March 2017 - ffmpeg-user - lists.ffmpeg.org

Compiling FFMPEG with ffplay support
by Stas Oskin 13 Jun '17

13 Jun '17

Hi. Did the compilation parameters changed in recent FFMPEG version? Configuring the FFMPEG like this: ./configure --prefix=/usr --enable-pthreads Does not produce ffplay anymore. I tried to add --enable-ffplay, without any result. Any idea what I'm missing? Thanks.

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Create thumbnails from RTMP stream
by Grant Curtis 09 Jun '17

09 Jun '17

If I use Flash Media Live Encoder to send a stream to an RTMP server, and then, on that server, run the below command, I can create thumbnails from the stream. But when I try to do the same when I stream the file to the server using ffmpeg, I get errors, and no thumbnails are created Server command and output: $ /usr/bin/ffmpeg -report -i "rtmp://localhost:1935/osc-live-packager/cumin?offsetFromLive=10,userID=testUser1/mp4:rebroadcast_testUser1" -y -r 29.97 -threads 4 -c:v copy -c:a copy /mnt_ephemeral1/media/cumin/cumin_1000.mp4 -filter:v "scale=w=178:h=100" -f image2 -vf "fps=fps=1/5" "/mnt_ephemeral1/media/cumin/cumin/thumbs/thumb_%04d.png" ffmpeg started on 2016-04-25 at 18:28:39 Report written to "/var/log/ffmpeg/ffmpeg-20160425-182839.log" ffmpeg version 2.7.2-static http://johnvansickle.com/ffmpeg/ Copyright (c) 2000-2015 the FFmpeg developers built with gcc 4.9.3 (Debian 4.9.3-1) configuration: --enable-gpl --enable-version3 --disable-shared --disable-debug --enable-runtime-cpudetect --enable-libmp3lame --enable-libx264 --enable-libx265 --enable-libwebp --enable-libspeex --enable-libvorbis --enable-libvpx --enable-libfreetype --enable-fontconfig --enable-libxvid --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-gray --enable-libopenjpeg --enable-libopus --enable-libass --enable-gnutls --enable-libvidstab --enable-libsoxr --cc=gcc-4.9 libavutil 54. 27.100 / 54. 27.100 libavcodec 56. 41.100 / 56. 41.100 libavformat 56. 36.100 / 56. 36.100 libavdevice 56. 4.100 / 56. 4.100 libavfilter 5. 16.101 / 5. 16.101 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 2.100 / 1. 2.100 libpostproc 53. 3.100 / 53. 3.100 [NULL @ 0x31b65e0] missing picture in access unit with size 110664 [h264 @ 0x31b65e0] no frame! [h264 @ 0x31b65e0] missing picture in access unit with size 2338 [h264 @ 0x31b65e0] no frame! [h264 @ 0x31b65e0] missing picture in access unit with size 2645 [h264 @ 0x31b65e0] no frame! [h264 @ 0x31b65e0] missing picture in access unit with size 2600 [h264 @ 0x31b65e0] no frame! [h264 @ 0x31b65e0] missing picture in access unit with size 8421 [h264 @ 0x31b65e0] no frame! [h264 @ 0x31b65e0] missing picture in access unit with size 2895 [h264 @ 0x31b65e0] no frame! [h264 @ 0x31b65e0] missing picture in access unit with size 1759 [h264 @ 0x31b65e0] no frame! [h264 @ 0x31b65e0] missing picture in access unit with size 776 [h264 @ 0x31b65e0] no frame! [h264 @ 0x31b65e0] missing picture in access unit with size 13391 [h264 @ 0x31b65e0] no frame! -truncated a bunch more similar messages- [flv @ 0x31b3c00] decoding for stream 0 failed [flv @ 0x31b3c00] Could not find codec parameters for stream 0 (Video: h264, none): unspecified size Consider increasing the value for the 'analyzeduration' and 'probesize' options Input #0, flv, from 'rtmp://localhost:1935/osc-live-packager/cumin?offsetFromLive=10,userID=testUser1/mp4:rebroadcast_testUser1': Metadata: moovPosition : 36 avcprofile : 100 avclevel : 31 videoframerate : 25 audiochannels : 2 timescale : 1000 length : 839 sampletype : .mp3 Duration: 00:00:00.84, start: 0.040000, bitrate: N/A Stream #0:0: Video: h264, none, 25 tbr, 1k tbn, 2k tbc Stream #0:1: Subtitle: text Stream #0:2: Audio: mp3, 44100 Hz, stereo, s16p, 128 kb/s Output #1, image2, to '/mnt_ephemeral1/media/cumin/cumin/thumbs/thumb_%04d.png': Metadata: moovPosition : 36 avcprofile : 100 avclevel : 31 videoframerate : 25 audiochannels : 2 timescale : 1000 length : 839 sampletype : .mp3 Output file #1 does not contain any stream Received signal 2: terminating. --- Here's how I start the stream from my local machine: --- ffmpeg -re -i testVideo.mp4 -ar 44100 -c:v libx264 -f flv rtmp://xxxxxxxx:1935/xxxxx/xxxxx/xxxxx_1000.mp4 ffmpeg version 2.8.4 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-lzma --enable-decklink --enable-zlib libavutil 54. 31.100 / 54. 31.100 libavcodec 56. 60.100 / 56. 60.100 libavformat 56. 40.101 / 56. 40.101 libavdevice 56. 4.100 / 56. 4.100 libavfilter 5. 40.101 / 5. 40.101 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 2.101 / 1. 2.101 libpostproc 53. 3.100 / 53. 3.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'testVideo.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 creation_time : 1970-01-01 00:00:00 encoder : Lavf53.24.2 Duration: 00:01:02.32, start: 0.000000, bitrate: 1347 kb/s Stream #0:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], 959 kb/s, 25 fps, 25 tbr, 12800 tbn, 50 tbc (default) Metadata: creation_time : 1970-01-01 00:00:00 handler_name : VideoHandler Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, 5.1, fltp, 383 kb/s (default) Metadata: creation_time : 1970-01-01 00:00:00 handler_name : SoundHandler [libx264 @ 03052340] using SAR=1/1 [libx264 @ 03052340] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX FMA3 AVX2 LZCNT BMI2 [libx264 @ 03052340] profile High, level 3.1 [libx264 @ 03052340] 264 - core 148 r2638 7599210 - H.264/MPEG-4 AVC codec - Copyleft 2003-2015 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=12 lookahead_threads=2 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00 Output #0, flv, to 'rtmp://xxxxxx:1935/xxxx/xxxxx/xxxxx_1000.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 encoder : Lavf56.40.101 Stream #0:0(und): Video: h264 (libx264) ([7][0][0][0] / 0x0007), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], q=-1--1, 25 fps, 1k tbn, 25 tbc (default) Metadata: creation_time : 1970-01-01 00:00:00 handler_name : VideoHandler encoder : Lavc56.60.100 libx264 Stream #0:1(und): Audio: mp3 (libmp3lame) ([2][0][0][0] / 0x0002), 44100 Hz, stereo, fltp (default) Metadata: creation_time : 1970-01-01 00:00:00 handler_name : SoundHandler encoder : Lavc56.60.100 libmp3lame Stream mapping: Stream #0:0 -> #0:0 (h264 (native) -> h264 (libx264)) Stream #0:1 -> #0:1 (aac (native) -> mp3 (libmp3lame)) Press [q] to stop, [?] for help frame= 1336 fps= 25 q=28.0 size= 7310kB time=00:00:53.63 bitrate=1116.6kbits/s

