancillary data can be sent in a SDI stream. That data can be all kinds of things, including audio, closed captioning, camera control, or private data between devices. The Blackmagik Decklink drivers support the capture and play of this data. The ffmpeg decklink driver sends no VANC data, makes no connections to a subtitle stream, and does not support 10bit YUV which using the VANC requires. It is not really a "problem", in that support has never been added. Not a bug, but a design choice. The choice of outgoing VANC lines for CC does not mater much. It only maters if you are capturing VANC and some other device is putting it on one of those lines mentioned in your link in which case it sounds like you would need the Decklink Extreme card. Do you know what line your device is putting the VANC on? Deron ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
[FFmpeg-user] Best way to stream to remote server
I'm outputting HLS (with a couple different bit rates) to a local drive for viewing. However, now I need to move it to a remote server with more bandwidth (but less processing power). I know I can use ftp from within ffmpeg to save to the remote network drive, and I could also use sshfs to do it (and would prefer something a bit more secure anyway) but I'm wondering if anyone has recommendation on most efficient way. Can sshfs keep up (yes, the bandwidth is not a problem), does ftp cause problems? Thanks for any thoughts, Deron ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
[FFmpeg-user] Loosing "sync" between HLS streams
I'm using ffmpeg to generate an HLS stream from a tv tuner capture card, and while it generates the sub-streams nicely, it seems that the two streams get out of sync pretty quickly for some stations. What I mean by that is the segments numbers (and the EXT-X-MEDIA-SEQUENCE) differ. When the iPhone decides to switch from one to the other, it will "hang" for several seconds before resuming. It seems (hard to reproduce because it almost always gets out of sync) shows that it doesn't do that when the EXT-X-MEDIA-SEQUENCE is the same between the sub-streams. The test command is: ffmpeg -f lavfi -i movie=/dev/dvb/adapter3/dvr0:f=mpegts:s=dv+da[out0+subcc][out1] -force_key_frames expr:gte(t,n_forced*2) -c:v libx264 -vprofile baseline -vf scale=1280:720 -s 1280x720 -aspect 1280:720 -pix_fmt yuv420p -r 30 -preset veryfast -b:v 1200k -c:a libfaac -ar 48000 -ac 2 -async 2 -af volume=volume=11dB -flags:a -global_header -f hls -hls_time 2 -hls_list_size 10 -hls_wrap 2 -hls_flags delete_segments -hls_segment_filename /var/www/html/stream/wfgc/v.mid.%d.ts -y /var/www/html/stream/wfgc/v.mid.m3u8 -force_key_frames expr:gte(t,n_forced*2) -c:v libx264 -vprofile baseline -vf scale=640:360 -s 640x360 -aspect 640:360 -pix_fmt yuv420p -r 30 -preset veryfast -b:v 200k -c:a libfaac -b:a 50k -ar 24000 -ac 1 -async 2 -af volume=volume=11dB -flags:a -global_header -f hls -hls_time 2 -hls_list_size 10 -hls_wrap 2000 -hls_flags delete_segments -hls_segment_filename /var/www/html/stream/wfgc/v.portrait.%d.ts -y /var/www/html/stream/wfgc/v.portrait.m3u8 I can provide console output, but I've not seen anything useful after piles of output. I'm guessing that chopping into 2 second segments, then passing to be scale/encode in two sub-streams (instead of two sub-streams chopping into 2 second segments) would solve the problem but I don't know of any way to do that except write an libav* app? Any suggestions? Deron ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] Loosing "sync" between HLS streams
On 1/10/16 4:29 PM, Deron wrote:
I'm using ffmpeg to generate an HLS stream from a tv tuner capture
card, and while it generates the sub-streams nicely, it seems that the
two streams get out of sync pretty quickly for some stations. What I
mean by that is the segments numbers (and the EXT-X-MEDIA-SEQUENCE)
differ. When the iPhone decides to switch from one to the other, it
will "hang" for several seconds before resuming.
The test command is:
ffmpeg -f lavfi -i
movie=/dev/dvb/adapter3/dvr0:f=mpegts:s=dv+da[out0+subcc][out1]
-force_key_frames expr:gte(t,n_forced*2) -c:v libx264 -vprofile
baseline -vf scale=1280:720 -s 1280x720 -aspect 1280:720 -pix_fmt
yuv420p -r 30 -preset veryfast -b:v 1200k -c:a libfaac -ar 48000 -ac 2
-async 2 -af volume=volume=11dB -flags:a -global_header -f hls
-hls_time 2 -hls_list_size 10 -hls_wrap 2 -hls_flags
delete_segments -hls_segment_filename
/var/www/html/stream/wfgc/v.mid.%d.ts -y
/var/www/html/stream/wfgc/v.mid.m3u8 -force_key_frames
expr:gte(t,n_forced*2) -c:v libx264 -vprofile baseline -vf
scale=640:360 -s 640x360 -aspect 640:360 -pix_fmt yuv420p -r 30
-preset veryfast -b:v 200k -c:a libfaac -b:a 50k -ar 24000 -ac 1
-async 2 -af volume=volume=11dB -flags:a -global_header -f hls
-hls_time 2 -hls_list_size 10 -hls_wrap 2000 -hls_flags
delete_segments -hls_segment_filename
/var/www/html/stream/wfgc/v.portrait.%d.ts -y
/var/www/html/stream/wfgc/v.portrait.m3u8
I can provide console output, but I've not seen anything useful after
piles of output.
I'm guessing that chopping into 2 second segments, then passing to be
scale/encode in two sub-streams (instead of two sub-streams chopping
into 2 second segments) would solve the problem but I don't know of
any way to do that except write an libav* app?
Any suggestions?
Deron
No replies, so maybe I can ask a more specific question that someone has
an answer for.
Right now, I presume the above command does something like this (and
that force_key_frames is for some reason choosing different places
between the two process paths)
input stream ---> [force_key_frames] [scale] etc [segment] -> output1
|--> [force_key_frames] [scale] etc [segment] -> output2
Is there some way to cause this instead:
input stream [force_key_frames] ---> [scale] etc [segment] -> output1
|--> [scale] etc [segment] -> output2
Or better yet:
input stream [force_key_frames] [segment] ---> [scale] etc -> output1
|--> [scale] etc -> output2
?
Thanks!
Deron
--
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KD0AYO Western South Dakota
Dog Star Kennel http://www.dogstarkennel.com
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[FFmpeg-user] How to put SCTE-35 markers in a video
Hello, I see the latest patches for SCTE-35 insertion into HLS (even if the important parts are not applied yet), but how do I actually generate an mpegts with SCTE-35 markers? Thanks, Deron ___ 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] Need info on ATSC3 stream capture going wrong
Hello, this is a really strange one for me that is way outside my understanding. I've been using ffmpeg to stream tv shows and I've run across a TV station I can not capture! The symptoms are after 12:55 or so minutes the audio goes silent in the output, and shortly after that ffmpeg quits. This works with some 40 other tv channels! I had to bump up the loglevel to 90(!) to see something that might explain this. Obviously 13 minutes of loglevel 90 is a huge amount of output, so I'm hoping I have identified the important part and make it obviously enough that someone might have some guidance on what is wrong. What appears to happen is some audio timestamp drift error is noticed, compensation is applied that perhaps has a math sign error, a whole lot of silence is inserted, the incoming data is ignored because of that silence, and the input buffer is overflowed. Happy to provide complete uncut console data, but it is 50mb! Hopefully this is enough that someone can guide me to a solution! Thanks, Deron Command (without proper shell quoting because it is not issued from the shell): ffmpeg -loglevel 90 -f lavfi -i movie=/dev/dvb/adapter0/dvr0:f=mpegts:s=dv+da[out0+subcc][out1] -force_key_frames expr:gte(t,n_forced*2) -c:v libx264 -profile:v high -level:v 4.0 -vf scale=1280:720 -s 1280x720 -aspect 1280:720 -pix_fmt yuv420p -r 30 -preset veryfast -b:v 1200k -c:a libfdk_aac -cutoff 18000 -b:a 128k -ar 48000 -ac 2 -async 2 -flags:a -global_header -af volume=volume=10.000dB:precision=float -f hls -hls_time 2 -hls_list_size 10 -hls_wrap 99 -hls_flags delete_segments -hls_segment_filename /var/www/html/stream/v.mid.