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

Split video at mid-stream codec/resolution changes
by Peter B. 16 Mar '17

16 Mar '17

Hi all! Is there any way to use ffmpeg to split/cut a video at positions where the codec or e.g. resolution changes mid-stream, but without transcoding the audio/video? (="-c copy")? Maybe by using ffprobe to detect where these codec/property changes are in the file... I've searched the Internet, but couldn't find anything suitable. Thank you very much in advance, Peter B.

1 0

Achieving key rotation for aes-128 encryption for HLS with ffmpeg
by Aameer Rafiq Wani 15 Mar '17

15 Mar '17

Hey Guys, I was trying to figure out how to achieve key rotation while doing aes-128 encryption for HLS trans-coding using ffmpeg . As it is mentioned here ( http://stackoverflow.com/a/34244568/2833695) it should be possible. Moreover service providers like zencoder.com ( https://app.zencoder.com/docs/api/encoding/encryption/encryption-key-rotati…) also do similar thing, so I am guessing that it is an industry standard, but unfortunately I couldn't figure it out nor could I find any useful material online . Till now I was able to do the encryption and decryption (hls transcoded video), sample command which is working for me is as under. I am doing many other things but the encryption bit is highlighted: ffmpeg -i sintel_trailer-720p.mp4 -profile:v baseline -level 4.0 -vf "scale=-2:360,subtitles='dynamic_subtitle_test.ass':force_style='FontName=common/indee_cloud/Aaargh.ttf,PrimaryColour=&H664c4c4c" -start_number 0 -hls_time 10 -hls_list_size 0 -*hls_key_info_file file.keyinfo* -f hls test_360_.m3u8 bash script which I am using to generate .key and .keyinfo files : #!/bin/sh BASE_URL="file.key" openssl rand 16 > file.key echo $BASE_URL > file.keyinfo echo file.key >> file.keyinfo echo $(openssl rand -hex 16) >> file.keyinfo Any help especially code snippets describing how to achieve the key rotation (i.e use of multiple keys based on different segments) would be ideal, otherwise any suggestions are more than welcome.Thanking all of you in anticipation of a quick and favorable response. Regards, -- Regards, AAMEER RAFIQ WANI email: aameer.rafiq(a)gmail.com websites: aameer.github.io <aameer(a)github.io>

1 0

Add blur-in at a particular time on a video
by Yariv Azatchi 15 Mar '17

15 Mar '17

Hi, I am trying to blur a video after 12 second. But I also want the video to gradually blur for 2 seconds sort of "blur-in" (like you would fade a video in by changing alpha over time). I was only able to do a simple blur for the 12 second and on: ffmpeg -i input.mp4 -vf "boxblur=enable='gte(t,12)'" -strict -2 output.mp4 Can someone help? Thanks for the help, Yariv

1 0

some question on how to use ffmpeg
by guanli 13 Mar '17

13 Mar '17

I encounter some question on how to use ffmpeg , request for your help ,The problem I wrote it in the attachment! File named “ffmpeg_confige_libgsm_error” list the config error . File named ”ffplay_paly_error” list ffplay error. I will be grateful thanks for receiving your response!