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Re: [FFmpeg-user] FFMPEG & Panasonic OP-1B MXF decode issue
by Jeff Smith 06 Jun '17

06 Jun '17

> 2017-03-10 9:20 GMT+01:00 <jeff at jksmith.com <http://ffmpeg.org/mailman/listinfo/ffmpeg-user>>: > >/Happy to provide a sample. Is there an FTP site or something I /> >/can upload it to? /> > You can usehttp://streams.videolan.org/upload/ but you absolutely > have to provide the file name, the service is extremely impractical. > > Carl Eugen Hi Carl, I'm assuming from radio silence that you couldn't get the sample I uploaded. I've posted it on an anonymous FTP server here: ftp://5.56.49.144/085FV4BR.MXF . Grateful if you could acknowledge if this works for you. Thanks, Jeff --- This email has been checked for viruses by Avast antivirus software. https://www.avast.com/antivirus

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Float formatting in drawtext
by Wolfgang Hugemann 29 May '17

29 May '17

Hey, seems to be an easy question, but I have been searching for the answer quite a while without success: If I perform a float calculation via text=%{expr\:whatever} is there any way to format the output the printf way? I know, the result could be converted to integer and be formatted via text=%{eif\:whatever\:d} but this is restricted to integer results. Wolfgang Hugemann

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How do I encode DVB Subtitles?
by Wagner Patriota 02 May '17

02 May '17

*QUESTION 1:* How do I add a STR file to a MPEG-TS file with the "DVB subtitle codec"? If so, could you give me a sample command line? Thank. I am trying: ffmpeg -i video.ts *-i subtitle.srt* -vcodec copy -acodec copy *-scodec dvbsub* output.ts The stream is created because I can see it with MediaInfo, but the subtitles don't appear! *QUESTION 2:* How do I use the command *slang*? I try to use it, but ffmpeg says the command doesn't exist. Thank you.