%06d.ts -y /var/www/html/stream/v.mid.m3u8 -sn -force_key_frames expr:gte(t,n_forced*2) -c:v libx264 -profile:v high -level:v 4.0 -vf scale=640:360 -s 640x360 -aspect 640:360 -pix_fmt yuv420p -r 30 -preset veryfast -b:v 200k -c:a libfdk_aac -profile:a aac_he -cutoff 18000 -b:a 96k -ar 44100 -ac 1 -flags:a -global_header -af volume=volume=10.000dB:precision=float -f hls -hls_time 2 -hls_list_size 10 -hls_wrap 99 -hls_flags delete_segments -hls_segment_filename /var/www/html/stream/v.portrait.%06d.ts -y /var/www/html/stream/v.portrait.m3u8 -sn -vn -c:a libfdk_aac -profile:a aac_he -cutoff 18000 -b:a 96k -ar 44100 -ac 2 -flags:a -global_header -af volume=volume=10.000dB:precision=float -f hls -hls_time 2 -hls_list_size 10 -hls_wrap 99 -hls_flags delete_segments -hls_segment_filename /var/www/html/stream/a.hi.%06d.ts -y /var/www/html/stream/a.hi.m3u8 -sn -vn -c:a libfdk_aac -profile:a aac_he -cutoff 18000 -b:a 60k -ar 32000 -ac 1 -flags:a -global_header -af volume=volume=10.000dB:precision=float -f hls -hls_time 2 -hls_list_size 10 -hls_wrap 99 -hls_flags delete_segments -hls_segment_filename /var/www/html/stream/a.low.%06d.ts -y /var/www/html/stream/a.low.m3u8 -vf scale=426:240 -s 426x240 -aspect 426:240 -f image2 -vf fps=fps=2 -y /var/www/html/stream/t.%06d.jpg -vn -map a:0 -c:a pcm_s16le -ar 48000 -ac 2 -f stream_segment -segment_time 2 -segment_format s16le -y /var/www/html/stream/a.loudness.%06d.pcm ffmpeg version N-93562-g3e10223385 Copyright (c) 2000-2019 the FFmpeg developers built with gcc 7 (Ubuntu 7.2.0-8ubuntu3) configuration: --enable-shared --enable-gpl --enable-nonfree --enable-libxvid --enable-libx264 --enable-libmp3lame --enable-libvorbis --enable-libfdk-aac --disable-stripping libavutil 56. 26.100 / 56. 26.100 libavcodec 58. 50.100 / 58. 50.100 libavformat 58. 27.102 / 58. 27.102 libavdevice 58. 7.100 / 58. 7.100 libavfilter 7. 48.100 / 7. 48.100 libswscale 5. 4.100 / 5. 4.100 libswresample 3. 4.100 / 3. 4.100 libpostproc 55. 4.100 / 55. 4.100 < ... snip ...> [graph_1_aresample_in_0_1 @ 0x56452dc9ad00] [SWR @ 0x56452dc9b600] compensating audio timestamp drift:-0.001021 compensation:-2 in:48000 [graph_1_aresample_in_0_1 @ 0x56452dc9ad00] [SWR @ 0x56452dc9b600] Using fltp internally between filters [graph_3_aresample_in_0_1 @ 0x56452dc5e480] [SWR @ 0x56452ddb9600] compensating audio timestamp drift:-0.001021 compensation:-2 in:48000 [graph_3_aresample_in_0_1 @ 0x56452dc5e480] [SWR @ 0x56452ddb9600] Using fltp internally between filters [graph_4_aresample_in_0_1 @ 0x56452dda0380] [SWR @ 0x56452dda1080] compensating audio timestamp drift:-0.001021 compensation:-2 in:48000 [graph_4_aresample_in_0_1 @ 0x56452dda0380] [SWR @ 0x56452dda1080] Using fltp internally between filters [graph_5_aresample_in_0_1 @ 0x56452f0c9200] [SWR @ 0x56452f0c9c40] compensating audio timestamp drift:-0.001021 compensation:-2 in:48000 [graph_5_aresample_in_0_1 @ 0x56452f0c9200] [SWR @ 0x56452f0c9c40] Using fltp internally between filters [graph_7_aresample_in_0_1 @ 0x56452f18a740] [SWR @ 0x56452f18b180] compensating audio timestamp drift:-0.001021 compensation:-2 in:48000 [graph_7_aresample_in_0_1 @ 0x56452f18a740] [SWR @ 0x56452f18b180] Using fltp inter
Re: [FFmpeg-user] Need info on ATSC3 stream capture going wrong
On 4/14/19 10:42 AM, Carl Zwanzig wrote: On 4/14/2019 7:14 AM, Deron wrote: What appears to happen is some audio timestamp drift error is noticed, compensation is applied that perhaps has a math sign error, a whole lot of silence is inserted, the incoming data is ignored because of that silence, and the input buffer is overflowed. I wonder, and this is just a WAG, if that stream has enough uncorrected bit errors spread around that the demux/decoder can't figure out where the data blocks start and then picks a likely spot that's wrong. Can you pick out the error stats for different channels and compare them? z! Could errors in the data stream produce such repeatable results? I don't know how accurate azap's status is, but I see no uncorrected blocks and reported signal strength is perfect. status 1f | signal | snr 0130 | ber | unc | FE_HAS_LOCK status 1f | signal | snr 012a | ber | unc | FE_HAS_LOCK status 1f | signal | snr 012a | ber | unc | FE_HAS_LOCK Here is some other station's output that do not suffer this problem. status 1f | signal | snr 0143 | ber | unc | FE_HAS_LOCK status 1f | signal | snr 0143 | ber | unc | FE_HAS_LOCK status 1f | signal | snr 0147 | ber | unc | FE_HAS_LOCK status 1f | signal eb84 | snr 00e3 | ber | unc | FE_HAS_LOCK status 1f | signal eb84 | snr 00e8 | ber | unc | FE_HAS_LOCK status 1f | signal eb84 | snr 00e8 | ber | unc | FE_HAS_LOCK status 1f | signal | snr 0140 | ber | unc | FE_HAS_LOCK status 1f | signal | snr 0140 | ber | unc | FE_HAS_LOCK status 1f | signal | snr 0140 | ber | unc | FE_HAS_LOCK ... status 1f | signal | snr 0117 | ber | unc | FE_HAS_LOCK status 1f | signal | snr 0116 | ber | unc | FE_HAS_LOCK status 1f | signal | snr 0119 | ber | unc | FE_HAS_LOCK Thanks for your WAG. Anyone with another? Deron ___ 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] Need info on ATSC3 stream capture going wrong
Hello, this is a really strange one for me that is way outside my understanding. I've been using ffmpeg to stream tv shows and I've run across a TV station I can not capture! The symptoms are after 12:55 or so minutes the audio goes silent in the output, and shortly after that ffmpeg quits. This works with some 40 other tv channels! I've managed to narrow down my problem to -async. Put simply, these commands work: ffmpeg -i "/dev/dvb/adapter5/dvr0" -y out.ts ffmpeg -i "/dev/dvb/adapter5/dvr0" -async 1 -y out.ts and this does not: ffmpeg -i "/dev/dvb/adapter5/dvr0" -async 2000 -y out.ts And I get similar output from my previous report: [graph_1_aresample_in_0_0 @ 0x563ce00fa0c0] [SWR @ 0x563ce063a8c0] compensating audio timestamp drift:-0.001021 compensation:-49 in:48000 [graph_1_aresample_in_0_0 @ 0x563ce00fa0c0] [SWR @ 0x563ce063a8c0] Using fltp internally between filters [graph_1_aresample_in_0_0 @ 0x563ce00fa0c0] [SWR @ 0x563ce063a8c0] adding 16384 audio samples of silence [graph_1_aresample_in_0_0 @ 0x563ce00fa0c0] [SWR @ 0x563ce063a8c0] adding 15312 audio samples of silence /dev/dvb/adapter5/dvr0: Value too large for defined data type [out_0_1 @ 0x563ce00f9d40] EOF on sink link out_0_1:default. /dev/dvb/adapter5/dvr0: Value too large for defined data type [out_0_0 @ 0x563ce05928c0] EOF on sink link out_0_0:default. No more output streams to write to, finishing. Looking for some ideas on the nature of my problem. The log output that talks about "compensating audio timestamp drift" clearly gets it -49 value as number of samples needed to compensate for the drift (-.001021 seconds * 48000 samples/second = -49 samples). So why does it try to add 16384 + 15312 samples? Is that it can only compensate in a positive direction (it should be dropping 49 samples not inserting 49)? Thanks for any help, Deron ___ 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 generate h.264 raw data
