March 2014 - ffmpeg-user - lists.ffmpeg.org

constant rtp stram
by the... 25 Mar '14

25 Mar '14

We see in the tcpdump that when we stream a wav file *ffmpeg -analyzeduration 0 -re -i /tmp/test.wav -vol 512 -c g722 -ar 16000 -f rtp rtp://[destIP]:[destPort]* That the stream is sending in burst in to the wire. First about 12 packages in 0.6ms then about 144ms delta and then the next 12 package in 0.6ms delta for about 144ms and so on... Is it possible to send the stream with a constant delta between the packages? Thanks Roman

2 7

STDOUT and STDERR choices....
by Sam Marrocco 25 Mar '14

25 Mar '14

I've been using ffmpeg and ffprobe on WindowsOS am wondering why the choices that have been made as to what goes to STDOUT vs. STDERR? For example: -A normally running ffmpeg task seems to send all it's output (even when there are no errors) to STDERR even with no errors. -A normally running ffprobe task sends some of it's output to STDOUT and some of it's output to STDERR even with no errors. I'm curious why they wouldn't send all their normal output to STDOUT, and only Error messages to STDERR. -- sam marrocco | chief technical officer ringside.cutters.picnic.moonlink 248 548 2500 w 248 910 3344 c ringsidecreative.com <http://ringsidecreative.com/>

5 10

How to determine the real value of gamma correction?
by konstantin 24 Mar '14

24 Mar '14

I would like to transcode a video file with FFmpeg, and apply some gamma correction because the video is too dark. In FFmpeg I have to use this option syntax to apply the color filter: -vf 'mp=eq2=1.7:1.0:0.0:1.0:1.0:1.0:1.0' Where the first real number (1.7) is the value of the gamma correction parameter. Other programs also use similar notation to define gamma correction. However for example in SMPlayer the gamma correction parameter is an integer number, for example 22. I would like to preview the video with SMplayer and to correct gamma, and than convert the applied gamma value to the format of FFMpeg. So my question is how can I determine the real value of gamma (usually in range 1.0-2.0.) correction with any similar player, eg. VLC? -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/How-to-determine-the-real-value-of… Sent from the FFmpeg-users mailing list archive at Nabble.com.

1 0

Crisp, exact color match for lossless screen-cast
by thangalin 24 Mar '14

24 Mar '14

Hi, I would like to record an audio-less screen-cast. The following image shows a side-by-side comparison of the video (left) versus the desktop (right): http://i.stack.imgur.com/lSXXY.png The screen-cast is recorded using: ffmpeg -r 30 -f x11grab -s $WIN_GEO -i :0.0+$WIN_XY -vcodec huffyuv -preset ultrafast -qp 0 output.mkv Where $WIN_GEO has the browser dimensions (1030x794) and $WIN_XY has the browser location (1020x221). The problem is that the video colours differ from the desktop colours, resulting in a slightly blurry playback (the whites are off-white... washed out, slightly tarnished). I have read that this could be due to conversion between RGB and YUV colours. How do you create a desktop video (screen-cast) that when played back (e.g., using mplayer, VLC, or ffplay) results in an *exact* match to the original desktop colours and crispness? Thank you! -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Crisp-exact-color-match-for-lossle… Sent from the FFmpeg-users mailing list archive at Nabble.com.

3 3

ffmpeg splitting files
by Marcus Engene 24 Mar '14

24 Mar '14

Hi, I use ffprobe to do scene detection: ffprobe -show_frames -of compact=p=0 -f lavfi "movie=file.mov,select=gt(scene\,.4)" > scenes.csv After some contemplating from ffprobe's part I get a file with rows looking like this (prettyprinted): media_type=video key_frame=1 pkt_pts=7386000 pkt_pts_time=308.058058 pkt_dts=7386000 pkt_dts_time=308.058058 pkt_duration=1000 pkt_duration_time=0.041708 pkt_pos=3124352768 pkt_size=N/A width=1920 height=1080 pix_fmt=rgb24 sample_aspect_ratio=1:1 pict_type=I coded_picture_number=0 display_picture_number=0 interlaced_frame=0 top_field_first=0 repeat_pict=0 tag:lavfi.scene_score=0.520000 Where can I see what these values mean? Ie, what values do i massage and put into ffmpeg with -ss and -t? Thanks, Marcus

1 0

Adaptive Bitrate Encoding with HLS wrapping
by Vishnu Prassad 24 Mar '14

24 Mar '14

Hello, I have some free on channel streams that are in form of RTMP and UDP. I want to encode it in multi bit rate and also wrap it to HLS(HTTP live streaming) and then i need to publish it to my wowza streaming engine. I could encode Video clips in multi bit rate and wrap it to HLS using ffmpeg. So it would be appreciated if any one one can help me to encode my channel streams for Adaptive Bitrate streaming Regards, Vishnu Prasad

1 0

using concat and duration?
by Tom Horsley 24 Mar '14

24 Mar '14

The docs are a little light on the use of the 'duration' directive in a concat filter. If I want to explicitly give duration (to see if drifting audio sync can be helped), what exactly does the duration directive mean? Is it the duration of the previous file? Is it the duration of the next file? Is it the duration of the concatenated files so far? Is is another meaning I can't think of? Anyone know?

2 1

Scenecut detection and consistent GOP size - adaptive streaming
by tarun singhal 24 Mar '14

24 Mar '14

Hi, Sample Command: -map 0:0 -f mp4 -vcodec libx264 -preset slow -profile:v main -vf scale="640:trunc(ow/a/2)*2" -vb 700k -minrate 650k -maxrate 750k -bufsize 10000k -an -g 48 -x264opts keyint=48:min-keyint=10:scenecut=40 -flags +cgop -sc_threshold 40 -pix_fmt yuv420p -threads 0 -y There is as such no error in encoding, But I wanted to understand following points- 1) the above command will ensure that range of GOP size is {10,48}, and if any scene change value (frame1 to fame2) is >40%, then a keyframe is introduced there? 2) So that means in a 3 hours of source video, there is no guarantee that GOP size will remain same 3) No consider, I am creating 7 MP4 files each with different bitrate and resolution. (These Mp4s will be encoded to smooth in later stage). Hence i am targeting adaptive streaming. But when I did that, I found that if GOP sizes were not consistent across each bitrates. What I mean by this is, for ex: if in Bitrate1 - GOP size is like 10, 20, 48 and so on, in other bitrates it wasnt in the same sequence. I hope my question makes sense. So is there a way to ensure that GOP size may vary across one single output. But it should be consistent across each bitrate provided that the source is same? Also for adaptive streaming is scenecut detection advisable? Regards, Tarun Singhal

