TAudioConverter 0.9.5-Multithreaded Audio Converter-Extractor CD ripper

[ozok]

Update:

--0.8.3
-Added: Support for fdkaac (thx to o-l-a-v for binaries)
-Added: Support for .aiff and .aif files (thx to eahm)
-Added: Custom output folder structures (thx to NappyHead and eahm)
-Added: New textboxes, trackbars, etc (some settings are reseted)
-Added: New progress view layout
-Fixed: Summary didn't show if custom tags were used
-Fixed: Some texts couldn't be seen if skins were disabled
-Updated: AlphaControls full version 8.31 Beta (thx for donations)

[ozok]

Update:

--0.8.4
-Added: More skins
-Added: Option to disable keeping encoder logs (thx to o-l-a-v)
-Added: Ability to convert to uncompressed aiff files (thx to eahm)
-Added: Temp. wav files will be deleted right after encoding is done (thx to NappyHead)
-Fixed: Too long log save times when encoding to MP3 (Lame) (thx to o-l-a-v)
-Fixed: Mp3 (Lame) progress wasn't shown properly (thx to o-l-a-v)
-Fixed: "Invalid floating point operation" errors
-Improved: Lowered GUI's CPU usage while converting
-Removed: CPU and memory usage bars
-Updated: FLAC to 1.2.1 mod4 by Case (thx to eahm)
-Updated: MadExcept to 4.0.6
-Updated: QAAC and refalac to 2.18

[ozok]

Update:

--0.8.5
-Added: Using pipes with encoders that support it
-Added: New skin "DarkMetro"
-Added: Option to skip a file if it's output exists (thx to henrikk)
-Fixed: FFMpeg AAC encoding bitrate was always 128kbps problem (thx to norc426)
-Fixed: It took too long to save logs
-Fixed: A possible error which caused source files to be deleted
-Fixed: Ctrl+A combination caused strange behaviour (thx to o-l-a-v)
-Fixed: Some codec settings obscured other controls
-Removed: SoX
-Removed: Some audio filters
-Updated: OpusEnc to 0.1.6 using libopus 1.1.x [2013-04-23]
-Updated: AltiVec/SSE optimized lame (thx to Brazil2)
-Updated: madExcept to 4.0.7

[ozok]

Some users reported that they get "a,bc is not a float" error. I fixed it. If you have that error too, you can try TAudioConverter.exes below:

For installed version: https://dl.dropboxusercontent.com/u/9617171/TAudioConverter.7z
For portable version: https://dl.dropboxusercontent.com/u/9617171/TAudioConverter_portable.7z

If you cannot download these files now, please try them later. My upload speed seems to be very low.

[ozok]

Update:

--0.8.6
-Added: "Volume" filter is now percentage based
-Fixed: "x,yy is not a float" error when TAC started
-Fixed: Encoding failed if effects were actived but not changed from defaults (thx to Glenn Gomes)
-Updated: FFmpeg to latest static build from ffmpeg.zeranoe.com/builds/

[ozok]

Update:

--0.8.6
-Added: "Volume" filter is now percentage based
-Fixed: "x,yy is not a float" error when TAC started
-Fixed: Encoding failed if effects were actived but not changed from defaults (thx to Glenn Gomes)
-Updated: FFmpeg to latest static build from ffmpeg.zeranoe.com/builds/

[ozok]

Update:

--0.8.7
-Added: Seperate 32bit and 64bit builds
-Added: Temp. wav files will be used again instead of stdin/stdout
-Added: SoX is added to package
-Added: Portable TAC will now save logs under "logs" folder (thx to o-l-a-v)
-Added: Some changes to info window
-Added: TAC will now delet temp wav files after encoding is done or stopped
-Added: TAC now won't write date to empty log entries
-Added: Support for sampling rates 96khz and 192khz (thx to o-l-a-v)
-Added: Option to jump latest files in progress list
-Fixed: Using FFMpeg as encoder caused channel problems
-Updated: FLAC to 1.3.0

Homepage: http://taudioconverter.sourceforge.net/

[ozok]