Is it possible to output just raw h.264 data? Better yet, both AnnexB and AVCC? Or is there a simple muxer that I can use that won't be so hard to look through to see the h.264? I tried ffmpeg -i "test.avi" -c:v libx264 -bsf h264_mp4toannexb -f h264 -an -y "test.h264" But the data it generated did not look correct, as far as I understand it (which is what I am trying to learn, so my understanding is very limited!). Thanks, Deron ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] How to generate h.264 raw data
On 9/25/14 10:39 AM, Carl Eugen Hoyos wrote: Deron pagestream.org> writes: I tried ffmpeg -i "test.avi" -c:v libx264 -bsf h264_mp4toannexb -f h264 -an -y "test.h264" Why did you add h264_mp4toannexb? Did you try without? (Note that you do not have to specify -an, -f h264 and -vcodec libx264 if your output filename ends on ".h264".) Carl Eugen As you might expect, I tried lots of different options. Looking at the output again, it looks like my problem is interpreting the hex dump. It appears that it put it into 16 bit intel words which is why nothing looked right. 0100 6467 etc instead of 00 00 00 01 67 64 Which makes a whole lot more sense to me. Sorry for the noise, and thanks! Last question, is it possible to dump avcc/mp4 stream? The opposite of h264_mp4toannexb? Deron ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
[FFmpeg-user] Multithreaded multioutput problem
I would like to generate 4 different HLS (sub)streams from a single DVB capture, but ffmpeg fails to keep up with more than 2. Yes, I have the CPU power and I'm writing to a ram disk (or SSD, doesn't matter). I have a dual tuner capture card and can generate two different HLS substreams per tuner (total of 4) with plenty of CPU left to burn. I would guess it is some kind of thread limit but I can't seem to get past it. Any suggestions? Below is an example command and output. If I cut out one of the outputs, it works fine. Thanks for your suggestions! Deron ffmpeg -f mpegts -i /dev/dvb/adapter0/dvr0 -c:v libx264 -vf scale=1280:720 -s 1280x720 -aspect 1280:720 -pix_fmt yuv420p -r 29.97 -preset veryfast -b:v 2400k -c:a libfaac -ar 48000 -ac 2 -async 2 -f hls -hls_time 2 -hls_list_size 10 -hls_wrap 20 -y "/var/www/html/stream/WPBF/wpbf.hi.m3u8" -c:v libx264 -vf scale=1280:720 -s 1280x720 -aspect 1280:720 -pix_fmt yuv420p -r 29.97 -preset veryfast -b:v 1200k -c:a libfaac -ar 48000 -ac 2 -async 2 -f hls -hls_time 2 -hls_list_size 10 -hls_wrap 20 -y "/var/www/html/stream/WPBF/wpbf.mid.m3u8" -c:v libx264 -vf scale=1280:720 -s 1280x720 -aspect 1280:720 -pix_fmt yuv420p -r 29.97 -preset veryfast -b:v 600k -c:a libfaac -ar 48000 -ac 2 -async 2 -f hls -hls_time 2 -hls_list_size 10 -hls_wrap 20 -y "/var/www/html/stream/WPBF/wpbf.low.m3u8" -vn -c:a libfaac -b:a 30k -ar 24000 -ac 1 -f hls -hls_time 2 -hls_list_size 10 -hls_wrap 20 -y "/var/www/html/stream/WPBF/wpbf.a.m3u8" ffmpeg version N-69206-g4b46ce8 Copyright (c) 2000-2015 the FFmpeg developers built on Jan 22 2015 18:39:59 with gcc 4.8 (Ubuntu 4.8.2-19ubuntu1) configuration: --enable-shared --enable-gpl --enable-nonfree --enable-libxvid --enable-libx264 --enable-libmp3lame --enable-libvorbis --enable-libfaac --enable-libopenjpeg libavutil 54. 17.100 / 54. 17.100 libavcodec 56. 20.100 / 56. 20.100 libavformat56. 19.100 / 56. 19.100 libavdevice56. 4.100 / 56. 4.100 libavfilter 5. 8.101 / 5. 8.101 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 1.100 / 1. 1.100 libpostproc53. 3.100 / 53. 3.100 [mpeg2video @ 0x1c430a0] Invalid frame dimensions 0x0. Input #0, mpegts, from '/dev/dvb/adapter0/dvr0': Duration: N/A, start: 94543.371178, bitrate: N/A Stream #0:0[0x31]: Video: mpeg2video (Main), yuv420p(tv), 1920x1080 [SAR 1:1 DAR 16:9], max. 24000 kb/s, 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc Stream #0:1[0x34]: Audio: ac3, 48000 Hz, 5.1(side), fltp, 384 kb/s -async is forwarded to lavfi similarly to -af aresample=async=2:min_hard_comp=0.10:first_pts=0. Last message repeated 3 times [libx264 @ 0x1db48a0] using SAR=1/1 [libx264 @ 0x1db48a0] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX [libx264 @ 0x1db48a0] profile High, level 3.1 [libx264 @ 0x1c4b780] using SAR=1/1 [libx264 @ 0x1c4b780] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX [libx264 @ 0x1c4b780] profile High, level 3.1 [libx264 @ 0x1c4f400] using SAR=1/1 [libx264 @ 0x1c4f400] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX [libx264 @ 0x1c4f400] profile High, level 3.1 Output #0, hls, to '/var/www/html/stream/WPBF/wpbf.hi.m3u8': Metadata: encoder : Lavf56.19.100 Stream #0:0: Video: h264 (libx264), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], q=-1--1, 2400 kb/s, 29.97 fps, 90k tbn, 29.97 tbc Metadata: encoder : Lavc56.20.100 libx264 Stream #0:1: Audio: aac (libfaac), 48000 Hz, stereo, s16, 128 kb/s Metadata: encoder : Lavc56.20.100 libfaac Output #1, hls, to '/var/www/html/stream/WPBF/wpbf.mid.m3u8': Metadata: encoder : Lavf56.19.100 Stream #1:0: Video: h264 (libx264), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], q=-1--1, 1200 kb/s, 29.97 fps, 90k tbn, 29.97 tbc Metadata: encoder : Lavc56.20.100 libx264 Stream #1:1: Audio: aac (libfaac), 48000 Hz, stereo, s16, 128 kb/s Metadata: encoder : Lavc56.20.100 libfaac Output #2, hls, to '/var/www/html/stream/WPBF/wpbf.low.m3u8': Metadata: encoder : Lavf56.19.100 Stream #2:0: Video: h264 (libx264), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], q=-1--1, 600 kb/s, 29.97 fps, 90k tbn, 29.97 tbc Metadata: encoder : Lavc56.20.100 libx264 Stream #2:1: Audio: aac (libfaac), 48000 Hz, stereo, s16, 128 kb/s Metadata: encoder : Lavc56.20.100 libfaac Output #3, hls, to '/var/www/html/stream/WPBF/wpbf.a.m3u8': Metadata: encoder : Lavf56.19.100 Stream #3:0: Audio: aac (libfaac), 24000 Hz, mono, s16, 30 kb/s Metadata: encoder : Lavc56.20.100 libfaac Stream mapping: Stream #0:0 -> #0:0 (mpeg2video (native) -> h264 (libx264)) Stream #0:1 -> #0:1 (ac3 (native) -> aac (libfaac)) Stream #0:0 -> #1:0 (mpeg2video (native) -> h264 (
Re: [FFmpeg-user] Multithreaded multioutput problem
On 1/23/15 9:28 AM, Deron wrote: I would like to generate 4 different HLS (sub)streams from a single DVB capture, but ffmpeg fails to keep up with more than 2. Yes, I have the CPU power and I'm writing to a ram disk (or SSD, doesn't matter). I have a dual tuner capture card and can generate two different HLS substreams per tuner (total of 4) with plenty of CPU left to burn. I would guess it is some kind of thread limit but I can't seem to get past it. Any suggestions? Below is an example command and output. If I cut out one of the outputs, it works fine. Thanks for your suggestions! Deron Reading this, I obviously did not make the problem very clear. I am trying to do this (simplified): ffmpeg -i "/dev/dvb/adapter0/dvr0" -b:v 2400k "hi.m3u8" -b:v 1200k "med.m3u8" -b:v 600k "low.m3u8" -vn "audio.m3u8" And it fails with: /dev/dvb/adapter0/dvr0: Value too large for defined data type But if I remove any 2 outputs like: ffmpeg -i "/dev/dvb/adapter0/dvr0" -b:v 2400k "hi.m3u8" -vn "audio.m3u8" It works. I can see fps is very close (not sure if it is a because of HLS working in "real time" or some thread limitation), but cpu usage is very light. Since I have 2 adapters, I can actually run ffmpeg -i "/dev/dvb/adapter0/dvr0" -b:v 2400k "hi.m3u8" -b:v 1200k "med.m3u8" ffmpeg -i "/dev/dvb/adapter1/dvr0" -b:v 600k "low.m3u8" -vn "audio.m3u8" And that runs fine (again with light cpu usage). Obviously a threading issue. What can I do to fix this? Thanks, Deron ffmpeg -f mpegts -i /dev/dvb/adapter0/dvr0 -c:v libx264 -vf scale=1280:720 -s 1280x720 -aspect 1280:720 -pix_fmt yuv420p -r 29.97 -preset veryfast -b:v 2400k -c:a libfaac -ar 48000 -ac 2 -async 2 -f hls -hls_time 2 -hls_list_size 10 -hls_wrap 20 -y "/var/www/html/stream/WPBF/wpbf.hi.m3u8" -c:v libx264 -vf scale=1280:720 -s 1280x720 -aspect 1280:720 -pix_fmt yuv420p -r 29.97 -preset veryfast -b:v 1200k -c:a libfaac -ar 48000 -ac 2 -async 2 -f hls -hls_time 2 -hls_list_size 10 -hls_wrap 20 -y "/var/www/html/stream/WPBF/wpbf.mid.m3u8" -c:v libx264 -vf scale=1280:720 -s 1280x720 -aspect 1280:720 -pix_fmt yuv420p -r 29.97 -preset veryfast -b:v 600k -c:a libfaac -ar 48000 -ac 2 -async 2 -f hls -hls_time 2 -hls_list_size 10 -hls_wrap 20 -y "/var/www/html/stream/WPBF/wpbf.low.m3u8" -vn -c:a libfaac -b:a 30k -ar 24000 -ac 1 -f hls -hls_time 2 -hls_list_size 10 -hls_wrap 20 -y "/var/www/html/stream/WPBF/wpbf.a.m3u8" ffmpeg version N-69206-g4b46ce8 Copyright (c) 2000-2015 the FFmpeg developers built on Jan 22 2015 18:39:59 with gcc 4.8 (Ubuntu 4.8.2-19ubuntu1) configuration: --enable-shared --enable-gpl --enable-nonfree --enable-libxvid --enable-libx264 --enable-libmp3lame --enable-libvorbis --enable-libfaac --enable-libopenjpeg libavutil 54. 