1 0

-vf tinterlace=6 not behaving as expected
by Andy Furniss 23 Mar '14

23 Mar '14

I see a recent commit fixed some aspects of this and referred to some discussions outlining behavior. http://git.videolan.org/gitweb.cgi/ffmpeg.git/?p=ffmpeg.git;a=commit;h=0d5a… http://forum.xbmc.org/showthread.php?tid=81834 https://github.com/mpv-player/mpv/issues/624#issuecomment-37685195 also https://www.ffmpeg.org/ffmpeg-filters.html#tinterlace describes what the filter should do, the trouble is when I test it as below with eg the scrolling letters burosch1.mpg (available in ffmpeg or mplayer samples IIRC). I don't see what I expect when frame stepping the result in that both fields move each frame rather than one being the same. The file does seem to have x2 frames and each frame has half the motion per step of the master, so I can only assume that the fields are getting swapped which is not the described behavior. ffm -loglevel verbose -i burosch1.mpg -vf tinterlace=interlacex2 -an -sn -vcodec rawvideo out.nut ffmpeg version N-61589-g37c07d4 Copyright (c) 2000-2014 the FFmpeg developers built on Mar 18 2014 21:32:47 with gcc 4.8.2 (GCC) configuration: --prefix=/usr --enable-gpl --enable-libmp3lame --enable-libx264 --enable-x11grab libavutil 52. 67.100 / 52. 67.100 libavcodec 55. 52.102 / 55. 52.102 libavformat 55. 34.101 / 55. 34.101 libavdevice 55. 11.100 / 55. 11.100 libavfilter 4. 3.100 / 4. 3.100 libswscale 2. 5.101 / 2. 5.101 libswresample 0. 18.100 / 0. 18.100 libpostproc 52. 3.100 / 52. 3.100 Input #0, mpeg, from 'burosch1.mpg': Duration: 00:00:29.92, start: 0.500000, bitrate: 8051 kb/s Stream #0:0[0x1e0]: Video: mpeg2video (Main), yuv420p(tv), 720x576 [SAR 64:45 DAR 16:9], max. 8000 kb/s, 25 fps, 25 tbr, 90k tbn, 50 tbc File 'out.nut' already exists. Overwrite ? [y/N] y [graph 0 input from stream 0:0 @ 0x2ab4880] w:720 h:576 pixfmt:yuv420p tb:1/90000 fr:25/1 sar:64/45 sws_param:flags=2 [Parsed_tinterlace_0 @ 0x2ab42c0] mode:6 filter:off h:576 -> h:576 Output #0, nut, to 'out.nut': Metadata: encoder : Lavf55.34.101 Stream #0:0: Video: rawvideo (I420 / 0x30323449), yuv420p, 720x576 [SAR 64:45 DAR 16:9], q=2-31, 200 kb/s, 51200 tbn, 50 tbc Stream mapping: Stream #0:0 -> #0:0 (mpeg2video -> rawvideo) Press [q] to stop, [?] for help No more output streams to write to, finishing.=00:00:28.60 bitrate=248844.0kbits/s frame= 1498 fps=332 q=0.0 Lsize= 910086kB time=00:00:29.96 bitrate=248846.0kbits/s video:910035kB audio:0kB subtitle:0 data:0 global headers:0kB muxing overhead 0.005641%

