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• AAC encoder : ANMR, avoid empty search ranges

  5 décembre 2015, par Claudio Freire

  AAC encoder : ANMR, avoid empty search ranges
  Avoid generating broken paths when some bands have an
  
  empty search space.
  

  • [DH] libavcodec/aaccoder.c

• 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

• Recording RTP VP8 packets with FFMPEG with named pipe

  6 août 2015, par Filipe Apolinário

  I’m developing a WebRTC video session recorder, in a gateway developed in C++, where I only have access to individual RTP packets.

  When a session starts, I create two threads one that initializes a named pipe and an other that starts FFMPEG to fetch data from that pipe and store it in a matroska file, with the command :

  ffmpeg -i \\.\pipe\screenRec -f matroska D:\djhfifj.mkv

  Whenever I receive an RTP packet I send it through the pipe to FFMPEG. Although all communication is working fine, FFMPEG does not seem to be recognizing the RTP packet :

  ffmpeg version N-73633-gdfc5858 Copyright (c) 2000-2015 the FFmpeg developers
  
    built with gcc 4.9.2 (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-libbs2b --enable-libcaca --enable-libdcadec --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --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-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-lzma --enable-decklink --enable-zlib
  
    libavutil      54. 28.100 / 54. 28.100
  
    libavcodec     56. 47.100 / 56. 47.100
  
    libavformat    56. 40.100 / 56. 40.100
  
    libavdevice    56.  4.100 / 56.  4.100
  
    libavfilter     5. 21.100 /  5. 21.100
  
    libswscale      3.  1.101 /  3.  1.101
  
    libswresample   1.  2.100 /  1.  2.100
  
    libpostproc    53.  3.100 / 53.  3.100
  
  [aac @ 031b3fc0] Format aac detected only with low score of 1, misdetection possible!
  
  [aac @ 031bd820] More than one AAC RDB per ADTS frame is not implemented. Update your FFmpeg version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented.
  
  [aac @ 031bd820] channel element 3.13 is not allocated
  
  [aac @ 031bd820] Reserved bit set.
  
  [aac @ 031bd820] Number of bands (26) exceeds limit (9).
  
  [aac @ 031bd820] Reserved bit set.
  
  [aac @ 031bd820] Number of bands (60) exceeds limit (44).
  
  [aac @ 031bd820] Number of bands (6) exceeds limit (4).
  
  [aac @ 031bd820] Reserved bit set.
  
  [aac @ 031bd820] Number of bands (30) exceeds limit (23).
  
  [aac @ 031bd820] Sample rate index in program config element does not match the sample rate index configured by the container.
  
  [aac @ 031bd820] Inconsistent channel configuration.
  
  [aac @ 031bd820] get_buffer() failed
  
  [aac @ 031bd820] Assuming an incorrectly encoded 7.1 channel layout instead of a spec-compliant 7.1(wide) layout, use -strict 1 to decode according to the specification instead.
  
  [aac @ 031bd820] Reserved bit set.
  
  [aac @ 031bd820] Number of bands (16) exceeds limit (11).
  
  [aac @ 031bd820] Dependent coupling is not supported together with LTP
  
      Last message repeated 9 times
  
  [aac @ 031bd820] channel element 3.5 is not allocated
  
  [aac @ 031bd820] channel element 3.13 is not allocated
  
  [aac @ 031bd820] channel element 3.3 is not allocated
  
  [aac @ 031bd820] Number of bands (16) exceeds limit (14).
  
  [aac @ 031bd820] channel element 3.10 is not allocated
  
  [aac @ 031bd820] channel element 3.2 is not allocated
  
  [aac @ 031bd820] Reserved bit set.
  
  [aac @ 031bd820] Number of scalefactor bands in group (61) exceeds limit (43).
  
  [aac @ 031bd820] Reserved bit set.
  
  [aac @ 031bd820] Number of bands (29) exceeds limit (13).
  
  [aac @ 031bd820] Sample rate index in program config element does not match the sample rate index configured by the container.
  
  [aac @ 031bd820] Inconsistent channel configuration.
  
  [aac @ 031bd820] get_buffer() failed
  
  [aac @ 031bd820] channel element 0.7 is not allocated
  
  [aac @ 031bd820] Number of bands (24) exceeds limit (15).
  
  [aac @ 031bd820] channel element 1.1 is not allocated
  
  [aac @ 031bd820] channel element 2.0 is not allocated
  
  [aac @ 031bd820] Reserved bit set.
  
  [aac @ 031bd820] Number of scalefactor bands in group (62) exceeds limit (41).
  
  [aac @ 031bd820] Reserved bit set.
  
  [aac @ 031bd820] Number of bands (15) exceeds limit (13).
  
