May 2020 - ffmpeg-user - lists.ffmpeg.org

Re: [FFmpeg-user] JPGs->video1 ; MOV -> Mp4 video2 ; Concatenate - video2 audio OK but video is slow . .
by Philip Rhoades 18 Mar '23

18 Mar '23

Carl, There has been some improvement - I will top-post for clarity but the slow motion video has disappeared and the only remaining problem is that the audio and video are out of sync: I stuck with concat because it alllows me to specify image files more conveniently in the concat_demuxer?.txt file: ffmpeg -f concat -i concat_demuxer1.txt -vf scale=1024:576 -r 30000/1001 output1.mp4 ffmpeg -f concat -i concat_demuxer2.txt -vf scale=1024:576 -r 30000/1001 output2.mp4 ffmpeg -f concat -i concat_demuxer3.txt -vf scale=1024:576 -r 30000/1001 output3.mp4 - see attached output. for i in 1 2 3 do ffmpeg -ar 44100 -acodec pcm_s16le -f s16le -ac 2 -i /dev/zero \ -i output${i}.mp4 \ -vcodec copy \ -acodec aac \ -shortest output${i}_silent.mp4 done - see attached output. #!/bin/sh ffmpeg -f concat -i inputs.txt -vcodec copy -acodec copy 2019-03-22_#1_XR_ClimateEmergencyDeclarationDayNSW_LoRes.mp4 - see attached output. On 2019-03-27 10:48, Carl Eugen Hoyos wrote: > 2019-03-27 0:23 GMT+01:00, Philip Rhoades <phil(a)pricom.com.au>: > >> On 2019-03-27 07:22, Carl Eugen Hoyos wrote: >>> 2019-03-26 21:15 GMT+01:00, Philip Rhoades <phil(a)pricom.com.au>: >>> >>>> I converted 21 JPGs to a video with: >>> >>> Usually, converting jpgs with the image2 input works better than >>> concat. >> >> Right - I will try that. > > This is probably not your issue but the concat demuxer > has the disadvantage that the console output is not > very helpful. > >>>> ffmpeg -f concat -i concat_demuxer1.txt -vf scale=1024:576 -r >>>> 29.97 >>> >>> This framerate is never correct, a possible frame rate is 30000/1001 >> >> Hmm . . I was just using what was reported for the frame for the MOVs >> . >> . so using: >> >> "-r 30000/1001" >> >> is OK? > > It is better if that is the framerate of the mov (as I assume). > >>> You can add the audio already in this step. >> >> I did have a few attempts at that but couldn't get it working so I >> resorted to a two-step process - I will have another look. >> >> >>>> output1.mp4 >>>> >>>> I then added a silent audio track with: >>>> >>>> ffmpeg -ar 44100 -acodec pcm_s16le -f s16le -ac 2 -channel_layout >>>> 2.1 >>> >>> The channel number and layout do not match. >> >> Should it be just "stereo"? > > Or drop channel_layout, for -ac 2 it should not be needed. > >>>> -i /dev/zero \ >>>> -i output${i}.mp4 \ >>>> -vcodec copy \ >>>> -acodec aac \ >>>> -shortest output${i}_silent.mp4 >>> >>>> I had another MOV file (fr=29.97) that I converted with: >>>> >>>> newfn=`basename $fn .MOV` >>>> ffmpeg -i $fn -vf scale=1024:576 ./${newfn}.mp4 >>>> >>>> When I concatenate them with (and other converted JPGs and MOVS on >>>> the >>>> end) using: >>>> >>>> ffmpeg -f concat -i inputs.txt -vcodec copy -acodec copy \ >>>> 2019-03-22_#1_XR_ClimateEmergencyDeclarationDayNSW_LoRes.mp4 >>>> >>>> the concat succeeds OK but when I play the video, the audio is OK >>>> but >>>> then when it gets to the converted MOV section, people are moving in >>>> slow motion . . >>>> >>>> Another odd thing, the first 21 converted JPGs at 4 sec each >>>> correctly >>>> shows 84 sec when viewed with mpv but when the converted MOV starts >>>> it >>>> jumps to ~3m 50s . . >>>> >>>> So there is at least one problem but maybe two? Can anyone tell me >>>> what >>>> I am missing? >>> >>> You may have to re-encode, feel free to provide the complete, uncut >>> console output. > > ..feel free to provide the complete, uncut console output for the > command line you already tried and also ffmpeg -i output for > the input files (as concat doesn't show it). > > Carl Eugen -- Philip Rhoades PO Box 896 Cowra NSW 2794 Australia E-mail: phil(a)pricom.com.au

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Recording audio from mic and speakers from commandline
by Mohan G 28 Sep '21

