TAudioConverter, Multithreaded Audio Converter and Extractor Options

[ozok]

@dreamliner77

1) Yes, tags will be copied to mp3 files. But I suggets you do a test and see if it works as you expected. Also note that artwork cannot be copied at the moment.
2) TAC does not have that feature yet I'm afraid.

[dreamliner77]

What I'm trying to do is convert a mix of flac and mpc to mp3 for portable use. I would like to duplicate everything (art in folder, txt files, etc) to the identical directory structure but not have to manually delete the (original) source files in the new directory. I know that might not make sense...

None of my business, but I would strongly advise against deleting your lossless source files. Once you convert to a lossy format, you're basically stuck with that format forever (converting one lossy format to another loses even more audio information). If you must delete them, make sure you delete to a large Recycle Bin in case you make a mistake and need to re-do the conversion.

I won't being deleting my archive FLAC's. I've collected those over many years and ripping sessions. Unfortunately they now sit on DVD+R's. They WILL resurface on hard disk soon once I have the time to copy them all over to 3 or 4 TB drives (in duplicate). What I'm attempting to do now is copy my music to a 1 TB 2.5" drive that will be housed in my car and read by a unit that only plays back mp3, wma and aac.

[dreamliner77]

@dreamliner77

1) Yes, tags will be copied to mp3 files. But I suggets you do a test and see if it works as you expected. Also note that artwork cannot be copied at the moment.
2) TAC does not have that feature yet I'm afraid.

Thanks! I'm hoping to test it out this weekend. I'm still debating if I should copy the directory structure, convert and re-add artwork or copy the entire source, convert to the existing directory and then recursively delete mpc and flac...

[kokuyo]

Hi ozok,

I want you to add a mp4 and m4b extension setting option.

Thanks, good app!

[ozok]

Update:

--0.7.2
-Added: option to extract embedded artworks to output folder (jpg, png) (thx to gottkaiser)
-Added: A better command creation algorithm
-Added: Info balloon will now show encoding time and speed
-Added: More skins
-Added: Some shortcuts (del, shift+del etc)
-Added: Option to select between extensions for aac output (thx to kokuyo)
-Added: Support for m4b and aac files
-Fixed: Adding files with shell content menu will be disabled while encoding
-Fixed: Could not write tags to TAK files (thx to jkauff)
-Fixed: Change log could not be viewed with installer version

[kokuyo]

Update:

--0.7.2
-Added: option to extract embedded artworks to output folder (jpg, png) (thx to gottkaiser)
-Added: A better command creation algorithm
-Added: Info balloon will now show encoding time and speed
-Added: More skins
-Added: Some shortcuts (del, shift+del etc)
-Added: Option to select between extensions for aac output (thx to kokuyo)
-Added: Support for m4b and aac files
-Fixed: Adding files with shell content menu will be disabled while encoding
-Fixed: Could not write tags to TAK files (thx to jkauff)
-Fixed: Change log could not be viewed with installer version

Thank you!!

[gottkaiser]

-Added: option to extract embedded artworks to output folder (jpg, png) (thx to gottkaiser)

Thanks for the work done. It's a good feature. But I didn't mean exporting the embedded artwork.
I meant copying the artwork to the newly created file tags (id3v2, vorbis comments or APEv2).

[ozok]

Yes, I know. But until I find a way to embed artwork to output file, this is the solution I could find.

Also if you guys get "Invalid floating point operation" after updating to 0.7.2, you can go to settings window and uncheck "Extract and save artwork to output" checkbox. It looks like there is a bug with MediaInfo.

[gottkaiser]

With lame it is actually quite easy. Just use the "--ti" switch to include any jpg or png file.
To include an image in vorbis comments you have to encode it with Base64 to text before you include it.

[ozok]

With lame it is actually quite easy. Just use the "--ti" switch to include any jpg or png file.
To include an image in vorbis comments you have to encode it with Base64 to text before you include it.

I was planning to find out for more encoders and implement it, but I can add only for these two as well.

[Stop the Noise]

Hi Ozok,

Just want to say THANK YOU! for all the hard work you've put into this program.

