February 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

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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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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/

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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

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problem transcoding raw g729 to wav format
by Dave Horton 20 Aug '18

20 Aug '18

I am trying to decode a raw g729 input stream to wav format like so: ffmpeg -loglevel debug -threads 1 -y -acodec g729 -f g729 -i pipe:0 -ar 44100 -acodec pcm_s32le -f wav pipe:1 As you can see, the raw g729 is piped to stdin and the wav output should be piped to stdout. I am seeing this output / errors in the log, and I get an empty wav file: ffmpeg version N-87327-g18821e3ba1-static http://johnvansickle.com/ffmpeg/ Copyright (c) 2000-2017 the FFmpeg developers built with gcc 6.4.0 (Debian 6.4.0-5) 20170906 configuration: --enable-gpl --enable-version3 --enable-static --disable-debug --disable-ffplay --disable-indev=sndio --disable-outdev=sndio --cc=gcc-6 --enable-fontconfig --enable-frei0r --enable-gnutls --enable-gray --enable-libfribidi --enable-libass --enable-libvmaf --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librubberband --enable-librtmp --enable-libsoxr --enable-libspeex --enable-libvorbis --enable-libopus --enable-libtheora --enable-libvidstab --enable-libvo-amrwbenc --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxvid --enable-libzimg libavutil 55. 75.100 / 55. 75.100 libavcodec 57.106.101 / 57.106.101 libavformat 57. 82.100 / 57. 82.100 libavdevice 57. 8.101 / 57. 8.101 libavfilter 6.105.100 / 6.105.100 libswscale 4. 7.103 / 4. 7.103 libswresample 2. 8.100 / 2. 8.100 libpostproc 54. 6.100 / 54. 6.100 Splitting the commandline. Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) with argument 'debug’. Reading option '-threads' ... matched as AVOption 'threads' with argument '1’. option '-y' ... matched as option 'y' (overwrite output files) with argument '1’. Reading option '-acodec' ... matched as option 'acodec' (force audio codec ('copy' to copy stream)) with argument 'g729’. Reading option '-f' ... matched as option 'f' (force format) with argument 'g729’. Reading option '-i' ... matched as input url with argument 'pipe:0’. Reading option '-ar' ... matched as option 'ar' (set audio sampling rate (in Hz)) with argument '44100’. Reading option '-acodec' ... matched as option 'acodec' (force audio codec ('copy' to copy stream)) with argument 'pcm_s32le’. Reading option '-f' ... matched as option 'f' (force format) with argument 'wav’. Reading option 'pipe:1' ... matched as output url. Finished splitting the commandline. Parsing a group of options: global . Applying option loglevel (set logging level) with argument debug. Applying option y (overwrite output files) with argument 1. Successfully parsed a group of options. Parsing a group of options: input url pipe:0. Applying option acodec (force audio codec ('copy' to copy stream)) with argument g729. Applying option f (force format) with argument g729. Successfully parsed a group of options. Opening an input file: pipe:0. [g729 @ 0x3e25da0] Opening 'pipe:0' for reading [pipe @ 0x3e265a0] Setting default whitelist 'crypto' [g729 @ 0x3e25da0] Before avformat_find_stream_info() pos: 0 bytes read:0 seeks:0 nb_streams:1 [g729 @ 0x3e25da0] After avformat_find_stream_info() pos: 0 bytes read:0 seeks:0 frames:0 Guessed Channel Layout for Input Stream #0.0 : mono Input #0, g729, from 'pipe:0': Duration: N/A, bitrate: 8 kb/s Stream #0:0, 0, 1/8000: Audio: g729, 8000 Hz, mono, s16 Successfully opened the file. Parsing a group of options: output url pipe:1. Applying option ar (set audio sampling rate (in Hz)) with argument 44100. Applying option acodec (force audio codec ('copy' to copy stream)) with argument pcm_s32le. Applying option f (force format) with argument wav. Successfully parsed a group of options. Opening an output file: pipe:1. [pipe @ 0x3e2aba0] Setting default whitelist ‘crypto' Successfully opened the file. Stream mapping: Stream #0:0 -> #0:0 (g729 (native) -> pcm_s32le (native)) cur_dts is invalid (this is harmless if it occurs once at the start per stream) No more output streams to write to, finishing. Finishing stream 0:0 without any data written to it. detected 8 logical cores [graph_0_in_0_0 @ 0x3e2daa0] Setting 'time_base' to value '1/8000’ [graph_0_in_0_0 @ 0x3e2daa0] Setting 'sample_rate' to value '8000’ graph_0_in_0_0 @ 0x3e2daa0] Setting 'sample_fmt' to value 's16’ [graph_0_in_0_0 @ 0x3e2daa0] Setting 'channel_layout' to value '0x4’ [graph_0_in_0_0 @ 0x3e2daa0] tb:1/8000 samplefmt:s16 samplerate:8000 chlayout:0x4 format_out_0_0 @ 0x3e25ca0] Setting 'sample_fmts' to value 's32’ [format_out_0_0 @ 0x3e25ca0] Setting 'sample_rates' to value '44100’ [format_out_0_0 @ 0x3e25ca0] auto-inserting filter 'auto_resampler_0' between the filter 'Parsed_anull_0' and the filter 'format_out_0_0' [AVFilterGraph @ 0x3e2c640] query_formats: 4 queried, 6 merged, 3 already done, 0 delayed [auto_resampler_0 @ 0x3e3fd20] [SWR @ 0x3e400c0] Using fltp internally between filters [auto_resampler_0 @ 0x3e3fd20] ch:1 chl:mono fmt:s16 r:8000Hz -> ch:1 chl:mono fmt:s32 r:44100Hz Output #0, wav, to 'pipe:1’: Metadata: ISFT : Lavf57.82.100 Stream #0:0, 0, 1/44100: Audio: pcm_s32le ([1][0][0][0] / 0x0001), 44100 Hz, mono, s32, 1411 kb/s Metadata: encoder : Lavc57.106.101 pcm_s32le size= 0kB time=00:00:00.00 bitrate=N/A speed= 0x video:0kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown Input file #0 (pipe:0): Input stream #0:0 (audio): 0 packets read (0 bytes); 0 frames decoded (0 samples); Total: 0 packets (0 bytes) demuxed Output file #0 (pipe:1): Output stream #0:0 (audio): 0 frames encoded (0 samples); 0 packets muxed (0 bytes); Total: 0 packets (0 bytes) muxed Output file is empty, nothing was encoded (check -ss / -t / -frames parameters if used) 0 frames successfully decoded, 0 decoding errors Is there something wrong with my command? Interestingly, if I transcode to flac instead of wav, while I get similar errors, an accurate output file _is_ created in that case. Dave

