Recherche avancée

Sur d’autres sites (4802)

• FFmpeg : Error occurred while encoding audio stream from ac3 to aac

  4 décembre 2015, par akki_2891

  I am using following command to encode my video to h264 and aac audio codec :

  ffmpeg -i sample.mp4 -codec:v libx264 -profile:v high -level:v 4.0 -codec:a libvo_aacenc -b:a 128k output_file.mp4

  Following is a trace of the error i am getting at the console.

  Input #0, mpegts, from 'sample.mp4':
  
    Duration: 00:00:58.08, start: 1.000033, bitrate: 17290 kb/s
  
    Program 1
  
      Stream #0:0[0x1011]: Video: h264 (High) (HDMV / 0x564D4448), yuv420p, 1920x1
  
  080 [SAR 1:1 DAR 16:9], 29.97 fps, 59.94 tbr, 90k tbn, 59.94 tbc
  
      Stream #0:1[0x1100]: Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, 5.1(side), fl
  
  tp, 448 kb/s
  
      Stream #0:2[0x1200]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090)
  
  [libx264 @ 003dc660] using SAR=1/1
  
  [libx264 @ 003dc660] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX
  
  [libx264 @ 003dc660] profile High, level 4.0
  
  [libx264 @ 003dc660] 264 - core 130 r2274 c832fe9 - H.264/MPEG-4 AVC codec - Cop
  
  yright 2003-2013 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 de
  
  block=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 chr
  
  oma_qp_offset=-2 threads=12 lookahead_threads=2 sliced_threads=0 nr=0 decimate=1
  
   interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=
  
  1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scen
  
  ecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmi
  
  n=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
  
  [libvo_aacenc @ 03c3c640] Unable to set encoding parameters
  
  Output #0, mp4, to 'output_file.mp4':
  
      Stream #0:0: Video: h264, yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], q=-1--1, 90
  
  k tbn, 29.97 tbc
  
      Stream #0:1: Audio: aac, 48000 Hz, 5.1(side), s16, 128 kb/s
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (h264 -> libx264)
  
    Stream #0:1 -> #0:1 (ac3 -> libvo_aacenc)
  
  Error while opening encoder for output stream #0:1 - maybe incorrect parameters
  
  such as bit_rate, rate, width or height

  I am really not able to find what possibly could be wrong. I dont want to copy audio codec , i wish to encode it from ac3 to aac

  Also any suggestion to make it lossless

• Ffmpeg not fully functioning

  26 août 2014, par Joshua

  Hello i have problem with ffmpeg i’m using phpvibe cms already asked in their forums but they don’t know the answer which leads me that it’s probably my fault so the problem is when i upload :

  6 seconds avi video which weights 817 KB it uploads perfectlly plays like it should run.

  However when i try upload 12mb avi sample i see that it does not work in folder where should be that video converted it shows 0kb size

  That means it didn’t start converting somewhow i think this is ffmpeg issue also i have spoted that my ffmpeg is placed in /usr/local/bin/ and in phpvibe cms systems there is different location for ffmpeg it looks like this : Img link

  Why when i upload small avi video it converts good and when i try longer it fails maybe it converts while upload process is going on but than why it shows 0b in folder where converted videos should be placed ?

  P.s i tried couple avi videos that weights more than 10 mb same thing and my code for ffmpeg conversation is :

  $output ="{ffmpeg-cmd} -i {input} -vcodec libx264 -s {ffmpeg-vsize} -threads 4 -movflags faststart {output}.mp4";

  Also tried :

  output ="{ffmpeg-cmd} -i {input} -vcodec libx264 -s {ffmpeg-vsize} -threads 4 {output}.mp4 2>&1";

  Same thing + im using Debian 7 Wheezy. Thanks !

  EDIT :

  i have made a test script it gives this output :

  array(21) {
  
    [0]=>
  
    string(83) "ffmpeg version git-2014-08-25-bb29896 Copyright (c) 2000-2014 the FFmpeg developers"
  
    [1]=>
  
    string(61) "  built on Aug 25 2014 19:52:12 with gcc 4.7 (Debian 4.7.2-5)"
  
    [2]=>
  
    string(62) "  configuration: --enable-shared --enable-libx264 --enable-gpl"
  
    [3]=>
  
    string(40) "  libavutil      54.  7.100 / 54.  7.100"
  
    [4]=>
  
    string(40) "  libavcodec     56.  0.101 / 56.  0.101"
  
    [5]=>
  
    string(40) "  libavformat    56.  3.100 / 56.  3.100"
  
    [6]=>
  
    string(40) "  libavdevice    56.  0.100 / 56.  0.100"
  
