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

FFProbe unable to find a media tag
by Senthil 10 Mar '17

10 Mar '17

This can be reproduced with any iTunes movie which has audio descriptions. The movie I tested against had 3 english streams - a stereo audio stream, a 5.1 audio stream and an audio description. ffmpeg version: ffmpeg-3.2.2-win64-staticMediaInfo version: 0.7.93 I used the following command on the iTunes movie and was UNABLE to find the media tag on the audio stream (output attached).ffprobe.exe -v error -show_streams -select_streams a -print_format json "Z:\Music\iTunes\iTunes Media\Movies\A Bug's Life\A Bug's Life (1080p HD).m4v" I then ran mediainfo on the same iTunes movie and I'm able to see the tag with value "tagc : public.accessibility.describes-video" (output attached). I'd like to know how to fetch this tag information via ffprobe ? Note: You can read this Apple page https://support.apple.com/en-us/HT204657 to know about the media characteristics tag "public.accessibility.describes-video” and how to add it to the audio stream. ThanksSenthil.

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Help
by Eduardo Souza 09 Mar '17

09 Mar '17

Dear Friends, Iwould like your help about convertion MXF to MP4 (codec H265). The convertedvideo present freeze in little interval of time on the images aftertransistions. Thecommand that I use is: ffmpeg -i "input.mxf" -map 0:v -map0:a -c:v libx265 -x265-params crf=26:psy-rd=1 "output.mp4" Inattached link for download of video: https://drive.google.com/file/d/0Bx5h-n3CtmU-cnNycE15aVE1b2s/view?usp=shari… Couldyou help me? Thank you, Eduardo Souza

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FFMPEG & Panasonic OP-1B MXF decode issue
by jeffrey.k.smith＠thomsonreuters.com 09 Mar '17

