Softpedia >Windows >Multimedia >Audio >Audio Converters >Awave Audio >  Changelog

Awave Audio Changelog

What's new in Awave Audio 11.3

Jun 6, 2022
  • Added support for image meta data (cover image album art). Supports ID3v2-tags (.AAC, .AIFF, .FLAC, .MP2, .MP3, .WAV), APE-tags (.AAC, .APE, .MP3, .WV), as well as some format specific picture meta data (.FLAC, .MKA, .OGG, .WMA)
  • The file info dialog has a new "Image" tab where you can view, override, or export the image for an input file
  • There's also an "Override image..." command if you right-click on a set of selected input files, allowing setting an image for multiple files in one go
  • Added support for EBU R 128 loudness range calculation according to Tech note 3342. This is a piece of meta data that, in addition to the normal R 128 loudness measure, tells how much a clip varies in loudness. It can currently only be saved to a BEXT-chunk (Broadcast Wave Extension), available when saving to BWF-type .WAV or .W64. It is calculated whenever you enable normalization using EBU R 128
  • Added "Direct Stream Copy" for Ogg Vorbis compressed data. Supported for .OGG (read and write), .MKV and .WAV (read only). This allows copying from one format to another, or modifying meta data without having to recompress the audio (since earlier already supported for MPEG and AAC data)
  • The "format options" page now has a sample rate selection for G721/G723/G726 files, for overriding the default 8000 Hz when reading or writing
  • Support for DirectX plug-ins has been removed as they are largely deprecated
  • The program and the installer executables are now digitally signed for improved security

New in Awave Audio 11.2 (Sep 2, 2020)

  • Awave Audio is now available in both 64-bit native and 32-bit versions.
  • The UI now supports high-dpi screens (without blurry upscaling).
  • The "Add file" dialog now use an up-to-date style (with a folder browser to the left).
  • The Audio Player now has a ">>" button which jumps to the next file in the input list, and a "<<" button, which either rewinds the current file, or jumps to the previous (when already rewinded).
  • Added detection and optional auto-merging of multi-channel files from mono files with file names ending with a speaker indication, e.g. "-l", ".r", " lfe" or "_sr". This is similar to the “merge split mono to stereo” function previously available, but now supports more channels.
  • Added support for reading and writing Opus files (.opus).
  • Added support for reading and writing raw ITU G.722 data. If you use a .G722 file extension, it assumes the 64-kbit/s mode. Use .G722-6, .G722-7 or .G722-8 to explicitly indicate 48, 56 or 64 kbit/s modes (i.e. 6, 7 or 8 bits per codeword, with two samples encoded per word). Use the "Format options" dialog to select between big-endian and little-endian packing of the codewords. It can also read G.722 data from .AU and .WAV-files now.
  • Added support for reading MIDI SDS sysex dump files (.SDS).
  • Added support for reading compacted MIDI SDS data as saved by the SDX program (.SDX).
  • Added support for reading SmartSound SDS files (.SDS).
  • Added support for reading ID3v2 tags embedded in Microsoft wave files (.WAV).
  • Added support for optionally writing ID3v2 tags to .AIFF and .WAV files. You can enable this in the "Format options" dialog (it is disabled by default as the data chunks are not standard).
  • Added support for BWF version 2 (EBU tech 3285), allowing EBU R 128 loudness and peak level meta data to be stored in Broadcast wave files.
  • When writing .OGG files, you can now select between "VBR Quality", "ABR" and "SBR" bit rate modes.
  • If you read or write multichannel audio to or from AAC, FLAC, OGG, OPUS or WAV, the audio channel order will now be rearranged to conform with the respective formats standard order (e.g. when converting a 5.1 WAV to Opus it will reorder from L R C LFE RL RR into L C R RL RR LFE).
  • The mixing dialog now indicate the speaker designation of the out-channel of each mix entry. Note that the channel numbering refers to the Microsoft standard order, after reordering has been done from the input file format and before any is done for the output file format.
  • Added a "Sampler" tab to the File options dialog, where looping, tuning and volume adjustment parameters for sampler synthesizers are now collected.
  • Added EBU R 128 loudness, loudness range and peak value fields to the metadata tab on the File options dialog (NB not editable - these should always be calculated algorithmically). Also added a "Cue points" field (which can be edited).
  • Miscellaneous bug-fixes.