28 Sep '21

Hi, I am using ffmpeg version 2.4.7 on Fedora-21 workstation. I want to record audio, simultaneously from mic and speakers as separate streams. I am using the following command for it. ffmpeg -f alsa -i pulse -f alsa -i default -acodec libmp3lame -map 0:0 -map 1:0 outfile.mkv But after executing this command in terminal, I have to goto "Volume Control" settings in GUI, then goto "Recording" tab, change the "capture from" of one channel to "Built-in Audio Analog Stereo" (i.e., Mic) and the "capture from" of another channel to "Monitor of Built-in Audio Analog Stereo" (i.e., Speakers). But how can this be done completely from command-line, for automating it to use in scripts. - Mohan

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Audio out of sync after many Non-monotonous DTS in output stream warning
by Yang Zhang 02 Feb '21

02 Feb '21

when I transcoding some live stream that comes from DVB, I got many warnings bellow and after a few hours, the audio out of sync. [tee @ 0x2998780] Non-monotonous DTS in output stream 0:0; previous: 27931, current: 27929; changing to 27932. This may result in incorrect timestamps in the output file. [tee @ 0x2998780] Non-monotonous DTS in output stream 0:0; previous: 27935, current: 27933; changing to 27936. This may result in incorrect timestamps in the output file. And here is the ffmpeg command we used. Any suggestions for how to solve the audio out of sync issue? ffmpeg -vsync passthrough -copytb 1 -hwaccel cuvid -hwaccel_device 3 -probesize 10000000 -analyzeduration 10000000 -c:v h264_cuvid -gpu 3 -deint adaptive -drop_second_field 1 -fifo_size 713924 -overrun_nonfatal 1 -buffer_size 67108864 -i udp://127.0.0.1:48210 -filter_complex [0:i:0xab]split=outputs=1[map:v:0] -map [map:v:0] -c:v:0 h264_nvenc -g 250 -flags:v +global_header+cgop -preset:v medium -profile:v:0 high -gpu 3 -b:v:0 3500k -maxrate:v:0 5000k -bufsize:v:0 10000k -map 0:i:0x92 -c:a:0 aac -ac 2 -ar 44100 -b:a:0 128k -metadata service_name="LA 1 HD" -metadata service_provider="Movisrta+" -max_muxing_queue_size 512 -f tee [select=\'v:0,a:0\':bsfs/v=dump_extra=freq=keyframe:f=hls:hls_time=5:hls_list_size=10:hls_flags=delete_segments:var_stream_map=\'v:0,a:0\':hls_segment_filename=cache/hls /2/playlist_%v_%d.ts]cache/hls/2/index.m3u8

3 2

'showspectrumpic' filter: scaling of the frequency axis and fft windows size
by Wolfgang Hugemann 29 Sep '20

29 Sep '20

Today, I have been experimenting with the 'showspectrum' filter, mostly with 'showspectrumpic'. I think there is a bug in the log scaling of the frequency axis if (and only if) it doesn't start at zero: The ratios of the frequency axis' labelings is not constant, they grow with larger values. I had a quick look at the source code and gained the impression that the fft windows size is based on the height of video created (?). At least it cannot by specified in the filter options. Kind regards Wolfgang Hugemann

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Demuxing HLS stream of fmp4 segments
by Pedro Pedrosa 27 Aug '20

27 Aug '20

Hello all I'm trying to demux an HLS stream containing fmp4 segments. It works fine if I demux from the start of the input but as soon as I add the -ss flag to skip to a timecode it no longer produces any output. I have tested on my own ffmpeg-generated streams and also on a public file at https://bitmovin-a.akamaihd.net/content/MI201109210084_1/m3u8s-fmp4/f08e80d… Exact command that does not work: ffmpeg -ss 00:03:00 -i https://bitmovin-a.akamaihd.net/content/MI201109210084_1/m3u8s-fmp4/f08e80d… -y -frames 1 -f image2 test.jpg Usage that works (omit -ss): ffmpeg -i https://bitmovin-a.akamaihd.net/content/MI201109210084_1/m3u8s-fmp4/f08e80d… -y -frames 1 -f image2 test.jpg Any help would be greatly appreciated

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How to have a secure UDP BroadCast
by hassan shatnawi 19 Aug '20

19 Aug '20

Dears, How to have a secure connection with username and password while broadcasting "UPD" I've use this command ffmpeg -f dshow -video_size 1280x720 -rtbufsize 702000K -framerate 30 -i video="Video (00-0 Pro Capture Quad HDMI)" -r 30 -threads 4 -vcodec libx264 -crf 0 -preset ultrafast -f mpegts "udp://username:password@192.168.1.12:6666" but it didn't work can you advice me plz. Thanks in advance. *Hassan Shatnawi* Software Developer *Software Development Department* *Haupshy Establishment for Electronics* Al Madina Al Monawara St. bld 31 | P.O.Box 6875 Amman 11118, Jordan. *T: *+962 6 5820 927* | F: *+962 6 5820 926* | M: *+962 78 5792 096 *E*: hassan(a)haupshy.com *|* www.haupshy.com

