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  • why type casting on non-pointer struct give syntax error

    1er avril 2020, par Sany Liew

    I am using Visual C++ express 2008 try to compile code similar to below :

    



    no problem

    



    {
  ...
  AVRational test = {1, 1000};
  ...
}


    



    but has problem when it is as below :

    



    {
  ...
  AVRational test = (AVRational){1, 1000};
  ...
}


    



    gave errors :

    



    1>..\..\..\projects\test\xyz.cpp(1139) : error C2059: syntax error : '{'
1>..\..\..\projects\test\xyz.cpp(1139) : error C2143: syntax error : missing   ';' before '{'
1>..\..\..\projects\test\xyz.cpp(1139) : error C2143: syntax error : missing ';' before '}'


    



    where AVRational (ffmpeg.org library) is defined as :

    



    typedef struct AVRational{
    int num; ///< numerator
    int den; ///< denominator
} AVRational;


    



    FFmpeg come with some pre-define value such as

    



    #define AV_TIME_BASE_Q (AVRational){1, AV_TIME_BASE}


    



    which is used as below

    



    av_rescale_q(seek_target, AV_TIME_BASE_Q, pFormatCtx->streams[stream_index]->time_base);


    



    will failed to compile on Visual C++ express 2008

    



    It seem like the same code will be compiled with no error/warning on gcc compiler. Why I get this error on VC++ ? Is it a C/C++ standard way to do casting on struct value ? Anyway I can avoid this error while still able to use the defined AV_TIME_BASE_Q ?

    


  • FFmpeg : A few errors including invalid frame size and incomplete frame

    8 février 2019, par Abs

    I use this ffmpeg command to take clips from a large VOB files that is on a DVD :

    ffmpeg.exe -i E:\VTS_02_2.vob -ss 00:00:57 -t 00:16:15 vids\VTS_02_2.vob

    It worked first for a VOB file but then I got these errors :

    c:\ffmpeg>ffmpeg.exe -i E:\VTS_02_2.vob -ss 00:00:57 -t 00:16:15 vids\VTS_02_2.vob
    
FFmpeg version SVN-r15625, Copyright (c) 2000-2008 Fabrice Bellard, et al.
    
  configuration: --enable-memalign-hack --enable-postproc --enable-swscale --enable-g
    
l --enable-libfaac --enable-libfaad --enable-libgsm --enable-libmp3lame --enable-libv
    
rbis --enable-libtheora --enable-libx264 --enable-libxvid --disable-ffserver --disabl
    
-vhook --enable-avisynth --enable-pthreads
    
  libavutil     49.12. 0 / 49.12. 0
    
  libavcodec    52. 0. 0 / 52. 0. 0
    
  libavformat   52.22. 1 / 52.22. 1
    
  libavdevice   52. 1. 0 / 52. 1. 0
    
  libswscale     0. 6. 1 /  0. 6. 1
    
  libpostproc   51. 2. 0 / 51. 2. 0
    
  built on Oct 16 2008 05:40:02, gcc: 4.2.4 (TDM-1 for MinGW)
    
Input #0, mpeg, from 'E:\VTS_02_2.vob':
    
  Duration: 00:17:11.61, start: 1031.828511, bitrate: 8326 kb/s
    
    Stream #0.0[0x1e0]: Video: mpeg2video, yuv420p, 720x576 [PAR 16:15 DAR 4:3], 9640
    
kb/s, 25.00 tb(r)
    
    Stream #0.1[0x80]: Audio: ac3, 48000 Hz, stereo, s16, 192 kb/s
    
File 'vids\VTS_02_2.vob' already exists. Overwrite ? [y/N] y
    
Output #0, svcd, to 'vids\VTS_02_2.vob':
    
    Stream #0.0: Video: mpeg2video, yuv420p, 720x576 [PAR 16:15 DAR 4:3], q=2-31, 200
    
kb/s, 25.00 tb(c)
    
    Stream #0.1: Audio: mp2, 48000 Hz, stereo, s16, 64 kb/s
    
Stream mapping:
    
  Stream #0.0 -> #0.0
    
  Stream #0.1 -> #0.1
    
Press [q] to stop encoding
    
[ac3 @ 026D3990]frame sync error
    
Error while decoding stream #0.1
    
frame=    0 fps=  0 q=0.0 size=       0kB time=10000000000.00 bitrate=   0.0kbits/s
    
[mpeg2video @ 0141B780]ac-tex damaged at 40 2
    
[mpeg2video @ 0141B780]invalid mb type in B Frame at 3 12
    
[mpeg2video @ 0141B780]00 motion_type at 15 16
    
[mpeg2video @ 0141B780]mb incr damaged
    
[mpeg2video @ 0141B780]00 motion_type at 2 21
    
[mpeg2video @ 0141B780]00 motion_type at 44 22
    
[mpeg2video @ 0141B780]ac-tex damaged at 43 24
    
[mpeg2video @ 0141B780]Warning MVs not available
    
[mpeg2video @ 0141B780]concealing 765 DC, 765 AC, 765 MV errors
    
frame=    0 fps=  0 q=0.0 size=       0kB time=10000000000.00 bitrate=   0.0kbits/s
    
frame=    0 fps=  0 q=0.0 size=       0kB time=10000000000.00 bitrate=   0.0kbits/s
    
frame=    0 fps=  0 q=0.0 size=       0kB time=10000000000.00 bitrate=   0.0kbits/s
    
frame=    0 fps=  0 q=0.0 size=       0kB time=10000000000.00 bitrate=   0.0kbits/s
    