2 5

ffmpeg+Xcode5编译install-ffmpeg.sh文件出现问题
by 姜彬彬 23 Mar '14

23 Mar '14

Last login: Fri Mar 21 16:21:14 on ttys001 localhost:~ mac$ /Users/mac/Desktop/ffmpegc-master-1/install-ffmpeg.sh ; exit; install gas-* perl script /Users/mac/Desktop/ffmpegc-master-1/install-ffmpeg.sh: line 11: ./install-gas.sh: No such file or directory Downloading ffmpeg-2.0.2.tar.bz2 % Total % Received % Xferd Average Speed Time Time Time Current Dload Upload Total Spent Left Speed 100 6046k 100 6046k 0 0 65733 0 0:01:34 0:01:34 --:--:-- 113k Building without ccache ./configure: line 2870: /Volumes/Xcode/Xcode.app/Contents/Developer/Toolchains/XcodeDefault.xctoolchain/usr/bin/clang: No such file or directory ./configure: line 2870: /Volumes/Xcode/Xcode.app/Contents/Developer/Toolchains/XcodeDefault.xctoolchain/usr/bin/clang: No such file or directory /Volumes/Xcode/Xcode.app/Contents/Developer/Toolchains/XcodeDefault.xctoolchain/usr/bin/clang 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. logout 我运行的是Xcode5 install-ffmpeg.sh文件里面的内容是： #!/bin/bash ########################################################################### # Choose your ffmpeg version and your currently-installed iOS SDK version: # VERSION="2.0.2" SDKVERSION="7.0" echo "install gas-* perl script" ./install-gas.sh # # ########################################################################### # # Don't change anything under this line! # ########################################################################### # No need to change this since xcode build will only compile in the # necessary bits from the libraries we create ARCHS="armv7 armv7s i386" DEVELOPER=`xcode-select -print-path` cd "`dirname \"$0\"`" REPOROOT=$(pwd) # Where we'll end up storing things in the end OUTPUTDIR="${REPOROOT}/dependencies" mkdir -p ${OUTPUTDIR}/include mkdir -p ${OUTPUTDIR}/lib mkdir -p ${OUTPUTDIR}/bin BUILDDIR="${REPOROOT}/build" mkdir -p $BUILDDIR # where we will keep our sources and build from. SRCDIR="${BUILDDIR}/src" mkdir -p $SRCDIR # where we will store intermediary builds INTERDIR="${BUILDDIR}/built" mkdir -p $INTERDIR ######################################## cd $SRCDIR # Exit the script if an error happens set -e if [ ! -e "${SRCDIR}/ffmpeg-${VERSION}.tar.bz2" ]; then echo "Downloading ffmpeg-${VERSION}.tar.bz2" curl -LO http://ffmpeg.org/releases/ffmpeg-${VERSION}.tar.bz2 else echo "Using ffmpeg-${VERSION}.tar.bz2" fi tar zxf ffmpeg-${VERSION}.tar.bz2 -C $SRCDIR cd "${SRCDIR}/ffmpeg-${VERSION}" set +e # don't bail out of bash script if ccache doesn't exist CCACHE=`which ccache` if [ $? == "0" ]; then echo "Building with ccache: $CCACHE" CCACHE="${CCACHE} " else echo "Building without ccache" CCACHE="" fi set -e # back to regular "bail out on error" mode for ARCH in ${ARCHS} do if [ "${ARCH}" == "i386" ]; then PLATFORM="iPhoneSimulator" EXTRA_CONFIG="--arch=i386 --disable-asm --enable-cross-compile --target-os=darwin --cpu=i386" EXTRA_CFLAGS="-arch i386" EXTRA_LDFLAGS="-I${DEVELOPER}/Platforms/${PLATFORM}.platform/Developer/SDKs/${PLATFORM}${SDKVERSION}.sdk/usr/lib -mfpu=neon" else PLATFORM="iPhoneOS" EXTRA_CONFIG="--arch=arm --target-os=darwin --enable-cross-compile --cpu=cortex-a9 --disable-armv5te" EXTRA_CFLAGS="-w -arch ${ARCH} -mfpu=neon" EXTRA_LDFLAGS="-mfpu=neon" fi mkdir -p "${INTERDIR}/${ARCH}" ./configure --prefix="${INTERDIR}/${ARCH}" --disable-ffmpeg --disable-ffplay --disable-ffprobe --disable-ffserver --disable-iconv --disable-bzlib --enable-avresample --sysroot="${DEVELOPER}/Platforms/${PLATFORM}.platform/Developer/SDKs/${PLATFORM}${SDKVERSION}.sdk" --cc="${DEVELOPER}/Toolchains/XcodeDefault.xctoolchain/usr/bin/clang" --as='/usr/local/bin/gas-preprocessor.pl' --extra-cflags="${EXTRA_CFLAGS} -miphoneos-version-min=${SDKVERSION} -I${OUTPUTDIR}/include" --extra-ldflags="-arch ${ARCH} ${EXTRA_LDFLAGS} -isysroot /Applications/Xcode.app/Contents/Developer/Platforms/${PLATFORM}.platform/Developer/SDKs/${PLATFORM}${SDKVERSION}.sdk -miphoneos-version-min=${SDKVERSION} -L${OUTPUTDIR}/lib" ${EXTRA_CONFIG} --enable-pic --extra-cxxflags="$CPPFLAGS -I${OUTPUTDIR}/include -isysroot ${DEVELOPER}/Platforms/${PLATFORM}.platform/Developer/SDKs/${PLATFORM}${SDKVERSION}.sdk" make && make install && make clean done mkdir -p "${INTERDIR}/universal/lib" cd "${INTERDIR}/armv7/lib" for file in *.a do cd ${INTERDIR} xcrun -sdk iphoneos lipo -output universal/lib/$file -create -arch armv7 armv7/lib/$file -arch armv7s armv7s/lib/$file -arch i386 i386/lib/$file echo "Universal $file created." done cp -r ${INTERDIR}/armv7/include ${INTERDIR}/universal/ echo "Done." 希望尽快回复，谢谢！！！

2 1

stepping through a video frame by frame
by meino.cramer＠gmx.de 23 Mar '14

23 Mar '14

Hi, I take videos of birds with a MOBIUS camera, which creates 1980x1080 h264 (High) files. The videos played with mplayer got stuttered periodically. WIth ffply they just run fine. With mplayer I can step frame by frame through the video. Is this possible with ffplay also? How? Best regards, mcc

2 1

Is there an FFmpeg video filter similar to Avisynth Ylevels plugin?
by konstantin 22 Mar '14

22 Mar '14

Hi! I would like to encode a video with ffmpeg, but the video is too dark. Before I used avisynth and virtualdub to encode such videos, and I simply applied the ylevels(gamma=1.2) video filter, which is based on the MaskTools avisynth plugin. Is there a similer purpose fileter for ffmpeg which can do this task? -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Is-there-an-FFmpeg-video-filter-si… Sent from the FFmpeg-users mailing list archive at Nabble.com.

2 1

MKV muxer and ass subtitles issue
by ValdikSS 22 Mar '14

22 Mar '14

Hello! I have uploaded 2 files to FTP: mkv_subtitles.mkv and mkv_subtitles_ffmpeg.mkv. The first file (mkv_subtitles.mkv) is splitted with mkvmerge, the second one (mkv_subtitles_ffmpeg.mkv) is remuxed with ffmpeg: ffmpeg -i mkv_subtitles.mkv -map 0 -map_metadata 0 -c copy mkv_subtitles_ffmpeg.mkv Subtitles in the first file are played correctly with all effects while subtitle effects in ffmpeg-remuxed file are not (in every player except mplayer). If you extract subtitles with mkvextract tracks mkv_subtitles.mkv 1:sub_orig.ass mkvextract tracks mkv_subtitles_ffmpeg.mkv 1:sub_ffmpeg.ass you can see the difference between them (with diff) That's strange since -c copy is set and log clearly says it's copied.