17.100 / 54. 17.100 libavcodec 56. 20.100 / 56. 20.100 libavformat56. 19.100 / 56. 19.100 libavdevice56. 4.100 / 56. 4.100 libavfilter 5. 8.101 / 5. 8.101 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 1.100 / 1. 1.100 libpostproc53. 3.100 / 53. 3.100 [mpeg2video @ 0x1c430a0] Invalid frame dimensions 0x0. Input #0, mpegts, from '/dev/dvb/adapter0/dvr0': Duration: N/A, start: 94543.371178, bitrate: N/A Stream #0:0[0x31]: Video: mpeg2video (Main), yuv420p(tv), 1920x1080 [SAR 1:1 DAR 16:9], max. 24000 kb/s, 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc Stream #0:1[0x34]: Audio: ac3, 48000 Hz, 5.1(side), fltp, 384 kb/s -async is forwarded to lavfi similarly to -af aresample=async=2:min_hard_comp=0.10:first_pts=0. Last message repeated 3 times [libx264 @ 0x1db48a0] using SAR=1/1 [libx264 @ 0x1db48a0] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX [libx264 @ 0x1db48a0] profile High, level 3.1 [libx264 @ 0x1c4b780] using SAR=1/1 [libx264 @ 0x1c4b780] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX [libx264 @ 0x1c4b780] profile High, level 3.1 [libx264 @ 0x1c4f400] using SAR=1/1 [libx264 @ 0x1c4f400] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX [libx264 @ 0x1c4f400] profile High, level 3.1 Output #0, hls, to '/var/www/html/stream/WPBF/wpbf.hi.m3u8': Metadata: encoder : Lavf56.19.100 Stream #0:0: Video: h264 (libx264), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], q=-1--1, 2400 kb/s, 29.97 fps, 90k tbn, 29.97 tbc Metadata: encoder : Lavc56.20.100 libx264 Stream #0:1: Audio: aac (libfaac), 48000 Hz, stereo, s16, 128 kb/s Metadata: encoder : Lavc56.20.100 libfaac Output #1, hls, to '/var/www/html/stream/WPBF/wpbf.mid.m3u8': Metadata: encoder : Lavf56.19.100 Stream #1:0: Video: h264 (libx264), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], q=-1--1, 1200 kb/s, 29.97 fps, 90k tbn,
Re: [FFmpeg-user] Multithreaded multioutput problem
On 1/23/15 2:30 PM, Deron wrote: On 1/23/15 9:28 AM, Deron wrote: I would like to generate 4 different HLS (sub)streams from a single DVB capture, but ffmpeg fails to keep up with more than 2. Yes, I have the CPU power and I'm writing to a ram disk (or SSD, doesn't matter). I have a dual tuner capture card and can generate two different HLS substreams per tuner (total of 4) with plenty of CPU left to burn. I would guess it is some kind of thread limit but I can't seem to get past it. Any suggestions? Below is an example command and output. If I cut out one of the outputs, it works fine. Thanks for your suggestions! Deron Reading this, I obviously did not make the problem very clear. I am trying to do this (simplified): ffmpeg -i "/dev/dvb/adapter0/dvr0" -b:v 2400k "hi.m3u8" -b:v 1200k "med.m3u8" -b:v 600k "low.m3u8" -vn "audio.m3u8" And it fails with: /dev/dvb/adapter0/dvr0: Value too large for defined data type But if I remove any 2 outputs like: ffmpeg -i "/dev/dvb/adapter0/dvr0" -b:v 2400k "hi.m3u8" -vn "audio.m3u8" It works. I can see fps is very close (not sure if it is a because of HLS working in "real time" or some thread limitation), but cpu usage is very light. Since I have 2 adapters, I can actually run ffmpeg -i "/dev/dvb/adapter0/dvr0" -b:v 2400k "hi.m3u8" -b:v 1200k "med.m3u8" ffmpeg -i "/dev/dvb/adapter1/dvr0" -b:v 600k "low.m3u8" -vn "audio.m3u8" And that runs fine (again with light cpu usage). Obviously a threading issue. What can I do to fix this? To further complicate maters, I tried copying 10 minutes of data to a file using: cp /dev/dvb/adapter0/dvr0 data.ts and then ffmpeg -i data.ts -b:v 2400k "hi.m3u8" -b:v 1200k "med.m3u8" -b:v 600k "low.m3u8" -vn "audio.m3u8" Which works. In fact, I can run 8 copies of this at the same time and it still runs faster than real time on this computer. where ffmpeg -i "/dev/dvb/adapter0/dvr0" -b:v 2400k "hi.m3u8" -b:v 1200k "med.m3u8" -b:v 600k "low.m3u8" -vn "audio.m3u8" and ffmpeg -i "/dev/dvb/adapter0/dvr0" -b:v 2400k "hi.m3u8" -b:v 1200k "med.m3u8" Fails as above (and below) with: /dev/dvb/adapter0/dvr0: Value too large for defined data type and ffmpeg -i "/dev/dvb/adapter0/dvr0" -b:v 2400k "hi.m3u8" -vn "audio.m3u8" Works. Does anyone have an idea on what could be causing this grief? What is kicking out this message? Could it be simply that ffmpeg can't get going fast enough and the dvb device is loosing initial data?? Ie, the first frame takes too long? Thanks, Deron Thanks, Deron ffmpeg -f mpegts -i /dev/dvb/adapter0/dvr0 -c:v libx264 -vf scale=1280:720 -s 1280x720 -aspect 1280:720 -pix_fmt yuv420p -r 29.97 -preset veryfast -b:v 2400k -c:a libfaac -ar 48000 -ac 2 -async 2 -f hls -hls_time 2 -hls_list_size 10 -hls_wrap 20 -y "/var/www/html/stream/WPBF/wpbf.hi.m3u8" -c:v libx264 -vf scale=1280:720 -s 1280x720 -aspect 1280:720 -pix_fmt yuv420p -r 29.97 -preset veryfast -b:v 1200k -c:a libfaac -ar 48000 -ac 2 -async 2 -f hls -hls_time 2 -hls_list_size 10 -hls_wrap 20 -y "/var/www/html/stream/WPBF/wpbf.mid.m3u8" -c:v libx264 -vf scale=1280:720 -s 1280x720 -aspect 1280:720 -pix_fmt yuv420p -r 29.97 -preset veryfast -b:v 600k -c:a libfaac -ar 48000 -ac 2 -async 2 -f hls -hls_time 2 -hls_list_size 10 -hls_wrap 20 -y "/var/www/html/stream/WPBF/wpbf.low.m3u8" -vn -c:a libfaac -b:a 30k -ar 24000 -ac 1 -f hls -hls_time 2 -hls_list_size 10 -hls_wrap 20 -y "/var/www/html/stream/WPBF/wpbf.a.m3u8" ffmpeg version N-69206-g4b46ce8 Copyright (c) 2000-2015 the FFmpeg developers built on Jan 22 2015 18:39:59 with gcc 4.8 (Ubuntu 4.8.2-19ubuntu1) configuration: --enable-shared --enable-gpl --enable-nonfree --enable-libxvid --enable-libx264 --enable-libmp3lame --enable-libvorbis --enable-libfaac --enable-libopenjpeg libavutil 54. 17.100 / 54. 17.100 libavcodec 56. 20.100 / 56. 20.100 libavformat56. 19.100 / 56. 19.100 libavdevice56. 4.100 / 56. 4.100 libavfilter 5. 8.101 / 5. 8.101 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 1.100 / 1. 1.100 libpostproc53. 3.100 / 53. 3.100 [mpeg2video @ 0x1c430a0] Invalid frame dimensions 0x0. Input #0, mpegts, from '/dev/dvb/adapter0/dvr0': Duration: N/A, start: 94543.371178, bitrate: N/A Stream #0:0[0x31]: Video: mpeg2video (Main), yuv420p(tv), 1920x1080 [SAR 1:1 DAR 16:9], max. 24000 kb/s, 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc Stream #0:1[0x34]: Audio: ac3, 48000 Hz, 5.1(side), fltp, 384 kb/s -async is forwarded to lavfi similarly to -af aresample=async=2:min_hard_comp=0.10
[FFmpeg-user] Troubles segmenting WebVTT