3 2

lut 3d trouble
by Bouke / VideoToolShed 13 Mar '17

13 Mar '17

Hi guys, Toying around on holiday, trying to apply a lut on OSX, having no luck. Is it my FFmpeg version? (I’m on very limited bandwidth at the moment, no option to download a recent version for the next days…) Any other stupid simple thing I’m missing? thx, Bouke Boukes-portable:3d bouke$ ffmpeg -threads 0 -i /Users/bouke/Desktop/input.mov -vf lut3d=/Users/bouke/Desktop/LegalLuts/3d/Hard_Clip_Legal_3D.cube -y -an -vcodec prores /Users/bouke/Desktop/HARD.mov ffmpeg version 2.6.1 Copyright (c) 2000-2015 the FFmpeg developers built with Apple LLVM version 6.1.0 (clang-602.0.49) (based on LLVM 3.6.0svn) configuration: --enable-ffplay libavutil 54. 20.100 / 54. 20.100 libavcodec 56. 26.100 / 56. 26.100 libavformat 56. 25.101 / 56. 25.101 libavdevice 56. 4.100 / 56. 4.100 libavfilter 5. 11.102 / 5. 11.102 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 1.100 / 1. 1.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/Users/bouke/Desktop/input.mov': Metadata: major_brand : qt minor_version : 512 compatible_brands: qt encoder : Lavf57.56.100 Duration: 00:00:10.00, start: 0.000000, bitrate: 21173 kb/s Stream #0:0(eng): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, bt709), 1920x1080 [SAR 1:1 DAR 16:9], 21172 kb/s, 25 fps, 25 tbr, 12800 tbn, 25600 tbc (default) Metadata: handler_name : DataHandler [AVFilterGraph @ 0x7fdf30d03580] Error initializing filter 'lut3d' with args '/Users/bouke/Desktop/LegalLuts/3d/Hard_Clip_Legal_3D.cube' Error opening filters! Boukes-portable:3d bouke$

2 6

Convert Audio Track To Numbers and Back
by Michele Monteferrante 13 Mar '17

13 Mar '17

I have a very simple task. Given an audio track (aiff or mp3) I want to convert it in series of real numbers. I do some kind of analysis on this numbers to obtain another series of real numbers (Corresponding to another audio file of the same duration). At this point I want to convert it back to audio file. That you. Michele Monteferrante

4 5

Re: [FFmpeg-user] FFMPEG & Panasonic OP-1B MXF decode issue
by jeff＠jksmith.com 13 Mar '17

13 Mar '17

On 10/03/2017, 09:05, "ffmpeg-user on behalf of Carl Eugen Hoyos" <ffmpeg-user-bounces(a)ffmpeg.org on behalf of ceffmpeg(a)gmail.com> wrote: >2017-03-10 9:20 GMT+01:00 <jeff(a)jksmith.com>: >> Happy to provide a sample. Is there an FTP site or something I >> can upload it to? >You can use https://urldefense.proofpoint.com/v2/url?u=http-3A__streams.videolan.org_up… but you absolutely have to provide the file name, the service is extremely impractical. > Carl Eugen Carl, I did post a reply online to this but it hasn’t appeared in the mail subscription so I’m guessing something went wrong. Anyway, I uploaded the file via the form you provided - 085FV4BR.MXF (approx 17MB).

1 0

Re: [FFmpeg-user] Convert iPhone MOV to Android MP4
by Richard Duran 11 Mar '17

11 Mar '17

On Mar 8, 2017 3:22 PM, "Cley Faye" <cleyfaye(a)gmail.com> wrote: 2017-03-08 23:06 GMT+01:00 Richard Duran <ricdur2(a)gmail.com>: > > Example #2, input with pcm_s16le, input bitrate of 705 kb/s. Output bitrate > defaults to 69 kb/s. I would have expected the output bitrate to match that > of the input. Does AAC support that high of a bitrate? What are you trying to do and to what end? Matching the bitrates between pcm (uncompressed) and AAC (lossy compression) doesn't make much sense. Ah! That distinction didn't even occur to me. To answer your question: I'm trying to retain as much of the source data as possible in moving video from an iPhone to an Android-based phone. I'll try my luck with ALAC instead of AAC. Thanks for the input.

