June 2017 - ffmpeg-user - lists.ffmpeg.org

amix filter hanging
by Jonathan Girven 29 Aug '17

29 Aug '17

I am trying to merge the audio of my video with a music track, whilst keeping the picture part the same. To do this I am trying to use the amix filter, but FFmpeg seems to hang when I run the command. Am I doing something wrong? $ ffmpeg -y -i aae635a604ff16c9a320376398ac121b_0.000_5.015.mp4 -i full_dillon_in_pro_audio.mp3 -filter_complex "[1:a][0:a]amix=inputs=2[m]" -map 0:v -map [m] -c:v libx264 -preset ultrafast -c:a aac 4615491572977860225_PLzGS5hkLTwTfxlgkk0.mp4 ffmpeg version N-86394-g66cf78e Copyright (c) 2000-2017 the FFmpeg developers built with gcc 4.9.2 (Debian 4.9.2-10) configuration: --prefix=/usr/local --extra-cflags=-I/usr/local/include --extra-ldflags=-L/usr/local/lib --bindir=/usr/local/bin --disable-doc --disable-static --enable-shared --disable-ffplay --extra-libs=-ldl --enable-version3 --enable-libfreetype --enable-libx264 --enable-gpl --enable-openssl --enable-nonfree --disable-debug 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. 91.100 / 6. 91.100 libswscale 4. 7.101 / 4. 7.101 libswresample 2. 8.100 / 2. 8.100 libpostproc 54. 6.100 / 54. 6.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'aae635a604ff16c9a320376398ac121b_0.000_5.015.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 encoder : Lavf57.71.100 Duration: 00:00:05.04, start: 0.000000, bitrate: 656 kb/s Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], 522 kb/s, 30 fps, 30 tbr, 15360 tbn, 60 tbc (default) Metadata: handler_name : VideoHandler Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 127 kb/s (default) Metadata: handler_name : SoundHandler [mp3 @ 0xf20a60] Estimating duration from bitrate, this may be inaccurate Input #1, mp3, from 'full_dillon_in_pro_audio.mp3': Metadata: title : Dillon in Pro Audio artist : Jingle Punks album : YouTube Audio Library genre : Dance & Electronic encoder : Google Duration: 00:01:18.84, start: 0.000000, bitrate: 320 kb/s Stream #1:0: Audio: mp3, 44100 Hz, stereo, s16p, 320 kb/s Stream mapping: Stream #0:1 (aac) -> amix:input1 (graph 0) Stream #1:0 (mp3) -> amix:input0 (graph 0) Stream #0:0 -> #0:0 (h264 (native) -> h264 (libx264)) amix (graph 0) -> Stream #0:1 (aac) Press [q] to stop, [?] for help [libx264 @ 0xf338e0] using SAR=1/1 [libx264 @ 0xf338e0] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX FMA3 AVX2 LZCNT BMI2 [libx264 @ 0xf338e0] profile Constrained Baseline, level 3.1 [libx264 @ 0xf338e0] 264 - core 148 - H.264/MPEG-4 AVC codec - Copyleft 2003-2016 - http://www.videolan.org/x264.html - options: cabac=0 ref=1 deblock=0:0:0 analyse=0:0 me=dia subme=0 psy=1 psy_rd=1.00:0.00 mixed_ref=0 me_range=16 chroma_me=1 trellis=0 8x8dct=0 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=0 threads=12 lookahead_threads=2 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=0 weightp=0 keyint=250 keyint_min=25 scenecut=0 intra_refresh=0 rc=crf mbtree=0 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=0 Output #0, mp4, to '4615491572977860225_PLzGS5hkLTwTfxlgkk0.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 encoder : Lavf57.72.101 Stream #0:0(und): Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv420p(progressive), 1280x720 [SAR 1:1 DAR 16:9], q=-1--1, 30 fps, 15360 tbn, 30 tbc (default) Metadata: handler_name : VideoHandler encoder : Lavc57.96.101 libx264 Side data: cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: -1 Stream #0:1: Audio: aac (LC) ([64][0][0][0] / 0x0040), 44100 Hz, stereo, fltp, 128 kb/s (default) Metadata: encoder : Lavc57.96.101 aac <hang for more than 10 minutes> I can email the files if helpful.