I'm trying to capture ATSC with a tuner card and stream it out via HLS. I've got it more or less working, and now I would like to try my hand and streaming out closed captioning in WebVTT utilizing the eia 608 demuxer. First, I captured 10 minutes of video to my HD. Then I used: ffmpeg -f lavfi -i "movie=WPEC.ts[out0+subcc]" -map s WPEC.vtt which generated a good .vtt file and then ffmpeg -i WPEC.vtt -f segment -segment_time 10 -segment_format webvtt -scodec copy WPEC-%05d.vtt to generate segments 0 to 00038 and it works just fine. However, ffmpeg -f lavfi -i "movie=WPEC.ts[out0+subcc]" -map s -f segment -segment_time 10 -segment_format webvtt WPEC-%05d.vtt fails to generate segments. What it generates is just one segment named WPEC-0.vtt. What am I doing wrong? Deron Complete uncut output: # ffmpeg -f lavfi -i "movie=WPEC.ts[out0+subcc]" -map s -f segment -segment_time 10 -segment_format webvtt WPEC-%05d.vtt ffmpeg version N-69558-g7f02fcd Copyright (c) 2000-2015 the FFmpeg developers built with gcc 4.9.1 (Ubuntu 4.9.1-16ubuntu6) configuration: --enable-shared --enable-gpl --enable-nonfree --enable-libxvid --enable-libx264 --enable-libmp3lame --enable-libvorbis --enable-libfaac --enable-libopenjpeg libavutil 54. 18.100 / 54. 18.100 libavcodec 56. 21.102 / 56. 21.102 libavformat56. 19.100 / 56. 19.100 libavdevice56. 4.100 / 56. 4.100 libavfilter 5. 9.103 / 5. 9.103 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 1.100 / 1. 1.100 libpostproc53. 3.100 / 53. 3.100 [mpeg2video @ 0x1961bc0] Invalid frame dimensions 0x0. Last message repeated 24 times [mpegts @ 0x195eb80] PES packet size mismatch Input #0, lavfi, from 'movie=WPEC.ts[out0+subcc]': Duration: N/A, start: 16458.590289, bitrate: N/A Stream #0:0: Video: rawvideo (I420 / 0x30323449), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 29.97 fps, 29.97 tbr, 90k tbn, 90k tbc Stream #0:1: Subtitle: eia_608 Output #0, segment, to 'WPEC-%05d.vtt': Metadata: encoder : Lavf56.19.100 Stream #0:0: Subtitle: webvtt Metadata: encoder : Lavc56.21.102 webvtt Stream mapping: Stream #0:1 -> #0:0 (eia_608 (cc_dec) -> webvtt (native)) Press [q] to stop, [?] for help [mpeg2video @ 0x1961bc0] invalid cbp -1 at 113 46 [mpeg2video @ 0x1961bc0] Warning MVs not available [mpeg2video @ 0x1961bc0] concealing 2640 DC, 2640 AC, 2640 MV errors in B frame size=N/A time=00:06:23.48 bitrate=N/A video:0kB audio:0kB subtitle:2kB other streams:0kB global headers:0kB muxing overhead: unknown ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] Troubles segmenting WebVTT
On 2/10/15 3:20 AM, Stefano Sabatini wrote: On date Saturday 2015-02-07 22:15:58 -0700, Deron wrote: I'm trying to capture ATSC with a tuner card and stream it out via HLS. I've got it more or less working, and now I would like to try my hand and streaming out closed captioning in WebVTT utilizing the eia 608 demuxer. First, I captured 10 minutes of video to my HD. Then I used: ffmpeg -f lavfi -i "movie=WPEC.ts[out0+subcc]" -map s WPEC.vtt which generated a good .vtt file and then ffmpeg -i WPEC.vtt -f segment -segment_time 10 -segment_format webvtt -scodec copy WPEC-%05d.vtt to generate segments 0 to 00038 and it works just fine. However, ffmpeg -f lavfi -i "movie=WPEC.ts[out0+subcc]" -map s -f segment -segment_time 10 -segment_format webvtt WPEC-%05d.vtt fails to generate segments. What it generates is just one segment named WPEC-0.vtt. What am I doing wrong? Deron Complete uncut output: # ffmpeg -f lavfi -i "movie=WPEC.ts[out0+subcc]" -map s -f segment -segment_time 10 -segment_format webvtt WPEC-%05d.vtt Use -loglevel debug, segment will generate log messages showing the input PTSs. Also, are you sure you have many key-frames in your input file? I can't imagine how PTS or i-frames can be defined in a .vtt ? But I'm certainly looking for understanding! In a private email, Anshul Maheshwari suggested that it was incomplete in his WebVTT implementation and he hopes to get it resolved some day. I've started trying to just trace along to figure out what it is doing but it is a large amount of stuff to absorb at once. I wanted to learn more about HLS, so I'm not apposed to trying to solve the problem if it needs to be. Thanks for your input, Deron ffmpeg -loglevel debug -f lavfi -i "movie=WPEC.ts[out0+subcc]" -map s -f segment -segment_time 10 -segment_format webvtt WPEC-%05d.vtt ffmpeg version N-69558-g7f02fcd Copyright (c) 2000-2015 the FFmpeg developers built with gcc 4.9.1 (Ubuntu 4.9.1-16ubuntu6) configuration: --enable-shared --enable-gpl --enable-nonfree --enable-libxvid --enable-libx264 --enable-libmp3lame --enable-libvorbis --enable-libfaac --enable-libopenjpeg libavutil 54. 18.100 / 54. 18.100 libavcodec 56. 21.102 / 56. 21.102 libavformat56. 19.100 / 56. 19.100 libavdevice56. 4.100 / 56. 4.100 libavfilter 5. 9.103 / 5. 9.103 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 1.100 / 1. 1.100 libpostproc53. 3.100 / 53. 3.100 Splitting the commandline. Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) with argument 'debug'. Reading option '-f' ... matched as option 'f' (force format) with argument 'lavfi'. Reading option '-i' ... matched as input file with argument 'movie=WPEC.ts[out0+subcc]'. Reading option '-map' ... matched as option 'map' (set input stream mapping) with argument 's'. Reading option '-f' ... matched as option 'f' (force format) with argument 'segment'. Reading option '-segment_time' ... matched as AVOption 'segment_time' with argument '10'. Reading option '-segment_format' ... matched as AVOption 'segment_format' with argument 'webvtt'. Reading option 'WPEC-%05d.vtt' ... matched as output file. Finished splitting the commandline. Parsing a group of options: global . Applying option loglevel (set logging level) with argument debug. Successfully parsed a group of options. Parsing a group of options: input file movie=WPEC.ts[out0+subcc]. Applying option f (force format) with argument lavfi. Successfully parsed a group of options. Opening an input file: movie=WPEC.ts[out0+subcc]. detected 32 logical cores [Parsed_movie_0 @ 0xf4a8e0] Setting 'filename' to value 'WPEC.ts' [mpegts @ 0xf4ab80] Format mpegts probed with size=2048 and score=100 [mpegts @ 0xf4ab80] Before avformat_find_stream_info() pos: 0 bytes read:5013504 seeks:1 [mpegts @ 0xf4ab80] stream=0 stream_type=0 pid=31 prog_reg_desc= [mpegts @ 0xf4ab80] stream=1 stream_type=0 pid=34 prog_reg_desc= [mpegts @ 0xf4ab80] probing stream 0 pp:2500 [mpegts @ 0xf4ab80] Probe with size=56014, packets=1 detected ico with score=25 [mpegts @ 0xf4ab80] probing stream 0 pp:2499 [mpegts @ 0xf4ab80] Probe with size=112027, packets=2 detected ico with score=25 [mpegts @ 0xf4ab80] probing stream 0 pp:2498 [mpegts @ 0xf4ab80] Probe with size=168053, packets=3 detected ico with score=25 [mpegts @ 0xf4ab80] probing stream 0 pp:2497 [mpegts @ 0xf4ab80] probing stream 1 pp:2500 [mpegts @ 0xf4ab80] Probe with size=7680, packets=1 detected ac3 with score=51 [mpegts @ 0xf4ab80] probed stream 1 [mpegts @ 0xf4ab80] probing stream 0 pp:2496 [mpegts @ 0xf4ab80] Probe with size=280096, packets=5 detected
[FFmpeg-user] How to get only audio from "rawvideo" stream
I'm trying to generate a number of HLS streams from a single source including using the new webvtt for cc. The following command: ffmpeg -i test.ts -vn -f hls -hls_segment_filename "a.%d.ts" -y "a.m3u8" Works as expected, but ffmpeg -f lavfi -i "movie=test.ts[out0+subcc]" -vn -sn -f hls -hls_segment_filename "a.%d.ts" -y "a.m3u8" Fails with: ffmpeg version N-70148-g418be7c Copyright (c) 2000-2015 the FFmpeg developers built with gcc 4.8 (Ubuntu 4.8.2-19ubuntu1) configuration: --enable-shared --enable-gpl --enable-nonfree --enable-libxvid --enable-libx264 --enable-libmp3lame --enable-libvorbis --enable-libfaac --enable-libopenjpeg libavutil 54. 19.100 / 54. 19.100 libavcodec 56. 26.100 / 56. 26.100 libavformat56. 23.104 / 56. 23.104 libavdevice56. 4.100 / 56. 4.100 libavfilter 5. 11.101 / 5. 11.101 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 1.100 / 1. 1.100 libpostproc53. 3.100 / 53. 3.100 [mpeg2video @ 0x1c777a0] Invalid frame dimensions 0x0. Last message repeated 35 times Input #0, lavfi, from 'movie=test.ts[out0+subcc]': Duration: N/A, start: 9602.931211, bitrate: N/A Stream #0:0: Video: rawvideo (I420 / 0x30323449), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 29.97 fps, 29.97 tbr, 90k tbn, 90k tbc Stream #0:1: Subtitle: eia_608 Output #0, hls, to 'a.m3u8': Output file #0 does not contain any stream I get that I surpressed the video stream, and the audio is merged with it. How do I get it out without changing the input? I ultimately need to generate multiple streams including cc from a single input, so I can't remove cc from the input. Thanks, Deron ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] How to get only audio from "rawvideo" stream