8 18

How to increase buffer for rtmp input
by Tung Tran 10 Mar '17

10 Mar '17

Dear all I have a lot of below alert that I guest there is not enough buffers of the incoming RTMP streams, * Last message repeated 4 times* *frame=107366 fps= 30 q=-0.0 size=N/A time=00:59:42.44 bitrate=N/A dup[decklink @ 0x4488d80] There's no buffered audio. Audio will misbehave!* * Last message repeated 4 times* *frame=107381 fps= 30 q=-0.0 size=N/A time=00:59:42.94 bitrate=N/A dup[decklink @ 0x4488d80] There's no buffered audio. Audio will misbehave!* * Last message repeated 4 times* *frame=107396 fps= 30 q=-0.0 size=N/A time=00:59:43.44 bitrate=N/A dup[decklink @ 0x4488d80] There's no buffered audio. Audio will misbehave!* so I would like to as if we can increase the input buffer for RTMP input.? My command: /root/bin/ffmpeg -vsync 1 -i "rtmp://something" -vf 'setpts=PTS+10000' -af loudnorm='I=-24:LRA=11:tp=-2:linear=true' -ar 48000 -f decklink -pix_fmt uyvy422 "DeckLink SDI (1)" > $log_file 2>&1 & Thank you --- Best regards, *Tung Tran*

1 0

FFMPEG & Panasonic OP-1B MXF decode issue
by jeff＠jksmith.com 10 Mar '17

10 Mar '17

On 09/03/2017, 22:31, "ffmpeg-user on behalf of Carl Eugen Hoyos" <ffmpeg-user-bounces(a)ffmpeg.org on behalf of ceffmpeg(a)gmail.com> wrote: 2017-03-09 11:24 GMT+01:00 <jeffrey.k.smith(a)thomsonreuters.com>: > 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 Please provide a sample and please remove the footer from your emails (you cannot send confidential emails to a public mailing list). Carl Eugen _______________________________________________ Hi Carl, I've flipped to my personal email as I can't control the footer being added through our system. Happy to provide a sample. Is there an FTP site or something I can upload it to? Many thanks, Jeff.

3 2

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.

1 0

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

2 1

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/>

2 1

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.

4 6

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

3 5

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

1 0

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.

2 1

[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

1 0

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)???

2 2

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.

1 2

h264_mp4toannexb unable to split h264_qsv-encoded files?
by Hugo Garbez 03 Mar '17

03 Mar '17

Hello, I'm using FFmpeg 3.2.4 on Debian 8.7, and I have some problems with h264_qsv and HLS. I'm splitting my files for HLS with the following command: $ ffmpeg -i example.mp4 -codec copy -map 0 -bsf:v h264_mp4toannexb -hls_time 5 -hls_init_time 1 -hls_list_size 0 example.m3u8 When example.mp4 is created with libx264, there's no problem with splitting files with HLS: $ ffmpeg -i original.mp4 -codec:v libx264 example.mp4 $ [previous HLS command] $ cat example.m3u8 #EXTM3U #EXT-X-VERSION:3 #EXT-X-TARGETDURATION:5 #EXT-X-MEDIA-SEQUENCE:0 #EXTINF:2.200000, example-0.ts #EXTINF:2.480000, example-1.ts #EXTINF:4.800000, example-2.ts #EXTINF:4.440000, example-3.ts With libx264, -hls_time 5 works (EXT-X-TARGETDURATION and EXTINF look good). But that's not the case if example.mp4 was encoded with h264_qsv instead of libx264: $ ffmpeg -i original.mp4 -codec:v h264_qsv example.mp4 $ [previous HLS command] $ cat example.m3u8 #EXTM3U #EXT-X-VERSION:3 #EXT-X-TARGETDURATION:10 #EXT-X-MEDIA-SEQUENCE:0 #EXTINF:10.000000, example-0.ts #EXTINF:10.000000, example-1.ts #EXTINF:10.000000, example-2.ts #EXTINF:10.000000, example-3.ts #EXTINF:9.640000, As you can see, it looks like "-hls_time 5" has no effect when the video is encoded with h264_qsv instead of libx264. Is there anything I can do about this? Thank you very much.