  [aac @ 031bd820] Reserved bit set.
  
  [aac @ 031bd820] Number of bands (23) exceeds limit (2).
  
  [aac @ 031bd820] channel element 1.4 is not allocated
  
  [aac @ 031bd820] Assuming an incorrectly encoded 7.1 channel layout instead of a spec-compliant 7.1(wide) layout, use -strict 1 to decode according to the specification instead.
  
  [aac @ 031bd820] channel element 1.2 is not allocated
  
  [aac @ 031bd820] channel element 1.8 is not allocated
  
  [aac @ 031bd820] channel element 3.7 is not allocated
  
  [aac @ 031bd820] channel element 2.9 is not allocated
  
  [aac @ 031bd820] channel element 3.8 is not allocated
  
  [aac @ 031bd820] Reserved bit set.
  
  [aac @ 031bd820] Number of scalefactor bands in group (48) exceeds limit (43).
  
  [aac @ 031bd820] channel element 3.8 is not allocated
  
  [aac @ 031bd820] channel element 2.13 is not allocated
  
  [aac @ 031bd820] channel element 3.4 is not allocated
  
  [aac @ 031bd820] Dependent coupling is not supported together with LTP
  
      Last message repeated 13 times
  
  [aac @ 031bd820] channel element 2.14 is not allocated
  
  [aac @ 031bd820] SBR was found before the first channel element.
  
  [aac @ 031bd820] Sample rate index in program config element does not match the sample rate index configured by the container.
  
  [aac @ 031bd820] Inconsistent channel configuration.
  
  [aac @ 031bd820] get_buffer() failed
  
  [aac @ 031bd820] Number of bands (6) exceeds limit (5).
  
  [aac @ 031bd820] channel element 3.0 is not allocated
  
  [aac @ 031bd820] channel element 1.5 is not allocated
  
  [aac @ 031bd820] channel element 1.13 is not allocated
  
  [aac @ 031bd820] channel element 1.7 is not allocated
  
  [aac @ 031bd820] channel element 2.0 is not allocated
  
  [aac @ 031bd820] Dependent coupling is not supported together with LTP
  
      Last message repeated 13 times
  
  [aac @ 031bd820] channel element 3.0 is not allocated
  
  [aac @ 031bd820] Assuming an incorrectly encoded 7.1 channel layout instead of a spec-compliant 7.1(wide) layout, use -strict 1 to decode according to the specification instead.
  
  [aac @ 031bd820] SBR was found before the first channel element.
  
  [aac @ 031bd820] Reserved bit set.
  
  [aac @ 031bd820] Number of bands (31) exceeds limit (30).
  
  [aac @ 031bd820] channel element 1.12 is not allocated
  
  [aac @ 031bd820] Reserved bit set.
  
  [aac @ 031bd820] Number of bands (29) exceeds limit (10).
  
  [aac @ 031bd820] channel element 3.2 is not allocated
  
  [aac @ 031bd820] channel element 3.15 is not allocated
  
  [aac @ 031bd820] channel element 1.5 is not allocated
  
  [aac @ 031bd820] channel element 2.7 is not allocated
  
  [aac @ 031bd820] channel element 1.9 is not allocated
  
  [aac @ 031bd820] Number of bands (54) exceeds limit (34).
  
  [aac @ 031bd820] channel element 1.6 is not allocated
  
  [aac @ 031bd820] channel element 1.2 is not allocated
  
  [aac @ 031bd820] channel element 3.7 is not allocated
  
  [aac @ 031bd820] Reserved bit set.
  
  [aac @ 031bd820] ms_present = 3 is reserved.
  
  [aac @ 031b3fc0] decoding for stream 0 failed
  
  [aac @ 031b3fc0] Could not find codec parameters for stream 0 (Audio: aac (LTP), 4.0, fltp, 1506 kb/s): unspecified sample rate
  
  Consider increasing the value for the 'analyzeduration' and 'probesize' options
  
  \\.\pipe\screenRec96: could not find codec parameters
  
  Input #0, aac, from '\\.\pipe\screenRec96':
  
    Duration: N/A, bitrate: 1506 kb/s
  
      Stream #0:0: Audio: aac (LTP), 4.0, fltp, 1506 kb/s
  
  [abuffer @ 0435cd00] Value inf for parameter 'time_base' out of range [0 - 2.14748e+009]
  
      Last message repeated 3 times
  
  [abuffer @ 0435cd00] Error setting option time_base to value 1/0.
  
  [graph 0 input from stream 0:0 @ 0319afe0] Error applying options to the filter.
  
  Error opening filters!

  Is it possible to make FFMPEG understand that the packet sent is RTP with VP8 ?