October 2018 - ffmpeg-user - lists.ffmpeg.org

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

3 10

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

7 7

HDR specification transfert
by Rollie Tyler 10 Feb '19

10 Feb '19

Hello everybody, I am trying to use FFMPEG ( 3.4 version) to transfert HDR video specifications (x265.mp4 videos). I have seen on other forums command lines that I tried to test for me but I never have succesfull results. I want to use Nvidia options (NVENC/SCALE_NPP/or other) because I have a P4 graphic card. Here is my file (Media Info) General Complete name : /storage/input_HDR.mp4 Format : MPEG-4 Format profile : Base Media Codec ID : isom (isom/iso2/mp41) File size : 293 MiB Duration : 54 s 422 ms Overall bit rate mode : Variable Overall bit rate : 45.1 Mb/s Writing application : Lavf56.15.103 Video ID : 1 Format : HEVC Format/Info : High Efficiency Video Coding Format profile : Main 10@L5.1@High Codec ID : hev1 Codec ID/Info : High Efficiency Video Coding Duration : 54 s 375 ms Bit rate : 44.8 Mb/s Width : 3 840 pixels Height : 2 160 pixels Display aspect ratio : 16:9 Frame rate mode : Constant Frame rate : 24.000 FPS Original frame rate : 23.976 (24000/1001) FPS Color space : YUV Chroma subsampling : 4:2:0 (Type 2) Bit depth : 10 bits Bits/(Pixel*Frame) : 0.225 Stream size : 290 MiB (99%) Writing library : ATEME Titan KFE 3.7.0 (4.7.0.2002) Color range : Limited Color primaries : BT.2020 Transfer characteristics : SMPTE ST 2084 Matrix coefficients : BT.2020 non-constant Mastering display color primaries : R: x=0.680000 y=0.320000, G: x=0.265000 y=0.690000, B: x=0.150000 y=0.060000, White point: x=0.312700 y=0.329000 Mastering display luminance : min: 0.0200 cd/m2, max: 1200.0000 cd/m2 Audio ID : 2 Format : AAC Format/Info : Advanced Audio Codec Format profile : LC Codec ID : 40 Duration : 54 s 422 ms Bit rate mode : Variable Bit rate : 384 kb/s Channel(s) : 2 channels Channel(s)_Original : 6 channels Channel positions : Front: L C R, Side: L R, LFE Sampling rate : 48.0 kHz Frame rate : 46.875 FPS (1024 SPF) Compression mode : Lossy Stream size : 2.47 MiB (1%) Default : Yes Alternate group : 1 And here is the command that I tried: time /FFmpeg/3.4/bin/ffmpeg -hwaccel cuvid -c:v hevc_cuvid -i /storage/input_HDR.mp4 -c:v nvenc_hevc -filter_complex [0:0]scale_npp=colormatrix=bt2020:h_chr_pos=0:v_chr_pos=0,format=yuv420p10le -pix_fmt yuv420p10le -x265-params "colorprim=bt2020:transfer=smpte-st-2084:colormatrix=bt2020nc" -crf 22 -c:a copy /storage/OUT/testHDR.mp4 When I ran the command I have this error message ffmpeg version 3.4 Copyright (c) 2000-2017 the FFmpeg developers built with gcc 4.8.5 (GCC) 20150623 (Red Hat 4.8.5-16) configuration: --prefix=/data/TOOL-FFMPEG/target --pkg-config-flags=--static --extra-cflags=-I/data/TOOL-FFMPEG/target/include --extra-ldflags=-L/data/TOOL-FFMPEG/target/lib --enable-static --enable-pic --disable-ffserver --disable-debug --disable-shared --disable-doc --enable-gpl --enable-fontconfig --enable-version3 --enable-libass --enable-libfdk-aac --enable-libfreetype --enable-libmp3lame --enable-libopus --enable-libtheora --enable-libvorbis --enable-libopenjpeg --enable-libvpx --enable-libx264 --enable-libx265 --enable-libvidstab --enable-runtime-cpudetect --enable-libfribidi --enable-libsoxr --enable-libvmaf --enable-vaapi --enable-libzimg --enable-openssl --enable-nvenc --enable-cuvid --enable-libnpp --enable-cuda --enable-cuda-sdk --enable-hwaccel=hevc_vaapi --enable-hwaccel=h264_vaapi --enable-hwaccel=mpeg2_vaapi --enable-hwaccel=vp9_vaapi --enable-librtmp --enable-nonfree libavutil 55. 78.100 / 55. 78.100 libavcodec 57.107.100 / 57.107.100 libavformat 57. 83.100 / 57. 83.100 libavdevice 57. 10.100 / 57. 10.100 libavfilter 6.107.100 / 6.107.100 libswscale 4. 8.100 / 4. 8.100 libswresample 2. 9.100 / 2. 9.100 libpostproc 54. 7.100 / 54. 7.100 [aac @ 0x2ce6920] This stream seems to incorrectly report its last channel as LFE[3], mapping to LFE[0] Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/storage/input_HDR.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2mp41 encoder : Lavf56.15.103 Duration: 00:00:54.42, start: 0.000000, bitrate: 45136 kb/s Stream #0:0(und): Video: hevc (Main 10) (hev1 / 0x31766568), yuv420p10le(tv, bt2020nc/bt2020/smpte2084), 3840x2160 [SAR 1:1 DAR 16:9], 44788 kb/s, 24 fps, 24 tbr, 24k tbn, 23.98 tbc (default) Metadata: handler_name : VideoHandler Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, 5.1, fltp, 381 kb/s (default) Metadata: handler_name : SoundHandler *[Parsed_scale_npp_0 @ 0x2ccb940] Option 'colormatrix' not found [AVFilterGraph @ 0x2cc86c0] Error initializing filter 'scale_npp' with args 'colormatrix=bt2020:h_chr_pos=0:v_chr_pos=0' Error initializing complex filters. Option not found * Why I cannot use the colomatrix option? /PS:I have tried with "scale_cuda" and it didn't work too./ -- Sent from: http://www.ffmpeg-archive.org/