    [7]=>
  
    string(40) "  libavfilter     5.  0.103 /  5.  0.103"
  
    [8]=>
  
    string(40) "  libswscale      3.  0.100 /  3.  0.100"
  
    [9]=>
  
    string(40) "  libswresample   1.  1.100 /  1.  1.100"
  
    [10]=>
  
    string(40) "  libpostproc    53.  0.100 / 53.  0.100"
  
    [11]=>
  
    string(31) "Input #0, avi, from 'test.avi':"
  
    [12]=>
  
    string(11) "  Metadata:"
  
    [13]=>
  
    string(36) "    encoder         : Nandub v1.0rc2"
  
    [14]=>
  
    string(60) "  Duration: 00:01:09.78, start: 0.000000, bitrate: 1517 kb/s"
  
    [15]=>
  
    string(126) "    Stream #0:0: Video: msmpeg4v3 (DIV3 / 0x33564944), yuv420p, 640x352, 1279 kb/s, 23.98 fps, 23.98 tbr, 23.98 tbn, 23.98 tbc"
  
    [16]=>
  
    string(83) "    Stream #0:1: Audio: mp3 (U[0][0][0] / 0x0055), 48000 Hz, stereo, s16p, 222 kb/s"
  
    [17]=>
  
    string(87) "[libx264 @ 0x6c31e0] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.2 AVX"
  
    [18]=>
  
    string(44) "[libx264 @ 0x6c31e0] profile High, level 3.0"
  
    [19]=>
  
    string(649) "[libx264 @ 0x6c31e0] 264 - core 124 - H.264/MPEG-4 AVC codec - Copyleft 2003-2012 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=4 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=23 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"
  
    [20]=>
  
    string(131) "[aac @ 0x6c3c20] The encoder 'aac' is experimental but experimental codecs are not enabled, add '-strict -2' if you want to use it."
  
  }
  
  int(1)

  i get this output with this :

  $output ="{ffmpeg-cmd} -y -i {input} {output}.mp4 2<&1"";

  simple ffmpeg command also tried :

  $output ="{ffmpeg-cmd} -i {input} -vcodec libx264 -s {ffmpeg-vsize} -threads 4 {output}.mp4 2>&1";

  With this command it gives this output :

  array(21) {
  
    [0]=>
  
    string(83) "ffmpeg version git-2014-08-25-bb29896 Copyright (c) 2000-2014 the FFmpeg developers"
  
    [1]=>
  
    string(61) "  built on Aug 25 2014 19:52:12 with gcc 4.7 (Debian 4.7.2-5)"
  
    [2]=>
  
    string(62) "  configuration: --enable-shared --enable-libx264 --enable-gpl"
  
    [3]=>
  
    string(40) "  libavutil      54.  7.100 / 54.  7.100"
  
    [4]=>
  
    string(40) "  libavcodec     56.  0.101 / 56.  0.101"
  
    [5]=>
  
    string(40) "  libavformat    56.  3.100 / 56.  3.100"
  
    [6]=>
  
    string(40) "  libavdevice    56.  0.100 / 56.  0.100"
  
    [7]=>
  
    string(40) "  libavfilter     5.  0.103 /  5.  0.103"
  
    [8]=>
  
    string(40) "  libswscale      3.  0.100 /  3.  0.100"
  
    [9]=>
  
    string(40) "  libswresample   1.  1.100 /  1.  1.100"
  
    [10]=>
  
    string(40) "  libpostproc    53.  0.100 / 53.  0.100"
  
    [11]=>
  
    string(31) "Input #0, avi, from 'test.avi':"
  
    [12]=>
  
    string(11) "  Metadata:"
  
    [13]=>
  
    string(36) "    encoder         : Nandub v1.0rc2"
  
    [14]=>
  
    string(60) "  Duration: 00:01:09.78, start: 0.000000, bitrate: 1517 kb/s"
  
    [15]=>
  
    string(126) "    Stream #0:0: Video: msmpeg4v3 (DIV3 / 0x33564944), yuv420p, 640x352, 1279 kb/s, 23.98 fps, 23.98 tbr, 23.98 tbn, 23.98 tbc"
  
    [16]=>
  
    string(83) "    Stream #0:1: Audio: mp3 (U[0][0][0] / 0x0055), 48000 Hz, stereo, s16p, 222 kb/s"
  
    [17]=>
  
    string(88) "[libx264 @ 0x13046c0] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.2 AVX"
  
    [18]=>
  
    string(45) "[libx264 @ 0x13046c0] profile High, level 3.0"
  
    [19]=>
  
    string(650) "[libx264 @ 0x13046c0] 264 - core 124 - H.264/MPEG-4 AVC codec - Copyleft 2003-2012 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=4 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=23 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"
  
    [20]=>
  
    string(132) "[aac @ 0x13055a0] The encoder 'aac' is experimental but experimental codecs are not enabled, add '-strict -2' if you want to use it."
  