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RTMP handshaking behind a proxy
by Verachten Bruno 25 Jun '20

25 Jun '20

Hi there, I unfortunately have to stream from behind a proxy. It works on one of my machines thanks to redsocks. The RTMP handshake goes well, and I can stream to twitch or youtube. On another machine, with the same redsocks configuration, the handshake never finishes, be it with youtube or twitch. The ffmpeg command is just a copy of an already h.264 encoded stream from rtmp to rtmp. My question is: is there a random port opened for the negotiation? Where should I look for the blocking of the handshake? Here is the command I have and the output: ffmpeg -i "$SOURCE" -c:v copy -c:a copy -f flv "$YOUTUBE_URL/$KEY" -loglevel debug [...] Splitting the commandline. Reading option '-i' ... matched as input url with argument 'rtmp:// 192.168.1.217:1935/live/stream15'. Reading option '-c:v' ... matched as option 'c' (codec name) with argument 'copy'. Reading option '-c:a' ... matched as option 'c' (codec name) with argument 'copy'. Reading option '-f' ... matched as option 'f' (force format) with argument 'flv'. Reading option 'rtmp://live-cdg.twitch.tv/app/live_of_myself' ... matched as output url. Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) with argument 'debug'. 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 url rtmp:// 192.168.1.217:1935/live/stream15. Successfully parsed a group of options. Opening an input file: rtmp://192.168.1.217:1935/live/stream15. [NULL @ 0xc401d0] Opening 'rtmp://192.168.1.217:1935/live/stream15' for reading [rtmp @ 0xc40990] No default whitelist set [tcp @ 0xc41010] No default whitelist set [tcp @ 0xc41010] Original list of addresses: [tcp @ 0xc41010] Address 192.168.1.217 port 1935 [tcp @ 0xc41010] Interleaved list of addresses: [tcp @ 0xc41010] Address 192.168.1.217 port 1935 [tcp @ 0xc41010] Starting connection attempt to 192.168.1.217 port 1935 [tcp @ 0xc41010] Successfully connected to 192.168.1.217 port 1935 [rtmp @ 0xc40990] Handshaking... [rtmp @ 0xc40990] Type answer 3 [rtmp @ 0xc40990] Server version 13.14.10.13 [rtmp @ 0xc40990] Proto = rtmp, path = /live/stream15, app = live, fname = stream15 [rtmp @ 0xc40990] Window acknowledgement size = 5000000 [rtmp @ 0xc40990] Max sent, unacked = 5000000 [rtmp @ 0xc40990] New incoming chunk size = 20480 [rtmp @ 0xc40990] Creating stream... [rtmp @ 0xc40990] Sending play command for 'stream15' [flv @ 0xc401d0] Format flv probed with size=2048 and score=100 [flv @ 0xc401d0] Before avformat_find_stream_info() pos: 13 bytes read:32868 seeks:0 nb_streams:0 [flv @ 0xc401d0] Unknown type |RtmpSampleAccess [NULL @ 0xc432a0] nal_unit_type: 7(SPS), nal_ref_idc: 3 [NULL @ 0xc432a0] nal_unit_type: 8(PPS), nal_ref_idc: 3 [NULL @ 0xc432a0] unknown SEI type 229 [h264 @ 0xc432a0] nal_unit_type: 7(SPS), nal_ref_idc: 3 [h264 @ 0xc432a0] nal_unit_type: 8(PPS), nal_ref_idc: 3 [h264 @ 0xc432a0] nal_unit_type: 7(SPS), nal_ref_idc: 3 [h264 @ 0xc432a0] nal_unit_type: 8(PPS), nal_ref_idc: 3 [h264 @ 0xc432a0] nal_unit_type: 6(SEI), nal_ref_idc: 0 [h264 @ 0xc432a0] nal_unit_type: 5(IDR), nal_ref_idc: 3 [h264 @ 0xc432a0] unknown SEI type 229 [h264 @ 0xc432a0] Format yuv420p chosen by get_format(). [h264 @ 0xc432a0] Reinit context to 1920x1088, pix_fmt: yuv420p [h264 @ 