July 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

2 2

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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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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Can FFmpeg convert from ArriRaw MXF sources?
by Harvey Pikelberger 19 Nov '19

19 Nov '19

Is FFmpeg capable of transcoding from ArriaRaw to h.264?

6 11

denoise after scaling or before?
by Jon bae 27 Oct '19

27 Oct '19

Hello, I just have an general question: As I know denoiser are specialized to specific noise pattern. Makes it a different when I denoise after scaling a video, or is the quality better when I denoise before scaling. In terms of speed it would be better for me, denoising after scaling, but when the quality is better, then I would go that way. Have a nice day! Jonathan

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[avi @ 0x9873ba0] Application provided invalid, non monotonically increasing dts to muxer in stream 1: 1104 >= 1104 [avi @ 0x987d2a0] Application provided invalid, non monotonically increasing dts to muxer only when using tee muxer
by Michael Kohne 11 Sep '19

11 Sep '19

I'm using ffmpeg 4.1 to grab an rtsp stream (includes both H.264 video and G.711 audio) from a Sony EP580 IP camera. If I transcode to mpeg4v2 and aac to a single avi file, things seem to work fine. This command line will go for as long as I like (I regularly do 15 minutes at a go). ffmpeg -min_port 62000 -max_port 62004 -i rtsp://192.168.0.113/media/video1 -codec:v msmpeg4v2 -codec:a ac3 -ar 44100 -map 0 -f avi /data/PatientData/vidtmp/test.avi If I instead use the tee muxer to make two avi files: ffmpeg -min_port 62000 -max_port 62004 -i rtsp://192.168.0.113/media/video1 -codec:v msmpeg4v2 -codec:a ac3 -ar 44100 -map 0 -f tee "[f=avi]/data/PatientData/vidtmp/one.avi|[f=avi]/data/PatientData/vidtmp/two.avi" Then within 2 minutes of starting the stream, it fails with the errors: [avi @ 0x9873ba0] Application provided invalid, non monotonically increasing dts to muxer in stream 1: 1104 >= 1104 [avi @ 0x987d2a0] Application provided invalid, non monotonically increasing dts to muxer in stream 1: 1104 >= 1104 I've repeated these tests over and over for a week or so now, so I am fairly confident that it's not simply that the camera is janky some of the time. I've also tested using the rtsp stream 'rtsp:// 184.72.239.149/vod/mp4:BigBuckBunny_175k.mov'. That stream does not produce any errors with the tee muxer. Note that those errors both appear to come from the avi muxers, not the tee muxer, which is why I find it surprising: if the camera's output causes this, then why doesn't it happen with the single avi muxer? Test note: the tee invocation is the simplest command line that I could come up with that produces the problem. In the real world one avi muxer goes to a pipe for a live output display, and the other goes to the segment muxer to produce output files in manageable chunks. I did also try this with the latest ffmpeg snapshot (as of yesterday morning), and there was no difference in behavior. Any clues as to how to debug this would be greatly appreciated. ffmpeg version 4.1 Copyright (c) 2000-2018 the FFmpeg developers built with gcc 4.4.7 (GCC) 20120313 (Red Hat 4.4.7-11) configuration: --prefix=/data/SHR4634/ffmpeg/build --extra-cflags=-I/data/SHR4634/ffmpeg/build/include --extra-ldflags=-L/data/SHR4634/ffmpeg/build/lib --extra-libs='-lm -ldl -lpthread -lrt' --enable-gpl --enable-nonfree --disable-libfdk_aac --enable-libmp3lame --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libfreetype --enable-libspeex --enable-libtheora --cpu=i686 --enable-runtime-cpudetect libavutil 56. 22.100 / 56. 22.100 libavcodec 58. 35.100 / 58. 35.100 libavformat 58. 20.100 / 58. 20.100 libavdevice 58. 5.100 / 58. 5.100 libavfilter 7. 40.101 / 7. 40.101 libswscale 5. 3.100 / 5. 3.100 libswresample 3. 3.100 / 3. 3.100 libpostproc 55. 3.100 / 55. 3.100 OS is CentOS 6, 32 bit. Michael Kohne Senior Software Engineer Office: 215.283.0860 x208 mhkohne(a)moberg.com -- Celebrating 20 Years Transforming Neurocritical Care Moberg Research, Inc. 224 S Maple Street, Ambler, PA 19002 24/7 Customer Support: 888.662.7246 www.moberg.com <https://www.moberg.com/>