[mpeg2video @ 0141B780]ac-tex damaged at 5 9
    
[mpeg2video @ 0141B780]00 motion_type at 21 13
    
[mpeg2video @ 0141B780]ac-tex damaged at 1 20
    
[mpeg2video @ 0141B780]mb incr damaged
    
[mpeg2video @ 0141B780]00 motion_type at 30 30
    
[mpeg2video @ 0141B780]Warning MVs not available
    
[mpeg2video @ 0141B780]concealing 450 DC, 450 AC, 450 MV errors
    
[ac3 @ 026D3990]frame CRC mismatch
    
[ac3 @ 026D3990]incomplete frame
    
[ac3 @ 026D3990]invalid frame size
    
frame=    0 fps=  0 q=0.0 size=       0kB time=10000000000.00 bitrate=   0.0kbits/s
    
[mpeg2video @ 0141B780]00 motion_type at 38 15
    
[mpeg2video @ 0141B780]00 motion_type at 30 34
    
[mpeg2video @ 0141B780]Warning MVs not available
    
[mpeg2video @ 0141B780]concealing 254 DC, 254 AC, 254 MV errors
    
frame=    0 fps=  0 q=0.0 size=       0kB time=10000000000.00 bitrate=   0.0kbits/s
    
frame=    0 fps=  0 q=0.0 size=       0kB time=10000000000.00 bitrate=   0.0kbits/s
    
[mpeg2video @ 0141B780]ac-tex damaged at 7 6
    
[mpeg2video @ 0141B780]concealing 1350 DC, 1350 AC, 1350 MV errors
    
frame=    0 fps=  0 q=0.0 size=       0kB time=10000000000.00 bitrate=   0.0kbits/s
    
[ac3 @ 026D3990]incomplete frame
    
[ac3 @ 026D3990]invalid frame size
    
frame=    0 fps=  0 q=0.0 size=       0kB time=10000000000.00 bitrate=   0.0kbits/s
    
frame=    0 fps=  0 q=0.0 Lsize=       0kB time=10000000000.00 bitrate=   0.0kbits/s
    

    
video:0kB audio:0kB global headers:0kB muxing overhead -1.#IND00%

    Is there a way to force ffmpeg to just give me an output ?

  • Create HLS streamable audio file from mp3

    15 août 2023, par isADon

    I am using following command to create a hls aac audio file for web streaming

    



    ffmpeg -y -i song.mp3 -c:a aac -b:a 128k -f hls -hls_time 7 -hls_list_size 0 -hls_segment_filename file%d.m4a playlist.m3u8


    



    This command works only with some audio files. With many mp3 files I receive following output :

    