4 5

uncompressed RGB within MXF
by Kieran O Leary 27 Jan '19

27 Jan '19

Hi, I see that uncompressed RGB is not supported within MXF. I know that Resolve can create 10-bit RGB but only at HD resolution. I didn't see anything relating to uncompressed RGB in the UL list in mxfenc.c and I don't have an SMPTE account to get beyond the Paywall that would let me know what RGB pixel formats are possible within MXF in a general sense. I'm assuming that there isn't some option that I'm missing out on here, it just isn't supported? $ ./ffmpeg -f lavfi -i testsrc -pix_fmt gbrp10le -c:v copy out.mxf ffmpeg version N-92260-g3cc3cb663b Copyright (c) 2000-2018 the FFmpeg developers built with gcc 7 (Ubuntu 7.3.0-27ubuntu1~18.04) configuration: libavutil 56. 20.100 / 56. 20.100 libavcodec 58. 33.102 / 58. 33.102 libavformat 58. 19.102 / 58. 19.102 libavdevice 58. 4.106 / 58. 4.106 libavfilter 7. 37.100 / 7. 37.100 libswscale 5. 2.100 / 5. 2.100 libswresample 3. 2.100 / 3. 2.100 Input #0, lavfi, from 'testsrc': Duration: N/A, start: 0.000000, bitrate: N/A Stream #0:0: Video: rawvideo (RGB[24] / 0x18424752), rgb24, 320x240 [SAR 1:1 DAR 4:3], 25 tbr, 25 tbn, 25 tbc [mxf @ 0x56227a627880] track 0: could not find essence container ul, codec not currently supported in container Could not write header for output file #0 (incorrect codec parameters ?): Operation not permitted Stream mapping: Stream #0:0 -> #0:0 (copy) Last message repeated 1 times Best, Kieran.