On 2/25/15 11:00 AM, Deron wrote: I'm trying to generate a number of HLS streams from a single source including using the new webvtt for cc. The following command: ffmpeg -i test.ts -vn -f hls -hls_segment_filename "a.%d.ts" -y "a.m3u8" Works as expected, but ffmpeg -f lavfi -i "movie=test.ts[out0+subcc]" -vn -sn -f hls -hls_segment_filename "a.%d.ts" -y "a.m3u8" Fails with: Input #0, lavfi, from 'movie=test.ts[out0+subcc]': Duration: N/A, start: 9602.931211, bitrate: N/A Stream #0:0: Video: rawvideo (I420 / 0x30323449), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 29.97 fps, 29.97 tbr, 90k tbn, 90k tbc Stream #0:1: Subtitle: eia_608 Output #0, hls, to 'a.m3u8': Output file #0 does not contain any stream I get that I surpressed the video stream, and the audio is merged with it. How do I get it out without changing the input? I ultimately need to generate multiple streams including cc from a single input, so I can't remove cc from the input. Thanks, Deron After playing with this for several more hours, I'm no closer than before. Simplified, what I need is to turn: ffmpeg -f lavfi -i "movie=test.ts[out0+subcc]" ... Stream #0:0: Video: rawvideo (I420 / 0x30323449), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 29.97 fps, 29.97 tbr, 90k tbn, 90k tbc Stream #0:1: Subtitle: eia_608 Into something like this: Stream #0:0[0x31]: Video: mpeg2video (Main), yuv420p(tv), 1920x1080 [SAR 1:1 DAR 16:9], max. 24000 kb/s, 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc Stream #0:1[0x34]: Audio: ac3, 48000 Hz, 5.1(side), fltp, 384 kb/s Stream #0:2: Subtitle: eia_608 Is there some filter or method to break that rawvideo stream into independent video and audio streams?? The ultimate goal is to have a single command that does something like: ffmpeg -f lavfi -i "movie=test.ts[out0+subcc]" parameters> "video.hi.m3u8" "video.low.m3u8" "audio.m3u8" Thanks, Deron ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] How to get only audio from "rawvideo" stream
On 2/25/15 3:58 PM, Deron wrote: On 2/25/15 11:00 AM, Deron wrote: I'm trying to generate a number of HLS streams from a single source including using the new webvtt for cc. The following command: ffmpeg -i test.ts -vn -f hls -hls_segment_filename "a.%d.ts" -y "a.m3u8" Works as expected, but ffmpeg -f lavfi -i "movie=test.ts[out0+subcc]" -vn -sn -f hls -hls_segment_filename "a.%d.ts" -y "a.m3u8" Fails with: Input #0, lavfi, from 'movie=test.ts[out0+subcc]': Duration: N/A, start: 9602.931211, bitrate: N/A Stream #0:0: Video: rawvideo (I420 / 0x30323449), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 29.97 fps, 29.97 tbr, 90k tbn, 90k tbc Stream #0:1: Subtitle: eia_608 Output #0, hls, to 'a.m3u8': Output file #0 does not contain any stream I get that I surpressed the video stream, and the audio is merged with it. How do I get it out without changing the input? I ultimately need to generate multiple streams including cc from a single input, so I can't remove cc from the input. Thanks, Deron Well, I decided that I didn't need the audio only stream if it was going to be this much trouble. As it turns out, someone could have saved me the aggravation and told me that the audio was not available at all and not "merged with the video" as I presumed. I thought it was a bit strange, but what do I know? I can't imagine why the audio would be dropped. As it turns out, "movie=test.ts[out0+subcc]" generates a rawvideo stream. No audio. So what happens to the audio, and how do I get it??? What good is subtitles if I loose the audio? Thanks, Deron ffmpeg -f lavfi -i "movie=test.ts[out0+subcc]" test.avi ffmpeg version N-70148-g418be7c Copyright (c) 2000-2015 the FFmpeg developers built with gcc 4.8 (Ubuntu 4.8.2-19ubuntu1) configuration: --enable-shared --enable-gpl --enable-nonfree --enable-libxvid --enable-libx264 --enable-libmp3lame --enable-libvorbis --enable-libfaac --enable-libopenjpeg --enable-librtmp --enable-libx265 --disable-stripping libavutil 54. 19.100 / 54. 19.100 libavcodec 56. 26.100 / 56. 26.100 libavformat56. 23.104 / 56. 23.104 libavdevice56. 4.100 / 56. 4.100 libavfilter 5. 11.101 / 5. 11.101 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 1.100 / 1. 1.100 libpostproc53. 3.100 / 53. 3.100 Input #0, lavfi, from 'movie=test.ts[out0+subcc]': Duration: N/A, start: 2.117867, bitrate: N/A Stream #0:0: Video: rawvideo (I420 / 0x30323449), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], 59.94 fps, 59.94 tbr, 90k tbn, 90k tbc Stream #0:1: Subtitle: eia_608 Output #0, avi, to 'test.avi': Metadata: ISFT: Lavf56.23.104 Stream #0:0: Video: mpeg4 (FMP4 / 0x34504D46), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], q=2-31, 200 kb/s, 59.94 fps, 59.94 tbn, 59.94 tbc Metadata: encoder : Lavc56.26.100 mpeg4 Stream mapping: Stream #0:0 -> #0:0 (rawvideo (native) -> mpeg4 (native)) Press [q] to stop, [?] for help frame=13086 fps=327 q=31.0 Lsize= 46145kB time=00:03:38.31 bitrate=1731.5kbits/s video:45826kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.695515% ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] How to get only audio from "rawvideo" stream
On 2/27/15 10:01 AM, Nicolas George wrote: Le nonidi 9 ventôse, an CCXXIII, Deron a écrit : >> As it turns out, "movie=test.ts[out0+subcc]" generates a rawvideo >> stream. No audio. >> >> So what happens to the audio, and how do I get it??? What good is >> subtitles if I loose the audio? > > Have you considered reading the fine documentation? No, I have not. I just type random letters on the keyboard until I get the results I want. Is that wrong? http://ffmpeg.org/ffmpeg-filters.html#movie-1 I don't know ffmpeg like the back of my hand, so what may be obvious to you is not to me. For those who might encounter this same problem, this is what I managed to decipher: "movie=test.ts:s=dv+da[out0+subcc][out1]" If a better/proper method exists, please let me know. The use of s= was relatively obvious and tried before, but the pads usage was not. Only by the assumption that is must be possible did I get it to work after randomly made choices. Thanks for your time (truly), Deron ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] Multithreaded multioutput problem