New in Awave Audio 11.1 (Mar 23, 2013)

  • Added support for using VST plug-ins for audio effects processing. This is integrated into the same page as the DX-plug-ins. In the "Effect setup" dialog (which shows the VST plug-ins GUI window), you can load and save plug-in preset files from the system menu (click on the small upper-left window icon of the dialog). In the "Format options" dialog, you can set one or more VST search path(s) (; separated - if left blank, it will retrieve the standard Steinberg path from the registry).
  • Added a new "channel format" option named "Split files; 0 1 ...". This splits multi-channel input files into multiple mono output files, adding the channel index to the file name (e.g. "x.0.wav", "x.1.wav" et c). Also, the old "Dual files 'L+R'" option has been replaced by "Split files; l r ..." which works similarly, but adds one or more letters indicating the speaker placement instead the index (e.g. "x.l.wav", "x.r.wav", "x.lfe.wav").
  • Added an "" dithering option. This selects "None (round to nearest)" if the output bit depth is = the input bit depth and no sample modifying audio processing is done (resampling, plug-ins, et c). In all other cases it selects "Noise-shaped dither". NB; Like in previous versions, the dithering option is disabled if the output data format is floating point or use a compression codec that accepts floating point data.
  • The conversion progress is now also shown in the taskbar (Windows 7 / 8 only).
  • The "Create log file" option now writes the log file to the out-path.
  • Updated EBU R 128 normalization to the 2011 revision of the spec (now 10 dB relative gate & 75% block overlap).
  • For writing BWF .WAV files (Broadcast Wave Files), the "Write EBU 'levl' chunk" option (in the "Format options" dialog) has been complemented by a block size selection and a choice of storing either only "Absolute peaks" or both "Pos. + Neg." peak values.
  • The output file format list is now formatted so that you can type in the first letter of the file extension to search in the list.

New in Awave Audio 11.0 (Apr 12, 2011)

  • Most file formats now have a new "" 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 "" 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 "" 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.

New in Awave Audio 10.5 (Nov 11, 2010)

  • Much improved resampling speed, and faster handling of uncompressed data formats

New in Awave Audio 10.4 (May 10, 2010)

  • Added a "Play selected" option to the "Add files" dialog where, when you select a file, you can hear its contents playing after 1s. The dialog is now also resizeable.
  • Added support for the Windows Audio Sessions API (WASAPI) for audio playback. This is the "native" audio interface for Windows 7 and Vista, providing low-latency, high quality audio playback (when running on Windows XP, the DirectSound API is used instead).
  • Added support for "performer" (band, orchestra) and "conductor" text meta data.
  • Added support for reading audio data from Matroska container format files (.MKV and .MKA). Currently the following audio codecs are supported: MPEG, AAC, Ogg, and PCM (sorry, no AC3 or DTS!)
  • Added support for reading audio from .AVI files that use OpenDML index list extensions (a.k.a. "AVI v2").
  • Added support for reading .WAV files containing Ogg Vorbis format compressed audio.
  • Added support for reading rare "non-seekeable" .OGG files.
  • Added support for reading and writing raw 12- and 20-bit signed linear PCM audio data (.L12 and .L20 files), packed as per RFC 1890 and RFC 3190.
  • Added support for reading 2/2.67/4-bit SoundBlaster/Creative Labs ADPCM compression from .VOC files.
  • Added support for reading and writing DAT LP (long play mode) 12-bit non-linear audio format (.DAT12 files), as per IEC 61119, and packed according to RFC 3190.
  • In total, including formats already supported in previous versions, the following file extensions borrowed from RTP payload encoding names (RFC 1890 and RFC 3190), are now recognized: .L8, .L16, .L20, .L24, .DAT12, .G721, .G728, .GSM, .MPA, .PCMA and .PCMU.

