May 2017 - ffmpeg-user - lists.ffmpeg.org

benching with -f null - fails on bluray sample since ffmpeg: init filtergraphs only after we have a frame on each input
by Andy Furniss 26 May '17

26 May '17

Sample https://drive.google.com/file/d/0BxP5-S1t9VEEcUtGa1d2U3VuZjQ/view?usp=shari… Since commit af1761f7b5b1b72197dc40934953b775c2d951cc Author: Anton Khirnov <anton(a)khirnov.net> Date: Fri May 27 12:14:33 2016 +0200 ffmpeg: init filtergraphs only after we have a frame on each input This makes sure the actual stream parameters are used, which is important mainly for hardware decoding+filtering cases, which would previously require various weird workarounds to handle the fact that a fake software graph has to be constructed, but never used. This should also improve behaviour in rare cases where avformat_find_stream_info() does not provide accurate information. This merges Libav commit a3a0230. It was previously skipped. The code in flush_encoders() which sets up a "fake" format wasn't in Libav. I'm not sure if it's a good idea, but it tends to give behavior closer to the old one in certain corner cases. The vp8-size-change gives different result, because now the size of the first frame is used. libavformat reported the size of the largest frame for some reason. The exr tests now use the sample aspect ratio of the first frame. For some reason libavformat determines 0/1 as aspect ratio, while the decoder returns the correct one. The ffm and mxf tests change the field_order values. I'm assuming another libavformat/decoding mismatch. Signed-off-by: wm4 <nfxjfg(a)googlemail.com> benching uploaded sample with -f null - fails andy [ffmpeg]$ ./ffmpeg -i ~/br-264-thd71.m2ts -f null - ffmpeg version N-86268-g43c394d Copyright (c) 2000-2017 the FFmpeg developers built with gcc 5.3.0 (GCC) configuration: --prefix=/usr --disable-doc --enable-gpl --enable-omx --enable-opencl --enable-libvpx --enable-libx265 --enable-libmp3lame --enable-libx264 --enable-gnutls libavutil 55. 63.100 / 55. 63.100 libavcodec 57. 96.101 / 57. 96.101 libavformat 57. 72.101 / 57. 72.101 libavdevice 57. 7.100 / 57. 7.100 libavfilter 6. 90.100 / 6. 90.100 libswscale 4. 7.101 / 4. 7.101 libswresample 2. 8.100 / 2. 8.100 libpostproc 54. 6.100 / 54. 6.100 [mpegts @ 0x3cec660] start time for stream 3 is not set in estimate_timings_from_pts [mpegts @ 0x3cec660] PES packet size mismatch [mpegts @ 0x3cec660] Could not find codec parameters for stream 3 (Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090)): unspecified size Consider increasing the value for the 'analyzeduration' and 'probesize' options Input #0, mpegts, from '/home/andy/br-264-thd71.m2ts': Duration: 00:00:05.13, start: 11.650667, bitrate: 30020 kb/s Program 1 Stream #0:0[0x1011]: Video: h264 (High) (HDMV / 0x564D4448), yuv420p(progressive), 1920x1080 [SAR 1:1 DAR 16:9], 23.98 fps, 23.98 tbr, 90k tbn, 47.95 tbc Stream #0:1[0x1100]: Audio: truehd (AC-3 / 0x332D4341), 48000 Hz, 7.1, s32 (24 bit) Stream #0:2[0x1100]: Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, 5.1(side), fltp, 640 kb/s Stream #0:3[0x1200]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090) Stream mapping: Stream #0:0 -> #0:0 (h264 (native) -> wrapped_avframe (native)) Stream #0:1 -> #0:1 (truehd (native) -> pcm_s16le (native)) Press [q] to stop, [?] for help Too many packets buffered for output stream 0:1. Conversion failed! Also fails with avconv andy [libav]$ ./avconv -i ~/br-264-thd71.m2ts -f null - avconv version v13_dev0-1134-g3d04051, Copyright (c) 2000-2017 the Libav developers built on May 26 2017 10:32:56 with gcc 5.3.0 (GCC) [mpegts @ 0x2606c40] start time is not set in estimate_timings_from_pts [mpegts @ 0x2606c40] PES packet size mismatch Input #0, mpegts, from '/home/andy/br-264-thd71.m2ts': Duration: 00:00:05.08, start: 11.650667, bitrate: 30266 kb/s Program 1 Stream #0:0[0x1011]: Video: h264 (High) [HDMV / 0x564D4448] yuv420p, 1920x1080 [PAR 1:1 DAR 16:9] 23.98 fps, 90k tbn Stream #0:1[0x1100]: Audio: truehd [AC[45]3 / 0x332D4341] 48000 Hz, 7.1, s32 Stream #0:2[0x1100]: Audio: ac3 [AC[45]3 / 0x332D4341] 48000 Hz, 5.1, fltp, 640 kb/s Stream #0:3[0x1200]: Subtitle: hdmv_pgs_subtitle Stream mapping: Stream #0:0 -> #0:0 (h264 (native) -> wrapped_avframe (native)) Stream #0:1 -> #0:1 (truehd (native) -> pcm_s16le (native)) Press ctrl-c to stop encoding Too many packets buffered for output stream 0:1.