0xc432a0] nal_unit_type: 1(Coded slice of a non-IDR picture), nal_ref_idc: 3 Last message repeated 5 times [flv @ 0xc401d0] All info found [flv @ 0xc401d0] rfps: 29.750000 0.015408 [flv @ 0xc401d0] rfps: 29.833333 0.010802 [flv @ 0xc401d0] rfps: 29.916667 0.008256 [flv @ 0xc401d0] rfps: 30.000000 0.007768 Last message repeated 1 times [flv @ 0xc401d0] rfps: 29.970030 0.007707 Last message repeated 1 times [flv @ 0xc401d0] Setting avg frame rate based on r frame rate [flv @ 0xc401d0] After avformat_find_stream_info() pos: 66491 bytes read:66491 seeks:0 frames:105 Input #0, flv, from 'rtmp://192.168.1.217:1935/live/stream15': Duration: N/A, start: 9159.905000, bitrate: N/A Stream #0:0, 41, 1/1000: Video: h264 (High), 1 reference frame, yuv420p(progressive, left), 1920x1080 (1920x1088), 0/1, 29.97 fps, 29.97 tbr, 1k tbn Stream #0:1, 64, 1/1000: Audio: aac (LC), 48000 Hz, stereo, fltp Successfully opened the file. Parsing a group of options: output url rtmp:// live-cdg.twitch.tv/app/live_of_myself. Applying option c:v (codec name) with argument copy. Applying option c:a (codec name) with argument copy. Applying option f (force format) with argument flv. Successfully parsed a group of options. Opening an output file: rtmp://live-cdg.twitch.tv/app/live_of_myself. [rtmp @ 0xc528c0] No default whitelist set [tcp @ 0xcacff0] No default whitelist set [tcp @ 0xcacff0] Original list of addresses: [tcp @ 0xcacff0] Address 52.223.195.16 port 1935 [tcp @ 0xcacff0] Address 52.223.195.119 port 1935 [tcp @ 0xcacff0] Address 52.223.195.169 port 1935 [tcp @ 0xcacff0] Address 52.223.195.205 port 1935 [tcp @ 0xcacff0] Address 52.223.195.9 port 1935 [tcp @ 0xcacff0] Interleaved list of addresses: [tcp @ 0xcacff0] Address 52.223.195.16 port 1935 [tcp @ 0xcacff0] Address 52.223.195.119 port 1935 [tcp @ 0xcacff0] Address 52.223.195.169 port 1935 [tcp @ 0xcacff0] Address 52.223.195.205 port 1935 [tcp @ 0xcacff0] Address 52.223.195.9 port 1935 [tcp @ 0xcacff0] Starting connection attempt to 52.223.195.16 port 1935 [tcp @ 0xcacff0] Successfully connected to 52.223.195.16 port 1935 [rtmp @ 0xc528c0] Handshaking... and I never get any other information, until CTRL+C. [rtmp @ 0xc528c0] Cannot read RTMP handshake response rtmp://live-cdg.twitch.tv/app/live_of_myself: Immediate exit requested [AVIOContext @ 0xc42a50] Statistics: 66491 bytes read, 0 seeks [rtmp @ 0xc40990] Deleting stream... Exiting normally, received signal 2. On the other machine, I go until the end of the handshaking: Opening an output file: rtmp://live-cdg.twitch.tv/app/live_of_myself. [rtmp @ 0x55a25d70e880] No default whitelist set [tcp @ 0x55a25d5ce220] No default whitelist set [rtmp @ 0x55a25d70e880] Handshaking... [rtmp @ 0x55a25d70e880] Type answer 3 [rtmp @ 0x55a25d70e880] Server version 1.2.3.4 [rtmp @ 0x55a25d70e880] Proto = rtmp, path = /app/ live_of_myself , app = app, fname = live_of_myself [rtmp @ 0x55a25d70e880] Window acknowledgement size = 2500000 [rtmp @ 0x55a25d70e880] Max sent, unacked = 2500000 [rtmp @ 0x55a25d70e880] New incoming chunk size = 4096 [rtmp @ 0x55a25d70e880] Releasing stream... [rtmp @ 0x55a25d70e880] FCPublish stream... [rtmp @ 0x55a25d70e880] Creating stream... [rtmp @ 0x55a25d70e880] Sending publish command for ' live_of_myself ' Successfully opened the file. Thanks. -- Bruno Verachten