  }
  
  int(1)

  And also tried thi command :

  $output ="{ffmpeg-cmd} -i {input} -vcodec libx264 -s {ffmpeg-vsize} -threads 4 -movflags faststart {output}.mp4";

  And with this it gives :

      array(0) {
  
  }
  
  int(1)

  in all cases file that weitghs 12 mb doest not convert it’s 0kb in ftp

• AAC encoder : Extensive improvements

  11 octobre 2015, par Claudio Freire

  AAC encoder : Extensive improvements
  This finalizes merging of the work in the patches in ticket #2686.
  Improvements to twoloop and RC logic are extensive.
  The non-exhaustive list of twoloop improvments includes :
  
   - Tweaks to distortion limits on the RD optimization phase of twoloop
  
   - Deeper search in twoloop
  
   - PNS information marking to let twoloop decide when to use it
  
     (turned out having the decision made separately wasn’t working)
  
   - Tonal band detection and priorization
  
   - Better band energy conservation rules
  
   - Strict hole avoidance
  For rate control :
  
   - Use psymodel’s bit allocation to allow proper use of the bit
  
     reservoir. Don’t work against the bit reservoir by moving lambda
  
     in the opposite direction when psymodel decides to allocate more/less
  
     bits to a frame.
  
   - Retry the encode if the effective rate lies outside a reasonable
  
     margin of psymodel’s allocation or the selected ABR.
  
   - Log average lambda at the end. Useful info for everyone, but especially
  
     for tuning of the various encoder constants that relate to lambda
  
     feedback.
  Psy :
  
   - Do not apply lowpass with a FIR filter, instead just let the coder
  
     zero bands above the cutoff. The FIR filter induces group delay,
  
     and while zeroing bands causes ripple, it’s lost in the quantization
  
     noise.
  
   - Experimental VBR bit allocation code
  
   - Tweak automatic lowpass filter threshold to maximize audio bandwidth
  
     at all bitrates while still providing acceptable, stable quality.
  I/S :
  
   - Phase decision fixes. Unrelated to #2686, but the bugs only surfaced
  
     when the merge was finalized. Measure I/S band energy accounting for
  
     phase, and prevent I/S and M/S from being applied both.
  PNS :
  
   - Avoid marking short bands with PNS when they’re part of a window
  
     group in which there’s a large variation of energy from one window
  
     to the next. PNS can’t preserve those and the effect is extremely
  
     noticeable.
  M/S :
  
   - Implement BMLD protection similar to the specified in
  
     ISO-IEC/13818:7-2003, Appendix C Section 6.1. Since M/S decision
  
     doesn’t conform to section 6.1, a different method had to be
  
     implemented, but should provide equivalent protection.
  
   - Move the decision logic closer to the method specified in
  
     ISO-IEC/13818:7-2003, Appendix C Section 6.1. Specifically,
  
     make sure M/S needs less bits than dual stereo.
  
   - Don’t apply M/S in bands that are using I/S
  Now, this of course needed adjustments in the compare targets and
  
  fuzz factors of the AAC encoder’s fate tests, but if wondering why
  
  the targets go up (more distortion), consider the previous coder
  
  was using too many bits on LF content (far more than required by
  
  psy), and thus those signals will now be more distorted, not less.
  The extra distortion isn’t audible though, I carried extensive
  
  ABX testing to make sure.
  A very similar patch was also extensively tested by Kamendo2 in
  
  the context of #2686.
  

  • [DH] Changelog
  • [DH] libavcodec/aac.h
  • [DH] libavcodec/aaccoder.c
  • [DH] libavcodec/aaccoder_trellis.h
  • [DH] libavcodec/aaccoder_twoloop.h
  • [DH] libavcodec/aacenc.c
  • [DH] libavcodec/aacenc.h
  • [DH] libavcodec/aacenc_is.c
  • [DH] libavcodec/aacenc_is.h
  • [DH] libavcodec/aacenc_pred.c
  • [DH] libavcodec/aacenc_quantization.h
  • [DH] libavcodec/aacenc_utils.h
  • [DH] libavcodec/aacpsy.c
  • [DH] libavcodec/mathops.h
  • [DH] libavcodec/mips/aaccoder_mips.c
  • [DH] libavcodec/psymodel.c
  • [DH] libavcodec/psymodel.h
  • [DH] tests/fate/aac.mak