Every update you make improves the app that much more, very impressed!

[eahm]

Agree, I followed this project from day 1 and it progressed extremely well. I think ozok is a very good developer and he knows what he's doing.

I don't use it but I will keep testing it, thanks for the effort ozok.

This post has been edited by eahm: Jan 14 2013, 08:44

[ozok]

Thank you guys, it means a lot.

[o-l-a-v]

Possible to add support for splitting audio files with attached cuesheets?

[sundance]

@ozok,
how do you think about implementing ReplayGain into TAudioConverter?
So if your (lossless) files have ReplayGain tags, you could select to use either track or album gain information, apply corresponding attenuation (via SoX or encoder command line) + apply addional gain (eg. if you want your mp3s @ 92dB reference level instead of standard 89dB) and thus have your target files all at the same loudness level, regardless if your player can't handle RG tags.

.sundance.

[ozok]

Possible to add support for splitting audio files with attached cuesheets?

Yes, this is in my to-do list. I do not know when, but I will add support for cuesheets.

@ozok,
how do you think about implementing ReplayGain into TAudioConverter?
So if your (lossless) files have ReplayGain tags, you could select to use either track or album gain information, apply corresponding attenuation (via SoX or encoder command line) + apply addional gain (eg. if you want your mp3s @ 92dB reference level instead of standard 89dB) and thus have your target files all at the same loudness level, regardless if your player can't handle RG tags.

.sundance.

I actually do not know much about replay gain. I think adding both keeping just in tag and applying replay gain with SoX sounds like a good idea. Added this to my to-do list.

[carpman]

Hi ozok

Just tried your portable version on Win 7 Ultimate x64. Tried to add both a folder of, and individual TAK files. Program wouldn't recognise them / allow me to add. Can the program not decode TAK files?

Also, on a seperate note -- would be really good if you could include lossyWAV.

Looks like a promising program. Personally I use fb2k, but when this is done I'll definitely recommend TAC to friends who don't want to dig into fb2k. I like that there's a portable version too.

Cheers,

C.

[ozok]

@carpman you are right, tak files were not recognised. I've added support for them. You can download TAC.exe with tak support from here https://dl.dropbox.com/u/9617171/TAudioConverter.7z.


BTW, if you get "invalid float operation" error with 0.7.2 while you don't get it with 0.7.1, just uncheck "Extract and copy artwork to output" in "Settings" window.

Edit: Typo

This post has been edited by ozok: Jan 22 2013, 11:29

[carpman]

I'll give it a go.
What are your thoughts re. lossyWAV?

C.

EDIT.
Well, I just encoded a TAK file to Opus which resulted in a 360 byte .opus file which foobar2000 said was a corrupted format.

Decoding failure at 0:00.000 (Unsupported format or corrupted file)

A couple of suggestions too:

EDIT 2 --- 1) It doesn't seem to remember the last directories chosen (input and output -- may be wrong about that?) sorry!
2) Haven't been able to drag and drop files (it would be nice if it supported that)

This post has been edited by carpman: Jan 22 2013, 11:47

[carpman]

Okay, did a little more testing.

• I can drag and drop individual MP3s and FLACs, but not TAK files.
• I can't drag and drop directories containing audio files.

C.

EDIT:

On the encoding from TAK:

All encodes from TAK result in < 1KB files (tested Opus, Lame MP3)
However encoding from MP3 or FLAC to Opus is fine, so it's an "encode from TAK" issue.

EDIT2:

Stream mapping:
Stream #0:0 -> #0:0 (tak -> pcm_s16le)
Press [q] to stop, [?] for help
[tak @ 0217b820] unsupported codec: 0
Error while decoding stream #0:0: Not yet implemented in FFmpeg, patches welcome

This post has been edited by carpman: Jan 22 2013, 12:07

[carpman]

One other thing you may want to look at:
TAC Portable seems to create a TAC directory in:

C:\Users\USERNAME\Documents\TAC

The folder is empty and I didn't see anywhere in the settings re. the use/creation of this folder. I don't see the need for it.

C.