1 0

Is mixing of audio channels happening by default in ffmpeg?
by Sam Marrocco 21 Mar '14

21 Mar '14

I have an uncompressed Quicktime with 8 tracks of audio (in addition to it's one video and one Timecode track). I am transcoding it to Quicktime Prores with the following command line (and results): "ffmpeg-20130507-git-399f6ef-win64-static\bin\ffmpeg.exe" -i "C:\Users\root\Desktop\MultiChannel.mov" -vcodec prores_ks -pix_fmt yuv444p10le -profile:v 3 -acodec copy -t 1 -f mov "D:\\MultiChannel\MultiChannel.mov" ffmpeg version N-52837-g399f6ef Copyright (c) 2000-2013 the FFmpeg developers built on May 7 2013 01:14:14 with gcc 4.7.3 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetype --enable-libgsm --enable-libilbc --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 52. 29.100 / 52. 29.100 libavcodec 55. 7.100 / 55. 7.100 libavformat 55. 4.101 / 55. 4.101 libavdevice 55. 0.100 / 55. 0.100 libavfilter 3. 63.101 / 3. 63.101 libswscale 2. 2.100 / 2. 2.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 3.100 / 52. 3.100 [mov,mp4,m4a,3gp,3g2,mj2 @ 000000000239e760] Stream #1: not enough frames to estimate rate; consider increasing probesize [mov,mp4,m4a,3gp,3g2,mj2 @ 000000000239e760] Stream #2: not enough frames to estimate rate; consider increasing probesize [mov,mp4,m4a,3gp,3g2,mj2 @ 000000000239e760] Stream #3: not enough frames to estimate rate; consider increasing probesize [mov,mp4,m4a,3gp,3g2,mj2 @ 000000000239e760] Stream #4: not enough frames to estimate rate; consider increasing probesize [mov,mp4,m4a,3gp,3g2,mj2 @ 000000000239e760] Stream #5: not enough frames to estimate rate; consider increasing probesize [mov,mp4,m4a,3gp,3g2,mj2 @ 000000000239e760] Stream #6: not enough frames to estimate rate; consider increasing probesize [mov,mp4,m4a,3gp,3g2,mj2 @ 000000000239e760] Stream #7: not enough frames to estimate rate; consider increasing probesize [mov,mp4,m4a,3gp,3g2,mj2 @ 000000000239e760] Stream #8: not enough frames to estimate rate; consider increasing probesize [mov,mp4,m4a,3gp,3g2,mj2 @ 000000000239e760] Stream #9: not enough frames to estimate rate; consider increasing probesize Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'C:\Users\root\Desktop\MultiChannel.mov': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt creation_time : 2013-07-17 21:37:47 Duration: 00:00:45.05, start: 0.000000, bitrate: 1199347 kb/s Stream #0:0(eng): Video: rawvideo (raw / 0x20776172), rgb24, 1920x1080, 1193199 kb/s, SAR 1:1 DAR 16:9, 23.98 fps, 23.98 tbr, 239760 tbn, 239760 tbc Metadata: creation_time : 2013-07-17 21:37:47 handler_name : Apple Alias Data Handler timecode : 00:59:45:00 Stream #0:1(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, mono, s16, 768 kb/s Metadata: creation_time : 2013-07-17 21:40:27 handler_name : Apple Alias Data Handler Stream #0:2(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, mono, s16, 768 kb/s Metadata: creation_time : 2013-07-17 21:40:28 handler_name : Apple Alias Data Handler Stream #0:3(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, mono, s16, 768 kb/s Metadata: creation_time : 2013-07-17 21:40:28 handler_name : Apple Alias Data Handler Stream #0:4(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, mono, s16, 768 kb/s Metadata: creation_time : 2013-07-17 21:40:29 handler_name : Apple Alias Data Handler Stream #0:5(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, mono, s16, 768 kb/s Metadata: creation_time : 2013-07-17 21:40:30 handler_name : Apple Alias Data Handler Stream #0:6(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, mono, s16, 768 kb/s Metadata: creation_time : 2013-07-17 21:40:30 handler_name : Apple Alias Data Handler Stream #0:7(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, mono, s16, 768 kb/s Metadata: creation_time : 2013-07-17 21:40:31 handler_name : Apple Alias Data Handler Stream #0:8(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, mono, s16, 768 kb/s Metadata: creation_time : 2013-07-17 21:40:31 handler_name : Apple Alias Data Handler Stream #0:9(eng): Data: none (tmcd / 0x64636D74) Metadata: creation_time : 2013-07-17 21:40:32 handler_name : Apple Alias Data Handler timecode : 00:59:45:00 Output #0, mov, to 'D:\\MultiChannel\MultiChannel.mov': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt encoder : Lavf55.4.101 Stream #0:0(eng): Video: prores (prores_ks) (apch / 0x68637061), yuv444p10le, 1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 200 kb/s, 11988 tbn, 23.98 tbc Metadata: creation_time : 2013-07-17 21:37:47 handler_name : Apple Alias Data Handler timecode : 00:59:45:00 Stream #0:1(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, mono, 768 kb/s Metadata: creation_time : 2013-07-17 21:40:27 handler_name : Apple Alias Data Handler Stream mapping: Stream #0:0 -> #0:0 (rawvideo -> prores_ks) Stream #0:1 -> #0:1 (copy) Press [q] to stop, [?] for help frame= 24 fps=2.5 q=0.0 Lsize= 8082kB time=00:00:01.00 bitrate=66139.3kbits/s dup=276 drop=0 video:7986kB audio:94kB subtitle:0 global headers:0kB muxing overhead 0.024802% The resulting quicktime appears to contain a single audio track with the contents of original 8 tracks mixed together. I have not provided any audio filtering/mixing options. Wouldn't the default condition to be passing through of the audio tracks exactly as the source contains it? If not, is it possible to pass each track through as such or must it be explicitely told to do so? Ideally if provided with 8 tracks, the resulting file would have 8 tracks. -- sam marrocco | chief technical officer ringside.cutters.picnic.moonlink 248 548 2500 w 248 910 3344 c ringsidecreative.com <http://ringsidecreative.com/>

