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FMJ-Software releases Awave Audio v11.0


FMJ-Software has announced that version 11.0 of Awave Audio has been released. This major update of the Audio file format batch converter adds Direct Stream Copy for select compressed formats - including support for computing gain adjustment meta data without modifying the audio. It also introduces a function to auto-select the best output data format, and support for the EBU R 128 integrated loudness measure.

Changes in Version 11.0:

  • Most file formats now have a new "<Auto select>" data format selection. Select this to let the file writer decide what data format to use. It does this by querying the file reader about the input file's data format - if it is a data format that the writer can also handle, then it will use that (thus preserving the data format of the input file). Otherwise, it will select a default data format (e.g. "PCM 16-bit").
  • Added "Direct Stream Copy" support for MPEG audio layer II, MPEG audio layer III and MPEG AAC compressed data (a.k.a. MP2, MP3 and AAC). NB, you *must* set the output data format to "<Auto select>" for this to work, and it will only work for file format for which the program normally supports writing the resp. data format. And just like "Direct Stream Copy" with uncompressed formats, it can only be used if the audio data is not modified in any way (so no resampling et c). When it can be used, it has the advantage of copying the compressed stream verbatim (e.g. from an.AVI file to a.MP3 file) without loosing audio quality due to recompression.
  • When the “Normalize” feature is set to "output meta data" (but not when set to "modify audio") it will now support "Direct Stream Copy" for MPEG audio layer II, MPEG audio layer III and MPEG AAC audio (in addition to uncompressed formats). The practical upside of this is that you can run MP2, MP3 or AAC data through the program and calculate gain adjustment (e.g. ReplayGain) which is added as meta data to the output file – without degrading the compressed audio.
  • Added support for reading and writing raw AC3 audio streams (.AC3), raw padded DTS audio streams (.DTS), and raw compact DTS audio streams (.CPT). Please note however, that the program does not contain any AC3 codec so by itself it can neither compress nor decompress these types of data. However, if you have installed a "Windows ACM filter" that can decode AC3 or DTS (e.g. the common "AC3Filter"), then it can use that to decompress such data. There's currently no way for the program to compress data to these formats. What you can now do though, is to copy compressed streams between files using new "Direct Stream Copy" support for AC3 and DTS. File formats that supports this are:.AC3,.AVI,.CPT,.DTS,.MKV,.MOV,.WAV (NB:.AVI,.MKV and.MOV are read only, and for the others you must select "AC3" or "<Auto select>" as output data format for the copy to work).
  • Added support for normalization per EBU R 128 (this is basically the ITU BS 1770 Leq(R2LB) loudness measure + two gating functions + definition of a "0 LU" reference level). NB, the EBU R128 target level of "0 LU = -23 LUFS" lies at approx. -5dB compared to the Replay Gain target. So if you wish to test to use EBU R 128 instead of ReplayGain, then you may want to enter a target value of 5 LU to compensate.
  • The normalization options "-20-Leq(RLB)" and "-20-Leq(R2LB)" have been replaced by "-Leq(RLB)" and "-Leq(R2LB)", with a default target level of -20 dBFS. These algorithms, both from ITU BS 1770, will now also work at sample rates other than 48KHz.
  • When normalizing the audio volume using the Replay Gain methods, the target value box now allows you select the desired reference level in dB(SPL). The original Replay Gain document specifies calibration against a 83 dB(SPL) reference level, but the majority of software today use 89 dB(SPL) instead (because the original value was deemed to be too low).
  • Added a normalization option to find the "True Peak Level". Whenever you enable any of the normalization types, the peak sample value is also determined, and is saved as meta data (if the output file format supports it). With true peak enabled, it will also examine the signal at time points between the original sample values (using a 16x oversampling filter - for better precision than the 4x demanded by EBU R 128). This comes closer to the true peak that a DAC will have to handle.
  • The channel icons in the input file list now better corresponds to actual the speaker layout (if known), not just the number of channels.
  • The "Mixing" tab of the file options dialog now allows you to indicate which speakers are be used (this info can be saved in.MOV,.W64, the "Microsoft extensible" version of.WAV,.WMA, and.WV.).
  • The "Format options" dialog box now allows you to select the sample rate for Rockwell ADPCM files (typically either 7200 or 8000 Hz).
  • For writing MPEG layer II compressed data, a new v1.3 of tooLameF.dll is required (available from our web-site).
  • Various minor file format-related improvements.


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