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FFmpeg single threaded bottleneck
by Gabriel Balaich 12 Jun '20

12 Jun '20

TLDR; FFmpeg seems to limit certain processes or aspects of a command to a single processing thread and it's limiting my ability to write / run complex commands. I'm trying to understand why that is the case and if there is a way to get around it. Hey there, about two years ago I inquired via this email list as to why I may be getting errors like this "real-time buffer [Capture Card] [video input] too full or near too full (62% of size: 2147480000 [rtbufsize parameter])! frame dropped!" constantly spammed in my console when trying to run large / complex FFmpeg commands with a multitude of inputs and outputs. I thought the issue may have been my inability to increase my buffer size passed max INT, but was told that that was likely not the issue. In the end, I ultimately accepted that commands this complex just weren't possible as I could never really pinpoint the issue. But recently I've very much wanted to record even more complex sources to multiple outputs via a single command, and I think I've come closer to finding what the issue is. For example, here is a stripped down version of the command I currently use: ffmpeg -y ` -thread_queue_size 9999 -indexmem 9999 -f dshow -rtbufsize 2147.48M -i audio="Analog (1+2) (RME Fireface UC)" ` -thread_queue_size 9999 -indexmem 9999 -f dshow -rtbufsize 2147.48M -i audio="ADAT (5+6) (RME Fireface UC)" ` -thread_queue_size 9999 -indexmem 9999 -f dshow -rtbufsize 2147.48M -video_size 1920x1080 -framerate 60 ` -pixel_format yuv420p -i video="Game Capture HD60 Pro (Video) (#01)":audio="Game Capture HD60 Pro (Audio) (#01)" ` -thread_queue_size 9999 -indexmem 9999 -f dshow -rtbufsize 2147.48M -video_size 3440x1440 -framerate 100 ` -pixel_format nv12 -i video="Video (00 Pro Capture HDMI 4K+)":audio="ADAT (3+4) (RME Fireface UC)" ` -thread_queue_size 9999 -indexmem 9999 -f dshow -rtbufsize 2147.48M -video_size 3840x2160 -framerate 60 ` -pixel_format nv12 -i video="AVerMedia HD Capture GC573 1":audio="SPDIF/ADAT (1+2) (RME Fireface UC)" ` -map 2:0,2:1 -map 2:1 -c:v h264_nvenc -preset: hp -r 60 -rc-lookahead 120 -pix_fmt yuv420p -b:v 100M -minrate 100M ` -maxrate 100M -bufsize 100M -c:a aac -ar 44100 -b:a 320k -af "aresample=async=250" ` -ss 00:00:02.482 -vsync 1 -max_muxing_queue_size 9999 C:\Users\gabri\Videos\FFmpeg\EL\EL.ts ` -map 3:0,3:1 -map 3:1 -c:v h264_nvenc -preset: hp -r 100 -rc-lookahead 200 -pix_fmt nv12 -b:v 288M -minrate 288M ` -maxrate 288M -bufsize 288M -c:a aac -ar 44100 -b:a 320k -af "atrim=0.034, asetpts=PTS-STARTPTS, aresample=async=250" ` -ss 00:00:01.904 -vsync 1 -max_muxing_queue_size 9999 C:\Users\gabri\Videos\FFmpeg\MW\MW.ts ` -c:v h264_nvenc -preset: hp -r 60 -rc-lookahead 120 -pix_fmt nv12 ` -b:v 288M -minrate 288M -maxrate 288M -bufsize 288M -c:a aac -ar 44100 -b:a 320k ` -filter_complex (('"[4:v]setpts=PTS-STARTPTS[v1]', '[4:a]atrim=1.615,asetpts=PTS-STARTPTS,aresample=async=250[a1]', '[0:a]atrim=4.879,asetpts=PTS-STARTPTS,aresample=async=250[a2]', '[1:a]atrim=4.199,asetpts=PTS-STARTPTS,aresample=async=250[a3]"') -join ';') ` -map "[v1]" -map "[a1]" -map "[a2]" -map "[a3]" ` -vsync 1 -max_muxing_queue_size 9999 C:\Users\gabri\Videos\FFmpeg\AM\AM%02d.ts Just to quickly explain I'm recording 3 video sources and 5 audio sources. For the video, I'm recording a camera (4k60), my main computer monitor (3440x1440@100FPS), and then a video game console (1920x1080p60, though I'd like to record 4k on this one too). And for the audio, I'm recording Discord (voice chat), my microphone EQd and compressed, my raw microphone at a lower gain (incase the treated one clips so I have something to fall back on), the audio from my desktop computer, and the audio from whatever game console I'm capturing at any given moment. The end result, I'm recording basically everything I'm doing at my setup simultaneously and synchronized for easy editing with something like Adobe Premiere in post, with the benefit of having every source recorded to its own output stream for individual modification. When I run this command I get the "real-time buffer [Capture Card] [video input] too full or near too full (62% of size: 2147480000 [rtbufsize parameter])! frame dropped!" spammed in my console when I start the command and when I end the command. But only at the beginning and end like I just described, for the most part, the command runs "normally" and in real-time, albeit just barely. Seeing as the error is pointing out that my buffer is too full I was under the impression the issue was the buffer (as previously stated), but upon further inspection, I can see that FFmpeg is completely saturating one of my processing threads when running the above command: [image: image.png] As a result, when I open something on the computer while running this command I drop frames, even something as simple as the calculator app, because anything that requests even a little bit of power from that thread takes it away from FFmpeg which is capping it out. Furthermore if I try to add further complexity to my command, like say another 4k capture card with a 4K input and output to the command (to replace the 1080p card) it just can't run in real-time, my RAM usage just keeps increasing until my RAM is completely saturated and my computer bluescreens. Inversely, if I run