6 25

ffmpeg fails joining two MPEG2 files
by Gilles 21 Mar '14

21 Mar '14

Hello Based on the FAQ, I used the following command to join two 2GB MPEG2 files (total time: about 1h) shot with my camcorder: ======================== c:\>ffmpeg -i concat:"M2U00075.MPG|M2U00076.MPG" -c copy intermediate_all.mpg ffmpeg version N-61663-g19139d8 Copyright (c) 2000-2014 the FFmpeg developers built on Mar 20 2014 22:01:15 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. 67.100 / 52. 67.100 libavcodec 55. 52.102 / 55. 52.102 libavformat 55. 34.101 / 55. 34.101 libavdevice 55. 11.100 / 55. 11.100 libavfilter 4. 3.100 / 4. 3.100 libswscale 2. 5.102 / 2. 5.102 libswresample 0. 18.100 / 0. 18.100 libpostproc 52. 3.100 / 52. 3.100 Input #0, mpeg, from 'concat:M2U00075.MPG|M2U00076.MPG': Duration: 01:01:12.92, start: 0.226767, bitrate: 9238 kb/s Stream #0:0[0x80]: Audio: ac3, 48000 Hz, 5.1(side), fltp, 448 kb/s Stream #0:1[0x1e0]: Video: mpeg2video (Main), yuv420p(tv), 720x576 [SAR 64:4 5 DAR 16:9], max. 9100 kb/s, 25 fps, 25 tbr, 90k tbn, 50 tbc [mpeg @ 03f11260] ac3 in MPEG-1 system streams is not widely supported, consider using the vob or the dvd muxer to force a MPEG-2 program stream. Output #0, mpeg, to 'intermediate_all.mpg': Metadata: encoder : Lavf55.34.101 Stream #0:0: Video: mpeg2video, yuv420p, 720x576 [SAR 64:45 DAR 16:9], q=2-3 1, max. 9100 kb/s, 25 fps, 90k tbn, 25 tbc Stream #0:1: Audio: ac3, 48000 Hz, 5.1(side), 448 kb/s Stream mapping: Stream #0:1 -> #0:0 (copy) Stream #0:0 -> #0:1 (copy) Press [q] to stop, [?] for help frame= 860 fps=0.0 q=-1.0 size= 36558kB time=00:00:34.36 bitrate=8716.0kbits/ frame= 1756 fps=0.0 q=-1.0 size= 75450kB time=00:01:10.20 bitrate=8804.6kbits/ frame= 2601 fps=1734 q=-1.0 size= 113680kB time=00:01:44.00 bitrate=8954.5kbits frame= 3442 fps=1721 q=-1.0 size= 151322kB time=00:02:17.64 bitrate=9006.3kbits frame= 3711 fps=141 q=-1.0 size= 163316kB time=00:02:28.40 bitrate=9015.4kbits/ frame= 3711 fps= 87 q=-1.0 Lsize= 163486kB time=00:02:28.40 bitrate=9024.8kbits /s video:154741kB audio:8107kB subtitle:0 data:0 global headers:0kB muxing overhead 0.392182% c:\> ============================= As shown above, ffmpeg gives up after about 2mn30 into the file. Here's what MediaInfo says about the first file: ========================= General Complete name : C:\M2U00075.MPG Format : MPEG-PS File size : 1.98 GiB Duration : 30mn 34s Overall bit rate mode : Variable Overall bit rate : 9 265 Kbps Video ID : 224 (0xE0) Format : MPEG Video Format version : Version 2 Format profile : Main@Main Format settings, BVOP : Yes Format settings, Matrix : Custom Format settings, GOP : M=3, N=12 Duration : 30mn 34s Bit rate mode : Variable Bit rate : 8 633 Kbps Maximum bit rate : 9 100 Kbps Width : 720 pixels Height : 576 pixels Display aspect ratio : 16:9 Frame rate : 25.000 fps Standard : PAL Color space : YUV Chroma subsampling : 4:2:0 Bit depth : 8 bits Scan type : Interlaced Scan order : Top Field First Compression mode : Lossy Bits/(Pixel*Frame) : 0.833 Time code of first frame : 00:00:00:00 Time code source : Group of pictures header Stream size : 1.84 GiB (93%) Audio ID : 189 (0xBD)-128 (0x80) Format : AC-3 Format/Info : Audio Coding 3 Mode extension : CM (complete main) Format settings, Endianness : Big Muxing mode : DVD-Video Duration : 30mn 33s Bit rate mode : Constant Bit rate : 448 Kbps Channel(s) : 6 channels Channel positions : Front: L C R, Side: L R, LFE Sampling rate : 48.0 KHz Bit depth : 16 bits Compression mode : Lossy Delay relative to video : -80ms Stream size : 97.9 MiB (5%) ========================= Any idea what's wrong with the files that prevents ffmpeg from running that task to completion? Thank you.

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Reducing end to end streaming latency (bmd->udp->bmd) < 0.5s
by Anusha Balan 21 Mar '14

21 Mar '14

Hello! I am trying to achieve a streaming latency of around 300ms (end to end) using a dedicated Gigabit Ethernet LAN of 1080i @ 30fps. When I run bmon on linux box based streaming side, it clearly shows that I am using only 1Mbps. Also, another prime source of delay is the receipt of first I-frame on recieving end. I have read the streaming guide and I am trying to incorporate almost everything mentioned there but I am not sure if I am even headed in a direction where this will be possible. What is the minimum latency achieved so far? I managed only about 1.5 - 2s which is extremely high for the interactive application for which I intend to use this. >From this article, it seems feasible that with a dedicated and powerful what I mentioned above is possible. (Given it was 2010, when it was written) http://x264dev.multimedia.cx/archives/249 This is what I am doing at the moment. Streaming Side (On Ubuntu 12.04) ./bmdcapture -C0 -m11 -V4 -v -F nut -f pipe:1 | ffmpeg -i - -vcodec libx264 -preset ultrafast -crf 22 -pix_fmt yuv420p -tune zerolatency -threads 8 -keyint_min 5 -maxrate 800000 -bufsize 26666 -intra-refresh 1 -f mpegts udp://10.0.0.2:6666 Receiving Side (Tested on Ubuntu 12.04 and Windows 7) ffmpeg -i udp://10.0.0.2:6666 -probesize 100 -analyzeduration 50 -f decklink -s 1920x1080 -fflags nobuffer -r 30000/1001 -g 10 -pix_fmt uyvy422 -an "DeckLink 4K Extreme" Is there any parameters I am actually feeding in wrong? Isn't bufsize = maxrate / framerate? Is there anything I can do to make this come down to less than 0.5s? Could really use some expert advice. Thanks, Anusha

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Timing control?
by Spencer Graves 20 Mar '14