3 7

could not find codec parameters for stream 0 unspecified size
by Ashish Lal 09 Aug '17

09 Aug '17

I am trying to convert a live rtsp stream (H264 + MPEG2 TS ) into MP4 using ffmpeg. I am getting an error "could not find codec parameters for stream 0 unspecified size". Any help in fixing this is appreciated. ffmpeg -rtsp_transport tcp -i rtsp://AA.BB.CC.DD:5555/mpeg2TransportStreamFr omUDPSourceTest -analyzeduration 999999999 -probesize 99999999 -vcodec libx264 -s 176x144 -b:v 256k -f mp4 -movflags frag_keyframe+empty_moov op.mp4 ffmpeg version 2.2.2 Copyright (c) 2000-2014 the FFmpeg developers built on May 27 2014 09:12:51 with gcc 4.7 (Debian 4.7.2-5) configuration: --enable-libx264 --enable-encoder=libx264 --enable-libvpx --pre fix=/home/ashish/ffmpeg-2.2.2/mybuild --enable-gpl --enable-debug --disable-stri pping --disable-optimizations --enable-memalign-hack libavutil 52. 66.100 / 52. 66.100 libavcodec 55. 52.102 / 55. 52.102 libavformat 55. 33.100 / 55. 33.100 libavdevice 55. 10.100 / 55. 10.100 libavfilter 4. 2.100 / 4. 2.100 libswscale 2. 5.102 / 2. 5.102 libswresample 0. 18.100 / 0. 18.100 libpostproc 52. 3.100 / 52. 3.100 [NULL @ 0x235cd60] non-existing PPS 0 referenced [h264 @ 0x235cd60] non-existing PPS 0 referenced [h264 @ 0x235cd60] decode_slice_header error [h264 @ 0x235cd60] no frame! [h264 @ 0x235cd60] non-existing PPS 0 referenced ......... ......... Last message repeated 1 times [h264 @ 0x235cd60] decode_slice_header error [h264 @ 0x235cd60] no frame! [h264 @ 0x235cd60] non-existing PPS 0 referenced [rtsp @ 0x232fc40] decoding for stream 0 failed [rtsp @ 0x232fc40] Could not find codec parameters for stream 0 (Video: h264 ([2 7][0][0][0] / 0x001B)): unspecified size Consider increasing the value for the 'analyzeduration' and 'probesize' options rtsp://AA.BB.CC.DD:5555/mpeg2TransportStreamFromUDPSourceTest: could not find codec parameters Conversion failed! Regards Ashish

3 3

Streaming Syntax error
by Mettavihari D 25 Jul '17

25 Jul '17

Greetings I run a server that is streaming out to a website on the Net I use the following command which is working. /root/bin/ffmpeg -re -i /data/video/sinhala/innput-15m.mp4 -acodec copy -vcodec copy -f flv -metadata streamName=xlarge rtmp://some.site.com. My client is complaining that the bit rate is varying too much and they want the bit-rate never to exceed 2m. They would also prefer if I can send them a constant bit rate. I looked at the Net and found the suggestion to use "-maxrate 1.5M -bufsize 3M" , I tried the following: /root/bin/ffmpeg -re -i /data/video/sinhala/innput-15m.mp4 -acodec copy -vcodec copy -maxrate 1.5M -bufsize 3M -f flv -metadata streamName=xlarge rtmp://some.site.com. but I get this error: Systax error. I do not know what the syntax error can be and would much appreciate to have a word of advice. regards Mettavihari

4 16

Create an AAC stream matching the Core Media Audio packet format / priming etc?
by Mark Burton 18 Jul '17

