February 2014 - ffmpeg-user - lists.ffmpeg.org

How to find if ffmpeg uses software decoder or hardware decoder for H.264?
by whoami Jils 24 Feb '14

24 Feb '14

HI, I have ported ffmpeg in android, compiled with libx264 library. how can i identify if ffmpeg uses hardware encoder or software encoder?

2 2

Error FFmpeg compile
by Karthi Keyan 24 Feb '14

24 Feb '14

Hello sir, When trying to compile ffmpeg with Android_NDK_r9c i got the following error plase help me to Clear this error. Thank you ./build_android.sh /home/karthikeyan/home/karthikeyan/MyFiles/datas/applications/inst-packs/android-ndk-r9c/toolchains/arm-linux-androideabi-4.8/prebuilt/linux-x86_64/bin/arm-linux-androideabi-gcc is unable to create an executable file. C compiler test failed. If you think configure made a mistake, make sure you are using the latest version from Git. If the latest version fails, report the problem to the ffmpeg-user(a)ffmpeg.org mailing list or IRC #ffmpeg on irc.freenode.net. Include the log file "config.log" produced by configure as this will help solving the problem. sed: can't read config.h: No such file or directory sed: can't read config.h: No such file or directory sed: can't read config.h: No such file or directory sed: can't read config.h: No such file or directory sed: can't read config.h: No such file or directory sed: can't read config.h: No such file or directory Makefile:2: config.mak: No such file or directory Makefile:45: /common.mak: No such file or directory Makefile:82: /libavutil/Makefile: No such file or directory Makefile:159: /doc/Makefile: No such file or directory Makefile:160: /tests/Makefile: No such file or directory make: *** No rule to make target `/tests/Makefile'. Stop. Makefile:2: config.mak: No such file or directory Makefile:45: /common.mak: No such file or directory Makefile:82: /libavutil/Makefile: No such file or directory Makefile:159: /doc/Makefile: No such file or directory Makefile:160: /tests/Makefile: No such file or directory make: *** No rule to make target `/tests/Makefile'. Stop. ./build_android.sh: line 59: /home/karthikeyan/home/karthikeyan/MyFiles/datas/applications/inst-packs/android-ndk-r9c/toolchains/arm-linux-androideabi-4.8/prebuilt/linux-x86_64/bin/arm-linux-androideabi-ar: No such file or directory ./build_android.sh: line 60: /home/karthikeyan/home/karthikeyan/MyFiles/datas/applications/inst-packs/android-ndk-r9c/toolchains/arm-linux-androideabi-4.8/prebuilt/linux-x86_64/bin/arm-linux-androideabi-ld: No such file or directory

2 1

audio is out of sync and "ALSA buffer xrun"
by YuGiOhJCJ Mailing-List 24 Feb '14

24 Feb '14

Hello, I am using ffmpeg version 1.2. To record (audio/video) my desktop with ffmpeg, I use this command: $ ffmpeg -f x11grab -r 15 -s 640x480 -i :0.0 -f alsa -i default out.mkv Unfortunately, when I am playing this "out.mkv" file on VLC, I can remark that the audio is out of sync. In the console, I can see some warnings: "ALSA buffer xrun". Do you see why ? Thank you. Best regards.

3 12

Slow conversion when using very low frame rates
by Tobias Rapp 24 Feb '14

24 Feb '14

Hi, I'm making some kind of "video preview" files from 25Hz SD/HD AVI files by scaling the frames and lowering the frame rate. During the process I noticed that when I use a command like: $ ffmpeg -i ffmpeg-huffyuv.avi -codec:v mjpeg -r 1:5 -an -y ffmpeg-test.avi the processing needs still around 80% of the original processing time even though only about 1% of the frames are passed through. It seems like FFmpeg is still reading and decompressing each image before dropping 99% of the frames. Is there another option to reduce frame-rate in a more optimized manner? Regards, Tobias

3 5

Changing Aspect Ratios
by Dirk van der Gouw 23 Feb '14

23 Feb '14

I am a newbie in the user-groups. Working with ffmpeg for 8 years already on my old Power Mac G5 (OS X 10.5.8) My ffmpeg version is 0.0.9 X. So not he commandline version but the (old) complete application. Mainly used it to convert video files to MPEG 2 (DVD). Works fine. But when I convert files to AVI (codec DIVX MPEG 4 and MP3 audio) and I burn these DIVX-files to a DVD for playing with my DVD-player and 16:9 flatscreen LCD the only video size that shows up on the sceen is 624 x 352. I tried a lot of sizes. But every other size there is only the audio and no picture. 624 x 352 works 640 x 360 not 720 x 405 not 960 x 540 not 1248 x 702 not 1280 x 720 not So my question: how can I calculate the videosize to ratios that work on my flatscreen? Thanks from, Oldy