3 3

Re: [FFmpeg-user] Chroma problem converting mp4 to prores
by David Cecco 03 Mar '17

03 Mar '17

Sorry about the slow response, that has fixed the issue, thanks chronek! The file still seems a bit soft in quality though. I have tried turning the -qscale right down to 1 and adjusting the bits_per_mb up to 2400 and the picture improved but it still wasn’t 100%, but the file grew to 2.5x the size. I tried with the prores codec as opposed to the prores_ks codec and it was better, but again not 100% interestingly the file size didn’t grow when using the bits_per_mb with prores. Here is an example output: $ ffmpeg -y -i "5082197_Soccer Enrique Upd.mp4" -acodec pcm_s16le -vcodec prores_ks -profile:v 1 -vf scale=interl=1 -sws_flags full_chroma_inp+full_chroma_int -copytb -1 "5082197_Soccer Enrique Upd.mov" ffmpeg version 3.2.2-tessus Copyright (c) 2000-2016 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. 34.100 / 55. 34.100 libavcodec 57. 64.101 / 57. 64.101 libavformat 57. 56.100 / 57. 56.100 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, mov,mp4,m4a,3gp,3g2,mj2, from '5082197_Soccer Enrique Upd.mp4': Metadata: major_brand : mp42 minor_version : 0 compatible_brands: mp42mp41 creation_time : 2017-03-01T23:52:21.000000Z Duration: 00:02:15.83, start: 0.000000, bitrate: 10170 kb/s Stream #0:0(eng): Video: h264 (Main) (avc1 / 0x31637661), yuv420p(tv), 1920x1080 [SAR 1:1 DAR 16:9], 9915 kb/s, 25 fps, 25 tbr, 50k tbn, 50 tbc (default) Metadata: creation_time : 2017-03-01T23:52:22.000000Z handler_name : Alias Data Handler encoder : AVC Coding Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 253 kb/s (default) Metadata: creation_time : 2017-03-01T23:52:22.000000Z handler_name : Alias Data Handler Output #0, mov, to '5082197_Soccer Enrique Upd.mov': Metadata: major_brand : mp42 minor_version : 0 compatible_brands: mp42mp41 encoder : Lavf57.56.100 Stream #0:0(eng): Video: prores (prores_ks) (apcs / 0x73637061), yuv422p10le, 1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 200 kb/s, 25 fps, 12800 tbn, 25 tbc (default) Metadata: creation_time : 2017-03-01T23:52:22.000000Z handler_name : Alias Data Handler encoder : Lavc57.64.101 prores_ks Stream #0:1(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, stereo, s16, 1536 kb/s (default) Metadata: creation_time : 2017-03-01T23:52:22.000000Z handler_name : Alias Data Handler encoder : Lavc57.64.101 pcm_s16le Stream mapping: Stream #0:0 -> #0:0 (h264 (native) -> prores (prores_ks)) Stream #0:1 -> #0:1 (aac (native) -> pcm_s16le (native)) Press [q] to stop, [?] for help frame= 3394 fps=7.7 q=-0.0 Lsize= 1522742kB time=00:02:15.80 bitrate=91852.6kbits/s speed=0.31x video:1497227kB audio:25464kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.003344% Thanks! dave > Try -sws_flags full_chroma_inp+full_chroma_int W dniu 2017-02-23 o 23:05, Dave pisze: > Hi guys, > > > > I am having a issue where all the clips that I convert have a chroma > > problem. The red is saturated on all clips and there are red horizontal > > lines in the chroma. > > > > $ ffmpeg -y -i 5079162_NHL\ Marleau.mp4 -acodec pcm_s16le -vcodec prores_ks > > -profile:v 1 -copytb -1 -vf yadif=1 -qscale:v 10 mp4-yadif1-q10.mov > > I thought this might be the chroma bug but I have experimented with a > > couple of different options replacing yadif=1 with the following: > > > > -vf scale=out_color_matrix=bt709 > > -pix_fmt yuv422p10le > > -vf colormatrix=bt601:bt709 > > -vf yadif > > > > The yadif=1 option provides the closest results and specifying multiple > > options seems to make it worse