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• lavf/matroskadec : Normalize noncompliant A_QUICKTIME/V_QUICKTIME private data

  10 janvier 2016, par Mats Peterson

  lavf/matroskadec : Normalize noncompliant A_QUICKTIME/V_QUICKTIME private data
  This patch adds a new static function get_qt_codec() that takes care of
  
  the initial retrieval of the fourcc and codec ID for A_QUICKTIME and
  
  V_QUICKTIME. It also normalizes noncompliant private data found in some
  
  older files that incorrectly starts with the fourcc by expanding/shifting
  
  the data by 4 bytes, and storing the data size at the start. This is
  
  necessary in order for the rest of the code in the A_QUICKTIME and
  
  V_QUICKTIME blocks (and most likely other code as well) to correctly
  
  parse the private data.
  Signed-off-by : Michael Niedermayer <michael@niedermayer.cc>
  

  • [DH] libavformat/matroskadec.c

• avcodec/mpegaudio_tablegen : more dynamic table creation speedups

  28 novembre 2015, par Ganesh Ajjanagadde

  avcodec/mpegaudio_tablegen : more dynamic table creation speedups
  This further speeds up runtime initialization, with identical generated tables.
  Sample benchmark (x86-64, Haswell, GNU/Linux) :
  old :
  
  34441423 decicycles in mpegaudio_tableinit,    8192 runs,      0 skips
  new :
  
  10776291 decicycles in mpegaudio_tableinit,    8192 runs,      0 skips
  Most low hanging fruit is taken care of here. For some idea, note that
  
  83,064 array elements totalling 233,722 bytes need to be initialized.
  
  Thus, with this patch, we average 12.9 cycles per element or 4.6
  
  cycles per byte.
  Reviewed-by : Derek Buitenhuis <derek.buitenhuis@gmail.com>
  
  Signed-off-by : Ganesh Ajjanagadde <gajjanagadde@gmail.com>
  

  • [DH] libavcodec/mpegaudio_tablegen.h

• Convert .mp3/.wav file into g729/amr/g711 codec file and vice versa using PJSIP

  19 septembre 2017, par NandhaKumar

  PJSIP is used for SIP Registering, audio/video Calling and supporting some other VoIP Call features. If you want to create an VoIP Application, you will create an application using their libraries.

  In this library, there are available lot of codecs and some of codecs are available as third party codecs and you can integrate into PJSIP library. Then you can able to support those codecs while calling another SIP user/client.

  Generally In mobile phones, the voice is record as audio file through
  mic and it has passed to PJSIP library. Then those codec/pjmedia library take care
  of other operations like converting into anyother codec format which
  they want to send.

  Instead of that, can we able to pass .mp3/.wav file into PJSIP library and convert it into codec files like .g729/.amr/.g711-u and vice versa

  I don’t about how codec conversion/codec engine internally works around.
  If you know about the codec conversion from .mp3 format is possible using PJSIP library, please suggest me how to solve this problem.

  Thanks in Advance !