09 Mar '17

Hi, First time post so apologies if I break any etiquette rules! We are trying to use FFMPEG to transcode video shot on a broadcast Panasonic camera that encodes to P2 Long GOP (OP-1B) MXF files. Each file contains a video stream, a couple of data timecode streams and four discreet, mono audio streams. Whenever we ask FFMPEG to process this file, it doesn’t seem to be able to identify all the streams correctly. For example, even if we don’t try to do any encoding at all, we see the following: ======================== ffmpeg -i 085FV4BR.MXF ffmpeg version N-82629-g9955123 Copyright (c) 2000-2016 the FFmpeg developers built with Apple LLVM version 8.0.0 (clang-800.0.42.1) configuration: --prefix=/usr/local --enable-gpl --enable-nonfree --enable-libass --enable-libfdk-aac --enable-libfreetype --enable-libmp3lame --enable-libopus --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid libavutil 55. 41.101 / 55. 41.101 libavcodec 57. 66.107 / 57. 66.107 libavformat 57. 58.100 / 57. 58.100 libavdevice 57. 2.100 / 57. 2.100 libavfilter 6. 67.100 / 6. 67.100 libswscale 4. 3.101 / 4. 3.101 libswresample 2. 4.100 / 2. 4.100 libpostproc 54. 2.100 / 54. 2.100 [mxf @ 0x7fd3ab002600] material track 3: no corresponding source track found [mxf @ 0x7fd3ab002600] material track 4: no corresponding source track found [mxf @ 0x7fd3ab002600] could not resolve material track strong ref [mxf @ 0x7fd3ab002600] index entry 49 + TemporalOffset 1 = 50, which is out of bounds [mxf @ 0x7fd3ab002600] IndexSID 101 starts at EditUnit 50 - seeking may not work as expected [mxf @ 0x7fd3ab002600] index entry 0 + TemporalOffset -2 = -2, which is out of bounds [mxf @ 0x7fd3ab002600] index entry 49 + TemporalOffset 1 = 50, which is out of bounds [mxf @ 0x7fd3ab002600] IndexSID 101 starts at EditUnit 50 - seeking may not work as expected [mxf @ 0x7fd3ab002600] index entry 0 + TemporalOffset -2 = -2, which is out of bounds [mxf @ 0x7fd3ab002600] index entry 49 + TemporalOffset 1 = 50, which is out of bounds [mxf @ 0x7fd3ab002600] IndexSID 101 starts at EditUnit 50 - seeking may not work as expected [mxf @ 0x7fd3ab002600] index entry 0 + TemporalOffset -2 = -2, which is out of bounds [mxf @ 0x7fd3ab002600] index entry 49 + TemporalOffset 1 = 50, which is out of bounds [mxf @ 0x7fd3ab002600] IndexSID 101 starts at EditUnit 50 - seeking may not work as expected [mxf @ 0x7fd3ab002600] index entry 0 + TemporalOffset -2 = -2, which is out of bounds [mxf @ 0x7fd3ab002600] index entry 49 + TemporalOffset 1 = 50, which is out of bounds [mxf @ 0x7fd3ab002600] got 30 index tables - only the first one (IndexSID 101) will be used [mxf @ 0x7fd3ab002600] error getting stream index 385942017 Last message repeated 49 times Guessed Channel Layout for Input Stream #0.3 : mono Guessed Channel Layout for Input Stream #0.4 : mono Guessed Channel Layout for Input Stream #0.5 : mono Input #0, mxf, from '085FV4BR.MXF': Metadata: uid : 49299363-f360-4654-a7b9-6901b73b9a46 generation_uid : cb55f1a6-46ca-4ca0-b98b-25d7d82ec736 company_name : Panasonic product_name : P2 product_version : 3.0 product_uid : 060e2b34-0401-0108-0e0e-010101010001 modification_date: 2017-02-17T06:48:17.000000Z material_package_umid: 0x060A2B340101010501010D4313000000AA601646017805E7008045822D4B0053 material_package_name: BART0897 timecode : 07:49:06:23 Duration: 00:00:09.24, start: 0.000000, bitrate: 15308 kb/s Stream #0:0: Video: h264 (High), yuv420p(tv, bt709, top first), 1920x1080 [SAR 1:1 DAR 16:9], 12.50 fps, 25 tbr, 25 tbn, 50 tbc Metadata: file_package_umid: 0x060A2B34010101050101064313000000AA601646017805E8008045822D4B0053 Stream #0:1: Data: none Metadata: file_package_umid: 0x060A2B340101010501010C4313000000AA601646017805E8008045822D4B0053 data_type : data Stream #0:2: Data: none Metadata: file_package_umid: 0x060A2B34010101050101084313000000AA601646017805E8008045822D4B0053 data_type : audio Stream #0:3: Audio: pcm_s16le, 48000 Hz, mono, s16, 768 kb/s Metadata: file_package_umid: 0x060A2B34010101050101084313000000AA601646017805E9008045822D4B0053 Stream #0:4: Audio: pcm_s16le, 48000 Hz, mono, s16, 768 kb/s Metadata: file_package_umid: 0x060A2B34010101050101084313000000AA601646017805EA008045822D4B0053 Stream #0:5: Audio: pcm_s16le, 48000 Hz, mono, s16, 768 kb/s Metadata: file_package_umid: 0x060A2B34010101050101084313000000AA601646017805EB008045822D4B0053 At least one output file must be specified ======================== You can see that it finds: Stream 0: Video Stream 1: Data Stream 2: Data Stream 3: Audio Stream 4: Audio Stream 5: Audio However, it’s missing one audio stream and I’m assuming it’s related to the message “error getting stream index 385942017” earlier in the output. If we try to do the transcode, we end up with audio being encoded that appears to be a concatenation of all 4 audio streams no matter how we try to use ‘-map’ to extract a single audio stream e.g. ffmpeg –I 085FV4BR.MXF –map 0:0 –c:v h264 –b:v 12M –flags +ildct+ilme+mv4 –top 1 –map 0:3 –c:a aac –b:a 256k –ac 2 –ar 48k –f mp4 –y out.mp4 By way of comparison, we’ve used both ‘MXF Inspect’ and ‘MediaInfo’ to interrogate the file and see all streams correctly. Here’s the output from MediaInfo: ======================== 085FV4BR.MXF General Complete name : 085FV4BR.MXF Format : MXF Format version : 1.3 Format profile : OP-1b Format settings : Closed / Complete File size : 16.9 MiB Duration : 9 s 240 ms Overall bit rate mode : Variable Overall bit rate : 15.3 Mb/s Package name : BART0897 Encoded date : 2017-02-17 06:48:17.000 Writing application : Panasonic P2 3.0 Video ID : 2 Format : AVC Format/Info : Advanced Video Codec Format profile : High@L4 Format settings, CABAC : Yes Format settings, ReFrames : 3 frames Format settings, GOP : M=3, N=24 Format settings, wrapping mode : Frame Codec ID : 0D01030102106001-0401020201314001 Duration : 9 s 240 ms Bit rate mode : Variable Bit rate : 12.2 Mb/s Maximum bit rate : 18.0 Mb/s Width : 1 920 pixels Height : 1 080 pixels Display aspect ratio : 16:9 Frame rate : 25.000 FPS Standard : Component Color space : YUV Chroma subsampling : 4:2:0 Bit depth : 8 bits Scan type : Interlaced Scan type, store method : Separated fields Scan order : Top Field First Bits/(Pixel*Frame) : 0.236 Stream size : 13.5 MiB (80%) Color range : Limited Color primaries : BT.709 Transfer characteristics : BT.709 Matrix coefficients : BT.709 Audio #1 ID : 3 Format : PCM Format settings, Endianness : Little Codec ID : 0D01030102060B00 Duration : 9 s 240 ms Bit rate mode : Constant Bit rate : 768 kb/s Channel(s) : 1 channel Sampling rate : 48.0 kHz Bit depth : 16 bits Stream size : 866 KiB (5%) Locked : Yes Audio #2 ID : 4 Format : PCM Codec ID : 0D01030102060B00 Duration : 9 s 240 ms Bit rate : 768 kb/s Channel(s) : 1 channel Sampling rate : 48.0 kHz Bit depth : 16 bits Stream size : 866 KiB (5%) Locked : Yes Audio #3 ID : 5 Format : PCM Codec ID : 0D01030102060B00 Duration : 9 s 240 ms Bit rate : 768 kb/s Channel(s) : 1 channel Sampling rate : 48.0 kHz Bit depth : 16 bits Stream size : 866 KiB (5%) Locked : Yes Audio #4 ID : 6 Format : PCM Codec ID : 0D01030102060B00 Duration : 9 s 240 ms Bit rate : 768 kb/s Channel(s) : 1 channel Sampling rate : 48.0 kHz Bit depth : 16 bits Stream size : 866 KiB (5%) Locked : Yes Other #1 ID : 1-Material Type : Time code Format : MXF TC Time code of first frame : 07:49:06:23 Time code settings : Material Package Time code, striped : Yes Other #2 ID : 1-Source Type : Time code Format : MXF TC Time code of first frame : 07:49:06:23 Time code settings : Source Package Time code, striped : Yes ======================== So my best guess is there’s a bug somewhere in mxfdec.c when it’s working against OP-1B content. But does anyone have any other ideas about what’s going on? Again, apologies for the ultra long message but very grateful for any input! Best Regards, Jeff. ________________________________ This e-mail is for the sole use of the intended recipient and contains information that may be privileged and/or confidential. If you are not an intended recipient, please notify the sender by return e-mail and delete this e-mail and any attachments. Certain required legal entity disclosures can be accessed on our website.<http://site.thomsonreuters.com/site/disclosures/>