not better. The problem is more obvious on > > a broadcast monitor. > > > > We are also having a problem where the converted clips are 1 frame shorter > > than the original, missing a black frame at the start of the clip. > > > > Here are the details of the clip I am trying to convert: > > > > $ ffprobe -v error -show_format -show_streams 5079162_NHL\ Marleau.mp4 > > [STREAM] > > index=0 > > codec_name=h264 > > codec_long_name=H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 > > profile=Main > > codec_type=video > > codec_time_base=1/50 > > codec_tag_string=avc1 > > codec_tag=0x31637661 > > width=1920 > > height=1080 > > coded_width=1920 > > coded_height=1080 > > has_b_frames=1 > > sample_aspect_ratio=1:1 > > display_aspect_ratio=16:9 > > pix_fmt=yuv420p > > level=41 > > color_range=tv > > color_space=unknown > > color_transfer=unknown > > color_primaries=unknown > > chroma_location=left > > field_order=unknown > > timecode=N/A > > refs=1 > > is_avc=true > > nal_length_size=4 > > id=N/A > > r_frame_rate=25/1 > > avg_frame_rate=25/1 > > time_base=1/50000 > > start_pts=0 > > start_time=0.000000 > > duration_ts=2354000 > > duration=47.080000 > > bit_rate=9081205 > > max_bit_rate=N/A > > bits_per_raw_sample=8 > > nb_frames=1177 > > nb_read_frames=N/A > > nb_read_packets=N/A > > DISPOSITION:default=1 > > DISPOSITION:dub=0 > > DISPOSITION:original=0 > > DISPOSITION:comment=0 > > DISPOSITION:lyrics=0 > > DISPOSITION:karaoke=0 > > DISPOSITION:forced=0 > > DISPOSITION:hearing_impaired=0 > > DISPOSITION:visual_impaired=0 > > DISPOSITION:clean_effects=0 > > DISPOSITION:attached_pic=0 > > DISPOSITION:timed_thumbnails=0 > > TAG:creation_time=2017-02-03T23:55:00.000000Z > > TAG:language=eng > > TAG:handler_name=Alias Data Handler > > TAG:encoder=AVC Coding > > [/STREAM] > > [STREAM] > > index=1 > > codec_name=aac > > codec_long_name=AAC (Advanced Audio Coding) > > profile=LC > > codec_type=audio > > codec_time_base=1/48000 > > codec_tag_string=mp4a > > codec_tag=0x6134706d > > sample_fmt=fltp > > sample_rate=48000 > > channels=2 > > channel_layout=stereo > > bits_per_sample=0 > > id=N/A > > r_frame_rate=0/0 > > avg_frame_rate=0/0 > > time_base=1/48000 > > start_pts=0 > > start_time=0.000000 > > duration_ts=2259840 > > duration=47.080000 > > bit_rate=253262 > > max_bit_rate=283875 > > bits_per_raw_sample=N/A > > nb_frames=2210 > > nb_read_frames=N/A > > nb_read_packets=N/A > > DISPOSITION:default=1 > > DISPOSITION:dub=0 > > DISPOSITION:original=0 > > DISPOSITION:comment=0 > > DISPOSITION:lyrics=0 > > DISPOSITION:karaoke=0 > > DISPOSITION:forced=0 > > DISPOSITION:hearing_impaired=0 > > DISPOSITION:visual_impaired=0 > > DISPOSITION:clean_effects=0 > > DISPOSITION:attached_pic=0 > > DISPOSITION:timed_thumbnails=0 > > TAG:creation_time=2017-02-03T23:55:01.000000Z > > TAG:language=eng > > TAG:handler_name=Alias Data Handler > > [/STREAM] > > [FORMAT] > > filename=5079162_NHL Marleau.mp4 > > nb_streams=2 > > nb_programs=0 > > format_name=mov,mp4,m4a,3gp,3g2,mj2 > > format_long_name=QuickTime / MOV > > start_time=0.000000 > > duration=47.146600 > > size=54956371 > > bit_rate=9325189 > > probe_score=100 > > TAG:major_brand=mp42 > > TAG:minor_version=0 > > TAG:compatible_brands=mp42mp41 > > TAG:creation_time=2017-02-03T23:54:57.000000Z > > [/FORMAT] > > _______________________________________________ > > ffmpeg-user mailing list > > ffmpeg-user at ffmpeg.org > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > > To unsubscribe, visit link above, or email > > ffmpeg-user-request at ffmpeg.org with subject "unsubscribe".

