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• avcodec/cfhd : Keep track of which subbands have been read

  3 avril 2021, par Michael Niedermayer

  avcodec/cfhd : Keep track of which subbands have been read
  This avoids use of uninitialized data
  
  also several checks are inside the band reading code
  
  so it is important that it is run at least once
  Fixes : out of array accesses
  
  Fixes : 28209/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_CFHD_fuzzer-5684714694377472
  
  Fixes : 32124/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_CFHD_fuzzer-5425980681355264
  
  Fixes : 30519/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_CFHD_fuzzer-4558757155700736
  Found-by : continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
  
  Signed-off-by : Michael Niedermayer <michael@niedermayer.cc>
  

  • [DH] libavcodec/cfhd.c
  • [DH] libavcodec/cfhd.h

• 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

• Looping Variables in a Batch File (for use with FFmpeg)

  16 avril 2021, par OrdinaryCarrot16

  I am creating a batch script that stores the input of the user into variables, that will later be used for creating converting files then adding metadata (via FFmpeg). It is working pretty good, now I'm just trying to figure out how to loop the FFmpeg line that uses all the variables the user inputted, and it will loop the command till it reaches the number specified in the 'track_amount' variable.

  


  Do you have any ideas on how to make this work ? Thanks !

  


  setlocal enabledelayedexpansion

:: Scan current directory for audio files
DIR *.mp3 OR *.m4a

set /P cover-art=Select the Audio/Image File with the Cover Art (ex. MP3/M4A or JPG/PNG): 
set /P artist=Artist/Band Name: 
set /P track_amount=How many tracks are in this album: 

if %track_amount% GTR 1 (
    set /P album=Album Name: 
)

set /P filename=Song File Name: 
set /P title=Song Name: 
set /P track=Track Number: 
set /P genre=Genre: 
set /P language=Language: 

ffmpeg -i "%cover-art%" -vf scale=512:512 -sws_flags bicubic cover_tmp.png

for %%t in (!track_amount!) do ffmpeg -i "%filename%" -i cover_tmp.png -map_metadata -1 -map 0:0 -map 1:0 -id3v2_version 3 -metadata artist="%artist%" -metadata album="%album%" -metadata title="%title%" -metadata track="%track%/%track_amount%" -metadata genre="%genre%" -metadata language="%language%" -c:1 png -disposition:1 attached_pic -c:a libmp3lame -ar 44100 -b:a 160k "%title%.mp3"

pause