6 15

Max rtbufsize Via dshow
by Gabriel Balaich 11 Jan '19

11 Jan '19

Up until a few days ago I had been running two capture cards, 1x4K card and 1x1080P card, I would encode both streams with a GTX 1080 no dropped frames no issues. But just recently I replaced the 1080P capture card with another 4K capture card to prep for a new camera and can't get things working gracefully. On my first test encoding 2x4K60 streams simultaneously I dropped frames starting the recording *and *ending the recording, but nothing in-between. Errors displayed: [dshow @ 000001499bb17180] real-time buffer [Video (00 Pro Capture HDMI 4K+)] [video input] too full or near too full (62% of size: 2147480000 [rtbufsize parameter])! frame dropped! [dshow @ 00000149944e7080] real-time buffer [AVerMedia HD Capture GC573 1] [video input] too full or near too full (62% of size: 2147480000 [rtbufsize parameter])! frame dropped! The obvious answer, according to the warning, would be increasing rtbufsize but I seem to have hit the cap... If I try to increase rtbufsize passed 2147.48M I get another error: [dshow @ 00000250df6c7080] Value 3000000000.000000 for parameter 'rtbufsize' out of range [0 - 2.14748e+09] [dshow @ 00000250df6c7080] Error setting option rtbufsize to value 3000M. video=AVerMedia HD Capture GC573 1:audio=SPDIF/ADAT (1+2) (RME Fireface UC): Result too large Is this a baked in limitation of FFmpeg or dshow? And if it is, why impose said limitation? Since then I've tried messing with other parts of my command and found that by re-arranging my inputs / outputs I can prevent frame drops upon starting a recording but I still can't end a recording without dropping a bunch of frames. Anyone know why re-arranging my inputs / outputs would prevent frames from dropping when starting a recording or affect anything? Is it possible to bypass the rtbufsize cap? Any other ideas? Here is my full command, Ignore the blank numbers next to -ss, will be used to sync outputs when everything is working: ffmpeg -y -hide_banner -thread_queue_size 9999 -indexmem 9999 -guess_layout_max 0 -f dshow -rtbufsize 2147.48M ` -i audio="Analog (1+2) (RME Fireface UC)" ` -thread_queue_size 9999 -indexmem 9999 -guess_layout_max 0 -f dshow -rtbufsize 2147.48M ` -i audio="ADAT (5+6) (RME Fireface UC)" ` -thread_queue_size 9999 -indexmem 9999 -r 25 -f lavfi -rtbufsize 2147.48M -i color=c=black:s=50x50 ` -thread_queue_size 9999 -indexmem 9999 -guess_layout_max 0 -f dshow -video_size 3840x2160 -rtbufsize 2147.48M ` -framerate 60 -pixel_format nv12 -i video="AVerMedia HD Capture GC573 1":audio="SPDIF/ADAT (1+2) (RME Fireface UC)" ` -thread_queue_size 9999 -indexmem 9999 -guess_layout_max 0 -f dshow -video_size 3840x2160 -rtbufsize 2147.48M ` -framerate 60 -pixel_format nv12 -i video="Video (00 Pro Capture HDMI 4K+)":audio="ADAT (3+4) (RME Fireface UC)" ` -map 2,0 -map 0 -c:v libx264 -r 25 -rc-lookahead 50 -forced-idr 1 -sc_threshold 0 -flags +cgop ` -force_key_frames "expr:gte(t,n_forced*2)" -preset ultrafast -pix_fmt nv12 -b:v 16K -minrate 16K -maxrate 16K -bufsize 16k ` -c:a aac -ar 44100 -b:a 384k -ac 2 -af "aresample=async=250" -vsync 1 -ss 00:00:00.000 ` -max_muxing_queue_size 9999 -f segment -segment_time 600 -segment_wrap 9 -reset_timestamps 1 ` -segment_format_options max_delay=0 C:\Users\djcim\Videos\Main\Discord\Discord%02d.ts ` -map 2,1 -map 1 -c:v libx264 -r 25 -rc-lookahead 50 -forced-idr 1 -sc_threshold 0 -flags +cgop ` -force_key_frames "expr:gte(t,n_forced*2)" -preset ultrafast -pix_fmt nv12 -b:v 16K -minrate 16K -maxrate 16K -bufsize 16k ` -c:a aac -ar 44100 -b:a 384k -ac 2 -af "aresample=async=250" -vsync 1 -ss 00:00:00.000 ` -max_muxing_queue_size 9999 -f segment -segment_time 600 -segment_wrap 9 -reset_timestamps 1 ` -segment_format_options max_delay=0 C:\Users\djcim\Videos\Main\Soundboard\Soundboard%02d.ts ` -map 3:0,3:1 -map 3:1 -c:v h264_nvenc -r 60 -rc-lookahead 120 -forced-idr 1 -strict_gop 1 -sc_threshold 0 -flags +cgop ` -force_key_frames "expr:gte(t,n_forced*2)" -preset: llhp -pix_fmt nv12 -b:v 250M -minrate 250M -maxrate 250M -bufsize 250M ` -c:a aac -ar 44100 -b:a 384k -ac 2 -af "pan=mono|c0=c0, aresample=async=250" -vsync 1 -ss 00:00:00.000 ` -max_muxing_queue_size 9999 -f segment -segment_time 600 -segment_wrap 9 -reset_timestamps 1 ` -segment_format_options max_delay=0 C:\Users\djcim\Videos\Main\Camera\Camera%02d.ts ` -map 4:0,4:1 -map 4:1 -c:v h264_nvenc -r 60 -rc-lookahead 120 -forced-idr 1 -strict_gop 1 -sc_threshold 0 -flags +cgop ` -force_key_frames "expr:gte(t,n_forced*2)" -preset: llhp -pix_fmt nv12 -b:v 250M -minrate 250M -maxrate 250M -bufsize 250M ` -c:a aac -ar 44100 -b:a 384k -ac 2 -af "atrim=0.086, asetpts=PTS-STARTPTS, aresample=async=250" -vsync 1 -ss 00:00:00.000 ` -max_muxing_queue_size 9999 -f segment -segment_time 600 -segment_wrap 9 -reset_timestamps 1 ` -segment_format_options max_delay=0 C:\Users\djcim\Videos\Main\Magewell\Magewell%02d.ts