2 1

CBR with libvpx using latest ffmpeg
by Manas Bhatnagar 23 Feb '14

23 Feb '14

Hello, I have been trying to transcode constant bitrate vp8 video files with libvorbis audio from a source in x264 video with ac3 audio. I have had mixed success after experimenting with the command line options but success is not reproducible on a different file. I am running the latest ffmpeg compiled with gcc49 on FreeBSD 9: *********************************************************************************************** > ffmpeg -version http://pastebin.com/Vr1Wzrib ************************************************************ All components were also compiled with gcc49. I use the video file located at https://mega.co.nz/#!0tcXGSSL!-4lY7eaEQHsiOCTIWIMhur8w87qlA3gUi92g3cxdfkM for the next few examples When I encode with vpxenc, I specify that I want a CBR file and set buffer sizes to be minimal to get as close to CBR as possible. I used ffmpeg to convert to y4m format then used vpxenc. I am able to get: ************************************************************************************************ > mediainfo video.mkv (video_mediainfo.txt) > vpxenc --end-usage=cbr --target-bitrate=450 --buf-initial-sz=1 --buf-optimal-sz=1 --buf-sz=1 --threads=4 --output=video_vpxenc.webm video.y4m > ffmpeg -i video_vpxenc.webm ffmpeg version git-2014-02-19-ca9f7e1 Copyright (c) 2000-2014 the FFmpeg developers built on Feb 20 2014 12:41:12 with gcc 4.9.0 (FeeBSD Pots Collection) 20140209 (expeimental) ... bitrate: 449 kb/s ... ************************************************************************************************ Notice "bitrate: 449 kb/s" (this is bitrate for the video only) Arguments were referenced from http://www.webmproject.org/docs/encoder-parameters/ So using the reference vpxenc encoder, a CBR video file is produced. Then I tried to transcode a 450 kbps CBR file through libvpx with ffmpeg. The setting which seems to work for this particular file is: ************************************************************ ************************************ > ffmpeg -i video.mkv -c:v libvpx -minrate:v 450k -maxrate:v 450k -b:v 450k -c:a libvorbis -b:a 48k -ac 2 video_ffmpeg10.webm > output.txt 2>&1 (output.txt) ************************************************************ ************************************ Notice: "bitrate 493.1 kbits/s" (for both video and audio) All options were referenced from http://www.ffmpeg.org/ffmpeg-all.html#libvpx & http://trac.ffmpeg.org/wiki/vpxEncodingGuide Notice that this is not as documented, I had to use -minrate:v 450k and -maxrate:v 450k. Using -minrate 450k & -maxrate 450k as suggested on the wiki page did not produce a CBR file: ************************************************************ ************************************ > ffmpeg -i video.mkv -c:v libvpx -minrate 450k -maxrate 450k -b:v 450k -c:a libvorbis -b:a 48k -ac 2 video_ffmpeg11.webm > output2.txt 2>&1 (output2.txt) ************************************************************ ************************************ Notice "bitrate= 898.7kbits/s" It seems to me that using -minrate 450k & -maxrate 450k affects the audio as this choice of arguments leads to a 3298 kB audio instead of 322 kB with -minrate:v 450k &-maxrate:v 450k. What I find confusing is that when I used the arguments which worked on video.mkv above on a different file of same codecs (x264, ac3), ffmpeg does not produce CBR output. The next file I use is 1.5 GB in size. It will not be easy to share this with you due to its size. I have included its mediainfo output and ffmpeg output as attachments. I also added -bufsize 128k as suggested in an example on the ffmpeg man page : ************************************************************ ************************************ > mediainfo video2.mkv (video2_mediainfo.txt) > ffmpeg -i video2.mkv -c:v libvpx -minrate:v 450k -maxrate:v 450k -b:v 450k -bufsize 128k -c:a libvorbis -b:a 48k -ac 2 -threads 4 video2_ffmpeg.webm > output3.txt 2>&1 (output3.txt)-> https://www.sugarsync.com/pf/D1124022_97834833_28599 ************************************************************ ************************************ Notice "bitrate= 975.0kbits/s" I used 4 threads here, the number of threads used does not seem to affect final bitrate. I also tried to encode video.mkv into a CBR vp8 video using -minrate 450k, -maxrate 450k, -b:v 450k & -bufsize 128k which did not work, a file with bitrate 800 kbits/s was produced. When I ran the ffmpeg command which was producing CBR output with the -v debug argument I saw these lines, among the other output: ************************************************************ ************************************ ... Reading option '-b:a' ... matched as option 'b' (video bitrate (please use -b:v)) with argument '48k'. ... Applying option b:a (video bitrate (please use -b:v)) with argument 48k. ... ************************************************************ ************************************ It seems like the option which I specified for audio bitrate is either not being interpreted correctly or it is programmed to print something incorrect while interpreting the option correctly. Thanks, Manas