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(PAID JOB) I'm looking for a programmer that would be able to develope for me a script that work with ffmpeg/osx
by nuvolablux 26 May '17

26 May '17

hello everybody is there anybody here that is be able to develope for me a script that work with ffmpeg/osx to split 1 hours long movie into few clips? This is a paid project. The script will have to work on mac osx I recommend using ffmpeg as a video engine Input File: Apple prores422hq MOV files / text file containing time code and file names The script must cut the videos with accuracy to the frame, the time code in the text file is in this format: HH: MM: SS: frames (1 second contains 25 frames - from 00 to 24 -) while ffmpeg uses different syntax: FFMPEG Time unit syntax Note that you can use two different time unit formats: sexagesimal (HOURS: MM: SS.MICROSECONDS, as in 01: 23: 45.678), or in seconds. If a fraction is used, such as 02: 30.05, this is interpreted as "5 100ths of a second", not as frame 5. For example, 02: 30.5 would be 2 minutes, 30 seconds and half a second Would be the same as using 150.5 in seconds. The ffmpeg command to use is the copy, which provides a crop and not the file regeneration. The script would be splitted into two parts PART A: Starting from a master video file (MOV apple prores 422HQ full HD) the script must be able to divide it into many files according to the time codes and names in the TXT file. Below you can download the sample video file and the EDL(TXT) file http://wetransfer.red-line.us/xzms4iU Looking at the text contained in the edl file (see below) the script will have to: Generate video file called "Video Source256485.mov" from video master from time code 00: 00: 00: 00 to 00: 00: 11: 14 Generate the second video file called "Video Source Copy 54.mov" from the video master file from 00: 00: 11: 14 to 00: 00: 23: 03 ...and so on... Up to file 11 with the name "Video Source copy 2342342.mov" which in this test file will be the last file that will process the script. I have more than 1000 files to split and name, hence the need to automate this process. TITLE: Video Source Copy FCM: NON-DROP FRAME 000001 001 V 00: 00: 00: 00 00: 00: 11: 14 00: 00: 00: 00 00: 00: 11: 14 * FROM CLIP NAME: Video Source256485.mov 000002 001 V C 00: 00: 00: 00 00: 00: 11: 14 00: 00: 11: 14 00: 00: 23: 03 * FROM CLIP NAME: Video Source Copy 54.mov 000003 001 V 00: 00: 00: 00 00: 00: 11: 14 00: 00: 23: 03 00: 00: 34: 17 * FROM CLIP NAME: Video Source copy 33.mov 000004 001 V C 00: 00: 00: 00 00: 00: 11: 14 00: 00: 34: 17 00: 00: 46: 06 * FROM CLIP NAME: Video Source Copy 433.mov 000005 001 V C 00: 00: 00: 00 00: 00: 11: 14 00: 00: 46: 06 00: 00: 57: 20 * FROM CLIP NAME: Video Source Copy 255.mov 000006 001 V C 00: 00: 00: 00 00: 00: 11: 14 00: 00: 57: 20 00: 01: 09: 09 * FROM CLIP NAME: Video Source Copy 6123.mov 000007 001 V 00: 00: 00: 00 00: 00: 11: 14 00: 01: 09: 09 00: 01: 20: 23 * FROM CLIP NAME: Video Source Copy 2357.mov 000008 001 V 00: 00: 00: 00 00: 00: 11: 14 00: 01: 20: 23 00: 01: 32: 12 * FROM CLIP NAME: Video Source 123434.mov copy 000009 001 V 00: 00: 00: 00 00: 00: 11: 14 00: 01: 32: 12 00: 01: 44: 01 * FROM CLIP NAME: Video Source 74345fdf.mov copy 000010 001 V C 00: 00: 00: 00 00: 00: 11: 14 00: 01: 44: 01 00: 01: 55: 15 * FROM CLIP NAME: Video Source copy 2342342.mov 000011 001 V C 00: 00: 00: 00 00: 00: 11: 14 00: 01: 55: 15 00: 02: 07: 04 * FROM CLIP NAME: Video Source copia.mov PART B) Starting with two distinct folders of video files, containing files named the same way and of the same duration eg: Folder A: Aaa.mov duration 00: 00: 10: 12 Bbb.mov duration 00: 01: 14: 10 Ccc.mov duration 00: 10: 00: 14 and so on... Folder B: Aaa.mov duration 00: 00: 10: 12 Bbb.mov duration 00: 01: 14: 10 Ccc.mov duration 00: 10: 00: 14 and so on... The script will have to: Extract an image file (png?) From each movie that contains the frame that is in the exact half of the movie. Create an image from the audio of movie files (https://trac.ffmpeg.org/wiki/ Waveform) Picture the exported frames and the audio image in a document according to a predefined pattern and add text such as the name of the files from which extra pictures and other standard texts have been extracted. . Get a pdf document or a offline reference database that can serve as a processing report. Consider that we speak about 1000 images, then consider the format that allows fast consultation (pdf too heavy?) And an alphanumeric string search. Clearly I'm available for any further explanation. nuvolablux(a)gmail.com I would need a quote with splitted times and costs for the two scripts. Thank you. S.