Another user has contacted me with the exact same problem hoping that I stumbled on a solution for the below problem. I did not, so I am adding some more extensive log output in hopes that someone might recongize the source of the problem. Thanks, Deron On 1/25/15 8:41 AM, Deron wrote: On 1/23/15 2:30 PM, Deron wrote: On 1/23/15 9:28 AM, Deron wrote: I would like to generate 4 different HLS (sub)streams from a single DVB capture, but ffmpeg fails to keep up with more than 2. Yes, I have the CPU power and I'm writing to a ram disk (or SSD, doesn't matter). I have a dual tuner capture card and can generate two different HLS substreams per tuner (total of 4) with plenty of CPU left to burn. I would guess it is some kind of thread limit but I can't seem to get past it. Any suggestions? Below is an example command and output. If I cut out one of the outputs, it works fine. Thanks for your suggestions! Deron Reading this, I obviously did not make the problem very clear. I am trying to do this (simplified): ffmpeg -i "/dev/dvb/adapter0/dvr0" -b:v 2400k "hi.m3u8" -b:v 1200k "med.m3u8" -b:v 600k "low.m3u8" -vn "audio.m3u8" And it fails with: /dev/dvb/adapter0/dvr0: Value too large for defined data type But if I remove any 2 outputs like: ffmpeg -i "/dev/dvb/adapter0/dvr0" -b:v 2400k "hi.m3u8" -vn "audio.m3u8" It works. I can see fps is very close (not sure if it is a because of HLS working in "real time" or some thread limitation), but cpu usage is very light. Since I have 2 adapters, I can actually run ffmpeg -i "/dev/dvb/adapter0/dvr0" -b:v 2400k "hi.m3u8" -b:v 1200k "med.m3u8" ffmpeg -i "/dev/dvb/adapter1/dvr0" -b:v 600k "low.m3u8" -vn "audio.m3u8" And that runs fine (again with light cpu usage). Obviously a threading issue. What can I do to fix this? To further complicate maters, I tried copying 10 minutes of data to a file using: cp /dev/dvb/adapter0/dvr0 data.ts and then ffmpeg -i data.ts -b:v 2400k "hi.m3u8" -b:v 1200k "med.m3u8" -b:v 600k "low.m3u8" -vn "audio.m3u8" Which works. In fact, I can run 8 copies of this at the same time and it still runs faster than real time on this computer. where ffmpeg -i "/dev/dvb/adapter0/dvr0" -b:v 2400k "hi.m3u8" -b:v 1200k "med.m3u8" -b:v 600k "low.m3u8" -vn "audio.m3u8" and ffmpeg -i "/dev/dvb/adapter0/dvr0" -b:v 2400k "hi.m3u8" -b:v 1200k "med.m3u8" Fails as above (and below) with: /dev/dvb/adapter0/dvr0: Value too large for defined data type and ffmpeg -i "/dev/dvb/adapter0/dvr0" -b:v 2400k "hi.m3u8" -vn "audio.m3u8" Works. Does anyone have an idea on what could be causing this grief? What is kicking out this message? Could it be simply that ffmpeg can't get going fast enough and the dvb device is loosing initial data?? Ie, the first frame takes too long? Thanks, Deron The below fails with (in about 15 seconds): movie=/dev/dvb/adapter6/dvr0:f=mpegts:s=dv+da[out0+subcc][out1]: Value too large for defined data type The data around the error is: libx264 @ 0x25d77a0] frame= 123 QP=14.82 NAL=2 Slice:P Poc:6 I:20 P:1028 SKIP:2552 size=5107 bytes *** 1 dup! [libx264 @ 0x25d77a0] scene cut at 174 Icost:2049115 Pcost:2006553 ratio:0.0208 bias:0.1387 gop:54 (imb:3186 pmb:168) [libx264 @ 0x246aca0] frame= 122 QP=21.41 NAL=2 Slice:P Poc:4 I:233 P:2646 SKIP:721 size=20496 bytes [libx264 @ 0x246aca0] frame= 123 QP=22.14 NAL=2 Slice:P Poc:6 I:26 P:541 SKIP:3033 size=2346 bytes movie=/dev/dvb/adapter6/dvr0:f=mpegts:s=dv+da[out0+subcc][out1]: Value too large for defined data type [libx264 @ 0x246aca0] scene cut at 174 Icost:2049115 Pcost:2006553 ratio:0.0208 bias:0.1387 gop:54 (imb:3186 pmb:168) [AVIOContext @ 0x27d36e0] Statistics: 0 seeks, 2 writeouts Any suggestions? Deron ffmpeg -loglevel 30 -f lavfi -i movie=/dev/dvb/adapter6/dvr0:f=mpegts:s=dv+da[out0+subcc][out1] -force_key_frames expr:gte\(t,n_forced*2\) -c:v libx264 -vf scale=1280:720 -s 1280x720 -aspect 1280:720 -pix_fmt yuv420p -r 59.94 -preset veryfast -b:v 1200k -c:a libfaac -ar 48000 -ac 2 -af volume=volume=11dB -flags:a -global_header -f hls -hls_time 2 -hls_allow_cache 0 -hls_list_size 10 -hls_wrap 2 -hls_flags delete_segments -hls_segment_filename /var/www/html/stream/test/v.mid.%d.ts -y /var/www/html/stream/test/v.mid.m3u8 -force_key_frames expr:gte\(t,n_forced*2\) -sn -c:v libx264 -vf scale=1280:720 -s 1280x720 -aspect 1280:720 -pix_fmt yuv420p -r 59.94 -preset veryfast -b:v 600k -c:a libfaac -ar 48000 -ac 2 -af volume=volume=11dB -flags:a -global_header -f hls -hls_time 2 -hls_allow_cache 0 -hls_list_size 10 -hls_wrap 2 -hls_flags delete_segments -hls_segment_filename /var/
Re: [FFmpeg-user] Multithreaded multioutput problem
On 4/20/15 1:48 PM, Marton Balint wrote: On Mon, 20 Apr 2015, Deron wrote: Another user has contacted me with the exact same problem hoping that I stumbled on a solution for the below problem. I did not, so I am adding some more extensive log output in hopes that someone might recongize the source of the problem. I have also experienced this problem. Depending on your use case you can face a similar situation with only one input and one output as well and a few cores. There is no buffering between the various stages of the ffmpeg encoder/filter/decoder pipeline, so even if the stages by themselves are multi threaded, you won't be able to scale up, because passing data between the stages is done in a single thread. Or at least that is what I think is going on. So as far as I know you can only scale up properly by running multiple ffmpeg instances. E.g: create a multicast and encode that. Regards, Marton What would be the best way to solve this then? I'm not sure I understand what you mean by multicast. Having multiple ffmpeg's decoding the same original mpegts input would be pretty slow. Is there some way to have ffmpeg output a multicast that is in a faster/raw format? Seems like more overhead and more ways for it to go wrong. Thanks for your input, Deron ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] Multithreaded multioutput problem
On 4/20/15 4:22 PM, Marton Balint wrote: On Mon, 20 Apr 2015, Deron wrote: On 4/20/15 1:48 PM, Marton Balint wrote: On Mon, 20 Apr 2015, Deron wrote: Another user has contacted me with the exact same problem hoping that I stumbled on a solution for the below problem. I did not, so I am adding some more extensive log output in hopes that someone might recongize the source of the problem. I have also experienced this problem. Depending on your use case you can face a similar situation with only one input and one output as well and a few cores. There is no buffering between the various stages of the ffmpeg encoder/filter/decoder pipeline, so even if the stages by themselves are multi threaded, you won't be able to scale up, because passing data between the stages is done in a single thread. Or at least that is what I think is going on. So as far as I know you can only scale up properly by running multiple ffmpeg instances. E.g: create a multicast and encode that. Regards, Marton What would be the best way to solve this then? I'm not sure I understand what you mean by multicast. Having multiple ffmpeg's decoding the same original mpegts input would be pretty slow. If that overhead is unacceptable, then I don't see any other way. You wrote you had plenty of free CPU :) Speaking of CPU and having been away from the problem for many weeks, I forget about the important part. The encoding process for the data runs at better than real time (nearly 2x) when pulling from a file instead of the /dvb device. So I don't think that this is thread bound for me but something else. I'm still missing a piece of the puzzle. Something else is causing me grief. Thanks for the help, Deron ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] Multithreaded multioutput problem