1 0

AVI format for huge V210 content
by Roberto Cano 23 Feb '14

23 Feb '14

Hi all, I'm using a custom compiled version of ffmpeg 2.0.2 to filter an original v210 movie with pcm_s24le audio track with 10 channels. The transcoding is done to v210 at the output and I'm using a custom video filter to overlay some letters. I use the following command: ffmpeg -i orig.avi -map0 -c copy -c:v v210 -vf custom=text=sample output *.avi* The original file is a 1:36:00 movie in v210 at 1920x1080, so really huge. The output file has around 700GB (similar to the input). The thing is that the output file lacks the audio track. What I see during the transcoding is the following: ffmpeg version 2.0.2 Copyright (c) 2000-2013 the FFmpeg developers built on Jan 21 2014 11:47:48 with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5) configuration: --enable-libx264 --enable-gpl --extra-cflags=-Ix264/ --extra-ldflags=-Lx264 libavutil 52. 38.100 / 52. 38.100 libavcodec 55. 18.102 / 55. 18.102 libavformat 55. 12.100 / 55. 12.100 libavdevice 55. 3.100 / 55. 3.100 libavfilter 3. 79.101 / 3. 79.101 libswscale 2. 3.100 / 2. 3.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 3.100 / 52. 3.100 [avi @ 0x24318a0] Stream #0: not enough frames to estimate rate; consider increasing probesize [avi @ 0x24318a0] Stream #1: not enough frames to estimate rate; consider increasing probesize Input #0, avi, from orig.avi': Duration: 01:36:15.18, start: 0.000000, bitrate: 1072272 kb/s Stream #0:0: Video: v210 (v210 / 0x30313276), yuv422p10le, 1920x1080, 23.98 tbr, 23.98 tbn, 23.98 tbc Stream #0:1: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, 10 channels (FL+FR+FC+LFE+BL+BR+FLC+FRC+BC+SL), s32, 11520 kb/s Output #0, avi, to 'output.avi': Metadata: ISFT : Lavf55.12.100 Stream #0:0: Video: v210 (v210 / 0x30313276), yuv422p10le, 1920x1080, q=2-31, 200 kb/s, 23.98 tbn, 23.98 tbc Stream #0:1: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, 10 channels (FL+FR+FC+LFE+BL+BR+FLC+FRC+BC+SL), 11520 kb/s Stream mapping: Stream #0:0 -> #0:0 (v210 -> v210) Stream #0:1 -> #0:1 (copy) Press [q] to stop, [?] for help frame= 5 fps=0.0 q=0.0 size= 10kB time=00:00:00.20 bitrate= 380.6kbits/s Invalid riff index 257 > 256 Invalid riff index 258 > 256 Invalid riff index 259 > 256 Invalid riff index 260 > 256 ...... Invalid riff index 718 > 256 frame=138466 fps=6.5 q=0.0 Lsize=755842213kB time=01:36:15.18 bitrate=1072148.9kbits/s video:747716400kB audio:8121355kB subtitle:0 global headers:0kB muxing overhead 0.000590% When checking the output file with ffmpeg again the output is: ffmpeg version 2.0.2 Copyright (c) 2000-2013 the FFmpeg developers built on Jan 21 2014 11:47:48 with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5) configuration: --enable-libx264 --enable-gpl --extra-cflags=-Ix264/ --extra-ldflags=-Lx264 libavutil 52. 38.100 / 52. 38.100 libavcodec 55. 18.102 / 55. 18.102 libavformat 55. 12.100 / 55. 12.100 libavdevice 55. 3.100 / 55. 3.100 libavfilter 3. 79.101 / 3. 79.101 libswscale 2. 3.100 / 2. 3.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 3.100 / 52. 3.100 Input #0, avi, from 'output.avi': Metadata: encoder : Lavf55.12.100 Duration: 01:36:15.18, start: 0.000000, bitrate: 1072149 kb/s Stream #0:0: Video: v210 (v210 / 0x30313276), yuv422p10le, 1920x1080, 23.98 tbr, 23.98 tbn, 23.98 tbc At least one output file must be specified So no audio in it. I see two things here: - AVI container complains about RIFF index. This is because every GB of file the container need to add a RIFF index to the file, but this file is huge and the maximum number of indices is 256, so after 256GB of file it starts complaining. Could this affect the audio track? - Generated file is 700GB. Is this too big for ffmpeg to add the audio track? Is the audio track added at the end, or interleaved? If I generate small snippets, around 3.7GB each then the audio track is there for the AVI files. Thank you very much for the help. Kind regards, Roberto Sosa Cano

2 2

Fwd: Can't concat two files which were encoded the same way.
by Zsolt 23 Feb '14

23 Feb '14

From: Zsolt <dqeswn(a)gmail.com> Date: 2014-02-22 21:12 GMT+01:00 Subject: Can't concat two files which were encoded the same way. To: ffmpeg-user(a)ffmpeg.org Hello! I always get errors like this and a borked file the lenght of the first one: > [matroska,webm @ 029b2a20] Unknown entry 0x1A45DFA301:05.66 > bitrate=34899.2kbits/s > [matroska,webm @ 029b2a20] Unknown entry 0x18538067 > I doesn't matter if I remux them (same format) first or if I only try to concat video/audio only. If I remux them to .avi first then I get: > [matroska @ 04517000] Can't write packet with unknown timestamp > av_interleaved_write_frame(): Invalid argument > [matroska @ 04517000] Can't write packet with unknown timestamp > If I try to concat to mkv again. or something like: > [matroska @ 04517000] Can't write packet with unknown timestamp > av_interleaved_write_frame(): Invalid argument [matroska @ 04517000] Can't > write packet with unknown timestamp > several times with changing values. What is the problem here? (attached mediainfo data) (resend, forgot to subscribe first)