2 7

Verifying lossless rewrapping/transcoding in one step?
by Peter B. 11 Aug '19

11 Aug '19

Hi everyone :) I was wondering if maybe there's a quicker way to verify lossless transcoding or container rewrapping than these 3 steps: 1. generate content hash during transcoding. 2. generate content hash of output file. 3. diff source-hash with output-hash (using whatever diff-tool). I'm doing this in bash scripts, but I'm wondering if it would be possible to tell ffmpeg something like this: "Please generate the output and compare source and target's content hash and only return exit code 0 if they match" Is that already implemented, or would it be possible to do that? Thank you very much in advance, Peter B.

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Dashing using ffmpeg
by Ibrahim Alzahrani 31 Jul '19

31 Jul '19

Hello there Hope this finds you well I have encoded videos in different qualities and different codecs (AVC, HEVC, VP9, AV1) and I want to dash them using segment size of 5 secs for each codec I have tried to find the answer but I couldn't apply them I wiil be grateful if you response me with the right way of the dashing so i get the mpd file for mpeg-dash streaming All regards, Ibrahim

1 0

Re: [FFmpeg-user] ffmpeg cannot capture audio from capture card
by Md Ismail, Zakiah 31 Jul '19

31 Jul '19

I did not get a valid wav file. The duration here is 0. $ ffprobe test.wav Input #0, wav, from 'test.wav': Metadata: encoder : Lavf58.12.100 Duration: 00:00:00.00, bitrate: 1688 kb/s Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, 2 channels, s16, 1536 kb/s Using arecord, I was able to get a valid wav file. For example here, I managed to get a 10 seconds wav file - but the recording is a complete silent. Please refer to my logs below. On target: $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC283 Analog [ALC283 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 9: HDMI 3 [HDMI 3] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 10: HDMI 4 [HDMI 4] Subdevices: 1/1 Subdevice #0: subdevice #0 $ sudo aplay -Dplughw:CARD=PCH,DEV=3 beep.wav Playing WAVE 'beep.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono On host - which was where I ran the ffmpeg command last time: $ arecord -v hw:CARD=nuc-03i,DEV=0 -d 10 arecord.wav Recording WAVE 'hw:CARD=nuc-03i,DEV=0' : Unsigned 8 bit, Rate 8000 Hz, Mono ALSA <-> PulseAudio PCM I/O Plugin Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : U8 subformat : STD channels : 1 rate : 8000 exact rate : 8000 (8000/1) msbits : 8 buffer_size : 4000 period_size : 1000 period_time : 125000 tstamp_mode : NONE tstamp_type : GETTIMEOFDAY period_step : 1 avail_min : 1000 period_event : 0 start_threshold : 1 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 9007199254740992000 Recording WAVE 'arecord.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono ALSA <-> PulseAudio PCM I/O Plugin Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : U8 subformat : STD channels : 1 rate : 8000 exact rate : 8000 (8000/1) msbits : 8 buffer_size : 4000 period_size : 1000 period_time : 125000 tstamp_mode : NONE tstamp_type : GETTIMEOFDAY period_step : 1 avail_min : 1000 period_event : 0 start_threshold : 1 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 9007199254740992000 Is there anything that I'm missing here? Am I supposed do some configurations?