18 Jul '17

I appreciate this is a tricky area and there appear to be different ways that some encoders create AAC streams with regards to the padding and remaining samples etc. I won’t pretend to fully understand all the factors, but I would like to ask a genuine question which purely comes from the point of view of wanting to create a file for my working environment - an environment dominated by Quicktime 7 and X playback / decoding tools. I’m no great fan of Quicktime and appreciate its not well loved here also. In my industry it is still very much the defecto playback engine though, so if I’m able to tailor a file for this decoder, it would be an enormous help. Let me also say, I am not accusing ffmpeg of having an issue, I have read a number of ‘bug’ reports surrounding ffmpeg and AAC priming and there being a sync discrepancy in the resultant encode when played back in certain decoders. I happen to see the exact same issue, but from some of the developers replies, I accept their position is that they feel ffmpeg is doing it the right way and its the decoders that are at fault. With that said, I’d like to approach this question purely from the point of view of finding out whether there is a way to tweak a command in order to change this way the aac stream is created to produce an mp4 or mov file using the native aac encoder which decodes in Quicktime 7 or X, in sync. Currently an encoded file plays back 1 frame out of sync (audio is early by approx. 1 frame). In VLC its about 1/2 a frame out of sync. The source file is a .mov, DNx115 24p (true 24p, not 23.976), PCM 24bit 48khz audio, which is in sync. This is film material where sync is crucial and always expected. Here is the basic command to reproduce. I have attached the uncut loglevel 99 console output for this command: ffmpeg -i SyncTest24p.mov -c:v libx264 -pix_fmt yuv420p -movflags faststart -c:a aac -b:a 128k ffmpeg.mp4 ffmpeg -i SyncTest24p.mov -c:v libx264 -pix_fmt yuv420p -movflags faststart -c:a aac -b:a 128k ffmpeg.mp4 ffmpeg version N-85343-gd0a3143-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-libzvbi --enable-version3 --disable-ffplay --disable-indev=qtkit libavutil 55. 60.100 / 55. 60.100 libavcodec 57. 92.100 / 57. 92.100 libavformat 57. 72.100 / 57. 72.100 libavdevice 57. 7.100 / 57. 7.100 libavfilter 6. 84.100 / 6. 84.100 libswscale 4. 7.100 / 4. 7.100 libswresample 2. 8.100 / 2. 8.100 libpostproc 54. 6.100 / 54. 6.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'SyncTest24p.mov': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt creation_time : 2017-04-14T13:37:46.000000Z timecode : 01:00:00:00 Duration: 00:00:02.00, start: 0.000000, bitrate: 118705 kb/s Stream #0:0(eng): Video: dnxhd (DNXHD) (AVdn / 0x6E645641), yuv422p(tv, bt709/unknown/unknown), 1920x1080, 116391 kb/s, SAR 1:1 DAR 16:9, 24 fps, 24 tbr, 24k tbn, 24k tbc (default) Metadata: creation_time : 2017-04-14T13:37:46.000000Z handler_name : Apple Alias Data Handler encoder : Avid DNxHD Codec Stream #0:1(eng): Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, stereo, s32 (24 bit), 2304 kb/s (default) Metadata: creation_time : 2017-04-14T13:37:46.000000Z handler_name : Apple Alias Data Handler Stream #0:2(eng): Data: none (tmcd / 0x64636D74), 0 kb/s (default) Metadata: creation_time : 2017-04-14T13:37:46.000000Z handler_name : Apple Alias Data Handler timecode : 01:00:00:00 Stream mapping: Stream #0:0 -> #0:0 (dnxhd (native) -> h264 (libx264)) Stream #0:1 -> #0:1 (pcm_s24le (native) -> aac (native)) Press [q] to stop, [?] for help [libx264 @ 0x7ff1ae809200] using SAR=1/1 [libx264 @ 0x7ff1ae809200] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX FMA3 AVX2 LZCNT BMI2 [libx264 @ 0x7ff1ae809200] profile High, level 4.0 [libx264 @ 0x7ff1ae809200] 264 - core 148 - H.264/MPEG-4 AVC codec - Copyleft 2003-2016 - 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=24 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, mp4, to 'ffmpeg.mp4': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt timecode : 01:00:00:00 encoder : Lavf57.72.100 Stream #0:0(eng): Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], q=-1--1, 24 fps, 12288 tbn, 24 tbc (default) Metadata: creation_time : 2017-04-14T13:37:46.000000Z handler_name : Apple Alias Data Handler encoder : Lavc57.92.100 libx264 Side data: cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: -1 Stream #0:1(eng): Audio: aac (LC) ([64][0][0][0] / 0x0040), 48000 Hz, stereo, fltp (24 bit), 128 kb/s (default) Metadata: creation_time : 2017-04-14T13:37:46.000000Z handler_name : Apple Alias Data Handler encoder : Lavc57.92.100 aac [mp4 @ 0x7ff1ae81ec00] Starting second pass: moving the moov atom to the beginning of the file frame= 48 fps=0.0 q=-1.0 Lsize= 24kB time=00:00:02.00 bitrate= 97.2kbits/s speed= 3.6x video:17kB audio:3kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 15.142209% [libx264 @ 0x7ff1ae809200] frame I:1 Avg QP:18.80 size: 4448 [libx264 @ 0x7ff1ae809200] frame P:14 Avg QP:16.87 size: 374 [libx264 @ 0x7ff1ae809200] frame B:33 Avg QP:23.31 size: 227 [libx264 @ 0x7ff1ae809200] consecutive B-frames: 2.1% 16.7% 6.2% 75.0% [libx264 @ 0x7ff1ae809200] mb I I16..4: 52.6% 46.2% 1.2% [libx264 @ 0x7ff1ae809200] mb P I16..4: 0.0% 0.0% 0.1% P16..4: 0.2% 0.1% 0.0% 0.0% 0.0% skip:99.6% [libx264 @ 0x7ff1ae809200] mb B I16..4: 0.0% 0.0% 0.0% B16..8: 0.2% 0.1% 0.0% direct: 0.0% skip:99.7% L0:60.9% L1:38.1% BI: 1.0% [libx264 @ 0x7ff1ae809200] 8x8 transform intra:45.8% inter:14.5% [libx264 @ 0x7ff1ae809200] coded y,uvDC,uvAC intra: 1.4% 0.0% 0.0% inter: 0.0% 0.0% 0.0% [libx264 @ 0x7ff1ae809200] i16 v,h,dc,p: 97% 0% 3% 0% [libx264 @ 0x7ff1ae809200] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 62% 2% 36% 0% 0% 0% 0% 0% 0% [libx264 @ 0x7ff1ae809200] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 30% 16% 33% 5% 2% 4% 2% 6% 2% [libx264 @ 0x7ff1ae809200] i8c dc,h,v,p: 100% 0% 0% 0% [libx264 @ 0x7ff1ae809200] Weighted P-Frames: Y:0.0% UV:0.0% [libx264 @ 0x7ff1ae809200] ref P L0: 70.5% 1.8% 16.3% 11.3% [libx264 @ 0x7ff1ae809200] ref B L0: 58.5% 32.1% 9.4% [libx264 @ 0x7ff1ae809200] ref B L1: 90.6% 9.4% [libx264 @ 0x7ff1ae809200] kb/s:68.72 [aac @ 0x7ff1ae80aa00] Qavg: 61514.836 Attached is an ffprobe log file showing the audio packets for this ffmpeg encode. Also, I ran the same source file through Apple Compressor v4.3.1 to the same output file spec and through YouTube and have attached ffprobe logs of their audio packets. Apologies if I’ve not gone about this the right way, I’m not trying to demonstrate a bug, simply want to highlight the differences between the initial packets of these files and see if there is a way to produce a file via ffmpeg which uses a packet ‘setup’ similar to the Compressor or even YouTube way as this plays very nice with Quicktime decoders. Perhaps the key here is that ffmpeg has a negative pts for the first packet and this is what Quicktime struggles with I would think. It also has SIDE DATA which Quicktime may also be struggling with, but I’ve not found much info on this and from my understanding Quicktime always assumes 2112 samples of delay at the start. https://developer.apple.com/library/content/technotes/tn2258/_index.html Start of ffmpeg file: [PACKET] codec_type=audio stream_index=1 pts=-1024 pts_time=-0.021333 dts=-1024 dts_time=-0.021333 duration=1024 duration_time=0.021333 convergence_duration=N/A convergence_duration_time=N/A size=300 pos=8722 flags=KD [SIDE_DATA] side_data_type=Skip Samples skip_samples=1024 discard_padding=0 skip_reason=0 discard_reason=0 [/SIDE_DATA] [/PACKET] [PACKET] codec_type=audio stream_index=1 pts=0 pts_time=0.000000 dts=0 dts_time=0.000000 duration=1024 duration_time=0.021333 convergence_duration=N/A convergence_duration_time=N/A size=410 pos=9351 flags=K_ [/PACKET] ... Start of Compressor file: [PACKET] codec_type=audio stream_index=0 pts=0 pts_time=0.000000 dts=0 dts_time=0.000000 duration=1024 duration_time=0.021333 convergence_duration=N/A convergence_duration_time=N/A size=6 pos=2320 flags=K_ [/PACKET] ... Start of YouTube file: [PACKET] codec_type=audio stream_index=1 pts=0 pts_time=0.000000 dts=0 dts_time=0.000000 duration=1024 duration_time=0.023220 convergence_duration=N/A convergence_duration_time=N/A size=557 pos=9043 flags=K_ [/PACKET] … Thank you if you have read this far, I really appreciate it. I’m slightly worried this is going to open a can of worms which will be above my pay grade! Anyway, assuming I have provided enough info (and I can send anyone the source file if they want it), are there any options / flags etc I could add to the command to bring the encode in line with the Compressor or YouTube outputs in terms of the aac audio stream? Many many thanks for any help, Regards Mark