2 12

"-vf mp=softpulldown" drops all frames
by Austin de Coup-Crank 23 Feb '14

23 Feb '14

Hi all,So far my mpeg2 conversion looks great, but some of my source material is 24p and hence I need 3:2 pulldown for NTSC DVDs. I see the mplayer filter "softpulldown" is available, but when invoked, it simply drops all frames. As far as I can tell, softpulldown has not been natively ported into libavfilter yet. Here is my ffmpeg script: #!/bin/bash SOURCE=$1 DESTINATION=$2 VB=$3 VF="mp=softpulldown",\ scale=720:480,\ hqdn3d=2:1:2,\ pp=ac" # Pass 1 ffmpeg \ -i $SOURCE \ -framerate 24000/1001 \ -vf "$VF" \ -aspect 16:9 \ -dc 10 \ -g 12 \ -b_strategy 2 \ -brd_scale 2 \ -bf 2 \ -lmin 0 \ -lmax '320*QP2LAMBDA' \ -mblmin 1 \ -qmin 1 \ -qmax 31 \ -maxrate 9800k \ -b:v $VB \ -bf 2 \ -pre_dia_size 5 \ -dia_size 5 \ -qcomp 0.7 \ -qblur 0 \ -preme 1 \ -me_method dia \ -sc_threshold 0 \ -sc_factor 4 \ -bidir_refine 4 \ -profile:v 4 \ -mbd rd \ -mbcmp satd \ -precmp satd \ -cmp satd \ -subcmp satd \ -skipcmp satd \ -pass 1 \ -passlogfile ffmpeg \ -threads 4 \ -an \ -f mpeg2video \ -y \ /dev/null # Pass 2 ffmpeg \ -i $SOURCE \ -vf "$VF" \ -framerate 24000/1001 \ -aspect 16:9 \ -dc 10 \ -g 12 \ -brd_scale 2 \ -bf 2 \ -lmin 0 \ -lmax '320*QP2LAMBDA' \ -mblmin 1 \ -qmin 1 \ -qmax 31 \ -maxrate 9800k \ -b:v $VB \ -bf 2 \ -pre_dia_size 5 \ -dia_size 5 \ -qcomp 0.7 \ -qblur 0 \ -preme 1 \ -me_method dia \ -sc_threshold 0 \ -sc_factor 4 \ -bidir_refine 4 \ -profile:v 4 \ -mbd rd \ -mbcmp satd \ -precmp satd \ -cmp satd \ -subcmp satd \ -skipcmp satd \ -pass 2 \ -passlogfile ffmpeg \ -threads 4 \ -an \ -y \ -f mpeg2video \ $DESTINATION Here is the output: ffmpeg version 2.1.3 Copyright (c) 2000-2013 the FFmpeg developers built on Feb 8 2014 08:54:30 with Apple LLVM version 5.0 (clang-500.2.79) (based on LLVM 3.3svn) configuration: --prefix=/usr/local/Cellar/ffmpeg/2.1.3 --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-nonfree --enable-hardcoded-tables --enable-avresample --enable-vda --cc=clang --host-cflags= --host-ldflags= --enable-libx264 --enable-libfaac --enable-libmp3lame --enable-libxvid libavutil 52. 48.101 / 52. 48.101 libavcodec 55. 39.101 / 55. 39.101 libavformat 55. 19.104 / 55. 19.104 libavdevice 55. 5.100 / 55. 5.100 libavfilter 3. 90.100 / 3. 90.100 libavresample 1. 1. 0 / 1. 1. 0 libswscale 2. 5.101 / 2. 5.101 libswresample 0. 17.104 / 0. 17.104 libpostproc 52. 3.100 / 52. 3.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'in/rm-sample.m4v': Metadata: major_brand : M4V minor_version : 512 compatible_brands: isomiso2avc1 encoder : Lavf55.19.104 Duration: 00:00:30.07, start: 0.000000, bitrate: 1536 kb/s Stream #0:0(eng): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, bt709), 1280x720 [SAR 2000:2000 DAR 16:9], 1406 kb/s, 23.98 fps, 23.98 tbr, 24k tbn, 48k tbc (default) Metadata: handler_name : VideoHandler Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 123 kb/s (default) Metadata: handler_name : SoundHandler [Parsed_mp_0 @ 0x7fb7f0c0bde0] 'softpulldown' is a wrapped MPlayer filter (libmpcodecs). This filter may be removed once it has been ported to a native libavfilter. [mpeg2video @ 0x7fb7f1023600] Automatically choosing VBV buffer size of 224 kbyte [mpeg2video @ 0x7fb7f1023600] [lavc rc] Using all of requested bitrate is not necessary for this video with these parameters. Output #0, mpeg2video, to 'out/rm-sample.m2v': Metadata: major_brand : M4V minor_version : 512 compatible_brands: isomiso2avc1 encoder : Lavf55.19.104 Stream #0:0(eng): Video: mpeg2video, yuv420p, 720x480 [SAR 32:27 DAR 16:9], q=1-31, pass 2, 8000 kb/s, 90k tbn, 23.98 tbc (default) Metadata: handler_name : VideoHandler Stream mapping: Stream #0:0 -> #0:0 (h264 -> mpeg2video) Press [q] to stop, [?] for help frame= 0 fps=0.0 q=0.0 Lsize= 0kB time=00:00:00.00 bitrate=N/A dup=0 drop=721 video:0kB audio:0kB subtitle:0 global headers:0kB muxing overhead nan% Output file is empty, nothing was encoded (check -ss / -t / -frames parameters if used) I have tried several variations of the above script but it ultimately drops all frames. I know mp filters are not really supported, I just want to know if I'm doing something wrong, or if there is another way to insert soft telecine flags into the mpeg2 stream so I can retain 24p material on a 29.97fps NTSC DVD. Regards, Austin