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FFserver doesn't work with HLS
by SD76 11 Jul '18

11 Jul '18

Hi, I try since a few days to stream a live with ffserver and HLS. My ffserver.conf run with ffserver -f ffserver.conf -d --------------- HTTPPort 8081 MaxClients 10 MaxBandwidth 5000000 <Feed plop.ffm> File /tmp/plop.ffm FileMaxSize 100M </Feed> <Stream s> Format status </Stream> <Stream plop.m3u8> Feed plop.ffm Format HLS VideoCodec libx264 VideoFrameRate 30 VideoBitRate 128 VideoSize 704*480 #AVOptionVideo hls_time 10 #AVOptionVideo hls_list_size 0 AVOptionVideo flags +global_header NoAudio </Stream> --------------- If i enable AVOptionVideo hls_time 10 and AVOptionVideo hls_list_size 0 ffserver say : Option not found: 'hls_time' Option not found: 'hls_list_size' I launch for send stream this command : ffmpeg -r 25 -s 640x480 -f video4linux2 -i /dev/video0 -hls_time 10 -hls_list_size 6 -hls_wrap 10 -start_number 1 http://192.168.0.16:8081/plop.ffm I use a remote player on my galaxy tab: <video controls width="480" height="270" src=" http://192.168.0.16:8081/plop.m3u8"> The stream doesn't work and ffserver say: Sun Nov 29 18:15:15 2015 [hls @ 0x1a20810]failed to rename file .tmp to Do HLS works with ffserver ? How ? I use ffserver version 2.8.1-1~bpo8+1 and ffmpeg version 2.8.1-1~bpo8+ on debian 8. Regard, Ben

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ffmpeg remuxing from MTS to MP4 now produces an unreadable audio stream
by Galen Menzel 20 Jun '18