4 13

Issues Live Streaming Audio Only
by Brett Garrett 03 Dec '18

03 Dec '18

I've been trying to stream an audio feed from a live source using the RTSP protocol. The main goal is to eventually utilize jitter buffers or other techniques to reduce/remove stutters and skips. Although, at the moment I can't seem to get any audio to transfer over the network at all. I do have a somewhat unique setup, and therefore don't have many other options to work with. I'm currently using Debian 9 with no gui for both the server and client(s). I would say the biggest hangup or issue is having to use Jack (which uses an ALSA device) for my output as it is used in several other applications that interact with the audio output. All of which are not running for the time being because I'm still just trying to get the stream from one computer to another across my local network. Here's what I'm hoping is enough information and that I'm not vomiting too much text. A little info on our audio card specs... root@test-9:~# *aplay -l* **** List of PLAYBACK Hardware Devices **** card 0: M44 [M Audio Delta 44], device 0: ICE1712 multi [ICE1712 multi] Subdevices: 1/1 Subdevice #0: subdevice #0 root@test-9:~# *arecord --dump-hw-params -D hw:0,0* Recording WAVE 'stdin' : Unsigned 8 bit, Rate 8000 Hz, Mono HW Params of device "hw:0,0": -------------------- ACCESS: MMAP_INTERLEAVED RW_INTERLEAVED FORMAT: S32_LE SUBFORMAT: STD SAMPLE_BITS: 32 FRAME_BITS: 384 CHANNELS: 12 RATE: [8000 96000] PERIOD_TIME: (20 341250] PERIOD_SIZE: [2 2730] PERIOD_BYTES: [96 131040] PERIODS: [1 1024] BUFFER_TIME: (20 682625] BUFFER_SIZE: [2 5461] BUFFER_BYTES: [96 262128] TICK_TIME: ALL -------------------- arecord: set_params:1299: Sample format non available Available formats: - S32_LE Audio comes in through an analog XLR connection. Verified audio input is good by using the following commands... root@test-9:~# /usr/bin/dbus-run-session ffmpeg -f jack -i ffmpeg -y output.wav root@test-9:~# jack_connect system:capture_1 ffmpeg:input_1 && jack_connect system:capture_2 ffmpeg:input_2 Waited a while then killed the tasks. Acquired the output.wav and it played with good audio. Although, I need to be able to get it from one computer to another through a live feed (not recording to some single file that will grow to crazy sizes). So here's my server config setup. Without the comments. root@test-9:~# *grep ^[^#] /etc/ffserver.conf* HTTPPort 8585 RTSPPort 15151 HTTPBindAddress 0.0.0.0 MaxHTTPConnections 2000 MaxClients 1000 MaxBandwidth 1000 CustomLog - <Feed feed1.ffm> File /tmp/feed1.ffm FileMaxSize 200K ACL allow 127.0.0.1 </Feed> <Stream test1-rtsp.ogg> Format rtp Feed feed1.ffm NoVideo AudioCodec aac AudioChannels 2 AudioBitRate 64 AudioSampleRate 48000 AVOptionAudio flags +global_header </Stream> <Stream stat.html> Format status ACL allow localhost ACL allow 192.168.0.0 192.168.255.255 </Stream> <Redirect index.html> URL http://www.ffmpeg.org/ </Redirect> root@test-9:~# *ffserver -loglevel debug &* [1] 11631 root@test-9:~# ffserver version 3.2.10-1~deb9u1 Copyright (c) 2000-2018 the FFmpeg developers built with gcc 6.3.0 (Debian 6.3.0-18) 20170516 configuration: --prefix=/usr --extra-version='1~deb9u1' --toolchain=hardened --libdir=/usr/lib/i386-linux-gnu --incdir=/usr/include/i386-linux-gnu --enable-gpl --disable-stripping --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libebur128 --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared libavutil 55. 