4 11

ffmpeg fails on mp2v codec
by Jan Ehrhardt 23 Feb '14

23 Feb '14

FFmpeg fails on transcoding a mov file with a mp2v video codec. Error messages: Unknown MOV field order 0x0200 [mov,mp4,m4a,3gp,3g2,mj2 @ 00000000003984e0] Could not find codec parameters for stream 0 (Video: none (mp2v / 0x7632706D), 720x480, 5388 kb/s): unknown codec Commandline: ffmpeg -i UEFC4AM.mov -acodec mp3 -vcodec h264 -y test.mp4 Complete output below The latest Subjunk MEncoder has no problems with this video. It reports the same errors but continues the transcoding. Commandline: mencoder UEFC4AM.mov -oac mp3lame -ovc x264 -o test.mp4 Complete output below Does anybody have an idea what is going wrong? Can it be fixed with a command line option? Jan FFMpeg output: C:\>ffmpeg -i UEFC4AM.mov -acodec mp3 -vcodec h264 -y test.mp4 ffmpeg version N-60813-g44b22bb Copyright (c) 2000-2014 the FFmpeg developers built on Feb 22 2014 03:16:53 with gcc 4.8.2 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab le-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetyp e --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --ena ble-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-l ibopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libsp eex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aa cenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavp ack --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable -zlib libavutil 52. 65.100 / 52. 65.100 libavcodec 55. 52.102 / 55. 52.102 libavformat 55. 33.100 / 55. 33.100 libavdevice 55. 10.100 / 55. 10.100 libavfilter 4. 1.103 / 4. 1.103 libswscale 2. 5.101 / 2. 5.101 libswresample 0. 17.104 / 0. 17.104 libpostproc 52. 3.100 / 52. 3.100 Unknown MOV field order 0x0200 [mov,mp4,m4a,3gp,3g2,mj2 @ 0000000002a284e0] Could not find codec parameters for stream 0 (Video: none (mp2v / 0x7632706D), 720x480, 5388 kb/s): unknown codec Consider increasing the value for the 'analyzeduration' and 'probesize' options Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'UEFC4AM.mov': Metadata: major_brand : qt minor_version : 0 compatible_brands: qt creation_time : 2014-02-12 03:27:28 Duration: 00:02:03.97, start: 0.000000, bitrate: 5839 kb/s Stream #0:0(und): Video: none (mp2v / 0x7632706D), 720x480, 5388 kb/s, SAR 4 0:33 DAR 20:11, 29.96 fps, 29.97 tbr, 90k tbn, 90k tbc (default) Metadata: creation_time : 2014-02-12 03:27:28 handler_name : Core Media Data Handler Stream #0:1(und): Audio: ac3 (ac-3 / 0x332D6361), 48000 Hz, 5.1(side), fltp, 448 kb/s (default) Metadata: creation_time : 2014-02-12 03:27:28 handler_name : Core Media Data Handler [buffer @ 0000000004fe5c60] Unable to parse option value "-1" as pixel format Last message repeated 1 times [buffer @ 0000000004fe5c60] Error setting option pix_fmt to value -1. [graph 0 input from stream 0:0 @ 0000000005035540] Error applying options to the filter. Error opening filters! MEncoder output: C:\>mencoder UEFC4AM.mov -oac mp3lame -ovc x264 -o test.mp4 MEncoder SB55 (C) 2000-2014 MPlayer Team Custom build by SubJunk, http://www.spirton.com WARNING: OUTPUT FILE FORMAT IS _AVI_. See -of help. success: format: 0 data: 0x0 - 0x564dc33 libavformat version 55.30.100 (internal) libavformat file format detected. Unknown MOV field order 0x0200 [mov,mp4,m4a,3gp,3g2,mj2 @ 0195f9c0]Could not find codec parameters for stream 0 (Video: none (mp2v / 0x7632706D), 720x480, 5388 kb/s): unknown codec Consider increasing the value for the 'analyzeduration' and 'probesize' options [lavf] stream 0: video (unknown), -vid 0 [lavf] stream 1: audio (ac3), -aid 0, -alang und VIDEO: [mp2v] 720x480 24bpp 29.970 fps 5388.6 kbps (657.8 kbyte/s) [V] filefmt:44 fourcc:0x7632706D size:720x480 fps:29.970 ftime:=0.0334 ========================================================================== Opening audio decoder: [ffmpeg] FFmpeg/libavcodec audio decoders libavcodec version 55.49.101 (internal) [ac3 @ 0198eb80]Channel layout '5.1(side)' with 6 channels does not match specif ied number of channels 2: ignoring specified channel layout AUDIO: 48000 Hz, 2 ch, floatle, 448.0 kbit/14.58% (ratio: 56000->384000) Selected audio codec: [ffac3] afm: ffmpeg (FFmpeg AC-3) ========================================================================== Opening video filter: [expand osd=1] Expand: -1 x -1, -1 ; -1, osd: 1, aspect: 0.000000, round: 1 ========================================================================== Opening video decoder: [ffmpeg] FFmpeg's libavcodec codec family Selected video codec: [ffmpeg2] vfm: ffmpeg (FFmpeg MPEG-2) ========================================================================== MP3 audio selected. Movie-Aspect is 1.82:1 - prescaling to correct movie aspect. x264 [info]: using SAR=109/90 x264 [info]: using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX x264 [info]: profile High, level 3.0 Pos: 0.0s 2f ( 2%) 0.00fps Trem: 0min 0mb A-V:0.003 [0:0] [VD_FFMPEG] DRI failure. Pos: 1.6s 50f ( 2%) 0.00fps Trem: 0min 0mb A-V:0.057 [0:231] Writing header... ODML: vprp aspect is 16384:9018. Setting audio delay to 0.072s. Writing header... ODML: vprp aspect is 16384:9018. Setting audio delay to 0.072s. Pos: 4.0s 120f ( 2%) 69.16fps Trem: 1min 24mb A-V:0.067 [836:235] Skipping frame! Pos: 123.9s 3715f (99%) 51.13fps Trem: 0min 29mb A-V:0.018 [1760:234] Skipping frame! Pos: 123.9s 3716f (99%) 51.14fps Trem: 0min 29mb A-V:0.018 [1760:234] Flushing video frames. Writing index... Writing header... ODML: vprp aspect is 16384:9018. Setting audio delay to 0.072s. Video stream: 1767.986 kbit/s (220998 B/s) size: 27379576 bytes 123.890 secs 3716 frames Audio stream: 234.285 kbit/s (29285 B/s) size: 3630240 bytes 123.960 secs x264 [info]: frame I:15 Avg QP:22.37 size: 47839 x264 [info]: frame P:1917 Avg QP:25.26 size: 11506 x264 [info]: frame B:1781 Avg QP:29.35 size: 2586 x264 [info]: consecutive B-frames: 9.5% 76.1% 10.3% 4.1% x264 [info]: mb I I16..4: 6.1% 81.6% 12.4% x264 [info]: mb P I16..4: 0.6% 3.6% 0.5% P16..4: 42.8% 20.3% 13.8% 0.0% 0 .0% skip:18.5% x264 [info]: mb B I16..4: 0.0% 0.1% 0.0% B16..8: 31.5% 5.4% 1.7% direct: 2.1% skip:59.3% L0:29.7% L1:53.0% BI:17.2% x264 [info]: 8x8 transform intra:77.7% inter:72.4% x264 [info]: coded y,uvDC,uvAC intra: 76.9% 74.7% 32.2% inter: 24.7% 20.5% 1.1% x264 [info]: i16 v,h,dc,p: 34% 14% 14% 37% x264 [info]: i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 31% 9% 21% 5% 6% 8% 5% 9% 6% x264 [info]: i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 31% 15% 17% 5% 6% 8% 5% 8% 5% x264 [info]: i8c dc,h,v,p: 51% 11% 31% 7% x264 [info]: Weighted P-Frames: Y:13.8% UV:3.9% x264 [info]: ref P L0: 64.0% 22.4% 9.6% 3.7% 0.4% x264 [info]: ref B L0: 92.0% 7.7% 0.3% x264 [info]: ref B L1: 98.6% 1.4% x264 [info]: kb/s:1767.99