20 Jun '18

Hi all, I have been using ffmpeg regularly for a couple years to remux MTS files from my video camera into mp4 files. I use the following simple command: ``` ffmpeg -i [MTS_FILE] -acodec copy -vcodec copy [MP4_FILE] ``` Although the command I use hasn’t changed, recently this has stopped working. The resulting mp4 now contains an audio stream in a format that VLC (and Quicktime Player) can’t play. (VLC gives the following error: `VLC could not decode the format "AC-3" (No description for this codec)`) Concretely, here is the audio stream information of an MTS file I’m working with, which plays just fine: Stream #0:1[0x1100]: Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, stereo, fltp, 256 kb/s Here is the audio stream information from the mp4 of this file, which I remuxed last year, and which also plays fine: Stream #0:1(und): Audio: ac3 (ac-3 / 0x332D6361), 48000 Hz, stereo, fltp, 256 kb/s (default) And here is the audio stream information from the mp4 of this file when I remux it now using the same command, which does not play: Stream #0:1(und): Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, stereo, fltp, 256 kb/s (default) As you can see, the file that I get now has the same encoding as the MTS file, but does not play, even though the MTS file does. And the file I generated last year has a different audio encoding (even though the commands that created these files were identical), but it plays fine. If I first do what should be a vacuous remux of the MTS file with ffmpeg -i [INPUT_MTS_FILE] -acodec copy -vcodec copy [OUTPUT_MTS_FILE] the resulting MTS file has the following audio stream information: Stream #0:1[0x101]: Audio: ac3 ([129][0][0][0] / 0x0081), 48000 Hz, stereo, fltp, 256 kb/s And when I remux *this* file into an mp4 container, the resulting file plays fine, and has the same audio stream as the file created last year: Stream #0:1(und): Audio: ac3 (ac-3 / 0x332D6361), 48000 Hz, stereo, fltp, 256 kb/s (default) But I would really rather not remux everything twice. I am using the following version of ffmpeg, which I installed with macports: ``` ffmpeg version 3.4.2 Copyright (c) 2000-2018 the FFmpeg developers built with Apple LLVM version 9.0.0 (clang-900.0.39.2) configuration: --prefix=/opt/local --enable-swscale --enable-avfilter --enable-avresample --enable-libmp3lame --enable-libvorbis --enable-libopus --enable-librsvg --enable-libtheora --enable-libopenjpeg --enable-libmodplug --enable-libvpx --enable-libsoxr --enable-libspeex --enable-libass --enable-libbluray --enable-lzma --enable-gnutls --enable-fontconfig --enable-libfreetype --enable-libfribidi --disable-jack --disable-libopencore-amrnb --disable-libopencore-amrwb --disable-libxcb --disable-libxcb-shm --disable-libxcb-xfixes --disable-indev=jack --enable-opencl --disable-outdev=xv --enable-audiotoolbox --enable-videotoolbox --enable-sdl2 --mandir=/opt/local/share/man --enable-shared --enable-pthreads --cc=/usr/bin/cc --arch=x86_64 --enable-x86asm --enable-libx265 --enable-gpl --enable-postproc --enable-libx264 --enable-libxvid --enable-nonfree --enable-libfdk-aac 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 libavresample 3. 7. 0 / 3. 7. 0 libswscale 4. 8.100 / 4. 8.100 libswresample 2. 9.100 / 2. 9.100 libpostproc 54. 7.100 / 54. 7.100 ``` This is the first time I’ve noticed this issue, but the last time I remuxed files (with no issue) was last October, so this has manifested sometime in the last four months or so. Does anyone have any ideas as to what is going on here? What do these various name/number pairs for the ac3 encoding mean? (e.g., `ac-3 / 0x332D6361` vs `AC-3 / 0x332D4341` vs `[129][0][0][0] / 0x0081`. I’d appreciate any help or insight on this issue. Thanks! Best regards, Galen

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4K 60Hz Directshow Video Capture
by Alex P 09 Jun '18

09 Jun '18

Windows 10 Intel i7-8700K GTX 1050Ti 16GB DDR4 SATA Samsung Evo I'm using a Yuan 4K60 capture card with the end goal of capturing the video in a lossless format. I need it to be lossless as the clips will be used to test hardware h264 encoders. Product page: http://www.yuan.com.tw/products/capture/4k/sc560n1_lv_hdmi2.htm Directshow is supported. Command: ffmpeg -f dshow -video_size 3840x2160 -framerate 59.9 -rtbufsize 2100000000 -pixel_format bgr24 -i video="MZ0380 PCI, Analog 01 Capture" -c:v libx265 -x265-params lossless=1 out1.avi I have also tried writing to null: ffmpeg -f dshow -video_size 3840x2160 -framerate 59.9 -rtbufsize 2100000000 -pixel_format bgr24 -i video="MZ0380 PCI, Analog 01 Capture" -c:v libx265 -x265-params lossless=1 -f null - I get the same error (see attatched) and I have to force quit by pressing Ctrl-C a bunch. Error also occurs if I write to a RAM disk. [dshow @ 000001dda20ca3a0] real-time buffer [MZ0380 PCI, Analog 01 Capture] [video input] too full or near too full (63% of size: 2100000000 [rtbufsize parameter])! frame dropped! Any help would be appreciated. 1080p capture right to rawvideo is perfect. I would like to use NVENC in lossless mode if anyone has experience with that.