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DVB H264 chunk transcoding => merging in MP4 => out of sync
by ffmpeg＠me.com 10 Apr '17

10 Apr '17

Viewing on VLC works great. Audio/Video perfectly in sync. Viewing in Quicktime (Version 10.4 MacOS sierra, Quicktime 7) the video get's out of sync after 3/4 minutes. Scrubbing in the timeline will fix the problem, then after 3/4 minutes I see the synchronization problem again. Using current FFmpeg version to transcode into H.264 and AAC TS chunks. ffmpeg -i "udp://224.5.2.5:10001?overrun_nonfatal=1&fifo_size=50000000" -map 0:3 -map 0:4 -s 1280x720 -acodec aac -b:a 192k -ac 2 -ar 48000 -vcodec libx264 -preset:v medium -r 50 -g 100 -vb 3200000 -y -fflags +genpts -f segment -segment_time 60 -segment_atclocktime 1 -strftime 1 "/videofiles/TV_HD_%Y-%m-%d_%H-%M-%S.ts Everything works fine... chunks are created. No errors in output. Audio/video TS pieces in sync. #### Details of a chunk file: ffprobe /videofiles/TV_HD_2017-03-13_14-25-00.ts ffprobe version 3.2.4-tessus Copyright (c) 2007-2017 the FFmpeg developers built with Apple LLVM version 8.0.0 (clang-800.0.42.1) configuration: --cc=/usr/bin/clang --prefix=/opt/ffmpeg --extra-version=tessus --enable-avisynth --enable-fontconfig --enable-gpl --enable-libass --enable-libbluray --enable-libfreetype --enable-libgsm --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopus --enable-libschroedinger --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzmq --enable-version3 --disable-ffplay --disable-indev=qtkit --disable-indev=x11grab_xcb 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 libswscale 4. 2.100 / 4. 2.100 libswresample 2. 3.100 / 2. 3.100 libpostproc 54. 1.100 / 54. 1.100 Input #0, mpegts, from '/Users/Andreas/Desktop/TEST/ARD_HD_2017-03-13_14-25-00.ts': Duration: 00:01:00.04, start: 115.402667, bitrate: 3720 kb/s Program 1 Metadata: service_name : Service01 service_provider: FFmpeg Stream #0:0[0x100]: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p(progressive), 1280x720 [SAR 1:1 DAR 16:9], 50 fps, 50 tbr, 90k tbn, 100 tbc Stream #0:1[0x101](deu): Audio: aac (LC) ([15][0][0][0] / 0x000F), 48000 Hz, stereo, fltp, 204 kb/s #### Merging together: ffmpeg -f concat -safe 0 -i files.txt -y -metadata creation_time=now -c:v copy -c:a copy -fflags +genpts -movflags faststart -use_editlist 0 -f mp4 tvexport.mp4 #### and output: ffmpeg version 3.2.4-tessus Copyright (c) 2000-2017 the FFmpeg developers built with Apple LLVM version 8.0.0 (clang-800.0.42.1) configuration: --cc=/usr/bin/clang --prefix=/opt/ffmpeg --extra-version=tessus --enable-avisynth --enable-fontconfig --enable-gpl --enable-libass --enable-libbluray --enable-libfreetype --enable-libgsm --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopus --enable-libschroedinger --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzmq --enable-version3 --disable-ffplay --disable-indev=qtkit --disable-indev=x11grab_xcb 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 libswscale 4. 2.100 / 4. 2.100 libswresample 2. 3.100 / 2. 3.100 libpostproc 54. 1.100 / 54. 1.100 Input #0, concat, from 'files.txt': Duration: N/A, start: 0.000000, bitrate: N/A Stream #0:0: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p(progressive), 1280x720 [SAR 1:1 DAR 16:9], 50 fps, 50 tbr, 90k tbn, 100 tbc Stream #0:1(deu): Audio: aac (LC) ([15][0][0][0] / 0x000F), 48000 Hz, stereo, fltp, 169 kb/s Output #0, mp4, to 'work_tmp0.mp4': Metadata: creation_time : now encoder : Lavf57.56.101 Stream #0:0: Video: h264 (High) ([33][0][0][0] / 0x0021), yuv420p(progressive), 1280x720 [SAR 1:1 DAR 16:9], q=2-31, 50 fps, 50 tbr, 90k tbn, 90k tbc Stream #0:1(deu): Audio: aac (LC) ([64][0][0][0] / 0x0040), 48000 Hz, stereo, 169 kb/s Stream mapping: Stream #0:0 -> #0:0 (copy) Stream #0:1 -> #0:1 (copy) Press [q] to stop, [?] for help [mp4 @ 0x7fff37800600] Starting second pass: moving the moov atom to the beginning of the filex frame=24000 fps=9825 q=-1.0 Lsize= 199633kB time=00:08:00.20 bitrate=3405.6kbits/s speed= 197x video:187817kB audio:11755kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.030912% What could be the problem?