2 1

Converting raw h264 file to MP4
by Varad Gunjal 03 Mar '17

03 Mar '17

Hello. I am posting this question here as this was a suggestion in the output message that I got while using ffmpeg. I have a file that has recorded an RTP stream in raw h264 format. Unfortunately, the data was stored in an unknown format (mjr, designed for the Janus WebRTC gateway). Can someone educate me as to if and how how this raw data can be converted to an mp4 file? Thanks a lot for your time. -- Best Regards, Varad Gunjal

2 1

filter: subtitles and special chars
by Mike F 02 Mar '17

02 Mar '17

Hello, Due to the fact that I cannot get "-i my.srt" and "sub_charenc" to work, even after ( I guess ) satisfying 'iconv' requirements during the compile of N-83701-g70ebc05-1 20170205, I found the "subtitles" filter and 'charenc=CP1250' perfectly doing the job. The only small issues is that the subtitle filename.srt can not use any characters that need to be escaped, such as '[' ']' etc. When issuing this option: -filter:v scale=960:-1,subtitles=myvidfile\[1of3\].srt:charenc=CP1250:force_style='Fontsize=24:original_size=hd480' the "myvidfile\[1of10\].srt" name causes ffmpeg to error out with: [Parsed_subtitles_1 @ 0x5635ef66d280] Unable to open myvidfile [AVFilterGraph @ 0x5635ef667560] Error initializing filter 'subtitles' with args 'myvidfile' wrapping the myvidfile[1of10].srt in double or single quotes does not help... Thank you! Mike

4 4

Re: [FFmpeg-user] filter: subtitles and special chars
by Mike F 02 Mar '17

02 Mar '17

Hello Nicolas, Let me start with this first, to clarify. < Hello. You just hijacked a thread. It is forbidden on this list. I suggest you re-read the posting guidelines carefully in order to not do anything forbidden again. > I am aware of the rules... and none of: "thread hijacking (that is, replying to a thread and changing the subject line to something completely unrelated that was not being discussed within the original thread)" has been done by me - a simple new "compose" and I pasted in the email address that i got in the initial email, after receiving the confirmation of the mailing list subscription. I am not sure how did you, or the system process that message, but "hijacking" was not done on my side... if there is a "real" problem with the msg showing up as "hijacked", i might think about dumping yahoo mail, or there is an issue on the opposite side. maybe I just subscribed to ffmpeg list, but I am a returning "ffmpeg user questions" mailing list subscriber - I was one between 2012-2014 before. going back to what you mentioned about being precise about the cmdline, and errors... I could provide the information in a "proper" form, I will try this now: ===================== $ ffmpeg -y -i myvid\[1of10\].mkv -c:v libx264 -coder 1 -bf 3 -refs 3 -subq 10 -partitions +parti4x4+parti8x8+partp8x8 -g 100 -keyint_min 25 -qmin 1 -qmax 51 -trellis 2 -sc_threshold 40 -threads 48 -filter:v scale=960:-1,subtitles=myvid\[1of10\].srt:charenc=CP1250:force_style='Fontsize=24:original_size=hd480' -c:a aac -ab 96k -ac 2 test.mp4 ffmpeg version N-83701-g70ebc05-1 Copyright (c) 2000-2017 the FFmpeg developers built with gcc 6.3.0 (Debian 6.3.0-6) 20170205 <obvious, snip> creation_time : 2017-03-02T16:34:40.000000Z handler_name : GPAC MPEG-4 Scene Description Handler [Parsed_subtitles_1 @ 0x564d1346b020] Shaper: FriBidi 0.19.7 (SIMPLE) HarfBuzz-ng 1.4.2 (COMPLEX) [Parsed_subtitles_1 @ 0x564d1346b020] Unable to open myvid [AVFilterGraph @ 0x564d132e5e40] Error initializing filter 'subtitles' with args 'myvid' Error opening filters! $ ====================== I would apologize, if I would hijack the thread... Thank you, Mike