2 9

errors building
by Raymond Blum 23 Feb '14

23 Feb '14

Hi all Any help with the below is of course greatly appreciated. I am following the installation guide here https://trac.ffmpeg.org/wiki/UbuntuCompilationGuide and have built the dependencies for ffmpeg and configured it with: ./configure --prefix="$HOME/Downloads/libffmpeg" --extra-cflags="-I$HOME/Downloads/libffmpeg/include" \ --extra-ldflags="-L$HOME/Downloads/libffmpeg/lib" --bindir="$HOME/bin" --extra-libs=-ldl --enable-gpl \ --enable-libass --enable-libfdk-aac --enable-libmp3lame --enable-libopus --enable-libtheora \ --enable-libvorbis --enable-libvpx --enable-libx264 --enable-nonfree --enable-x11grab But the make fails with: YASM libavfilter/x86/af_volume.o libavutil/x86/x86inc.asm:103: error: unknown 'cpu' type libavutil/x86/x86inc.asm:885: error: unknown 'cpu' type libavfilter/x86/af_volume.asm:39: error: unknown 'cpu' type libavfilter/x86/af_volume.asm:40: error: unknown 'cpu' type libavfilter/x86/af_volume.asm:99: error: unknown 'cpu' type libavfilter/x86/af_volume.asm:101: error: unknown 'cpu' type libavfilter/x86/af_volume.asm:104: error: unknown 'cpu' type libavfilter/x86/af_volume.asm:105: error: unknown 'cpu' type libavfilter/x86/af_volume.asm:112: error: unknown 'cpu' type libavfilter/x86/af_volume.asm:113: error: unknown 'cpu' type make: *** [libavfilter/x86/af_volume.o] Error 1 raymond@ocubut:~/Downloads/libffmpeg/ffmpeg$ uname -a Linux ocubut 3.2.0-32-generic #51-Ubuntu SMP Wed Sep 26 21:33:09 UTC 2012 x86_64 x86_64 x86_64 GNU/Linux

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Newbie: Need to stream output from audio capture device to VLC and it needs to be lossless
by Tom Blahovici 23 Feb '14

23 Feb '14

Hi, I currently have an application that takes the stereo output of an audio capture device (line in) and then in real time performs fast fourier transforms and converts the signal into a real time spectrum scope. What I would now like to do is take the output of the capture and stream it to VLC whose output will go directly into the line in of another PC effectively streaming the “data” to my application except remotely. Why I am using VLC is that the output of VLC can go directly into an audio device driver I use called a virtual sound cable. A virtual sound cable has a “Line” in and the output goes to my application. I have already tried this with using an mp3 stream…but the performance is terrible and I must have bit perfect transfer of the audio output since calculations will be done. So basically, and here is where the newbie in me shows, I need to take the output of a directshow audio device, and then send it with a lossless transfer to the VLC client. I don’t know what the output of the direct show filter is . Is it PCM? Wav? How do I encode it and encapsulate it to get to the vlc client? Any help would be really appreciated. I remember in the past spending days and days trying to use VLC for streaming and getting nowhere. I’m hoping that the ffmpeg route will be much more pleasant! Thanks, Tom

1 0

MJPEG: Deprecated pixel format warning
by Tobias Rapp 21 Feb '14

21 Feb '14

I am converting HuffYuv AVI files into MJPEG using the following command: $ ./ffmpeg -i ffmpeg-huffyuv.avi -codec:v mjpeg -an -y ffmpeg-mjpeg.avi this shows the following warning message (see attachment for the full log): [swscaler @ 00b78000] deprecated pixel format used, make sure you did set range correctly I tested to add "-color_range" with option value "0", "1" and "2" but this does not make any difference. If I try to add "-pix_fmt yuv422p" an additional warning is shown: Incompatible pixel format 'yuv422p' for codec 'mjpeg', auto-selecting format 'yuvj422p' Is there anything I can do to fix the issue that causes the warning? Regards, Tobias

2 2

ffprobe aac profile
by Young Kim 21 Feb '14

21 Feb '14

Hello, I'm currently trying to use ffprobe as a means for finding the metadata of videos. However, it doesn't seem as if it can display HE-AAC or LE-AAC for aac audio files. Is there a fix for this or some workaround? Thanks, Young Kim

3 2

Help with excessive CPU usage
by Loadlinx 20 Feb '14

20 Feb '14

I think I know what causes it but would like someone to confirm: I have two identical 16 core servers, identical ffmpeg lines, identical software installs. Server 1 does a very good job transcoding 20 inputs from DVB @ 75% per process Server 2 is on the same LAN and transcodes inputs from server 1 but CPU usage is through the roof. If I start with 1 process is starts at 75%, if I add 4 more all processes go up to 180%. The only difference between the two is input: DVB to http on localhost vs http remote host. Can someone shed some light if its possible to get server 2 to work as efficiently as server 1?

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Cimpilation trouble
by Alexandr Drachuk 20 Feb '14

20 Feb '14

Hi, Can you help me build shared library of FFMPEG for Android? I have exception when run ./configure_make_everything.sh: "/home/fedora/Develop/Projects/Eclipse/android-ffmpeg-x264/Project/jni/ffmpeg /home/fedora/Develop/Projects/Eclipse/android-ffmpeg-x264/Project/jni /home/fedora/Develop/Projects/Eclipse/android-ffmpeg-x264/Project/jni ERROR: libx264 not found"... I attached two log files. Thank you very mach! Best regards!