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Video scaling, padding and cropping together question
by TPCffmpeg＠mklab.ph.rhul.ac.uk 31 Mar '18

31 Mar '18

I am attempting to down-convert HD video (1280x720 with 16:9 aspect ratio) to 720x576 for use on an old PAL 4x3 display. I found two days of doing this, using, (1) -filter:v scale=iw*min(720/iw\,576/ih):ih*min(720/iw\,576/ih), pad=720:576:(720-iw*min(720/iw\,576/ih))/2:(576-ih*min(720/iw\,576/ih))/2 -target pal-dvd (2) -filter:v 'crop=ih/3*4:ih' -target pal-dvd Method 1 scales and pads to letterbox format, so I get the whole of the original picture width with bars above and below. Method 2 scales and crops, so I get to use the whole 4x3 display screen height (better resolution) but lose the LH & RH picture edges via the cropping. The thing I am trying to achieve is a compromise between these two extremes, eg. a method which gives less cropping at the LH & RH picture edges for the price of a little padding above and below the picture. I'm presuming this (ideally) involves scaling, padding and cropping together in a single filter. Could anyone show how to do this or point me in the right direction? Many thanks Tom Crane -- Tom Crane, Dept. Physics, Royal Holloway, University of London, Egham Hill, Egham, Surrey, TW20 0EX, England. Email: T.Crane(a)rhul.ac.uk Fax: +44 (0) 1784 472794

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Low latency H.264 streaming with ffmpeg
by Greger Burman 27 Mar '18

27 Mar '18

I would like to hear from anyone who has experience with low latency video streaming and ffmpeg. In my application I take video from a PCIe capture device and stream live over network. The end to end latency must be below 200ms otherwise it is not usable. Let us assume that the network latency is near zero. This question is about encoder+decoder+buffering latency. Very low latency can be achieved when using the x264 encoder by using settings like tune zerolatency. From my own testing with x264 (without ffmpeg) I can confirm that this works. I have measured approx 90 ms (3 frames) of latency added by encoder+decoder combined. Awesome. Sadly I have been unable to get anywhere near those numbers when using ffmpeg and libx264. As player I use either ffplay or a dot-net player built on the ffmpeg libraries, with about equal results. The lowest end to end latency I have seen is 400-500ms. That includes capture latency also. Yes, there are many unknowns here. To help pin point the problem I used mjpeg streaming for comparison. Same everything except I change ffmpeg vcodec to mjpeg and format to mjpeg. I change nothing with ffplay. With mjpeg I now get approx 150ms end to end latency. Observation: x264 codec + mpegts format compared to mjpeg codec + mjpeg format adds about 300 ms (10 frames) latency. I have tried every ffplay-, ffmpeg- and libx264 setting that I believe relevant. Most have not made any noticable difference in latency. I will not list a lot of settings here. Here are the basic settings and commands that I have used. I can post more details on request. x264-preset: vcodec=libx264 thread_type=slice slices=1 # x264 profile=baseline level=32 preset=superfast tune=zerolatency intra-refresh=1 crf=15 x264-params=vbv-maxrate=5000:vbv-bufsize=1:slice-max-size=1500:keyint=60 $ ffmpeg -r 30 -f dshow -i video="devicename" -pix_fmt yuv420p -an -vpre x264-preset -f mpegts udp://127.0.0.1:8888 $ ffplay -fflags nobuffer -flags low_delay -framedrop -strict experimental udp://127.0.0.1:8888 Everything is run on Windows 7 64-bit, with 32-bit executables. ffmpeg version N-76539-g480bad7 Copyright (c) 2000-2015 the FFmpeg developers built with gcc 5.2.0 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libdcadec --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-decklink --enable-zlib libavutil 55. 5.100 / 55. 5.100 libavcodec 57. 15.100 / 57. 15.100 libavformat 57. 14.100 / 57. 14.100 libavdevice 57. 0.100 / 57. 0.100 libavfilter 6. 15.100 / 6. 15.100 libswscale 4. 0.100 / 4. 0.100 libswresample 2. 0.101 / 2. 0.101 libpostproc 54. 0.100 / 54. 0.100 cheers -- Greger Burman

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