1 0

How to specify Audio bit depth for test sources?
by Basin Ilya 02 Mar '17

02 Mar '17

Hi. Different test sources have different bit depth: ffplay -f lavfi aevalsrc="0" # Audio: pcm_f64le ffplay -f lavfi anullsrc # Audio: pcm_u8 If I want to do some PCM math in the audio filter, I'll need to adapt the filter for each different input bit depth. There's no filter to change the depth. "aresample" only changes the sample rate. -acodec option won't do here, because it takes effect AFTER filtering and some tools don't have it (ffplay). -isf: I was unable to make it work The only workaround is to use /dev/zero as the source. Then you can specify the needed input format.

2 3

ogg encoder always adds 128-255 extra samples
by Basin Ilya 02 Mar '17

02 Mar '17

Hi. I'm trying to create a set of chainable ogg files out of a long wav file. However, the produced ogg files are always 128-255 samples longer than the source PCM data and the extra data looks like anti-click or fade-out. See the attached image. test0 was made directly. test1 was made via an intermediate ogg file. I tried to choose a duration whose sample count is divisible by 64 or 128, but that didn't help. # produce 10 seconds of noise ffmpeg -y -t 10 -f lavfi -i aevalsrc="random(0)*2-1" -c:a pcm_s16le test.wav ( </dev/null ffmpeg -y -ss 0 -t 0.995567 -i test.wav -af "ashowinfo" test0.wav </dev/null ffmpeg -y -ss 0 -t 0.995567 -i test.wav -af "ashowinfo" test1.ogg # pts:43008 nb_samples:897 </dev/null ffmpeg -y -i "test1.ogg" -af "ashowinfo" test1.wav # pts:44032 nb_samples:128 ) _ The only workaround I see is to make a slightly longer ogg, then use "-acodec copy" and "-t" to discard 1 or 2 unneeded ogg frames.

3 4

Video and Audio is off-sync on Decklink SDI output
by Tung Tran 02 Mar '17

02 Mar '17

Dear all I am testing to setup a decoder with ffmpeg, take live stream in (RTMP/HLS,.) and output to Decklink SDI. Its working but I have the issue of audio and video is off-sync on decklink SDI output, I tried to play around with vf setpts and af asetpts but couldnt make video/audio to sync. Any input will be very appreciated. Here is my command so far * ffmpeg -i rtmp://something -vf "setpts=PTS-STARTPTS" -af "asetpts=PTS-STARTPTS" -f decklink -pix_fmt uyvy422 "DeckLink SDI" * Video and audio of input rtmp is synced on VLC or other software. --- Best regards, *Tung Tran*

4 5

libass : ERROR NOT in Path in ffmpeg compile
by derekg 02 Mar '17

02 Mar '17

Dear Help , Hi , I am trying to set up a fully working version of ffmpeg on my Linux Mint Qiana 17 OS , using thecomprehensive guide https://trac.ffmpeg.org/wiki/CompilationGuide/Ubuntu . Please find attached a config.log complaining about a libass package not in the path . Also please see a second attached file pkg-configversion.txt showing releases data . I am an English old-age-pensioner and using Mint's Synaptic Package Manager I had Completely Removed ffmpeg version 7:2.4.3-1ubuntu1~trusty before following the guide . Any suggestions will be gratefully received and requests for more data will be acted upon . Thank you , Derek Hopley