2 1

Transport Stream (.ts) with ac3 audio
by relaht＠gmx.de 20 Feb '14

20 Feb '14

I have problems with .ts files including ac3 audio. I described the whole issue including sample files in this thread: http://ffmpeg.gusari.org/viewtopic.php?f=11&t=1264 I really hope someone can help me on this. Thanks, Michael

7 27

How to solve flickering image / corrupted image
by whoami Jils 20 Feb '14

20 Feb '14

i am writing simple video player in android , I looked at SDL Side, and there is no issue in rendering. But from ffmpeg side, i have a question. What i am trying to achieve is : 1) Get the complete frame from ffmpeg, 2) Update into texture & render. 3) I have hard coded the Height/Width value ,1280, 720, as i am getting the same from streaming server. [ i don't depend on it from context, as it does not provide initially unless i retrieve some packet ] Issue: Flickering image. I am not sure if there are any buffer issues i need to consider from ffmpeg side? Kindly advice. Here is my code: av_register_all(); if (SDL_Init(SDL_INIT_VIDEO | SDL_INIT_AUDIO | SDL_INIT_TIMER)) { LOGD ( "Could not initialize SDL - %s\n", SDL_GetError()); exit(1); } if (avformat_open_input(&pFormatCtx, "rtsp://<IP>:Port", NULL, NULL) != 0) return -1; // Couldn't open file // Retrieve stream information if (avformat_find_stream_info(pFormatCtx, NULL) < 0) return -1; // Couldn't find stream information // Find the first video stream videoStream = -1; for (i = 0; i < pFormatCtx->nb_streams; i++) if (pFormatCtx->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO) { videoStream = i; break; } if (videoStream == -1) return -1; // Get a pointer to the codec context for the video stream pCodecCtx = pFormatCtx->streams[videoStream]->codec; // Find the decoder for the video stream pCodec = avcodec_find_decoder(pCodecCtx->codec_id); if (pCodec == NULL) { LOGD ("Unsupported codec!\n"); return -1; // Codec not found } // Open codec if (avcodec_open2(pCodecCtx, pCodec, NULL) < 0) return -1; // Could not open codec // Allocate video frame pFrame = avcodec_alloc_frame(); screen = SDL_CreateWindow ("Testing..",SDL_WINDOWPOS_UNDEFINED, SDL_WINDOWPOS_UNDEFINED, 1280, 720, \ SDL_WINDOW_SHOWN|SDL_WINDOW_ALLOW_HIGHDPI ); if (!screen) { fprintf(stderr, "SDL: could not set video mode - exiting\n"); exit(1); } renderer = SDL_CreateRenderer (screen, -1, SDL_RENDERER_ACCELERATED| SDL_RENDERER_TARGETTEXTURE); bmp = SDL_CreateTexture (renderer, SDL_PIXELFORMAT_YV12, SDL_TEXTUREACCESS_STREAMING,1280, 720); i = 0; while (av_read_frame(pFormatCtx, &packet) >= 0) { if (packet.stream_index == videoStream) { avcodec_decode_video2(pCodecCtx, pFrame, &frameFinished, &packet); if (frameFinished) { SDL_UpdateYUVTexture(bmp, NULL, pFrame->data[0], pFrame->linesize[0], \ pFrame->data[1], pFrame->linesize[1], \ pFrame->data[2], pFrame->linesize[2]); } } av_free_packet(&packet); SDL_RenderClear(renderer); retcopy = SDL_RenderCopy(renderer, bmp, NULL, NULL); SDL_RenderPresent(renderer); SDL_PollEvent(&event); switch (event.type) { case SDL_QUIT: SDL_Quit(); exit(0); break; case SDL_KEYDOWN: SDL_Quit (); exit (0); default: break; } } av_free(pFrame); avcodec_close(pCodecCtx); avformat_close_input(&pFormatCtx); return 0; }

1 0

Re: [FFmpeg-user] FFmpeg h264 decode error
by Manas Bhatnagar 20 Feb '14

20 Feb '14

Hello The input video file is 1.5 GB. I have found another video with similar codecs that also produces errors. The file is hosted at: https://mega.co.nz/#!0tcXGSSL!-4lY7eaEQHsiOCTIWIMhur8w87qlA3gUi92g3cxdfkM The complete uncut console output is: https://mega.co.nz/#!Ap9RCJpK!wk1thHPlat_H_gCK_jRTzurySzE3oUjJG8QsjwUyQQo My message got rejected for being over the 250 KB size limit so I have hosted the two files at mega.co.nz. I hope this is ok. Thanks, Manas

3 5

Streaming fbdev
by Tomasz Łatka 20 Feb '14

20 Feb '14

Hi, I'm new in ffmpeg. I want to stream linux framebuffer, but I have a problem when I want open fbdev: / int ret = 0;// // char* error = (char*)malloc(1024);// // // const char *filename = "fbdev";// // av_register_all();// // AVFormatContext *avfc = avformat_alloc_context();// // ret = avformat_open_input(&avfc, filename, NULL, NULL);// // if (ret < 0)// // {// // av_strerror(ret, error, 1024);// // printf("\tERROR %s\n", error);// // abort();// // }/ I get a error "Invalid data found when processing input" Any sugestions? Thanks for the help. Tomek

