June 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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Get started with FFmpeg
by Jamenson Ferreira Espindula de Almeida Melo 26 Sep '18

26 Sep '18

Hi. I am new to FFmpeg. I am not an English native speaker. I am from Brazil, South America. I would like to get some tutorials about FFmpeg. The official documentation seems to me very abstract. In fact, I tried to convert some files from vob format to a file in mp4 format, but it did not work. It seems converting vob files to mp4 files is very very complex. Do you want to help me? Thank you. Jamenson Ferreira Espindula de Almeida Melo GNU/Linux user # 166197 https://linuxcounter.net/cert/166197.png Key fingerprint: 234D 1914 4224 7C53 BD13 6855 2AE0 25C0 08A8 6180

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Make Output Files More Digestible / Increase Output Compatibility
by Gabriel Balaich 20 Sep '18

20 Sep '18

Hey everyone, I've found that my video files captured with FFmpeg don't play nice with other programs. Many video players display incorrect duration, others allow no scrubbing, and some won't even open the files at all. But perhaps the most annoying incompatibility is with Adobe Premiere. Trying to edit the 3440x1440 footage outputted from FFMpeg in Premiere is almost impossible, it will play for around 2 seconds at a time before completely freezing up and just displaying one frame. It would appear it is only a problem with the preview in premiere as once I export the video everything plays back perfectly smooth. But the preview is obviously very important while editing video as you can't see if your edits look good otherwise. When I capture the same quality of footage (3440x1440, 100FPS, 250M bitrate) with something like OBS the files playback perfectly fine in every video player I test. And more importantly OBS output plays back very smooth in Adobe Premiere while editing. Now I am capturing MP4 not TS in OBS but when I encode MP4 with FFmpeg it screws with syncronization and I can confirm it doesn't improve compatability, at least in my experience. So I was just wondering if anyone knows a way to increase compatability or make the output files more digestible. My specs aren't an issue I'm running a 7900x, GTX 1080, 64GB of RAM, and store all the footage on a PCIe M.2 drive (1000mb+ read and write). And as mentioned before OBS videos of the same quality, file type, and sometimes even of larger size work perfectly fine. I wouldn't even have a problem re-encoding the files afterwards, but would prefer finding a solution that allows the initial output files to play nice with other programs, thanks for any advice. Output file I'm having the most problems with: ffmpeg -loglevel error -stats -y -thread_queue_size 9999 -indexmem 9999 -guess_layout_max 0 -f dshow -video_size 3440x1440 -rtbufsize 2147.48M -framerate 200 -pixel_format nv12 ^ -i video="Video (00 Pro Capture HDMI 4K+)":audio="SPDIF/ADAT (1+2) (RME Fireface UC)" ^ -map 0:0,0:1 -map 0:1 -c:v h264_nvenc -r 100 -rc-lookahead 200 -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=0.035, asetpts=PTS-STARTPTS, aresample=async=250" -vsync 1 -ss 00:00:01.096 ^ -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\PC\PC\PC%02d.ts Full block: ffmpeg -loglevel error -stats -y -thread_queue_size 9999 -indexmem 9999 -guess_layout_max 0 -f dshow -rtbufsize 2147.48M ^ -i audio="Analog (3+4) (RME Fireface UC)" ^ -thread_queue_size 9999 -indexmem 9999 -guess_layout_max 0 -f dshow -rtbufsize 2147.48M ^ -i audio="ADAT (3+4) (RME Fireface UC)" ^ -thread_queue_size 9999 -indexmem 9999 -guess_layout_max 0 -f dshow -video_size 3440x1440 -rtbufsize 2147.48M ^ -framerate 200 -pixel_format nv12 -i video="Video (00 Pro Capture HDMI 4K+)":audio="SPDIF/ADAT (1+2) (RME Fireface UC)" ^ -thread_queue_size 9999 -indexmem 9999 -guess_layout_max 0 -f dshow -video_size 1920x1080 -rtbufsize 2147.48M ^ -framerate 60 -pixel_format yuyv422 -i video="Game Capture HD60 Pro (Video) (#01)":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 ^ -map 4,0 -map 0 -c:v libx264 -r 25 -rc-lookahead 50 -forced-idr 1 -strict_gop 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 -vf "fps=25" -af "aresample=async=250" -vsync 1 -ss 00:00:02.456 ^ -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\PC\Theirs\TPC%02d.ts ^ -map 4,1 -map 1 -c:v libx264 -r 25 -rc-lookahead 50 -forced-idr 1 -strict_gop 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 -vf "fps=25" -af "aresample=async=250" -vsync 1 -ss 00:00:01.840 ^ -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\PC\Soundboard\SPC%02d.ts ^ -map 2:0,2:1 -map 2:1 -c:v h264_nvenc -r 100 -rc-lookahead 200 -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=0.035, asetpts=PTS-STARTPTS, aresample=async=250" -vsync 1 -ss 00:00:01.096 ^ -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\PC\PC\PC%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 yuv420p -b:v 40M -minrate 40M -maxrate 40M -bufsize 40M ^ -c:a aac -ar 44100 -b:a 384k -ac 2 -af "pan=mono|c0=c0, adelay=120|120, aresample=async=250" -vsync 1 ^ -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\PC\Camera\CPC%02d.ts