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Error Compiling with CUDA on PPC64le
by Rhett Ding 26 May '17

26 May '17

1. This is my configuration: ./configure --enable-cuda --enable-cuvid --enable-nvenc --enable-nonfree --enable-libnpp --extra-cflags=-I/usr/local/cuda/include --extra-ldflags=-L/usr/local/cuda/lib64; 2. This is the Errror reported: ERROR: cuvid requested, but not all dependencies are satisfied: cuda; 3. I installed cuda and I'm sure my extra-cflags is right, but it just failed. My environment is ppc64le architecture, ubuntu 16 Operating System, FFmpeg 3.3.1 source code, cuda 8.0 toolkit, NVIDIA driver 375, GPU is NVIDIA Tesla. My problem is just the same as this guy faced: http://ffmpeg.gusari.org/viewtopic.php?f=8&t=3594&p=10756&hilit=cuda#p10756 in the FFmpeg support forum. PLEASE help to solve the problem, it seems ffmpeg can't find right dir or it's not well supported on power architecture. THANKS a lot.

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Ignore source stream error options
by James Carter 25 May '17

25 May '17

Hello, just wondering if someone can let me know if there are some better switches than -err_detect ignore_err to use to help ffmpeg ignore live stream errors while hls stream transcoding on the fly. The command I use is; ffmpeg -i "myvideo.mp4" -err_detect ignore_err -c:a aac -ar 48000 -ab 64k -s 720x480 -vcodec libx264 -hls_time 240 myvideo.m3u8 but the -err_detect ignore_err switch doesn't seem to help that much, when there is a steam glitch, ffmpeg still craps out and audio and video output issues happen and I have to close it and restart it :( Thanks

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Streaming audio on demaind
by Patrick Shirkey 25 May '17

25 May '17

Hi, I am looking at options for on demand audio file streaming (no video). Currently my audio files are encoded as .m4a aac_he 64k. This format works but there is a noticeable lag while the file is downloaded before playback starts and I see inconsistent results on mobile devices. Is the correct approach for streaming audio files to use hls to separate the file into chunks with an m3u8 playlist or is there a better/alternate/more appropriate method? ex. https://www.ffmpeg.org/ffmpeg-formats.html#hls https://www.ffmpeg.org/ffmpeg-formats.html#segment -- Patrick Shirkey Boost Hardware Ltd

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Capture Audio only from an Output Device?
by Jan Petersen 24 May '17

24 May '17

Hey everyone I am playing with ffmpeg currently and I wonder if there is any way to capture a particular output from a given output device (preferably WASAPI) and stream it to another machine. Background: A software running at Windows is playing multiple signals to a specified output. Program 1 is played at Out 1, program 2 at Out 2 ... up to 64 programs when a MADI soundcard is involved, in theory unlimited with a virtual adapter. Can I use ffmpeg to capture these outputs and stream it in PCM wave to a Linux box for further processing? Or can ffmpeg only capture imputs? Basically I try to find a possibility to have AoIP from a Windows machine to a Linux box, preferably with only virtual soundcards involved. The ultimate goal is, to have the playout, the streams and the further processing virtualized. Maybe other ideas? I tried GStreamer also, but the DirectSound/WASAPI implementation is really not good. In ffmpeg at least DSound is usable. There is no WASAPI implementation at all, is that true? Or is there something available? Thanks in advance for your thoughts. Jan