each output from the first command as a separate process in three seperate instances of powershell as opposed to all in one like this: ffmpeg -y ` -thread_queue_size 9999 -indexmem 9999 -f dshow -rtbufsize 2147.48M -i audio="Analog (1+2) (RME Fireface UC)" ` -thread_queue_size 9999 -indexmem 9999 -f dshow -rtbufsize 2147.48M -i audio="ADAT (5+6) (RME Fireface UC)" ` -thread_queue_size 9999 -indexmem 9999 -f dshow -rtbufsize 2147.48M -video_size 3840x2160 -framerate 60 ` -pixel_format nv12 -i video="AVerMedia HD Capture GC573 1":audio="SPDIF/ADAT (1+2) (RME Fireface UC)" ` -c:v h264_nvenc -preset: hp -r 60 -rc-lookahead 120 -pix_fmt nv12 ` -b:v 288M -minrate 288M -maxrate 288M -bufsize 288M -c:a aac -ar 44100 -b:a 320k ` -filter_complex (('"[2:v]setpts=PTS-STARTPTS[v1]', '[2:a]atrim=1.615,asetpts=PTS-STARTPTS,aresample=async=250[a1]', '[0:a]atrim=4.879,asetpts=PTS-STARTPTS,aresample=async=250[a2]', '[1:a]atrim=4.199,asetpts=PTS-STARTPTS,aresample=async=250[a3]"') -join ';') ` -map "[v1]" -map "[a1]" -map "[a2]" -map "[a3]" ` -vsync 1 -max_muxing_queue_size 9999 C:\Users\gabri\Videos\AM.ts ffmpeg -y ` -thread_queue_size 9999 -indexmem 9999 -f dshow -rtbufsize 2147.48M -video_size 1920x1080 -framerate 60 ` -pixel_format yuv420p -i video="Game Capture HD60 Pro (Video) (#01)":audio="Game Capture HD60 Pro (Audio) (#01)" ` -map 0:0,0:1 -map 0:1 -c:v h264_nvenc -preset: hp -r 60 -rc-lookahead 120 -pix_fmt yuv420p -b:v 100M -minrate 100M ` -maxrate 100M -bufsize 100M -c:a aac -ar 44100 -b:a 320k -af "aresample=async=250" ` -ss 00:00:02.482 -vsync 1 -max_muxing_queue_size 9999 C:\Users\gabri\Videos\EL.ts ffmpeg -y ` -thread_queue_size 9999 -indexmem 9999 -f dshow -rtbufsize 2147.48M -video_size 3440x1440 -framerate 100 ` -pixel_format nv12 -i video="Video (00 Pro Capture HDMI 4K+)":audio="ADAT (3+4) (RME Fireface UC)" ` -map 0:0,0:1 -map 0:1 -c:v h264_nvenc -preset: hp -r 100 -rc-lookahead 200 -pix_fmt nv12 -b:v 288M -minrate 288M ` -maxrate 288M -bufsize 288M -c:a aac -ar 44100 -b:a 320k -af "atrim=0.034, asetpts=PTS-STARTPTS, aresample=async=250" ` -ss 00:00:01.904 -vsync 1 -max_muxing_queue_size 9999 C:\Users\gabri\Videos\MW.ts But still, at the same time (all three commands started and ended within less than half a second), the load seems to be more evenly distributed with no core consistently over 60% usage: [image: image.png] Furthermore, there are no real-time buffer errors or dropped frames at all, all three commands start and end gracefully, and if I open another basic application while running these commands separately they carry on without a hiccup. So it would seem pretty obvious to me that the problem was never the buffer size, but that a process within a single instance of FFmpeg is limited to a single processing thread, regardless of how many inputs and outputs you have. Basically, from what I can tell, FFmpeg is incapable of scaling vertically. My plan is to upgrade my capture PC with a 16 core Threadripper (1950X) and another 4K capture card so I can capture from three 4k cards simultaneously with FFmpeg while simultaneously running an instance of OBS that is streaming to Twitch and Discord at the same time. The reason for this is so I can stream at 1080p with the scene functionality of OBS, but then record each source at its native resolution / FPS to separate streams using FFmpeg for easy / high-quality post-production editing, as opposed to just recording with OBS in-which each video source is baked into one stream at a much lower resolution and framerate. This way if I wanted to remove my camera from the from a clip, grab something from a scene that wasn't currently displaying in the active OBS scene, or full screen my camera / gameplay everything stays crisp because with FFmpeg each video source is being outputted to its own video stream at full quality. However, I don't think it's possible for me to do that with a single FFmpeg command with how FFmpeg is currently functioning in accordance with my testing. My only option is to run each command separately which on the surface seems like a fairly simple thing to do, the only problem is... getting them to start at the same time is no simple task. Even when using a separate script to start each command as their own processes programmatically I have a possible drift between each source of about half a second, as opposed less than a tenth of a second when running everything as one command. Basically, it's hard to keep things synchronized. On top of that, it's difficult to manage the output of each command through a single terminal, and trying to end each process gracefully with "q" or [ctr+c] is nigh impossible without dipping into an actual programming language that allows you to start a process or multiple processes while retaining the ability to send input to the STDIN. So really my main questions are: 1) What part of an FFmpeg process is limited to a single processing thread? 2) Why is that process limited to a single processing thread? 3) Is it possible to force the said process to use multiple processing threads? I've attached the full console output for the two above scenarios I provided earlier (one command vs separate commands), keep in mind that while the second scenario is using multiple FFmpeg processes they are still being run simultaneously, so each input and output is being run at the same time, just like the first command.