1 0

about ffmpeg integrating x265
by 吴昌猛 20 Feb '14

20 Feb '14

FYI. Any reply is appreciated. Thanks very much! ________________________________________ 发件人: 吴昌猛发送时间: 2014年2月20日 10:50 收件人: ffmpeg-user-owner(a)ffmpeg.org 主题: 答复: Request to mailing list ffmpeg-user rejected Thanks for your reply. I have re-edit the config.log as your instruction. The content I have sent in the last email is : " I have a doubt about ffmpeg integrating x265 lib. I have downloaded the latest ffmpeg source from http://git.videolan.org/?p=ffmpeg.git and x265 source code from https://bitbucket.org/multicoreware/x265/wiki/Home and my shell command of ffmpeg below: ./configure --enable-static --enable-gpl --enable-nonfree --enable-postproc --enable-runtime-cpudetect --enable-libfaac --enable-libmp3lame --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-pthreads --enable-static --enable-libopencore-amrnb --enable-libtheora --enable-yasm --enable-version3 --enable-avfilter --enable-libopencore-amrwb --enable-libfdk_aac --extra-libs=-static --disable-ffprobe --disable-ffserver --enable-libass --enable-libfreetype but error happens: ERROR: x265 not found If you think configure made a mistake, make sure you are using the latest version from Git. If the latest version fails, report the problem to the ffmpeg-user(a)ffmpeg.org mailing list or IRC #ffmpeg on irc.freenode.net. Include the log file "config.log" produced by configure as this will help solving the problem. The config.log is in attachment. Thanks for your checking and replyment " ________________________________________ 发件人: ffmpeg-user-bounces(a)ffmpeg.org [ffmpeg-user-bounces(a)ffmpeg.org] 代表 ffmpeg-user-owner(a)ffmpeg.org [ffmpeg-user-owner(a)ffmpeg.org] 发送时间: 2014年2月20日 4:03 收件人: 吴昌猛主题: Request to mailing list ffmpeg-user rejected Your request to the ffmpeg-user mailing list Posting of your message titled "about ffmpeg integrating x265" has been rejected by the list moderator. The moderator gave the following reason for rejecting your request: "Message is over the size limit of 250k. Please reduce the size of your message or provide links to large files instead of attaching them and re-send your message. For config.log you can include just the first ~50 lines and the last ~50 lines." Any questions or comments should be directed to the list administrator at: ffmpeg-user-owner(a)ffmpeg.org

1 0

Re: [FFmpeg-user] x.265
by Rens Dijkshoorn 20 Feb '14

20 Feb '14

I have a question and it will be a blast if someone will answer it. This gentleman has stated that x.265 encoder is integrated in ffmpeg since February 12th. http://stackoverflow.com/a/21853796 however my build of ffmpeg that I have compiled, notifies me (Unknown encoder 'libx265') So can someone kindly give my some information on this, If it is not too much trouble. Thank You for your time. You first need to install x265 from x265.org as a shared library. Rens

2 1

desktop streaming
by nugroho 20 Feb '14

20 Feb '14

Hi, I have ffmpeg installed in debian linux. I also have a flash streaming server located at http://a.b.c.d/app/stream. Now, I would like to stream my desktop include the sound to the streaming server. So what kind of ffmpeg syntax should be? Is it like this one bellow? _ffmpeg -video_size 1024x768 -framerate 25 -f x11grab -i :0.0+100,200 -f flv rtmp://a.b.c.d/app/__stream_ thanks for help! --

3 3

Re: [FFmpeg-user] Levels and Primaries
by Rio Kierkels 19 Feb '14

19 Feb '14

> > >>> Rio Kierkels wrote: >>>> >>>> Thanks Andy, >>>>> >>>>> If this is true it's good news for ffmpeg, bad news for the other >>>>> apps and our current delivery pipeline. My next step would be to >>>>> create a very basic mxf parser to determine the Y' values but as >>>>> I've said my C skills are sub par to my python skills. Does anybody >>>>> have some other nice tool to determine levels to cross check this. >>>>> Or maybe someone who can verify that the ffmpeg mxf parser and >>>>> mpeg2video decoder doesn't do anything to the levels when converting >>>>> it to a waveform? >>>>> >>>>> >>>> My C skills aren't up to much either so maybe you should have more >>>> confidence in something you write in python. >>>> >>>> Attached is what I used to look (cue someone spots blindingly obvious >>>> bug) of course looking means decoding with >>>> ffmpeg to rawvideo either to a file or a named pipe. >>>> >>>> So if you compile the file - gcc -Wall yuvp-counter.c -o count-yuv >>>> >>>> My sanity test file is 100% bars (+ subsampling artifacts) from >>>> >>>> ftp://ftp.tek.com/tv/test/streams/Element/MPEG-Video/625/100b_400.m2v >>>> >>>> for 422 or for 420 >>>> >>>> ftp://ftp.tek.com/tv/test/streams/Element/MPEG-Video/625/100b_060.m2v >>>> >>>> Save & make a named pipe somewhere - >>>> >>>> mkfifo myfifo >>>> >>>> In one terminal/konsole/whatever run >>>> >>>> ffmpeg -i 100b_400.m2v -f rawvideo -y myfifo >>>> >>>> Look at the output to get width, height and format (must be planar) >>>> >>>> and from another term set wide so you have a couple of hundred chars run >>>> eg. >>>> >>>> ./count-yuv 704 576 422 myfifo >>>> >>>> To me the output from that looks sane so I assume my counter works >>>> sometimes at least :-) >>> >>> Nice example Andy, It's good to see ffmpeg stay true to the data on disk and not like a lot of apps I'm testing does some "smart" scaling of the values. I'm going to keep testing and playing around to check for values. Again any body chime in if they have any information that could be of relevance. Now I'm trying to match the metadata. It is causing all sorts of problems with some programs like Resolve Lite simply not seeing the file. Tested it on 2 different machines. It would be nice to be able to set the color metadata (color primaries, matrix, transfer) and the GOP structure data (M=3, N=12). I'll be trying to look at the ffmbc code, as it looks like they've solved this problem long a go. -- 42

7 18

Re ffserver memory consumption
by ota sverma 19 Feb '14

19 Feb '14

> >/ > > If a stop saveing or all clients are disconnected, /> >/ > > memory consumption is stoped but don`t free used />/ //> > > memory. /> >/ > /> >/ > This sounds like a leak that you can probably /> >/ > find with valgrind (use a very, very small /> >/ > resolution). />/ > /> >/ I tried very small resolution and situation is same... / > You mean you are still seeing the memleak? > That is expected, I just wanted to suggest a way that > allows using valgrind even if real-time usage is > required. Yes, i am still seeing memleak, memory usage is still growing up... Can you help me how setup Valgrind for proper output? Or is any other diagnostik tool? >/ > Valgrind didn`t help... / > What do you mean? Valgrind did not showed me any way how to solve where is problem with memleak. > Please fix your quoting! Sorry, it is my first mailing list i wrote to it. Is it better know?

2 1