On 4/21/15 2:51 AM, Marton Balint wrote: On Mon, 20 Apr 2015, Deron wrote: On 4/20/15 4:22 PM, Marton Balint wrote: On Mon, 20 Apr 2015, Deron wrote: On 4/20/15 1:48 PM, Marton Balint wrote: On Mon, 20 Apr 2015, Deron wrote: Another user has contacted me with the exact same problem hoping that I stumbled on a solution for the below problem. I did not, so I am adding some more extensive log output in hopes that someone might recongize the source of the problem. I have also experienced this problem. Depending on your use case you can face a similar situation with only one input and one output as well and a few cores. There is no buffering between the various stages of the ffmpeg encoder/filter/decoder pipeline, so even if the stages by themselves are multi threaded, you won't be able to scale up, because passing data between the stages is done in a single thread. Or at least that is what I think is going on. So as far as I know you can only scale up properly by running multiple ffmpeg instances. E.g: create a multicast and encode that. Regards, Marton What would be the best way to solve this then? I'm not sure I understand what you mean by multicast. Having multiple ffmpeg's decoding the same original mpegts input would be pretty slow. If that overhead is unacceptable, then I don't see any other way. You wrote you had plenty of free CPU :) Speaking of CPU and having been away from the problem for many weeks, I forget about the important part. The encoding process for the data runs at better than real time (nearly 2x) when pulling from a file instead of the /dvb device. So I don't think that this is thread bound for me but something else. I'm still missing a piece of the puzzle. Something else is causing me grief. In that case have you tried using the threaded input read support of ffmpeg? You have to specify more than one inputs (add an extra dummy input source) and play with -thread_queue_size option. Regards, Ma I've resolved this problem and would like to report my solution. The error evidently was coming from some part of azap/dvb device. When I increased the buffer size as per http://panteltje.com/panteltje/dvd/ the problem went away. I can now capture and encode 6 channels simultaneously, outputting 2 video,1 audio, and images for each on this computer. I presume the problem was simply that setup time in ffmpeg increased enough that it (temporarily) exceeded real time encoding and overflowed the buffer between azap and ffmpeg. I've considering writing an azap-like input module for ffmpeg. I'm sure this is generally frowned on, but in this case I see a couple of advantages. One is obviously this problem with too small a buffer. The other is when interference causes the stream to be degraded, ffmpeg will start consuming huge amounts of cpu until it actually freezes up the computer. Since azap knows when the signal tanks, it would appear to me it could deal with the encoding issue if it had access into the encoder itself. Maybe some way exists already to deal with it, or ffmpeg should simply be made more robust to deal with this problem? Anyway, thanks for the help! Deron ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
[FFmpeg-user] Past duration too large
I am encoding live video streams and generally it works well. The most common error in the logs seems to be something like: Past duration 0.77 too large I can get hundreds of them in a row, or even thousands. Can anyone shed light on what conditions can actually causes this error to be generated? What does it mean? Thanks! Deron ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
[FFmpeg-user] decklink output crashing
I decided to give the built in decklink output a test today and about 2/3 of the files I try and play cause it to crash. I'm hoping I'm just missing some vital parameter. I've included uncut console output from running under gdb. Deron ffmpeg -i "/work/THE AUTO CHANNEL/ACTIVE SPOTS/spot Chrysler 200 Miggy 60.wmv" -vf scale=1280:720 -vol 19 -vsync 1 -c:v rawvideo -pix_fmt uyvy422 -s 1280x720 -r 6/1001 -c:a pcm_s16le -ar 48000 -ac 2 -f decklink "DeckLink HD Extreme 3D+" [Thread debugging using libthread_db enabled] Using host libthread_db library "/lib/x86_64-linux-gnu/libthread_db.so.1". ffmpeg version N-72309-gde0d3fe Copyright (c) 2000-2015 the FFmpeg developers built with gcc 4.8 (Ubuntu 4.8.2-19ubuntu1) configuration: --enable-shared --enable-gpl --enable-nonfree --enable-libxvid --enable-libx264 --enable-libmp3lame --enable-libvorbis --enable-libfaac --enable-libopenjpeg --enable-librtmp --enable-decklink --disable-stripping libavutil 54. 23.101 / 54. 23.101 libavcodec 56. 40.100 / 56. 40.100 libavformat56. 33.101 / 56. 33.101 libavdevice56. 4.100 / 56. 4.100 libavfilter 5. 16.101 / 5. 16.101 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 1.100 / 1. 1.100 libpostproc53. 3.100 / 53. 3.100 [wmv3 @ 0x653720] Extra data: 8 bits left, value: 0 Guessed Channel Layout for Input Stream #0.0 : stereo Input #0, asf, from '/work/THE AUTO CHANNEL/ACTIVE SPOTS/spot Chrysler 200 Miggy 60.wmv': Metadata: SfOriginalFPS : 299700 WMFSDKVersion : 12.0.7601.17514 WMFSDKNeeded: 0.0.0. IsVBR : 0 DeviceConformanceTemplate: MP@HL Duration: 00:01:00.03, start: 0.00, bitrate: 12984 kb/s Stream #0:0(eng): Audio: wmav2 (a[1][0][0] / 0x0161), 48000 Hz, 2 channels, fltp, 192 kb/s Stream #0:1(eng): Video: wmv3 (Main) (WMV3 / 0x33564D57), yuv420p, 1440x1080, 16585 kb/s, SAR 4:3 DAR 16:9, 29.97 fps, 29.97 tbr, 1k tbn, 1k tbc [New Thread 0x7230a700 (LWP 43163)] [New Thread 0x71b09700 (LWP 43164)] [New Thread 0x71308700 (LWP 43165)] [New Thread 0x70b07700 (LWP 43166)] [New Thread 0x70306700 (LWP 43167)] [New Thread 0x7fffefb05700 (LWP 43168)] [New Thread 0x7fffef304700 (LWP 43169)] [New Thread 0x7fffeeb03700 (LWP 43170)] [New Thread 0x7fffee302700 (LWP 43171)] [New Thread 0x7fffedb01700 (LWP 43172)] [New Thread 0x7fffed300700 (LWP 43173)] [New Thread 0x7fffecaff700 (LWP 43174)] [New Thread 0x7fffec2fe700 (LWP 43175)] [New Thread 0x7fffebafd700 (LWP 43176)] [New Thread 0x7fffeb2fc700 (LWP 43177)] [New Thread 0x7fffeaafb700 (LWP 43178)] [New Thread 0x7fffea2fa700 (LWP 43179)] [New Thread 0x7fffe9af9700 (LWP 43180)] [New Thread 0x7fffe92f8700 (LWP 43181)] [New Thread 0x7fffe8af7700 (LWP 43182)] [New Thread 0x7fffe82f6700 (LWP 43183)] [New Thread 0x7fffe7af5700 (LWP 43184)] [New Thread 0x7fffe72f4700 (LWP 43185)] [New Thread 0x7fffe6af3700 (LWP 43186)] [New Thread 0x7fffe62f2700 (LWP 43187)] [New Thread 0x7fffe5af1700 (LWP 43188)] [New Thread 0x7fffe52f0700 (LWP 43189)] [New Thread 0x7fffe4aef700 (LWP 43190)] [New Thread 0x7fffe42ee700 (LWP 43191)] [New Thread 0x7fffe3aed700 (LWP 43192)] [New Thread 0x7fffe32ec700 (LWP 43193)] [New Thread 0x7fffe2aeb700 (LWP 43194)] [New Thread 0x7fffe22ea700 (LWP 43195)] [New Thread 0x7fffe1ae9700 (LWP 43196)] [New Thread 0x7fffe12e8700 (LWP 43197)] [New Thread 0x7fffe0ae7700 (LWP 43198)] [New Thread 0x7fffe02e6700 (LWP 43199)] [New Thread 0x7fffdfae5700 (LWP 43200)] [New Thread 0x7fffdf2e4700 (LWP 43201)] [New Thread 0x7fffdeae3700 (LWP 43202)] [New Thread 0x7fffde2e2700 (LWP 43203)] [New Thread 0x7fffddae1700 (LWP 43204)] [New Thread 0x7fffdd2e0700 (LWP 43205)] [New Thread 0x7fffdcadf700 (LWP 43206)] [New Thread 0x7fffdc2de700 (LWP 43207)] [New Thread 0x7fffdbadd700 (LWP 43208)] [New Thread 0x7fffdb2dc700 (LWP 43209)] [New Thread 0x7fffdaadb700 (LWP 43210)] [New Thread 0x7fffda2da700 (LWP 43211)] [New Thread 0x7fffd9ad9700 (LWP 43212)] [New Thread 0x7fffd92d8700 (LWP 43213)] [New Thread 0x7fffd8ad7700 (LWP 43214)] [New Thread 0x7fffd82d6700 (LWP 43215)] [New Thread 0x7fffd7ad5700 (LWP 43216)] [New Thread 0x7fffd72d4700 (LWP 43217)] [New Thread 0x7fffd6ad3700 (LWP 43218)] [New Thread 0x7fffd62d2700 (LWP 43219)] [New Thread 0x7fffd5ad1700 (LWP 43220)] [New Thread 0x7fffd52d0700 (LWP 43221)] [New Thread 0x7fffd4acf700 (LWP 43222)] [New Thread 0x7fffd42ce700 (LWP 43223)] [New Thread 0x7fffd3acd700 (LWP 43224)] [New Thread 0x7fffd32cc700 (LWP 43225)] [New Thread 0x7fffd2acb700 (LWP 43226)] [New Thread 0x7fffd22ca700 (LWP 43227)] [New Thread 0x7fffd1ac9700 (LWP 43228)] -vol has been deprecated. Use the volume audio filter instead. -vol is forwarded to lavfi similarly to -af volume=0.074219. [wmv3 @ 0x67e1a0] Extra data: 8 bits left, value: 0 [New Thread 0x7fffd053a700 (LWP 43229)] [New Thread 0x7fffcfd39700 (LWP 43230)] [decklink @ 0x674cc0] Found Decklink mode 1280 x 720 with rat