1 0

ffmpeg cannot capture audio from capture card
by Md Ismail, Zakiah 31 Jul '19

31 Jul '19

Hi all, I am using ffmpeg to record HDMI audio on another machine - using a capture card. The capture card looks as follow: https://www.amazon.com/EZCAP-Capture-Recorder-Windows-Streaming/dp/B075T8C1…. The capture card is not connected to any monitor - I use a HDMI dummy that looks as follow: https://www.amazon.com/NewerTech-Headless-Video-Accelerator-Dummy/dp/B01ASJ…. On the host, output of arecord -l is as follow: $ sudo arecord -l card 11: nuc-03i [ezcap U3 capture], device 0: USB Audio [USB Audio] Subdevices: 0/1 Subdevice #0: subdevice #0 card 12: nuc-03i-usb [USB Audio CODEC], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 As you can see, the HDMI capture card (ezcap U3 capture) is in the list. Then, I ran the following command on the host, while playing a video on the target: $ sudo ffmpeg -f alsa -i hw:CARD=nuc-03i,DEV=0 -t 30 test-03i-debug.wav -loglevel debug ffmpeg version 4.0.4 Copyright (c) 2000-2019 the FFmpeg developers built with gcc 8 (GCC) configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --docdir=/usr/share/doc/ffmpeg --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man --arch=x86_64 --optflags='-O2 -g -pipe -Wall -Werror=format-security -Wp,-D_FORTIFY_SOURCE=2 -Wp,-D_GLIBCXX_ASSERTIONS -fexceptions -fstack-protector-strong -grecord-gcc-switches -specs=/usr/lib/rpm/redhat/redhat-hardened-cc1 -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -m64 -mtune=generic -fasynchronous-unwind-tables -fstack-clash-protection -fcf-protection' --extra-ldflags='-Wl,-z,relro -Wl,-z,now -specs=/usr/lib/rpm/redhat/redhat-hardened-ld ' --extra-cflags=' ' --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libvo-amrwbenc --enable-version3 --enable-bzlib --disable-crystalhd --enable-fontconfig --enable-frei0r --enable-gcrypt --enable-gnutls --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libcdio --enable-libdrm --enable-indev=jack --enable-libfreetype --enable-libfribidi --enable-libgsm --enable-libmp3lame --enable-nvenc --enable-openal --enable-opencl --enable-opengl --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-librsvg --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libvorbis --enable-libv4l2 --enable-libvidstab --enable-libvmaf --enable-libvpx --enable-libx264 --enable-libx265 --enable-libxvid --enable-libzvbi --enable-avfilter --enable-avresample --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug --disable-stripping --shlibdir=/usr/lib64 --enable-libmfx --enable-runtime-cpudetect libavutil 56. 14.100 / 56. 14.100 libavcodec 58. 18.100 / 58. 18.100 libavformat 58. 12.100 / 58. 12.100 libavdevice 58. 3.100 / 58. 3.100 libavfilter 7. 16.100 / 7. 16.100 libavresample 4. 0. 0 / 4. 0. 0 libswscale 5. 1.100 / 5. 1.100 libswresample 3. 1.100 / 3. 1.100 libpostproc 55. 1.100 / 55. 1.100 Splitting the commandline. Reading option '-f' ... matched as option 'f' (force format) with argument 'alsa'. Reading option '-i' ... matched as input url with argument 'hw:CARD=nuc-03i,DEV=0'. Reading option '-t' ... matched as option 't' (record or transcode "duration" seconds of audio/video) with argument '30'. Reading option 'test-03i-debug.wav' ... 