11 50

Converting a cassette audio tape (Lecture) to mp3
by jd1008 14 Jul '17

14 Jul '17

Hi all, I am running fedora 20 with ffmpeg version N-51556-ge278500 Copyright (c) 2000-2013 the FFmpeg developers I also installed in /opt ffmpeg version N-72985-g7c3f7e6 Copyright (c) 2000-2015 the FFmpeg developers I need a set of steps/commands to convert this tape. I have plenty of audio tools installed - but not certain what else I might need. If anyone has done this before, please share your script. Kind regards, JD

6 15

Using FFMPEG to encode multiple outputs in 'parallel' (doesn't work?)
by Shalin Mehta 06 Jul '17

06 Jul '17

Hello Community members, Quoting from the ffmpeg's wiki for creating multiple outputs. https://trac.ffmpeg.org/wiki/Creating%20multiple%20outputs *Parallel encoding*: Outputting and re encoding multiple times in the same FFmpeg process will typically slow down to the "slowest encoder" in your list. Some encoders (like libx264) perform their encoding "threaded and in the background" so they will effectively allow for parallel encodings. Unfortunately, my results are not correlating with what is mentioned above. By that I mean, I am not seeing encoding being performed in parallel on multiple outputs even when using libx264. To simplify the problem and to understand it easily, here's what I am doing for strictly benchmarking purposes. Option1: Run 3 instances of ffmpeg using pipe. ffmpeg -i input.mp4 -c:v libx264 -b:v 12M -f null - | ffmpeg -i input.mp4 -c:v libx264 -b:v 12M -f null - | ffmpeg -i input.mp4 -c:v libx264 -b:v 12M -f null - Option2: run all three encodes in single ffmpeg process ffmpeg -i input.mp4 -c:v libx264 -b:v 12M -f null - -c:v libx264 -b:v 12M -f null - -c:v libx264 -b:v 12M -f null - I am executing this workload on 32 core platform on google cloud. Option1 gives me nearly 100% cpu utilization. In case of Option 2 cpu utilization doesn't go beyond 50% and also fps is about 70% of the option1. Then why don't I just use Option1? Because Option2 decodes the input only once. I need to generate many streams (up to 25 streams) from the same input file which could be 4K or even bigger. So I want to avoid the overhead of decoding the same input so many times in Option1. So basically, I want to make Option2 work but I am not understanding why it's not running all the encodes in parallel to give me better fps and maximum cpu utilization. After taking a quick look at the code, it doesn't seem like ffmpeg is creating multiple threads to run multiple x264 encode sessions in parallel. I know x264 library itself will create multiple threads but that's for a single output. What I am looking for is output level parallelism. Can some one explain if this is possible? Or may be, I can use some different kind of command line to achieve such output level parallelism? Shalin

8 9

Conversion wav -> mp3 -> wav
by Rodolfo Medina 02 Jul '17

02 Jul '17

Hi all. As an experiment, I converted a .wav file to mp3 format and then back into wav again, just to see what happens: $ ffmpeg -i file1.wav file1.mp3 $ ffmpeg -i file1.mp3 file2.wav I've always heard and read that the first step produces a loss in quality. So I would expect that to be seen in a reduction of size. Instead, I was suprised to see that file1.wav and file2.wav are both 154M large. Also the output of `ffmpeg -i' is almost the same for the two: in both cases, there is: Duration: 00:15:10.84, bitrate: 1411 kb/s Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s So I wonder, and am asking to you listers, in where that quality loss is shown and how it can be detected. Or maybe should we think and conclude that the original quality is restored with the second step...? Thanks for any help, Rodolfo