34.101 / 55. 34.101 libavcodec 57. 64.101 / 57. 64.101 libavformat 57. 56.101 / 57. 56.101 libavdevice 57. 1.100 / 57. 1.100 libavfilter 6. 65.100 / 6. 65.100 libavresample 3. 1. 0 / 3. 1. 0 libswscale 4. 2.100 / 4. 2.100 libswresample 2. 3.100 / 2. 3.100 libpostproc 54. 1.100 / 54. 1.100 Tue Oct 30 13:23:01 2018 [file @ 0xfe65c0]Setting default whitelist 'file,crypto' Tue Oct 30 13:23:01 2018 [ffm @ 0xfe39e0]Using AVStream.codec to pass codec parameters to muxers is deprecated, use AVStream.codecpar instead. Tue Oct 30 13:23:01 2018 writing recommended configuration: ac=2,b=64000,ar=48000,flags=+global_header Tue Oct 30 13:23:01 2018 [AVIOContext @ 0xfe6660]Statistics: 0 seeks, 1 writeouts Tue Oct 30 13:23:01 2018 FFserver started. root@test-9:~# */usr/bin/dbus-run-session ffmpeg -f jack -i ffmpeg http://127.0.0.1:8585/feed1.ffm <http://127.0.0.1:8585/feed1.ffm>* ffmpeg version 3.2.10-1~deb9u1 Copyright (c) 2000-2018 the FFmpeg developers built with gcc 6.3.0 (Debian 6.3.0-18) 20170516 configuration: --prefix=/usr --extra-version='1~deb9u1' --toolchain=hardened --libdir=/usr/lib/i386-linux-gnu --incdir=/usr/include/i386-linux-gnu --enable-gpl --disable-stripping --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libebur128 --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared libavutil 55. 34.101 / 55. 34.101 libavcodec 57. 64.101 / 57. 64.101 libavformat 57. 56.101 / 57. 56.101 libavdevice 57. 1.100 / 57. 1.100 libavfilter 6. 65.100 / 6. 65.100 libavresample 3. 1. 0 / 3. 1. 0 libswscale 4. 2.100 / 4. 2.100 libswresample 2. 3.100 / 2. 3.100 libpostproc 54. 1.100 / 54. 1.100 Cannot connect to server socket err = No such file or directory Cannot connect to server request channel jackdmp 1.9.11 Copyright 2001-2005 Paul Davis and others. Copyright 2004-2014 Grame. jackdmp comes with ABSOLUTELY NO WARRANTY This is free software, and you are welcome to redistribute it under certain conditions; see the file COPYING for details no message buffer overruns no message buffer overruns no message buffer overruns JACK server starting in realtime mode with priority 10 self-connect-mode is "Don't restrict self connect requests" audio_reservation_init Acquire audio card Audio0 creating alsa driver ... hw:0|hw:0|1024|2|48000|0|0|nomon|swmeter|-|32bit configuring for 48000Hz, period = 1024 frames (21.3 ms), buffer = 2 periods ALSA: final selected sample format for capture: 32bit integer little-endian ALSA: use 2 periods for capture ALSA: final selected sample format for playback: 32bit integer little-endian ALSA: use 2 periods for playback [jack @ 0x2494700] JACK client registered and activated (rate=48000Hz, buffer_size=1024 frames) Guessed Channel Layout for Input Stream #0.0 : stereo Input #0, jack, from 'ffmpeg': Duration: N/A, start: 1540927425.470458, bitrate: 3072 kb/s Stream #0:0: Audio: pcm_f32le, 48000 Hz, stereo, flt, 3072 kb/s Tue Oct 30 13:23:45 2018 [NULL @ 0xfe39e0]Opening '/tmp/feed1.ffm' for reading Tue Oct 30 13:23:45 2018 [file @ 0xfe46c0]Setting default whitelist 'file,crypto' Tue Oct 30 13:23:45 2018 [ffm @ 0xfe39e0]Format