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 hw:CARD=nuc-03i,DEV=0. Applying option f (force format) with argument alsa. Successfully parsed a group of options. Opening an input file: hw:CARD=nuc-03i,DEV=0. [alsa @ 0x5643d94d4400] All info found Guessed Channel Layout for Input Stream #0.0 : stereo Input #0, alsa, from 'hw:CARD=nuc-03i,DEV=0': Duration: N/A, start: 1564539037.140202, bitrate: 1536 kb/s Stream #0:0, 1, 1/1000000: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s Successfully opened the file. Parsing a group of options: output url test-03i-debug.wav. Applying option t (record or transcode "duration" seconds of audio/video) with argument 30. Successfully parsed a group of options. Opening an output file: test-03i-debug.wav. [file @ 0x5643d94e8240] Setting default whitelist 'file,crypto' Successfully opened the file. Stream mapping: Stream #0:0 -> #0:0 (pcm_s16le (native) -> pcm_s16le (native)) Press [q] to stop, [?] for help cur_dts is invalid (this is harmless if it occurs once at the start per stream) detected 4 logical cores [graph_0_in_0_0 @ 0x5643d94eb740] Setting 'time_base' to value '1/48000' [graph_0_in_0_0 @ 0x5643d94eb740] Setting 'sample_rate' to value '48000' [graph_0_in_0_0 @ 0x5643d94eb740] Setting 'sample_fmt' to value 's16' [graph_0_in_0_0 @ 0x5643d94eb740] Setting 'channel_layout' to value '0x3' [graph_0_in_0_0 @ 0x5643d94eb740] tb:1/48000 samplefmt:s16 samplerate:48000 chlayout:0x3 [format_out_0_0 @ 0x5643d94ec0c0] Setting 'sample_fmts' to value 's16' [AVFilterGraph @ 0x5643d94ea300] query_formats: 5 queried, 12 merged, 0 already done, 0 delayed Output #0, wav, to 'test-03i-debug.wav': Metadata: ISFT : Lavf58.12.100 Stream #0:0, 0, 1/48000: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, stereo, s16, 1536 kb/s Metadata: encoder : Lavc58.18.100 pcm_s16le [out_0_0 @ 0x5643d94ec840] EOF on sink link out_0_0:default.=0.244x No more output streams to write to, finishing. size= 1kB time=00:00:27.37 bitrate= 0.4kbits/s speed=0.243x video:0kB audio:1kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 5.685131% Input file #0 (hw:CARD=nuc-03i,DEV=0): Input stream #0:0 (audio): 8 packets read (1568 bytes); 8 frames decoded (392 samples); Total: 8 packets (1568 bytes) demuxed Output file #0 (test-03i-debug.wav): Output stream #0:0 (audio): 7 frames encoded (343 samples); 7 packets muxed (1372 bytes); Total: 7 packets (1372 bytes) muxed 8 frames successfully decoded, 0 decoding errors [AVIOContext @ 0x5643d94e83c0] Statistics: 4 seeks, 4 writeouts The audio capture was not successful - the output file is always a 1KB/2KB file that cannot be played at all. My knowledge on ffmpeg is limited - can any of you help or just give me clues of what this might be? Extra information: I have another capture card connected to target but this is to record audio from the audio jack instead of HDMI. I am able to record audio for that without any issue. $ sudo ffmpeg -f alsa -i hw:CARD=nuc-03i-usb,DEV=0 -t 30 test-03i-soundcard.wav -loglevel debug ffmpeg version 4.0.4 Copyright (c) 2000-2019 the FFmpeg developers built with gcc 8 (GCC) configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --docdir=/usr/share/doc/ffmpeg --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man --arch=x86_64 --optflags='-O2 -g -pipe -Wall -Werror=format-security -Wp,-D_FORTIFY_SOURCE=2 -Wp,-D_GLIBCXX_ASSERTIONS -fexceptions -fstack-protector-strong -grecord-gcc-switches -specs=/usr/lib/rpm/redhat/redhat-hardened-cc1 -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -m64 -mtune=generic -fasynchronous-unwind-tables -fstack-clash-protection -fcf-protection' --extra-ldflags='-Wl,-z,relro -Wl,-z,now -specs=/usr/lib/rpm/redhat/redhat-hardened-ld ' --extra-cflags=' ' --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libvo-amrwbenc --enable-version3 --enable-bzlib --disable-crystalhd --enable-fontconfig --enable-frei0r --enable-gcrypt --enable-gnutls --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libcdio --enable-libdrm --enable-indev=jack --enable-libfreetype --enable-libfribidi --enable-libgsm --enable-libmp3lame --enable-nvenc --enable-openal --enable-opencl --enable-opengl --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-librsvg --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libvorbis --enable-libv4l2 --enable-libvidstab --enable-libvmaf --enable-libvpx --enable-libx264 --enable-libx265 --enable-libxvid --enable-libzvbi --enable-avfilter --enable-avresample --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug --disable-stripping --shlibdir=/usr/lib64 --enable-libmfx --enable-runtime-cpudetect libavutil 56. 