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continuous streaming even if streaming source isn't available
by Lars Uhlmann 24 May '17

24 May '17

I use the following command to save a camera stream to disk: $ ffmpeg -t 3600 -vcodec copy -acodec copy \ -i rtsp://<IP:Port>/live.sdp \ current.mp4 Now if the network connection is lost or the camera is unavailable, »ffmeg« stops recording and exits with an error. Is there a away, to let »ffmeg« continuously try to reconnect to the stream source and only end if either the record duration is reached or a kill signal received? thanks Lars

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Questions regarding mutli-DRM with ffmpeg
by Aameer Rafiq Wani 24 May '17

24 May '17

Hey guys, Hope you are doing great. I was facing some issues in one of the projects I am working on; which involves ffmpeg and bento4. So I wanted your guidance on few things. Let me walk you through what I have achieved and what my plan is: I have a cloud transcoding system which transcodes videos and burns subtitles on them using ffmpeg. I have explained this in some details on my post here ( http://aameer.github.io/cloud-computing-101/) In crux what happens is on this system is explained by the following line of code ffmpeg -i $INPUT_VIDEO_FILE -profile:v baseline -level 4.0 -vf "scale=-2:360,subtitles='dynamic_subtitle_'$VIDEO_ID'. ass':force_style='FontName=Aaargh/Aaargh.ttf,PrimaryColour=&H664c4c4c" -start_number 0 -hls_time 10 -hls_list_size 0 -hls_key_info_file /mnt/s3_temp/$SERVER_NAME/$PROJECT_ID/videos/$VIDEO_ID/file360.keyinfo -f hls /mnt/s3_temp/$SERVER_NAME/$PROJECT_ID/videos/$VIDEO_ID/$VIDEO_ID"_360_.m3u8" where I transform an input file (mp4) into hls encrypted adaptive streams. Now to add to this system I want to add multi-DRM and for this my drm provider mentioned that this could be accomplished with bento4. Do you guys have any other open source alternatives which I should keep in mind? Moreover if bento4 would do the actual transcoding what purpose will ffmpeg serve? Could you guys share any code snippets (if any) related to multi-DRM with ffmpeg and bento4. Any relevant material which could help me in this regard would be appreciated Thanks, AAMEER RAFIQ WANI aameer.github.io <aameer(a)github.io>

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Unable to capture audio and video synced
by Leonardo Soares Müller 23 May '17