3 4

minimal nb_sambles in ogg frame
by Basin Ilya 02 Mar '17

02 Mar '17

Hi. I'm splitting a .wav file into a set of chainable .ogg files all having 1 second duration. I'm using input flags: "-ss n -t 1" When I concatenate these ogg files, the resulting clip is slightly longer than the original. Is it because the minimal nb_sambles in ogg frame is 128 and a set of ogg frames cannot contain exactly 44100 samples? </dev/null ffmpeg -y -ss 0 -t 1 -i test.wav -af "ashowinfo" test1.ogg # pts:44032 nb_samples:68 </dev/null ffmpeg -y -ss 1 -t 1 -i test.wav -af "ashowinfo" test2.ogg # pts:44032 nb_samples:68 </dev/null ffmpeg -y -i "concat:test1.ogg|test2.ogg" -af "ashowinfo" test5.wav # pts:88132 nb_samples:128 88132 + 128 != 88200 ffprobe test5.wav # Duration: 00:00:02.01

2 1

Stream from a dynamic collection
by Jorge Reyero 02 Mar '17

02 Mar '17

Hi, I'm new using ffmpeg so I don't know if what I want is possible or not, I hope you can tell. I have a video collection (place in the server filesystem) which is growing every minute. I want to simulate a live stream from this video collection and if the stream reaches the end of the videos it should start form the first one again. I know how to repeat a single video and how to play videos from static video collection but I have no idea how do it with this dynamic collection. Any idea? Is it possible to do that? Thanks

3 2

Difference in ffmpeg command execution time
by Vivek Jain 01 Mar '17

01 Mar '17

Hi, On trying to create video using ffmpeg as per command below, it takes much more time (around 6 minutes) on AWS lambda and EC2 instances where as it takes just 2 minutes on my local mac. I understand run time differs on CPU, but I am not able to figure out what needs to be done in order to get it run within 5 minutes on lambda. Or it could be due to differences in or missing libraries. Could someone please point me in right direction? Command: ffmpeg -safe 0 -f concat -i /tmp/config.txt -i /tmp/audio.mp3 -y -pix_fmt yuv444p -shortest /tmp/out_ffff.mp4 Attached are the logs of both the machines. -- Thanks, Vivek

2 3

CodecPrivateData is empty when h264_vaapi encoder used
by w_boba 01 Mar '17

01 Mar '17

I am trying to use new CPU with VA-API enabled h264 encoder on FFmpeg (ffmpeg version n3.2.4-4-g36fff6c). I have no issues encoding "offline" files with h264_vaapi encoder, but when I try to encode live streams and push output to SmoothStream server (using ISMV muxer) new streams do not work. Exactly same configuration works fine with libx264 encoder. So far I figured that after libx264 encoder ffmpeg correctly sets "CodecPrivateData" field data but h264_vaapi sends empty data field. Old command line (works fine): ffmpeg -y -i "udp://235.1.1.136:392" -g 30 -r ntsc -movflags isml+frag_keyframe -sc_threshold 0 -c:v libx264 -c:a aac -ac 2 -b:a 128k -b:v 3072k -f ismv "http://10.100.1.14/testenc1.isml/Streams(testenc1)" New command line (encoding works, but streams do not play): ffmpeg -y -vaapi_device /dev/dri/renderD128 -i "udp://235.1.1.136:392" -vf 'format=nv12,hwupload' -g 30 -r ntsc -movflags isml+frag_keyframe -sc_threshold 0 -c:v h264_vaapi -c:a aac -ac 2 -b:a 128k -b:v 3072k -f ismv "http://10.100.1.14/testenc1.isml/Streams(testenc1)" -- View this message in context: http://www.ffmpeg-archive.org/CodecPrivateData-is-empty-when-h264-vaapi-enc… Sent from the FFmpeg-users mailing list archive at Nabble.com.

4 9