12 23

How to convert 1 video type to another based on -i
by Dan Violet Sagmiller 01 Jul '17

01 Jul '17

I have a video player with poor documentation. I have one video that works (that came with a demo). Is there any documentation that makes it easy to look at the info from one video and convert another video to match it’s codec/audio? I’m relatively new to using ffmpeg. This is from the working video (A): Stream #0:0(eng): Audio: aac (LC) (mp4a / 0x6134706D), 22050 Hz, stereo, fltp, 65 kb/s (default) Metadata: creation_time : 2010-02-09T01:55:39.000000Z handler_name : Apple Sound Media Handler Stream #0:1(eng): Video: h264 (Constrained Baseline) (avc1 / 0x31637661), yuv420p(tv, smpte170m/smpte170m/bt709), 640x360, 612 kb/s, 23.96 fps, 24 tbr, 600 tbn, 1200 tbc (default) This is from the one that doesn’t work (B): Stream #0:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 3840x1920 [SAR 1:1 DAR 2:1], 60520 kb/s, 30.03 fps, 30 tbr, 15360 tbn, 60 tbc (default) Metadata: handler_name : VideoHandler Side data: spherical: equirectangular (0.000000/0.000000/0.000000) Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 193 kb/s (default) I want to figure out the right switches for this case, but also like to figure a better way to figure this out. 😊 * Thanks. (I apologize if this is the wrong contact group, but it seemed the best I could find after searching a while)

2 1

Crop and transpose filters combined produces large output
by Clay D. Montgomery 29 Jun '17

29 Jun '17

Hello, I have been experimenting with using the video crop and transpose filters with ffmpeg (version 2.5.2 on Windows) to convert 4K video into 1080p video without scaling. When I combine these filters in a 1 pass operation, I get an output (File4.mp4) that is about 4X larger than it should be. But, when I split the same operation into 2 passes, I get an output (File3.mp4) that seems to be the correct size. The details are: Crop and transpose in 1 pass: ffmpeg -i File1.mp4 -vf crop=1080:1920:0:0 -vf transpose=2 File4.mp4 File sizes in bytes: File1.mp4 79,889,077 (Input 4K Video) File4.mp4 25,918,758 (Excessive Size) Crop and transpose in 2 passes: ffmpeg -i File1.mp4 -vf crop=1080:1920:0:0 File2.mp4 ffmpeg -i File2.mp4 -vf transpose=2 File3.mp4 File sizes in bytes: File1.mp4 79,889,077 (Same Input 4K Video) File2.mp4 6,994,548 File3.mp4 5,777,801 (Correct Size) All of these files can be played with ffplay, but I use other video players and they have trouble playing File4.mp4. It is not a big problem to use 2 passes, but I'm curious if this is a known issue or am I missing something? Thanks, Clay

3 4

VideoStream from a Cloud VM with gdigrab window capture withut Remote Desktop
by Thadeu Antonio Ferreira de Melo 28 Jun '17

28 Jun '17

Hello. We are trying to create an automated cloud stream for a 3D application running on an Azure Windows VM. We use gdigrab to capture that window and FFMPEG streams it to a FFSERVER Our command is this ffmpeg -f gdigrab -i title="Win1 (64-bit, PCD3D_SM5)" -pix_fmt yuv420p -framerate 30 -draw_mouse 0 -vcodec libx264 -tune zerolatency -bf 0 -preset ultrafast -crf 25 -g 30 -an -refs 4 -x264-params vbv-maxrate=2000:vbv-bufsize=256:keyint=600 http://<ServerIP>/feed1.ffm The problem is that, so far, we can only run tests while we have a Remote Desktop instance opened. If one minimises the RD the stream stops. Our go is to have a script that could start this command from any cloud VM with diffent configurations for different instances of the 3D app. Our VM are Windows Server 2016. Thank you.

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