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SCTE-35 passthrough in transcoding process
by jamesed84.it＠gmail.com 12 Jun '20

12 Jun '20

Hello, I am transcoding cable sourced streams with an SCTE-35 data stream. The output seems to drop the SCTE-35 type and shows just bin_data on an ffprobe. I am wondering if there is a way to properly set that type on the output or preserve it. FFprobe of Source stream: hostname:~$ ffprobe udp://@239.1.1.12:59012 ffprobe version 4.2.1 Copyright (c) 2007-2019 the FFmpeg developers built with gcc 7 (Ubuntu 7.4.0-1ubuntu1~18.04.1) configuration: --prefix=/home/circle/ffmpeg_build --pkg-config-flags=--static --extra-cflags=-I/home/circle/ffmpeg_build/include --extra-ldflags=-L/home/circle/ffmpeg_build/lib --extra-libs='-lpthread -lm' --bindir=/home/circle/bin --enable-libsrt --enable-gpl --enable-libass --enable-libfdk-aac --enable-libfreetype --enable-libmp3lame --enable-libvorbis --enable-libx264 --enable-libx265 --enable-nonfree --enable-nvenc libavutil 56. 31.100 / 56. 31.100 libavcodec 58. 54.100 / 58. 54.100 libavformat 58. 29.100 / 58. 29.100 libavdevice 58. 8.100 / 58. 8.100 libavfilter 7. 57.100 / 7. 57.100 libswscale 5. 5.100 / 5. 5.100 libswresample 3. 5.100 / 3. 5.100 libpostproc 55. 5.100 / 55. 5.100 [mpeg2video @ 0x560dabb59040] Invalid frame dimensions 0x0. Last message repeated 28 times Input #0, mpegts, from 'udp://@239.1.1.12:59012': Duration: N/A, start: 10753.571022, bitrate: N/A Program 715 Stream #0:0[0x120]: Video: mpeg2video (Main) ([2][0][0][0] / 0x0002), yuv420p(tv, progressive), 1280x720 [SAR 1:1 DAR 16:9], Closed Captions, 59.94 fps, 59.94 tbr, 90k tbn, 119.88 tbc Stream #0:1[0x121](eng): Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, 5.1(side), fltp, 384 kb/s Stream #0:2[0x122](spa): Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, stereo, fltp, 192 kb/s Stream #0:3[0x140]: Data: scte_35 Unsupported codec with id 98305 for input stream 3 Transcoding command and process: hostname:~$ ffmpeg -hwaccel nvdec -i 'udp://@239.1.1.12:59012' -map 0 -vcodec h264_nvenc -b:v 1500K -s hd480 -acodec copy -f mpegts -metadata service_name="YES" -metadata service_provider="company" 'udp://@239.1.124.12:59012?pkt_size=1316' ffmpeg version 4.2.1 Copyright (c) 2000-2019 the FFmpeg developers built with gcc 7 (Ubuntu 7.4.0-1ubuntu1~18.04.1) configuration: --prefix=/home/circle/ffmpeg_build --pkg-config-flags=--static --extra-cflags=-I/home/circle/ffmpeg_build/include --extra-ldflags=-L/home/circle/ffmpeg_build/lib --extra-libs='-lpthread -lm' --bindir=/home/circle/bin --enable-libsrt --enable-gpl --enable-libass --enable-libfdk-aac --enable-libfreetype --enable-libmp3lame --enable-libvorbis --enable-libx264 --enable-libx265 --enable-nonfree --enable-nvenc libavutil 56. 31.100 / 56. 31.100 libavcodec 58. 54.100 / 58. 54.100 libavformat 58. 29.100 / 58. 29.100 libavdevice 58. 8.100 / 58. 8.100 libavfilter 7. 57.100 / 7. 57.100 libswscale 5. 5.100 / 5. 5.100 libswresample 3. 5.100 / 3. 5.100 libpostproc 55. 5.100 / 55. 