20 Mar '14

What's the preferred method for controlling the timing of individual frames? This list previously recommended the concat demuxer with a control file (thread "[FFmpeg-user] concatenate png files for different durations?"). Unfortunately, that failed to control the timing as advertised: A toy example played "duration 6" in less than 1 second followed by 5 seconds for a frame at "duration 1". (I want to mate frames with audio timed to hundreths of a second.) I've not been able to control the times even if all frames have officially the same time. Other posts suggest that I'm not the only person with this problem. It looks to me like there is a bug in the FFmpeg concat demuxer. However, another question asked about keyframes. Researching that led me to documentation saying that, "forcing too many keyframes is very harmful for the lookahead algorithms of certain encoders: using fixed-GOP options or similar would be more efficient." (http://ffmpeg.org/ffmpeg.html) Suggestions? Thanks, Spencer

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Copying streams from partial FLVs?
by Chris Pennello 20 Mar '14

20 Mar '14

I'm trying to copy the video and audio streams from a partial FLV, and it's able to copy only the audio. I'd like to figure out how to get the video out as well. This partial FLV is one that I've downloaded from a server that supports "pseudo-streaming". Such a file can be played in a Flash player. For example, when the user seeks ahead of buffered data, the Flash program consults a table of the keyframe time indexes and corresponding byte offsets, originally read from the video's metadata, and provides a query-string parameter to a "pseudo-streaming"-compatible web server containing the desired byte offset. The web server then supplies not only the video data starting at that byte offset, but it also prepends a 13-byte header which seems to identify the data as an FLV video. I have parsed and consulted this list of keyframe time indexes and corresponding byte offsets in order to obtain the FLV I'm feeding into FFmpeg. How can I get FFmpeg to be able to read the video out of this partial FLV? Find several invocations below. Much appreciated, Chris % ffprobe tmp/rangetest2.flv ffprobe version 2.0.1-tessus Copyright (c) 2007-2013 the FFmpeg developers built on Aug 10 2013 21:25:56 with llvm-gcc 4.2.1 (LLVM build 2336.1.00) configuration: --prefix=/Users/tessus/data/ext/ffmpeg/sw --as=yasm --extra-version=tessus --disable-shared --enable-static --disable-ffplay --enable-gpl --enable-pthreads --enable-postproc --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid --enable-libspeex --enable-bzlib --enable-zlib --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libxavs --enable-version3 --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvpx --enable-libgsm --enable-libopus --enable-fontconfig --enable-libfreetype --enable-libass --enable-libbluray --enable-filters --enable-runtime-cpudetect 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 [h264 @ 0x102014600] no frame! Last message repeated 133 times [h264 @ 0x102014600] non-existing PPS 1 referenced [h264 @ 0x102014600] decode_slice_header error [h264 @ 0x102014600] no frame! Last message repeated 16 times [h264 @ 0x102014600] non-existing SPS 0 referenced in buffering period [h264 @ 0x102014600] no frame! [flv @ 0x10200c000] decoding for stream 0 failed [flv @ 0x10200c000] Could not find codec parameters for stream 0 (Video: h264): unspecified size Consider increasing the value for the 'analyzeduration' and 'probesize' options Input #0, flv, from 'tmp/rangetest2.flv': Duration: 00:20:32.23, start: 1218.614000, bitrate: 5 kb/s Stream #0:0: Video: h264, 30 tbr, 1k tbn, 2k tbc Stream #0:1: Audio: aac, 44100 Hz, mono, fltp % ffmpeg -i tmp/rangetest2.flv -vcodec copy -acodec copy tmp/rangetest2.mp4 ffmpeg version 2.0.1-tessus Copyright (c) 2000-2013 the FFmpeg developers built on Aug 10 2013 21:25:56 with llvm-gcc 4.2.1 (LLVM build 2336.1.00) configuration: --prefix=/Users/tessus/data/ext/ffmpeg/sw --as=yasm --extra-version=tessus --disable-shared --enable-static --disable-ffplay --enable-gpl --enable-pthreads --enable-postproc --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid --enable-libspeex --enable-bzlib --enable-zlib --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libxavs --enable-version3 --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvpx --enable-libgsm --enable-libopus --enable-fontconfig --enable-libfreetype --enable-libass --enable-libbluray --enable-filters --enable-runtime-cpudetect 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 [h264 @ 0x102020c00] no frame! Last message repeated 133 times [h264 @ 0x102020c00] non-existing PPS 1 referenced [h264 @ 0x102020c00] decode_slice_header error [h264 @ 0x102020c00] no frame! Last message repeated 16 times [h264 @ 0x102020c00] non-existing SPS 0 referenced in buffering period [h264 @ 0x102020c00] no frame! [flv @ 0x102018600] decoding for stream 0 failed [flv @ 0x102018600] Could not find codec parameters for stream 0 (Video: h264): unspecified size Consider increasing the value for the 'analyzeduration' and 'probesize' options Input #0, flv, from 'tmp/rangetest2.flv': Duration: 00:20:32.23, start: 1218.614000, bitrate: 5 kb/s Stream #0:0: Video: h264, 30 tbr, 1k tbn, 2k tbc Stream #0:1: Audio: aac, 44100 Hz, mono, fltp File 'tmp/rangetest2.mp4' already exists. Overwrite ? [y/N] y Output #0, mp4, to 'tmp/rangetest2.mp4': Metadata: encoder : Lavf55.12.100 Stream #0:0: Audio: aac ([64][0][0][0] / 0x0040), 44100 Hz, mono Stream mapping: Stream #0:1 -> #0:0 (copy) Press [q] to stop, [?] for help [flv @ 0x102018600] Stream discovered after head already parsed [flv @ 0x102018600] New data stream 0:2 at pos:757450 and DTS:1231.23s size= 134kB time=00:00:14.14 bitrate= 77.6kbits/s video:0kB audio:131kB subtitle:0 global headers:0kB muxing overhead 2.402472%

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udp streaming transrate
by celon 20 Mar '14

