December 2019 - 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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'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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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 nvenc without cuda
by Johanna Nilson 29 Apr '20

29 Apr '20

nvidia-smi -L GPU 0: GRID M10-1B

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ADD REAL TIME STAMP TO RTSP STREAM SEGMENT
by Alejandro Escudero 28 Apr '20

28 Apr '20

Hi, I am trying to get a RTSP stream from an IP Camera and save that stream as several MP4 segments, but i need that each file segment name gets the real time stamp of the video stream. (If I use the -strftime 1, it gets the time of the local machine but i need the real rtsp time). How can I get the Real Time Stamp? I am suing this command: ffmpeg -rtsp_transport tcp -i "rtsp://admin:pass@myddns.dyndns.org:554/cam/realmonitor?channel=1&subtype=0" –f segment -segment_time 5 -c copy OUT%d.mp4 Thanks! Alejandro [https://ipmcdn.avast.com/images/icons/icon-envelope-tick-round-orange-anima…]<https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> Libre de virus. www.avast.com<https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…>

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delay time in live ultrasound converter
by Michael Koch 18 Apr '20

18 Apr '20

Hello, I'm using the following batch file (Windows 7) for a live ultrasound conversion. set "SR=44100" :: Sample rate set "F=14000" :: Subtracted frequency set "VOL=10" :: Volume factor set /a "N=4096*%F%/%SR%" :: N = 4096 * F / SR c:\ffmpeg\ffmpeg -f dshow -sample_rate %SR% -sample_size 16 -channels 2 -i audio="Microphone (SoundMAX Integrated" -af volume=%VOL%,afftfilt='real=if(lt(b+%N%,nb),real(b+%N%,ch),0)':'imag=if(lt(b+%N%,nb),imag(b+%N%,ch),0)' -f nut - | c:\ffmpeg\ffplay - FFmpeg gets the audio input from the computer's microphone input and passes the converted output to FFplay. This works fine for hearing bats, but it has about 1-2 seconds delay between input and output. Are there any options for minimizing the delay? The console output is copied below. Thanks, Michael C:\Users\mkoch\Desktop\Ultrasound Live>c:\ffmpeg\ffmpeg -f dshow -sample_rate 44 100 -sample_size 16 -channels 2 -i audio="Microphone (SoundMAX Integrated" -af v olume=10,afftfilt='real=if(lt(b+1300,nb),real(b+1300,ch),0)':'imag=if(lt(b+1300, nb),imag(b+1300,ch),0)' -f nut - | c:\ffmpeg\ffplay - ffmpeg version N-92857-g3e461f9d4c Copyright (c) 2000-2018 the FFmpeg developers built with gcc 8.2.1 (GCC) 20181201 configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfi g --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-lib freetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amr wb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy -- enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx --enable-l ibwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 -- enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --en able-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --en able-libxvid --enable-libaom --enable-libmfx --enable-amf --enable-ffnvcodec --e nable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enab le-avisynth --enable-libopenmpt libavutil 56. 25.100 / 56. 25.100 libavcodec 58. 42.104 / 58. 42.104 libavformat 58. 25.100 / 58. 25.100 ffplay version N-92857-g3e461f9d4c libavdevice 58. 6.101 / 58. 6.101 Copyr ight (c) 2003-2018 the FFmpeg developers libavfilter 7. 46.101 / 7. 46.101 built with gcc 8.2.1 (GCC) 20181201 configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfi g --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-lib freetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amr wb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy -- enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx --enable-l ibwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 -- enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --en able-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --en able-libxvid --enable-libaom --enable-libmfx --enable-amf --enable-ffnvcodec --e nable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enab le-avisynth --enable-libopenmpt libswscale 5. 4.100 / 5. 4.100 libswresample 3. 4.100 / 3. 4.100 libpostproc 55. 4.100 / 55. 4.100 libavutil 56. 25.100 / 56. 25.100 libavcodec 58. 42.104 / 58. 42.104 libavformat 58. 25.100 / 58. 25.100 libavdevice 58. 6.101 / 58. 6.101 libavfilter 7. 46.101 / 7. 46.101 libswscale 5. 4.100 / 5. 4.100 libswresample 3. 4.100 / 3. 4.100 libpostproc 55. 4.100 / 55. 4.100 Guessed Channel Layout for Input Stream #0.0 : stereo 0B f=0/0 Input #0, dshow, from 'audio=Microphone (SoundMAX Integrated': Duration: N/A, start: 1701.243000, bitrate: 1411 kb/s Stream #0:0: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s Stream mapping: Stream #0:0 -> #0:0 (pcm_s16le (native) -> vorbis (libvorbis)) Press [q] to stop, [?] for help Output #0, nut, to 'pipe:': aq= 0KB vq= 0KB sq= 0B f=0/0 Metadata: encoder : Lavf58.25.100 Stream #0:0: Audio: vorbis (libvorbis) (oV[0][0] / 0x566F), 44100 Hz, stereo , fltp Metadata: encoder : Lavc58.42.104 libvorbis Input #0, nut, from 'pipe:': Metadata: encoder : Lavf58.25.100 Duration: N/A, start: 0.000000, bitrate: N/A Stream #0:0: Audio: vorbis (oV[0][0] / 0x566F), 44100 Hz, stereo, fltp Metadata: encoder : Lavc58.42.104 libvorbis 16.27 M-A: 0.000 fd= 0 aq= 3KB vq= 0KB sq= 0B f=0/0

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DVD EIA-608 to ASS from stdin with custom style
by CGar 02 Apr '20

02 Apr '20

When extracting EIA-608 captions from a DVD to an ASS file I am unsure how to specify the style. I've seen snippets online about a force_style option but it seems to be about hardcoding them or specifying a subtitle file as input. Since I'm reading from stdin I'm unsure where or how to write the relevant part. My command line is as follows: [...] | ffmpeg -f lavfi -i 'movie=pipe\\:0[out0+subcc]' -map s sub_out.ass Where or how would I add something like: force_style=BorderStyle=1,Outline=2 -- -CGar-

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Ffmpeg is a Windows codec?
by jamie marchant 31 Mar '20

31 Mar '20

Is their a Windows codec available? I want to play Indeo video 3 files, which FFPlay can do but I want to play it through 'Video for Windows"(Windows NT/10 version). That way I can run an old piece of multimedia software. -- -Jamie Marchant

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