[ozok]

I'll release a new version fixing these issues tonight. Thanks for bug reports, carpman.

[carpman]

It's going to be good. I like the interface. I like that it's portable. I'd like LossyWav support (that would also help set it apart from the competition ).
EDIT: I also like that it maintains directory structure (if you want it to). Agree with posting above re. ReplayGain support would be nice.

Very nice work!

C.

This post has been edited by carpman: Jan 22 2013, 12:27

[carpman]

Another bug?

I just tried to encode some 44100 Hz 16 bit FLACs to the default LAME VBR @ ~200kbps, bitdepth (original) ....

They were resampled to 8000Hz !

Resampling: input 44.1 kHz output 8 kHz
polyphase lowpass filter disabled
Encoding C:\Users\USERNAME\AppData\Local\Temp\TAudioConverter\TEST.wav
to H:\TEST.mp3
Encoding as 8 kHz j-stereo MPEG-2.5 Layer III VBR(q=6.48399)

Redacted Log file below:

CODE

Outputs:
Encoder1:
ffmpeg version N-48425-g9e36d9e Copyright (c) 2000-2013 the FFmpeg developers
  built on Jan  3 2013 19:39:04 with gcc 4.7.2 (GCC)
  configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libtheora --enable-libtwolame --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib
  libavutil      52. 13.100 / 52. 13.100
  libavcodec     54. 85.101 / 54. 85.101
  libavformat    54. 59.102 / 54. 59.102
  libavdevice    54.  3.102 / 54.  3.102
  libavfilter     3. 30.102 /  3. 30.102
  libswscale      2.  1.103 /  2.  1.103
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  2.100 / 52.  2.100
[flac @ 022197e0] max_analyze_duration 5000000 reached at 5015510
Input #0, flac, from 'H:\TEST.flac':
  Metadata:
    ALBUM           : TEST
    ARTIST          : TEST
    COMMENT         : TEST
    DATE            : TEST
    GENRE           : TEST
    PERFORMER       : TEST
    TITLE           : TEST
    REPLAYGAIN_ALBUM_GAIN: +3.06 dB
    REPLAYGAIN_ALBUM_PEAK: 0.505005
    REPLAYGAIN_TRACK_GAIN: +3.06 dB
    REPLAYGAIN_TRACK_PEAK: 0.505005
  Duration: 00:04:00.12, bitrate: 695 kb/s
    Stream #0:0: Audio: flac, 44100 Hz, stereo, s16
Output #0, wav, to 'C:\Users\USERNAME\AppData\Local\Temp\TAudioConverter\TEST.wav':
  Metadata:
    IPRD            : TEST
    IART            : TEST
    ICMT            : TEST
    ICRD            : TEST
    IGNR            : TEST
    PERFORMER       : TEST
    INAM            : TEST
    REPLAYGAIN_ALBUM_GAIN: +3.06 dB
    REPLAYGAIN_ALBUM_PEAK: 0.505005
    REPLAYGAIN_TRACK_GAIN: +3.06 dB
    REPLAYGAIN_TRACK_PEAK: 0.505005
    ISFT            : Lavf54.59.102
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (flac -> pcm_s16le)
Press [q] to stop, [?] for help
size=   41364kB time=00:04:00.11 bitrate=1411.2kbits/s    
video:0kB audio:41364kB subtitle:0 global headers:0kB muxing overhead 0.000614%

LAME 3.99.5 64bits (http://lame.sf.net)
CPU features: SSE (ASM used), SSE2 (ASM used)
Resampling:  input 44.1 kHz  output 8 kHz
polyphase lowpass filter disabled
Encoding C:\Users\USERNAME\AppData\Local\Temp\TAudioConverter\TEST.wav
      to H:\TEST.mp3
Encoding as 8 kHz j-stereo MPEG-2.5 Layer III VBR(q=6.48399)

3 different FLAC files tested all were resampled to 8 kHz.

C.

[ozok]

Yes, it is a bug. Even if user disables effects-filters, if sample rate value is set something different than "Original", audio files sampled to that value.

This post has been edited by ozok: Jan 22 2013, 17:44