20 Mar '14

hi all, I'm using ffmpeg for transrate (down rate) UDP stream CLI is below: ffmpeg -f mpegts -i 'udp://238.1.1.111:8001?fifo_size=1000000&overrun_nonfatal=1' -acodec aac -strict experimental -ar 44100 -ac 2 -b:a 48k -vcodec h264 -pix_fmt yuv420p -profile:v main -s 640x480 -b:v 500k -f mpegts 'udp://238.1.1.202:8001' ============ Last message repeated 1 times [h264 @ 0x2346060] decode_slice_header error [h264 @ 0x2346060] no frame! [h264 @ 0x2346060] Missing reference picture, default is 6 [h264 @ 0x2346060] decode_slice_header error [h264 @ 0x2346060] number of reference frames (0+5) exceeds max (4; probably corrupt input), discarding one Input #0, mpegts, from 'udp://238.1.1.111:8001?fifo_size=1000000&overrun_nonfatal=1': Duration: N/A, start: 28093.147811, bitrate: 48 kb/s Program 1 Stream #0:0[0x121]: Video: h264 (Main) ([27][0][0][0] / 0x001B), yuvj420p(pc, bt470bg), 528x576 [SAR 16:11 DAR 4:3], 25 fps, 50 tbr, 90k tbn, 50 tbc Stream #0:1[0x122]: Audio: mp2 ([3][0][0][0] / 0x0003), 48000 Hz, mono, s16p, 48 kb/s [swscaler @ 0x231b0c0] deprecated pixel format used, make sure you did set range correctly [libx264 @ 0x23509a0] using SAR=1/1 [libx264 @ 0x23509a0] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.1 Cache64 [libx264 @ 0x23509a0] profile Main, level 3.0 Output #0, mpegts, to 'udp://238.1.1.202:8001': Metadata: encoder : Lavf55.34.101 Stream #0:0: Video: h264 (libx264), yuv420p, 640x480 [SAR 1:1 DAR 4:3], q=-1--1, 500 kb/s, 90k tbn, 25 tbc Stream #0:1: Audio: aac, 44100 Hz, stereo, fltp, 48 kb/s Stream mapping: Stream #0:0 -> #0:0 (h264 -> libx264) Stream #0:1 -> #0:1 (mp2 -> aac) Press [q] to stop, [?] for help [mp2 @ 0x2346b20] Header missing Error while decoding stream #0:1: Invalid data found when processing input [adts @ 0x2b9b3e0] Encoder did not produce proper pts, making some up. frame= 375 fps= 28 q=32.0 size= 1156kB time=00:00:14.32 bitrate= 661.1kbits/s dup=46 drop=0 ================================================== I occurs above errors and it's streaming but seems can't join my multicast group properly. please give your kind suggestion thank very much Celon -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/udp-streaming-transrate-tp4664414.… Sent from the FFmpeg-users mailing list archive at Nabble.com.

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Some help/ideas
by JULIAN GARDNER 20 Mar '14

20 Mar '14

I am working on some new infrasturcture which will allow the playing/overlaying of images on to the main video at predtermined times, i can already add/remove images, which was pretty simple, but what i want now is to be able to take an animated gif/png and play this into my system frame by frame so it looks correct when encoded to its final format. My code for the single frame image looks like this static int load_image(uint8_t **image_data, int *w, int *h, int *stride, const char *filename, void *log_ctx) { int ret; enum AVPixelFormat pix_fmt; uint8_t *src_data[4], *dst_data[4]; int src_linesize[4], dst_linesize[4]; /* load image from file */ if ((ret = ff_load_image(src_data, src_linesize, w, h, &pix_fmt, filename, log_ctx)) < 0) return ret; if ((ret = ff_scale_image(dst_data, dst_linesize, *w, *h, PIXEL_FORMAT, src_data, src_linesize, *w, *h, pix_fmt, log_ctx)) < 0) goto end; *stride = dst_linesize[0]; /* copy image_data to a newly allocated array */ *image_data = av_malloc(*stride * *h); if (!*image_data) ret = AVERROR(ENOMEM); av_image_copy_plane(*image_data, *stride, dst_data[0], dst_linesize[0], *w*4, *h); end: av_free(src_data[0]); av_free(dst_data[0]); return ret; } Now this returns a single frame of data containing the image. What i need is a way to load a multi frame gif/png and be able to play each frame individually into the system, and my system gets called every frame to add the image onto the main video plane. So if I can load an animated, then have a call and get the next image then that would be perfect. Any pointers as to how i can do this? I dont want to add this onto the command line as a '-i' as this is going to dynamic. joolz

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DVB subtitling
by Tadas Mačėnas 20 Mar '14

20 Mar '14

Hello, My question is about DVB subtitles. I want to encode and save DVB subtitles as dvb format file. Is it possible? “dvb_subtitle” codec is supported for decoding and encoding (ffmpeg -codecs), but I can’t find DVB format on supported formats list (ffmpeg -codecs). My command fails: [cid:image001.png@01CF3EB2.A70BE870] Thank you for any comments and suggestions. Best regards Tadas

2 1

legal/free to decode wma/mms ?
by Kaineda Dev 20 Mar '14

20 Mar '14

Hello, I'm looking for advice on the following question : is it legal (and royalties free) to use the ffmpeg library to decode wma/mms streams in a commercial ios app? In fact I wanted to use this sdk which uses ffmpeg : http://www.yakamozlabs.com/products/radiotunes but it is very unclear what is legal on not. I contacted them but they don't have any official inputs/link about this subject. thanks for your help! regards Kaineda

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How to cope with slow to open v4l2 devices
by LANGLOIS Olivier PIS -EXT 19 Mar '14