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Can't get a/v synchronisation right with mpeg transport stream that has gaps in it
by Erik Slagter 03 Apr '17

03 Apr '17

Hi people, I'm having this problem for years and years (so it's not a recent problem or regression). I make recordings on my enigma2 based settopbox. It saves the recordings as the original transport stream, but with filtering of only the requested service pids. There is a feature to edit the recorded services, in a basic way. Very helpful to edit out commercials. Problem is that it makes cuts in a very "rude" way, it justs cuts the file at a certain position, in the middle of all kinds of open packets and streams. On playback it isn't such a big deal, because the damaged packets are discarded and the DVB hardware resyncs on the timestamps pretty quickly. After the crude editing, I like to perform a real editing run, using kdenlive (or any other NLVE, for that case). I've found that all of them have problems when fed with a "damaged" transport stream, like this. So what I do, is transcode the stream to something with mjpeg and pcm codecs, intra frames only, very easy on the editor. This is the problem: after every timestamp discontinuity ("cut" or "gap"), ffmpeg resyncs audio and video BUT it doesn't do it 100% correctly. Sound remains a bit "in front" of the video, skewed a bit. Also, it gets worse at every cut, everytime about 200-300 ms is added to the skew. That's not what I'd expect? I've tried these options to force synchronisation: - -vsync 1 (not per sé required, but I want the audio and video streams in the container in such a way that they're not skewed, so if the editor decides to demux and throw away the timestamps, it still works) - -r 25 (material is 25 fps, tried with and without, doesn't matter) - filter:a " aresample=async=10000:min_comp=0.1:min_hard_comp=0.1:max_soft_comp=10000:first_pts=0" None of this helps. I can see ffmpeg dropping and dupping frames, but it doesn't help to fix the slow drift. I was under the impression that ffmpeg would automatically base the synchronisation of the (single) audio stream on the (single) video stream. Is this a correct assumption or do I need to specify an explicit "-map" statement? It looks like ffmpeg takes care that frames are dupped/dropped so the frame rate keeps "solid" 25 fps (or whatever the source frame rate is specified as or is specified with -r), but it seems it does not keep the audio stream synchronised. Thanks for help, Erik.

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T shirt
by David Whitaker 03 Apr '17

03 Apr '17

Hi, I am a big fan of ffmpeg, I like the logo so I decided to have a single white t shirt with the logo printed for my self. The company I asked to print for me wants me to have permission to use the logo first. Here is the logo I want to use https://upload.wikimedia.org/wikipedia/commons/thumb/7/76/FFmpeg_icon.svg/2… ------------------------------- Here is there contact info if needed Casey Maxwell CustomInk Satisfaction Assurance Associate 800-497-2067 service(a)customink.com http://www.customink.com FAST, FRIENDLY HELP 7 DAYS A WEEK Mon-Fri: 8am - 10:30pm Sat: 10am - 6pm Sun: 10am - 9pm (all times eastern) Helpful Links... Frequently Asked Questions: http://customink.com/help_center ------------------------------- If someone could help me find out if it possible to get permission to use this logo for 1 shirt id definitely appreciate the effort. Thanks Dave

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Building Error avcodec_send_packet
by יוסף אלון 03 Apr '17