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Convert iPhone MOV to Android MP4
by Richard Duran 08 Mar '17

08 Mar '17

I would've thought that this was a FAQ, but I have been unable to find it. I'm emailing this list because many solutions I find online still reference "-strict experimental" for AAC output. I would like to convert iPhone videos to MP4, and would like to retain the input audio bitrate and sampling rate. I had the impression that this would be the default behavior if the values weren't specified on the command line, but I've seen that this isn't the case. I will follow up this email with sample output.

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Desktop Capture 'Jerky'
by derekg 08 Mar '17

08 Mar '17

Dear Carl Eugen , I have two (2) different PCs and for the TEST process I have successfully followed with your help on my older machine https://trac.ffmpeg.org/wiki/CompilationGuide/Ubuntu , and with the addition of '--enable libasound' in the compilation stage . An ffmpeg capture process successfully captures both video and sound , however unsurprisingly on the older machine it is a little jerky . I've used ffmpeg -f alsa -f pulse -ac 2 -i default -f x11grab -s 1920x1080 -r 60 -i :0.0 -qscale 0 -f mpeg -ar 48000 -s wvga -y /media/sda5/sample.mkv to run this , but could you advise a little both on changes I might make firstly to achieve a less jerky one pass capture , and maybe on the appropriate two stages to create a lossless *.mkv . Obviously I want to become a little more competent before transferring to my 'live' TV PC . Thank you , from 'newbie/novice' , Derek