19 Mar '14

Hi, My v4l2 input is very slow (it is measured in seconds) to open and my other video input is a live feed produced by a process that will just bail out if ffmpeg is not able to consume the raw stream in real-time. The whole thing choke right at the beginning when ffmpeg opens the v4l2 device. My live video producer determine that ffmpeg is too slow and close its side of the pipe. ALSA generates buffer overruns. What is the usual strategy to deal with several live inputs with different initialization times? /dev/video0 is a old Logitech webcam. I'm not ruling the possibility to buy a faster device if it can solve my problem. I also have full control over the live feed video producer I am just not sure how to modify it to cope with the problem. (https://github.com/lano1106/glcs) I could make it drop frames but then how will it discern between a one time initialization slow down vs a permanent problem where the encoder is just too slow to process the stream in realtime? I do not like too much this idea as I wonder how I will ensure synchronization between the different streams. 2. If I was reordering the input declaration order from the slowest device to the fastest on the cmdline, could it help? 3. Can ffmpeg report back when its initialization is done and when it is ready to accept data by socket and/or pipe? Here is my super ffmpeg cmdline: ffmpeg -nostats -y \ -f rawvideo -video_size 1920x1080 -pixel_format bgra -framerate 30 -i /dev/stdin \ -f v4l2 -input_format mjpeg -video_size 320x240 -framerate 30 -i /dev/video0 \ -f alsa -acodec pcm_s16le -ar 44100 -ac 2 -i loop_capture \ -f alsa -acodec pcm_s16le -ar 44100 -ac 2 -i sysdefault:1 \ -filter_complex "overlay;[3:a]lowpass=f=4000[lp_out];[2:a][lp_out]amix" \ -c:a libfdk_aac -profile:a aac_low -b:a 128k -ar 44100 \ -c:v libx264 -preset superfast -profile:v main -level 4.1 -pix_fmt yuv420p \ -x264opts keyint=60:bframes=2 -maxrate 6000k -bufsize 12000k -shortest out.mkv Something that I missed a lot during my debugging was timestamps in the log msgs. I did create a small & dirty patch. I do not know if there would be interest from the devs to add it....: --- ffmpeg-2.1.4/libavutil/log.c.orig 2014-03-17 16:41:42.157039924 -0400 +++ ffmpeg-2.1.4/libavutil/log.c 2014-03-17 16:45:01.177223972 -0400 @@ -32,6 +32,7 @@ #if HAVE_IO_H #include <io.h> #endif +#include <time.h> #include <stdarg.h> #include <stdlib.h> #include "avutil.h" @@ -226,6 +227,7 @@ char line[LINE_SZ]; static int is_atty; int type[2]; + struct timespec ts; if (level > av_log_level) return; @@ -253,6 +255,8 @@ count = 0; } strcpy(prev, line); + clock_gettime(CLOCK_MONOTONIC, &ts); + fprintf(stderr, "[%d.%03d] ", ts.tv_sec, ts.tv_nsec/1000000); sanitize(part[0].str); colored_fputs(type[0], part[0].str); sanitize(part[1].str); With this patch, I have the following output (with a different webcam but this demonstrate that v4l2 devices are very slow to start): ffmpeg -y -loglevel debug -f v4l2 -input_format raw -input_format yuyv422 -video_size 320x240 -framerate 30 -i /dev/video0 -c:v libx264 -preset ultrafast out.mkv [23626.939] Opening an input file: /dev/video0. [23626.978] [video4linux2,v4l2 @ 0x955b300] fd:3 capabilities:84000001 [23626.979] [video4linux2,v4l2 @ 0x955b300] Current input_channel: 0, input_name: Camera 1, input_std: 0 [23626.981] [video4linux2,v4l2 @ 0x955b300] Setting time per frame to 1/30 [23627.726] [video4linux2,v4l2 @ 0x955b300] All info found [23627.726] Input #0, video4linux2,v4l2, from '/dev/video0': [23627.727] Duration: [23627.727] N/A[23627.727] , start: [23627.727] 23627.691989[23627.727] , bitrate: [23627.727] 36864 kb/s[23627.727] [23627.727] Stream #0:0[23627.728] , 1, 1/1000000[23627.728] : Video: rawvideo (YUY2 / 0x32595559), yuyv422, 320x240, 1/1000000, 36864 kb/s[23627.728] , 30 fps[23627.728] , 30 tbr[23627.728] , 1000k tbn[23627.728] , 1000k tbc[23627.728] [23627.728] Successfully opened the file. Please ignore the confidentiality notice below. It is automatically added without my consent. ________________________________ CONFIDENTIALITY : This e-mail and any attachments are confidential and may be privileged. If you are not a named recipient, please notify the sender immediately and do not disclose the contents to another person, use it for any purpose or store or copy the information in any medium.

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Trouble using -ss -t with dvcpro50 files
by Shlomo Morosow 19 Mar '14

19 Mar '14

Recently I've been encountering a strange problem which I don't seem to find any info online regarding anything similar, unless of course I'm using the wrong keywords while googling it (probably the case). I hope that perhaps someone here can shed some light on the matter: With a particular file(type) which is dvcpro50 contained within an avi, when I use any syntax (that otherwise works flawlessly on the given file) with the added parameters of -ss and -t it won't export anything but the audio unless I've set the -ss to anywhere in the first few minutes of the video which then it works fine. Further, I've set ffplay to play using the same parameters as for converting video and it has identical results: it won't seek to the referenced frame yet it can play the same file from the beginning or with an -ss set to somewhere in the first few minutes. Here's an example, but keep in mind I've tried this with many output formats and they all work fine until the time is entered (anywhere past the first few minutes) ffmpeg.exe -y -ss 0:10:35.700 -i input.AVI -t 0:02:35.600 -map 0 -c copy "export.mov" I've attached a .txt file with what I get from ffmpeg when trying the above syntax. I've been having this issue with multiple files and they're all of the same codec (dvcpro50) and I don't seem to have it on any other file types. Perhaps be something unique to this codec. I hope someone can help me with this, Thanks

2 2

Frames at a particular (interval) distance
by oana1404 19 Mar '14

19 Mar '14

Hi, I'm trying to extract key frames from a video using the following option: eq(pict_type\,I). However, there are lot of keyframes, and I wouldn't want to get all of them. Thus, I've been thinking to get the timestamps only for keyframes at a difference between 50 and 60 seconds. Is there a way to do that? For example, if the timestamps are the following: t:18.320000 t:24.160000 t:35.880000 t:37.800000 t:47.040000 t:52.840000 t:55.840000 t:64.200000 t:66.080000 t:77.720000 t:80.320000 I would like to have only t:52.840000. Thank you! -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Frames-at-a-particular-interval-di… Sent from the FFmpeg-users mailing list archive at Nabble.com.

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