ffm probed with size=2048 and score=101 Tue Oct 30 13:23:45 2018 [NULL @ 0xff0f20]Setting entry with key 'ac' to value '2' Tue Oct 30 13:23:45 2018 [NULL @ 0xff0f20]Setting entry with key 'b' to value '64000' Tue Oct 30 13:23:45 2018 [NULL @ 0xff0f20]Setting entry with key 'ar' to value '48000' Tue Oct 30 13:23:45 2018 [NULL @ 0xff0f20]Setting entry with key 'flags' to value '+global_header' Tue Oct 30 13:23:45 2018 writing recommended configuration: ac=2,b=64000,ar=48000,flags=+global_header Tue Oct 30 13:23:45 2018 127.0.0.1 - - [GET] "/feed1.ffm HTTP/1.1" 200 4175 Tue Oct 30 13:23:45 2018 [AVIOContext @ 0xfe6720]Statistics: 4096 bytes read, 0 seeks Output #0, ffm, to 'http://127.0.0.1:8585/feed1.ffm': Metadata: creation_time : now encoder : Lavf57.56.101 Stream #0:0: Audio: aac (LC), 48000 Hz, stereo, fltp, 64 kb/s Metadata: encoder : Lavc57.64.101 aac Stream mapping: Stream #0:0 -> #0:0 (pcm_f32le (native) -> aac (native)) Press [q] to stop, [?] for help Tue Oct 30 13:23:45 2018 [NULL @ 0xff0f20]Setting entry with key 'b' to value '64000' Tue Oct 30 13:23:45 2018 [NULL @ 0xff0f20]Setting entry with key 'ab' to value '64000' Tue Oct 30 13:23:45 2018 [NULL @ 0xff0f20]Setting entry with key 'flags' to value '0x00400000' Tue Oct 30 13:23:45 2018 [NULL @ 0xff0f20]Setting entry with key 'ar' to value '48000' Tue Oct 30 13:23:45 2018 [NULL @ 0xff0f20]Setting entry with key 'ac' to value '2' Tue Oct 30 13:23:45 2018 [NULL @ 0xff0f20]Setting entry with key 'frame_size' to value '1024' Tue Oct 30 13:23:45 2018 [NULL @ 0xff0f20]Setting entry with key 'profile' to value '1' Tue Oct 30 13:23:45 2018 [NULL @ 0xff0f20]Setting entry with key 'channel_layout' to value '3' Tue Oct 30 13:23:45 2018 [NULL @ 0xff0f20]Setting entry with key 'time_base' to value '1/48000' Tue Oct 30 13:23:45 2018 [NULL @ 0xff0f20]Setting entry with key 'delay' to value '1024' Tue Oct 30 13:23:45 2018 [NULL @ 0xff0f20]Setting entry with key 'pkt_timebase' to value '1/1000000' Changing text at the bottom makes it appear like 'somethings happening'. There is a slow bit transfer because I have not connected the jack inputs/outputs. size= 72kB time=00:01:07.08 bitrate= 8.8kbits/s speed= 1x root@test-9:~# *jack_connect system:capture_1 ffmpeg:input_1 && jack_connect system:capture_2 ffmpeg:input_2 && jack_lsp -c* system:capture_1 ffmpeg:input_1 system:capture_2 ffmpeg:input_2 system:capture_3 system:capture_4 system:capture_5 system:capture_6 system:capture_7 system:capture_8 system:capture_9 system:capture_10 system:capture_11 system:capture_12 system:playback_1 system:playback_2 system:playback_3 system:playback_4 system:playback_5 system:playback_6 system:playback_7 system:playback_8 system:playback_9 system:playback_10 ffmpeg:input_1 system:capture_1 ffmpeg:input_2 system:capture_2 size= 800kB time=00:03:31.24 bitrate= 31.0kbits/s speed= 1x The bitrate goes up, so I'm assuming this means it is receiving the audio alright and that everything's in place. Go to http://192.168.1.119:8585/stat.html the page comes up with... [see attached picture] Time to see if we can get the stream from another computer... root@test-17:~# ffplay -nodisp rtsp://192.168.1.119:15151/test1-rtsp.ogg ffplay version 3.2.12-1~deb9u1 Copyright (c) 2003-2018 the FFmpeg developers built with gcc 6.3.0 (Debian 6.3.0-18+deb9u1) 20170516 configuration: --prefix=/usr --extra-version='1~deb9u1' --toolchain=hardened --libdir=/usr/lib/i386-linux-gnu --incdir=/usr/include/i386-linux-gnu --enable-gpl --disable-stripping --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libebur128 --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared libavutil 55. 