23 May '17

Hello I am trying to record my computer's screen together with the sound from speakers and (occasionally) microphone. While the record is made successfully, it is problematic: when watching on VLC it is good, but on Kdenlive or YouTube audio and video are out of sync. I am using Xubuntu 16.04.2 and I have downloaded ffmpeg source and built it. To record the screen, I use x11grab. To record speakers and microphone, I use pulse. The command I used to record was: env PULSE_LATENCY_MSEC=20 ffmpeg -vaapi_device /dev/dri/renderD128 -hwaccel vaapi -hwaccel_output_format yuv420p -thread_queue_size 16384 -f pulse -sample_rate 44100 -channels 2 -i alsa_output.pci-0000_00_1f.3.analog-stereo.monitor -thread_queue_size 16384 -f x11grab -s 1366x768 -r 30 -i :0.0 -acodec libfdk_aac -b:a 160k -vf format=nv12,hwupload,scale_vaapi=w=960:h=540 -vcodec h264_vaapi -qp 20 -f flv -shortest TEST001.flv The output of the command was: ffmpeg version N-86126-ge434840 Copyright (c) 2000-2017 the FFmpeg developers built with gcc 5.4.0 (Ubuntu 5.4.0-6ubuntu1~16.04.4) 20160609 configuration: --enable-shared --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmodplug --enable-libmp3lame --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-librtmp --enable-libschroedinger --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-libzvbi --enable-openal --enable-opengl --enable-libdc1394 --enable-libiec61883 --enable-libzmq --enable-frei0r --enable-libx264 --enable-libopencv --enable-libfdk-aac --enable-libmfx --enable-vaapi --enable-nonfree --enable-gpl --enable-libxcb --enable-libxcb-shm --enable-libxcb-xfixes --enable-libxcb-shape libavutil 55. 63.100 / 55. 63.100 libavcodec 57. 96.101 / 57. 96.101 libavformat 57. 72.101 / 57. 72.101 libavdevice 57. 7.100 / 57. 7.100 libavfilter 6. 90.100 / 6. 90.100 libavresample 3. 6. 0 / 3. 6. 0 libswscale 4. 7.101 / 4. 7.101 libswresample 2. 8.100 / 2. 8.100 libpostproc 54. 6.100 / 54. 6.100 libva info: VA-API version 0.39.0 libva info: va_getDriverName() returns 0 libva info: Trying to open /usr/lib/x86_64-linux-gnu/dri/i965_drv_video.so libva info: Found init function __vaDriverInit_0_39 libva info: va_openDriver() returns 0 Guessed Channel Layout for Input Stream #0.0 : stereo Input #0, pulse, from 'alsa_output.pci-0000_00_1f.3.analog-stereo.monitor': Duration: N/A, start: 1495555903.842161, bitrate: 1411 kb/s Stream #0:0: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s [x11grab @ 0x18612e0] Stream #0: not enough frames to estimate rate; consider increasing probesize Input #1, x11grab, from ':0.0': Duration: N/A, start: 1495555904.112324, bitrate: N/A Stream #1:0: Video: rawvideo (BGR[0] / 0x524742), bgr0, 1366x768, 30 fps, 1000k tbr, 1000k tbn, 1000k tbc Stream mapping: Stream #1:0 -> #0:0 (rawvideo (native) -> h264 (h264_vaapi)) Stream #0:0 -> #0:1 (pcm_s16le (native) -> aac (libfdk_aac)) Press [q] to stop, [?] for help [swscaler @ 0x187b200] Warning: data is not aligned! This can lead to a speedloss Output #0, flv, to 'TEST001.flv': Metadata: encoder : Lavf57.72.101 Stream #0:0: Video: h264 (h264_vaapi) (High) ([7][0][0][0] / 0x0007), vaapi_vld(progressive), 960x540, q=0-31, 30 fps, 1k tbn, 30 tbc Metadata: encoder : Lavc57.96.101 h264_vaapi Stream #0:1: Audio: aac (libfdk_aac) ([10][0][0][0] / 0x000A), 44100 Hz, stereo, s16, 160 kb/s Metadata: encoder : Lavc57.96.101 libfdk_aac frame= 681 fps= 30 q=-0.0 Lsize= 2607kB time=00:00:22.64 bitrate= 943.1kbits/s speed=0.999x video:2134kB audio:443kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 1.161332% After I have recorded it, I have separated the video and audio from the resulting file in two files (the input being the file, using -an -vcodec copy and -vn acodec copy, for the respective outputs). The video only file was opened on Kdenlive and the audio file was opened on Audacity. Looking at the command, it would be expected a difference of 20 ms (0,02 s) between audio and video at maximum, but the length's difference was 460 ms (0,46 s). There are three different lengths for this file and, possibly, different multimedia implementations see the file with a different lengths. ffmpeg (from command line): 22,64 s Kdenlive: 22,22 s Audacity: 22,686 s The millisecond is only visible using Audacity, so I don't know the milliseconds from ffmpeg and Kdenlive. I have done this test some times, and the results appear to be random. Sometimes the delay is 80 ms, sometimes 1000 ms. There is not a fixed difference, so correcting the command with a fixed offset is not a options, unfortunately. I would like to know what should I do to capture the screen with sound synced, as the -shortest option was ineffective on this case because the inputs don't have an end. Thank you.

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(probably) the filter loop causing memory leak for version N-86175-g64ea4d1
by ibrahim 23 May '17

23 May '17

hi, i'd like to report that the following command prints "error marking filters as finished" at the end of the encoding and leaks memory until i manage to quit ffmpeg, or even cause system crash and reboot sometimes for version N-86175-g64ea4d1 (20170520-64ea4d1, 64 bit static), however the command works fine for one of the previous versions i have, version N-82324-g872b358. if you like, you may try to reproduce it quickly with the following -t 9 version of the original command i used for a much longer transcode (-t 57600, 16 hours). i have also reproduced it without the sound (aevalsrc), so i think the problem is with the filter loop. if you really need the complete uncut console output for this case i may reproduce the problem once more to supply it but for now, i really do not want to risk another system crash :) ffmpeg -f lavfi -i aevalsrc="sin((1000-t*0.0086805)*2*PI*t):s=48000:d=9" -i https://upload.wikimedia.org/wikipedia/commons/c/c4/PM5544_with_non-PAL_sig… -vf loop=size=1:loop=-1,hue=h=t*360/57600 -t 9 -pixel_format yuv420p -c:v libvpx-vp9 -lossless 1 -c:a libvorbis -qscale:a 10 test1.webm -- ibrahim GPG:F7DA A00A CFF3 9032 3E14 F8EE 1386 6EE6 29C9 E65F

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