    C:\ffmpeg>ffmpeg -y -i song.mp3 -c:a aac -b:a 128k -f hls -hls_time 7 -hls_list_size 0 -hls_segment_filename file%d.m4a playlist.m3u8
ffmpeg version git-2020-01-31-62d92a8 Copyright (c) 2000-2020 the FFmpeg developers
  built with gcc 9.2.1 (GCC) 20200122
  configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libdav1d --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libmfx --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt --enable-amf
  libavutil      56. 38.100 / 56. 38.100
  libavcodec     58. 67.100 / 58. 67.100
  libavformat    58. 37.100 / 58. 37.100
  libavdevice    58.  9.103 / 58.  9.103
  libavfilter     7. 72.100 /  7. 72.100
  libswscale      5.  6.100 /  5.  6.100
  libswresample   3.  6.100 /  3.  6.100
  libpostproc    55.  6.100 / 55.  6.100
[mp3 @ 0000027d800babc0] Estimating duration from bitrate, this may be inaccurate
Input #0, mp3, from 'song.mp3':
  Metadata:
    TSS             : Logic Pro 8.0.2
    iTunNORM        :  000000EE 000000ED 00000C34 00001135 000088F0 0000B505 000080FA 00007577 00009B82 00018F49
    iTunSMPB        :  00000000 00000210 00000A07 00000000008783E9 00000000 007AD4E6 00000000 00000000 00000000 00000000 00000000 00000000
    genre           : Rock
    TCM             : Kevin MacLeod
    album           : Funk and Blues
    TKE             : C
    TBP             : 101
    title           : Funkorama
    artist          : Kevin MacLeod
    date            : 2008-06-16 18:35
  Duration: 00:03:21.46, start: 0.000000, bitrate: 325 kb/s
    Stream #0:0: Audio: mp3, 44100 Hz, stereo, fltp, 320 kb/s
    Stream #0:1: Video: mjpeg (Baseline), yuvj444p(pc, bt470bg/unknown/unknown), 400x400 [SAR 72:72 DAR 1:1], 90k tbr, 90k tbn, 90k tbc (attached pic)
    Metadata:
      comment         : Other
Stream mapping:
  Stream #0:1 -> #0:0 (mjpeg (native) -> h264 (libx264))
  Stream #0:0 -> #0:1 (mp3 (mp3float) -> aac (native))
Press [q] to stop, [?] for help
[hls @ 0000027d80100c40] Frame rate very high for a muxer not efficiently supporting it.
Please consider specifying a lower framerate, a different muxer or -vsync 2
[libx264 @ 0000027d800c1280] using SAR=1/1
[libx264 @ 0000027d800c1280] MB rate (56250000) > level limit (16711680)
[libx264 @ 0000027d800c1280] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX FMA3 BMI2 AVX2
[libx264 @ 0000027d800c1280] profile High 4:4:4 Predictive, level 6.2, 4:4:4, 8-bit
[libx264 @ 0000027d800c1280] 264 - core 159 - H.264/MPEG-4 AVC codec - Copyleft 2003-2019 - 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=4 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=25 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, hls, to 'playlist.m3u8':
  Metadata:
    TSS             : Logic Pro 8.0.2
    iTunNORM        :  000000EE 000000ED 00000C34 00001135 000088F0 0000B505 000080FA 00007577 00009B82 00018F49
    iTunSMPB        :  00000000 00000210 00000A07 00000000008783E9 00000000 007AD4E6 00000000 00000000 00000000 00000000 00000000 00000000
    genre           : Rock
    TCM             : Kevin MacLeod
    album           : Funk and Blues
    TKE             : C
    TBP             : 101
    title           : Funkorama
    artist          : Kevin MacLeod
    date            : 2008-06-16 18:35
    encoder         : Lavf58.37.100
    Stream #0:0: Video: h264 (libx264), yuvj444p(pc, progressive), 400x400 [SAR 72:72 DAR 1:1], q=-1--1, 90k fps, 90k tbn, 90k tbc (attached pic)
    Metadata:
      comment         : Other
      encoder         : Lavc58.67.100 libx264
    Side data:
      cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: N/A
    Stream #0:1: Audio: aac (LC), 44100 Hz, stereo, fltp, 128 kb/s
    Metadata:
      encoder         : Lavc58.67.100 aac
[mp3float @ 0000027d80146580] overread, skip -7 enddists: -6 -6 speed=68.6x
[mp3float @ 0000027d80146580] overread, skip -6 enddists: -5 -5
[mp3float @ 0000027d80146580] overread, skip -6 enddists: -4 -4
    Last message repeated 2 times
[mp3float @ 0000027d80146580] overread, skip -7 enddists: -6 -6
    Last message repeated 2 times
[mp3float @ 0000027d80146580] overread, skip -5 enddists: -2 -2
[mp3float @ 0000027d80146580] overread, skip -7 enddists: -6 -6
[mp3float @ 0000027d80146580] overread, skip -6 enddists: -4 -4
    Last message repeated 1 times
[mp3float @ 0000027d80146580] overread, skip -7 enddists: -6 -6
    Last message repeated 1 times
[mp3float @ 0000027d80146580] overread, skip -6 enddists: -4 -4
[mp3float @ 0000027d80146580] overread, skip -5 enddists: -3 -3
[mp3float @ 0000027d80146580] overread, skip -6 enddists: -4 -4
[mp3float @ 0000027d80146580] overread, skip -7 enddists: -6 -6
    Last message repeated 2 times
[mp3float @ 0000027d80146580] overread, skip -5 enddists: -4 -4
[hls @ 0000027d80100c40] Opening 'file0.m4a' for writingate=N/A speed=64.1x
[hls @ 0000027d80100c40] Opening 'playlist.m3u8.tmp' for writing
frame=    1 fps=0.3 q=33.0 Lsize=N/A time=00:03:21.45 bitrate=N/A speed=63.7x
video:7kB audio:3209kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
[libx264 @ 0000027d800c1280] frame I:1     Avg QP:34.64  size:  6567
[libx264 @ 0000027d800c1280] mb I  I16..4: 19.5% 53.0% 27.5%
[libx264 @ 0000027d800c1280] 8x8 transform intra:53.0%
[libx264 @ 0000027d800c1280] coded y,u,v intra: 46.8% 26.1% 15.3%
[libx264 @ 0000027d800c1280] i16 v,h,dc,p: 38% 39%  9% 14%
[libx264 @ 0000027d800c1280] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 21% 14% 26%  8%  5%  6%  5%  7%  7%
[libx264 @ 0000027d800c1280] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 42% 16% 14%  7%  4%  5%  3%  4%  4%
[libx264 @ 0000027d800c1280] kb/s:4728240.00
[aac @ 0000027d800bcc40] Qavg: 2138.508


    



    Notice the "mp3float overread" message.

    



    It results in a single file0.m4a file without splitting it up after every 7 seconds as specified.
This is an example audio file I am trying to convert to a aac hls stream that results the mentioned problem : https://incompetech.com/music/royalty-free/index.html?isrc=USUAN1100474

    



    How can I convert an audio file to a web friendly hls stream with ffmpeg ?

    


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