14.100 / 56. 14.100 libavcodec 58. 18.100 / 58. 18.100 libavformat 58. 12.100 / 58. 12.100 libavdevice 58. 3.100 / 58. 3.100 libavfilter 7. 16.100 / 7. 16.100 libavresample 4. 0. 0 / 4. 0. 0 libswscale 5. 1.100 / 5. 1.100 libswresample 3. 1.100 / 3. 1.100 libpostproc 55. 1.100 / 55. 1.100 Splitting the commandline. Reading option '-f' ... matched as option 'f' (force format) with argument 'alsa'. Reading option '-i' ... matched as input url with argument 'hw:CARD=nuc-03i-usb,DEV=0'. Reading option '-t' ... matched as option 't' (record or transcode "duration" seconds of audio/video) with argument '30'. Reading option 'test-03i-soundcard.wav' ... 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 hw:CARD=nuc-03i-usb,DEV=0. Applying option f (force format) with argument alsa. Successfully parsed a group of options. Opening an input file: hw:CARD=nuc-03i-usb,DEV=0. [alsa @ 0x55f5103ae400] All info found Guessed Channel Layout for Input Stream #0.0 : stereo Input #0, alsa, from 'hw:CARD=nuc-03i-usb,DEV=0': Duration: N/A, start: 1564542205.434050, bitrate: 1536 kb/s Stream #0:0, 1, 1/1000000: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s Successfully opened the file. Parsing a group of options: output url test-03i-soundcard.wav. Applying option t (record or transcode "duration" seconds of audio/video) with argument 30. Successfully parsed a group of options. Opening an output file: test-03i-soundcard.wav. File 'test-03i-soundcard.wav' already exists. Overwrite ? [y/N] y [file @ 0x55f5103c2440] Setting default whitelist 'file,crypto' Successfully opened the file. Stream mapping: Stream #0:0 -> #0:0 (pcm_s16le (native) -> pcm_s16le (native)) Press [q] to stop, [?] for help cur_dts is invalid (this is harmless if it occurs once at the start per stream) detected 4 logical cores [graph_0_in_0_0 @ 0x55f5103c6000] Setting 'time_base' to value '1/48000' [graph_0_in_0_0 @ 0x55f5103c6000] Setting 'sample_rate' to value '48000' [graph_0_in_0_0 @ 0x55f5103c6000] Setting 'sample_fmt' to value 's16' [graph_0_in_0_0 @ 0x55f5103c6000] Setting 'channel_layout' to value '0x3' [graph_0_in_0_0 @ 0x55f5103c6000] tb:1/48000 samplefmt:s16 samplerate:48000 chlayout:0x3 [format_out_0_0 @ 0x55f5103c2980] Setting 'sample_fmts' to value 's16' [AVFilterGraph @ 0x55f5103c4bc0] query_formats: 5 queried, 12 merged, 0 already done, 0 delayed Output #0, wav, to 'test-03i-soundcard.wav': Metadata: ISFT : Lavf58.12.100 Stream #0:0, 0, 1/48000: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, stereo, s16, 1536 kb/s Metadata: encoder : Lavc58.18.100 pcm_s16le [out_0_0 @ 0x55f5103c7100] EOF on sink link out_0_0:default.=1.05x No more output streams to write to, finishing. size= 5625kB time=00:00:30.00 bitrate=1536.0kbits/s speed=1.05x video:0kB audio:5625kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.001354% Input file #0 (hw:CARD=nuc-03i-usb,DEV=0): Input stream #0:0 (audio): 13018 packets read (5760192 bytes); 13018 frames decoded (1440048 samples); Total: 13018 packets (5760192 bytes) demuxed Output file #0 (test-03i-soundcard.wav): Output stream #0:0 (audio): 13017 frames encoded (1439949 samples); 13017 packets muxed (5759796 bytes); Total: 13017 packets (5759796 bytes) muxed 13018 frames successfully decoded, 0 decoding errors [AVIOContext @ 0x55f5103c2600] Statistics: 4 seeks, 25 writeouts Regards, Zakiah

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