34.101 / 55. 34.101 libavcodec 57. 64.101 / 57. 64.101 libavformat 57. 56.101 / 57. 56.101 libavdevice 57. 1.100 / 57. 1.100 libavfilter 6. 65.100 / 6. 65.100 libavresample 3. 1. 0 / 3. 1. 0 libswscale 4. 2.100 / 4. 2.100 libswresample 2. 3.100 / 2. 3.100 libpostproc 54. 1.100 / 54. 1.100 nan : 0.000 fd= 0 aq= 0KB vq= 0KB sq= 0B f=0/0 Then it just sits there. No audio, but that's not surprising since normally I'd have to pipe the audio out through jack (no doable with ffmpeg, to my knowledge). So I re-enabled the integrated audio to see if it might automatically pipe the audio out to that device, but I get the same output noted above and still no audio. When I run the same command on server. I get the exact same output on the new ssh prompt I opened up. Even if I change the ip address to 'localhost'. Although, on the ssh prompt where I started up ffserver from I get the following output pop up along with the constantly updating bitrate... Wed Oct 30 14:37:35 2018 [NULL @ 0x128cfe0]Opening '/tmp/feed1.ffm' for reading Wed Oct 30 14:37:35 2018 [file @ 0x127be20]Setting default whitelist 'file,crypto' Wed Oct 30 14:37:35 2018 [ffm @ 0x128cfe0]Format ffm probed with size=2048 and score=101 Wed Oct 30 14:37:35 2018 [NULL @ 0x128e8a0]Setting entry with key 'strict' to value '-2' Wed Oct 30 14:37:35 2018 [NULL @ 0x128e8a0]Setting entry with key 'ac' to value '2' Wed Oct 30 14:37:35 2018 [NULL @ 0x128e8a0]Setting entry with key 'b' to value '128000' Wed Oct 30 14:37:35 2018 [NULL @ 0x128e8a0]Setting entry with key 'ar' to value '48000' Wed Oct 30 14:37:35 2018 [NULL @ 0x128e8a0]Setting entry with key 'flags' to value '+global_header' Wed Oct 30 14:37:35 2018 [rtp @ 0x128f440]No default whitelist set Wed Oct 30 14:37:35 2018 [udp @ 0x1290c20]No default whitelist set Wed Oct 30 14:37:35 2018 [udp @ 0x1290d60]No default whitelist set Wed Oct 30 14:37:35 2018 192.168.1.119:31830 - - "PLAY test1-rtsp.ogg/streamid=0 RTP/UDP" Wed Oct 30 14:37:35 2018 Failed to parse interval end specification '' I couldn't find much on the final line there about the "Failed to parse interval end specification ''" or of a way to fix it I mean. I've tried adding -analyzeduration and -probesize on both the ffmpeg command that uploads the audio to the feed, and also on the ffplay command that should play the feed. They did absolutely nothing to change anything. I'm pretty sure it's not firewall related. ufw and iptables are not installed. root@test-9:~# iptables -su: iptables: command not found root@test-9:~# ufw -su: ufw: command not found It feels like I'm missing some config option somewhere, but I can't figure out where. All I see everywhere are examples of people setting up streams with webcams or other video related formats and very little that just deals with audio only, nothing has worked for me so far. I'm open to any and all suggestions. Thank you for your time. -- Brett