New in Awave Audio 10.3 (Nov 6, 2009)

  • Added support for reading and writing Musepack (.MPC) compressed files (SV7 and SV8 types supported for reading, SV8 only for writing).
  • Added support for reading and writing files with Shorten (.SHN) lossless compression.
  • Several improvements to the normalization function:
  • Fixed a conformance problem with Replay Gain meta data calculations (if you have used an earlier version to calculate Replay Gain meta data for files, then it is recommended that you use a "tag editor" of your own choice to rescan them and calculate new Replay Gain data). This selection is now called "Replay Gain 'Standard'".
  • The new selection "Replay Gain 'ISO 226:2003'" adds a fairly accurate filter implementation based on the "75-phon equal loudness contour" from the "new" research standardized as ISO 226:2003. This is an alternative to the standard Replay Gain hearing filter (which is an approximation of the "80-dB F-weight curve", based on research as old as 1933!).
  • Also added new "Leq(RLB)" and "Leq(R2LB)" selections for loudness measurement (the latter is also know as ITU BS 1770). NB, these are currently only supported for 48KHz.
  • Several improvements to the "Dithering" option (called "Quantization" in previous versions):
  • Available dithering options are now: "None (round to nearest)", "White noise dither - Rectangular p.d.f. - +/-0.5 lsb", "White noise dither - Triangular p.d.f. - +/-1 lsb" (new in this version), and "Noise-shaped dither (IRSO 226:2003 HR) - Triangular p.d.f." (new in this version, better than old noise-shaped dither option that it replaces).
  • The selection box is now disabled when selecting a data format where no quantization has to be made (e.g. "Float 32-bit") or where the encoder accepts the data in floating point format and does its own quantization if necessary (e.g. .AAC, .MP2, .MP3, .MPC, .OGG).
  • The dithering selection is now stored in preset files.
  • Added Unicode support for file names.
  • Added Unicode support for text meta data (for file formats that support it). When saving to meta data formats (e.g. ID3v2 tags) that supports multiple character encodings, the program will now automatically select the most compact one (from Latin-1, UTF-8 or UTF-16) that can represent the text without losing any data.
  • Preset files (.aap) are now stored in app-data folder (they were previously stored in the program folder, where Vista UAC didn't like them to be created).
  • Added support for reading text meda data in .WAV files from a "cart chunk" used by some radio broadcasters.
  • When reading a .CDA file (Windows place-holder for an Audio CD-track), then text meta data (artist, album, track name, track no, year) are now automatically retrieved from the freedb internet database.
  • Added support for an "URL" (web-link) text meta data type.

New in Awave Audio 10.2 (Jun 25, 2008)

  • Added support for reading and writing Apples CoreAudio file format (.caf, .caff). Currently supported for reading are uncompressed PCM, floating point, IMA 4-bit ADPCM, MAC3 and MAC6, mu-law and A-law, (AAC, ALAC, MP1, MP2, MP3) data formats. Supported for writing are PCM, float, mu/A-law, and IMA ADPCM.
  • Added support for encoding 12 or 20-bits/sample FLAC files (in addition to 8, 16 or 24-bits/sample).
  • Added selections for iPhone ringtones (.m4r, same thing as the .m4a format).
  • Added ASIO support to the audio recording wizard.

New in Awave Audio 10.1 (Jan 28, 2008)

  • Added support for reading and writing Multichannel Broadcast wave format (.WAV) a.k.a. EBU RF64. This is an extension to the Broadcast wave format (which is in turn based on the normal Microsoft wave format). It adds features to support for file sizes > 4 GB, as well as for high bit-depth and multi-channel audio.
  • Added support for reading and writing WavPack (.WV) lossless compressed files. NB; to use this you need to install the free WavPack codec add-on (wavpackdll.dll).
  • Added support for reading Apple lossless (a.k.a. ALAC) compressed data (.M4A and .MOV files).
  • Added a "Pattern rename files" option (right click on the input files list to find it). This allows you to use wildcard patterns to automatically create output file name overrides for a lot of files.
  • Added support for using libFLAC.dll versions later than v1.1.2.
  • Added selection of data format type for .VAP files to the 'More options...' dialog.
  • Added a new 'Normalize' option: "Adjust volume by selected amount", which simply allows you to select by how many dB to change the volume of the input.

New in Awave Audio 10.00 (Jan 1, 2007)

  • New or improved support for important file formats (.MOV, .AVI, .M4A, .MP4, .3GP, .3G2 et c), support for writing ID3v2 format meta data, improved help file, and much more!