5.100 [mpeg2video @ 0x556ef51bc800] Invalid frame dimensions 0x0. Last message repeated 10 times Input #0, mpegts, from 'udp://@239.1.1.12:59012': Duration: N/A, start: 10015.331022, bitrate: N/A Program 715 Stream #0:0[0x120]: Video: mpeg2video (Main) ([2][0][0][0] / 0x0002), yuv420p(tv, progressive), 1280x720 [SAR 1:1 DAR 16:9], Closed Captions, 59.94 fps, 59.94 tbr, 90k tbn, 119.88 tbc Stream #0:1[0x121](eng): Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, stereo, fltp, 192 kb/s Stream #0:2[0x122](spa): Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, stereo, fltp, 192 kb/s Stream #0:3[0x140]: Data: scte_35 Stream mapping: Stream #0:0 -> #0:0 (mpeg2video (native) -> h264 (h264_nvenc)) Stream #0:1 -> #0:1 (copy) Stream #0:2 -> #0:2 (copy) Stream #0:3 -> #0:3 (copy) Press [q] to stop, [?] for help Output #0, mpegts, to 'udp://@239.1.124.12:59012?pkt_size=1316': Metadata: service_name : YES service_provider: company encoder : Lavf58.29.100 Stream #0:0: Video: h264 (h264_nvenc) (Main), nv12, 852x480 [SAR 640:639 DAR 16:9], q=-1--1, 1500 kb/s, 59.94 fps, 90k tbn, 59.94 tbc Metadata: encoder : Lavc58.54.100 h264_nvenc Side data: cpb: bitrate max/min/avg: 0/0/1500000 buffer size: 3000000 vbv_delay: -1 Stream #0:1(eng): Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, stereo, fltp, 192 kb/s Stream #0:2(spa): Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, stereo, fltp, 192 kb/s Stream #0:3: Data: scte_35 FFPROBE output information: hostname:~$ ffprobe udp://@239.1.124.12:59012 ffprobe version 4.2.1 Copyright (c) 2007-2019 the FFmpeg developers built with gcc 7 (Ubuntu 7.4.0-1ubuntu1~18.04.1) configuration: --prefix=/home/circle/ffmpeg_build --pkg-config-flags=--static --extra-cflags=-I/home/circle/ffmpeg_build/include --extra-ldflags=-L/home/circle/ffmpeg_build/lib --extra-libs='-lpthread -lm' --bindir=/home/circle/bin --enable-libsrt --enable-gpl --enable-libass --enable-libfdk-aac --enable-libfreetype --enable-libmp3lame --enable-libvorbis --enable-libx264 --enable-libx265 --enable-nonfree --enable-nvenc libavutil 56. 31.100 / 56. 31.100 libavcodec 58. 54.100 / 58. 54.100 libavformat 58. 29.100 / 58. 29.100 libavdevice 58. 8.100 / 58. 8.100 libavfilter 7. 57.100 / 7. 57.100 libswscale 5. 5.100 / 5. 5.100 libswresample 3. 5.100 / 3. 5.100 libpostproc 55. 5.100 / 55. 5.100 Input #0, mpegts, from 'udp://@239.1.124.12:59012': Duration: N/A, start: 19.224000, bitrate: N/A Program 1 Metadata: service_name : YES service_provider: company Stream #0:0[0x100]: Video: h264 (Main) ([27][0][0][0] / 0x001B), yuv420p(progressive), 852x480 [SAR 640:639 DAR 16:9], 59.94 fps, 59.94 tbr, 90k tbn, 119.88 tbc Stream #0:1[0x101](eng): Audio: ac3 ([129][0][0][0] / 0x0081), 48000 Hz, stereo, fltp, 192 kb/s Stream #0:2[0x102](spa): Audio: ac3 ([129][0][0][0] / 0x0081), 48000 Hz, stereo, fltp, 192 kb/s Stream #0:3[0x103]: Data: bin_data ([6][0][0][0] / 0x0006) Unsupported codec with id 100359 for input stream 3

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how to control quality of subtitles
by Peter Münster 11 Jun '20

11 Jun '20

Hi, I would like to learn, how to control the quality of the displayed subtitles. This is how I create the video: ffmpeg -i input.ts -map 0:0 -map 0:1 -map 0:2 -c:v libx264 -c:a copy -c:s dvdsub output.mkv This looks nice with VLC: http://pmrb.free.fr/tmp/ffmpeg-test/vlc.jpg But very ugly with a Samsung TV: http://pmrb.free.fr/tmp/ffmpeg-test/samsung.jpg How could I get nice looking subtitles on this kind of Samsung TV please? Thanks in advance for any hints, -- Peter

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