03 Apr '17

I would like to compile to following program: With the flages: -lavcodec -lavutil -lavfilter -lavformat -lswscale -lz #include "libavcodec/avcodec.h" #include "libavformat/avformat.h" #include "libswscale/swscale.h" #include <stdio.h> // The flush packet is a non-NULL packet with size 0 and data NULL int decode(AVCodecContext *avctx, AVFrame *frame, int *got_frame, AVPacket *pkt) { int ret; *got_frame = 0; if (pkt) { ret = avcodec_send_packet(avctx, pkt); // In particular, we don't expect AVERROR(EAGAIN), because we read all // decoded frames with avcodec_receive_frame() until done. if (ret < 0) return ret == AVERROR_EOF ? 0 : ret; } ret = avcodec_receive_frame(avctx, frame); if (ret < 0 && ret != AVERROR(EAGAIN) && ret != AVERROR_EOF) return ret; if (ret >= 0) *got_frame = 1; return 0; } int main(int argc, char *argv[]) { return 0; But I receive: undefined reference to `avcodec_send_packet' undefined reference to `avcodec_receive_frame' ffmpeg version is: ffmpeg version 3.2.4 Copyright (c) 2000-2017 the FFmpeg developers built with gcc 5.4.0 (Ubuntu 5.4.0-6ubuntu1~16.04.4) 20160609 configuration: 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 libswscale 4. 2.100 / 4. 2.100 libswresample 2. 3.100 / 2. 3.100 Does some one know, Whats the problem? -- Yosef Alon

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Maximum argument length for tee pseudo muxer
by luke harrison 02 Apr '17

02 Apr '17

I'm preparing a script to take a feed coming in over SDI and encode it into multiple video and audio streams, all for distribution to different stream paths using the tee muxer. I'm sending quite a few different mixes of the streams to an RTMP server which means that my tee muxer is taking quite a large argument. My command line is shown below. ffmpeg -threads 0 -f decklink -i 'UltraStudio Mini Recorder@11' -loglevel debug \ -filter_complex "[0:v]split=2[VID1][VID2];[0:a]channelsplit=channel_layout=stereo[AL][AR]" -map '[AR]' -map '[AL]' -map '[VID1]' -map '[VID2]' -flags +global_header -c:v libx264 -profile:v high -preset veryfast -pix_fmt yuv420p -r 25 -s:v:0 640x360 -s:v:1 426x240 -b:v:0 484k -b:v:1 215k -bufsize 1500k -coder 1 -g 125 -keyint_min 25 -c:a aac -b:a 64k -ar 44100 -async 1 -rtmp_flashver "FMLE/3.0(compatible;FMSc/1.0)" -rtmp_live live -f tee "[select=\'a:0,v:0\':f=flv]'${entrypoint}${streampath1}${datavid11}'| \ [select=\'a:1,v:0\':f=flv]'${entrypoint}${streampath2}${datavid11}'| \ [select=\'a:0,v:1\':f=flv]'${entrypoint}${streampath1}${datavid12}'| \ [select=\'a:1,v:1\':f=flv]'${entrypoint}${streampath2}${datavid12}'| \ [select=\'a:0\':f=flv]'${entrypoint}${streampath1}${dataaud11}'| \ [select=\'a:1\':f=flv]'${entrypoint}${streampath2}${dataaud11}'" All of the output streams up until the last one (with a:1) connect to the server and begin streaming. The issue comes when ffmpeg tries to connect to the final stream. The stream path is cut off for some reason. This results in ffmpeg trying to connect to a stream path that is incomplete and therefore failing. [tcp @ 0x7fa0dc414180] Failed to resolve hostname p.epXXXXXX.i.ak: nodename nor servname provided, or not known [rtmp @ 0x7fa0dc414100] Cannot open connection tcp://p.epXXXXX.i.ak:1935 [tee @ 0x7fa0de801800] Slave '[select='a:1':f=flv]rtmp://XXXXXXX:XXXXXXX@p.epXXXXXX.i.ak': error opening: Input/output error [tee @ 0x7fa0de801800] Slave muxer #5 failed, aborting. I have confirmed that removing characters within the inverted commas ("") after the tee statement allows more of the stream path to be processed by ffmpeg and inversely adding characters within the "" results in less of the stream path being processed by ffmpeg. The characters can be related to any of the commands within the "" and don't need to specifically be any part of the stream paths. To me this looks as if the tee muxer has an undocumented, maximum character input limit for the argument that it takes within the inverted commas. Would any devs or users be able to confirm this based on experience/a deeper knowledge of the code? -- Kind regards, Luke Harrison

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