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Optimization of command and why it stops conversion?
by Ankit Gupta 06 Mar '17

06 Mar '17

Hi All, I am using below command for RTSP to .m3u8 conversion and saving it to amazon s3 bucket. I am able to get camera view with a delay of approx 10-12 second. Any idea on further optimization of this command to minimize delay and flickering. ffmpeg -rtsp_transport tcp -i rtsp:// root:kastle@192.168.17.142/axis-media/media.amp -r 25 -crf 25 -preset faster -maxrate 3000k -bufsize 6000k -g 50 -c:v libx264 -hls_time 2 -hls_list_size 2 -hls_wrap 8 -start_number 1 -y Z:/Z/playlist.m3u8 Also this command auto stopped after running for few hours and there is no error in console, I can restart it but any clue on why it might have stopped. I verified that camera is up and running and camera RTSP is accessible. Regards, Ankit Kumar Gupta

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Missing bitrate information on AAC encoded videos
by Marco 04 Mar '17

04 Mar '17

Hello, i use to encode the audio of my movie using libfdk_aac, but in the result file doesn't appear the used bitrate (with mediainfo, ffprobe, etc.). I encode using CBR mode, in general matroska container. I have the same problem with Handbrake. I've written to the FDK site, this is the response: https://github.com/mstorsjo/fdk-aac/issues/58 The problem is that the missing information cause different issues with other software. For example i had to hack the libmedianfo per library in order to manage that. Can please someone explains how i can see the bitrate of the AAC audio encoded with ffmpeg using such library and, in general, how i can make this information visible to the other softwares ? Thanks for the attention and sorry for the bad english. Best regards. -- 3634753 They who dream by day are cognizant of many things which escape those who dream only by night.

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[CLT2017] FFmpeg at Chemnitzer Linux-Tage
by Alexander Strasser 04 Mar '17

04 Mar '17

Hi! FFmpeg has been accepted for CLT 2017, a FOSS event in Chemnitz, Germany! This "Chemnitzer Linux Tage" will take place from 11th to 12th of March. You can find more details here: https://chemnitzer.linux-tage.de/2017/en/ Thus once again, we will man a booth, happily waiting for users and fellow developers to get in touch with us! We would like to encourage everyone visiting CLT to bring sample files and/or command lines that show suspicious or buggy behavior -- see it as chance to get your bug fixed right away! :) See you in Chemnitz! Best regards, Alexander

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hflip so slooooooooow
by JD 04 Mar '17

04 Mar '17

Hi all, should horizontal flipping be as slow as this: frame=27067 fps= 12 q=28.0 size= 143993kB time=00:18:06.27 bitrate=1085.9kbits/s dup=0 drop=35 speed=0.47x the command was: $ ~/bin/ffmpeg.d/ffmpeg -i video_mnt13.mp4 -vf "hflip" -y video_mnt13-hflip-m1B6z-54q_0.mp4 ffmpeg version 3.2.4-static http://johnvansickle.com/ffmpeg/ Copyright (c) 2000-2017 the FFmpeg developers built with gcc 5.4.1 (Debian 5.4.1-5) 20170205 how can I speed it up? (I thought that flipping should not need transcoding)???

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h264_omx and non-existing PPS 0 referenced
by luke 03 Mar '17

03 Mar '17

Trying to stream a webcam through rtp with a raspberry pi. Cating the SDP file on my desktop, and launching ffplay: ffplay sdp.out then streaming out this way: ffmpeg -i /dev/video0 -c:v h264_omx -f rtp rtp://192.168.0.101:1234 works perfectly. But If I launch ffmpeg first THEN ffplay, ffplay stutters some error message about missing PPS: [h264 @ 0x7fd97c0008c0] decode_slice_header error [h264 @ 0x7fd97c0008c0] no frame! [h264 @ 0x7fd97c0008c0] non-existing PPS 0 referenced After looking for a while I understand some intermediary frame might be missing. However, using the -g option this way does not change anything: ffmpeg -i /dev/video0 -c:v h264_omx -g 30 -f rtp rtp://192.168.0.101:1234 It is a problem for me because I am using the raspberry as a securiy camera. My desktop can be restarted from time to time and I need ffplay to be able to catch long after streaming is started. Any help is appreciated.

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