3 3

Alpha Channel QT with SubTitles
by Gandharv Bhagat 05 Nov '18

05 Nov '18

Hi, I am trying to generate an alpha channel QT, with Apple animation codec, but the I get output has a white background. here is the command ffmpeg -r 23.976 -loop 1 -i Alpha_1920x1080.png -c:v qtrle -t 00:00:30.000 -vf subtitles=“Subtitle.srt:force_style='Fontsize=18,BorderStyle=1,Outline=0,Shadow=1,MarginV=25,FontName=Arial'" Subs2.mov Gandharv p.s. The PNG was created in Photoshop

4 9

Combining GIF slideshow and sound from MP4
by Markus Klyver 02 Nov '18

02 Nov '18

Hello, I would like to know the syntax for combining a slideshow of multiple GIF pictures and the sound of a MP4 file. I have timestamps for when a certain picture should be viewed, and I want to merge the pictures and the sound together into a single file. Best Regards, Markus Klyver

3 5

Question about rendering and photo dimensions.
by Hwaen Ch'uqi 02 Nov '18

02 Nov '18

Greetings All, I am a newbie to ffmpeg and am quite impressed at the program's capabilities. For the moment, I am simply combining an audio file with a single still image to create an mp4 file. I am using this command: ffmpeg -loop 1 -i FILE.jpg -i FILE.mp3 -c:v libx264 -c:a aac -strict experimental -b:a 192k -shortest FILE.mp4 >From what I can tell, it appears that picture files with smaller dimensions - 2960x2640, for example - render quite easily, whereas something like 4910x6054 causes the rendering to slow tremendously, even to the point of hanging/crashing my computer! Has this something to do with libx264 presets? What can I do to anticipate which picture files will have this undesired effect? Are there flags that I can call to help with these sizes? Any help would be appreciated! Hwaen Ch'uqi

4 13

slow encoding from raw rgb
by Daniel Oberhoff 02 Nov '18

02 Nov '18

Hello, So we are trying to encode raw rgb data from a pipe with ffmpeg. We found that ffmpeg implicitly converts the data to tv range bt601, which is suboptimal. Especially tv range loses more information than neccessary, also bt709 seems to be the better color space. We know we can fix this by adding -vf scale=out_color_matrix=bt709:out_range=pc. Unfortuntately thi makes encoding a lot slower (40 vs 60 fps) which effectively makes this approach too slow for us. Do yiz have any idea why this is the case? Best Daniel

2 3