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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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Anyone having success capturing hours of 4k video, reliably and with low loss, using ffmpeg?
by Jim DeLaHunt 08 Aug '18

08 Aug '18

Hello, folks: I'm working with a video production company that is thinking about using ffmpeg for a field production. They set up a 4k camera on top of a building (have electricity, but limited internet), and they need to capture 6-12 hours of 4k 29.92fps video from that camera reliably and with very low quality loss. We know we can use ffmpeg to excerpt and zoom in on parts of the main video afterwards, if the recording system doesn't crash in the middle of the shoot, and if the main video retains quality. Vendor A seems to be saying, buy our expensive system, and it will capture the video and put it on a file server which your computer can access. But my boss is saying, can't ffmpeg just capture and write out the video, and save us the expense of Vendor A? We tried a simple experiment. We set up a 4K camera in the office. A straightforward ffmpeg invocation did capture a couple of minutes of video. But there were nasty artifacts, such as 25% of the pixels being corrupted in the first second or so, and occasional speckles of corruption. This tells me that reliable capture by ffmpeg is easier than "impossible", but harder than "works without even trying". Is anyone on this list capturing hours of 4k video, reliably and with low quality loss? Or, has anyone tried it and discovered that it's really, really hard to do it well? Is ffmpeg up to the task of running for 12 hours of capturing 4k video without crashing? What bottlenecks or weak points in our capture card or computer do we need to watch out for? I looked through the archives of /ffmpeg-user/. The most relevant thread I found was this: "4K 60Hz Directshow Video Capture"[1] (Feb 2018). The conclusion I drew from this thread was circumstantial evidence that my task might be reasonable for short periods, and no evidence about longer periods like 12 hours. a) they are trying to capture 60Hz, and we only want 30Hz. b) it really matters that the capture card exports something which is accepted by the ffmpeg code you use. c) disk I/O is a primary bottleneck, CPU speed is a secondary bottleneck. d) the right GPU card can do the encoding, relieving the CPU speed bottleneck. The drawback of /ffmpeg-user/ is that generally people post here about problems, not about successes. If there are people quietly having success with ffmpeg for 12 hours continuously or ffmpeg for 4K video at 29.92 fps, I'd love to hear it. [1] <https://ffmpeg.org/pipermail/ffmpeg-user/2018-February/038895.html> -- --Jim DeLaHunt, jdlh(a)jdlh.com http://blog.jdlh.com/ (http://jdlh.com/) multilingual websites consultant 355-1027 Davie St, Vancouver BC V6E 4L2, Canada Canada mobile +1-604-376-8953

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how to use Intel qsv codec for ffmpeg 4.0
by qw 15 Jul '18

15 Jul '18

Hi, I want to use Intel qsv encoder/decoder for ffmpeg 4.0. I use '--enable-libmfx' to configure ffmpeg, and build it. What's ffmpeg command to use Intel qsv h.264 encoder/decoder to do transcoding? Thanks! Regards Andrew

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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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