Update 0.8.8:

-Added: Support for encoder FLACCL (thx to ChronoSphere)
-Added: Support for pre-processor lossyWAV
-Added: Support for ".dtsma" extension (thx to hubblec4)
-Added: Channel options for lame (thx to Trinket)
-Added: Replaygain option to FLAC encoder
-Added: Option to play a sound after encoding was done
-Added: Option to overwrite if output file alread exists (thx to ChronoSphere)
-Fixed: Temp wav files couldn't be deleted right after an encode
-Fixed: SoX temp files weren't deleted right after an encode (thx to kolpotoru)
-Fixed: Logs weren't saved in "logs" folder by portable TAC (thx to o-l-a-v)
-Fixed: SoX effects weren't shown in the summary
-Fixed: Output folder edit was enabled after an encode to source folder (thx to ChronoSphere)
-Fixed: "Clear All" button in log window didn't work
-Updated: TAK to 2.3.0
-Updated: AlphaControls to 8.40 Stable

[ozok]

Update:

--0.8.9
-Added: Trim tool
-Added: TAC will now ignore files in the list that do not exist
-Fixed: TAC tried to save some text files to "C:\" (thx to Capsbackup)
-Fixed: TAC will now use UTF-8 when saving logs (thx to ChronoSphere)

[ozok]

Update 0.9.0:

--0.9.0
-Added: New skins "Steam2" and "GPlus"
-Added: Hints to some options (thx to Trinket)
-Added: Option to pass blocksize option to lossless encoders for losswav (thx to elubron)
-Added: Support for extensions mpa, mp2 and mka (thx to kleen)
-Fixed: LossyWAV failed if temp folder path ended with "\"
-Fixed: FDKAAC always used CBR even if VBR was selected (NePaC)
-Fixed: Removing files from list didn't reset trim values (thx to elubron)
-Fixed: Moving files in the list caused problems with duration etc
-Fixed: Musepack, TAK, Wavpack couldn't overwrite files even if it was enabled
-Updated: SoX build with unicode by Lord_Mulder
-Updated: AlphaControls to 8.41 Stable
-Updated: MediaInfo to 0.7.64
-Updated: OpusEnc to 1.1-beta

[hello_hello]

I just downloaded TAudioConverter to give it a spin and it looks like a nice program. Thanks. However....

I can't open files using the Add Files menu. I can add them using the Add Folder menu or I can drag and drop them into the program, but the "Add Files" menu doesn't show a single file on my hard drive, supported or otherwise. I'm running XP. Also, there's a couple of entries in the "file type" drop down list which aren't legible.



Or sometimes it looks like this:

[ozok]

I think it is caused by skinning component. I fix it and everytime I add a new extension it get messed up again. Link below contain fix for it. They are portable versions and also include ttagger.exe which will be used to write tags. Extract TAC.exe and ttagger.exe to TAC folder.

https://dl.dropboxusercontent.com/u/9617171/TAudioConverter_32_portable.7z
https://dl.dropboxusercontent.com/u/9617171/TAudioConverter_64_portable.7z

[hello_hello]

That seems to have fixed the problem. Thanks!

[ozok]

No problem. Feel free to report other bugs or make suggestions.

[hello_hello]

Given I needed to time stretch some audio today and I'd just downloaded TAudioConverter..... but alas it doesn't seem to have such a function. Would it be worth implimenting? Even just basic PAL to NTSC conversion etc might be handy.

Every so often I need to time-stretch audio by a small amount (ie a percentage) which doesn't seem to be a function too many programs have. MeGUI's audio encoder will do "frame rate" conversions but you can't specify anything else. For those I use the SoundTouch plugin with foobar2000.

I do like being able to open audio with DirectShow for encoding. Sometimes ffdshow's audio decoder filters come in handy..... remapping channels "on the fly", adjusting volume, and quite often "problem" audio files will only encode/decode via DirectShow. Every so often I'll try to convert an audio stream which switches between 2ch and 5.1ch (ie TV captures) and most encoders/decoders don't seem to cope, whereas I can use ffdshow's mixer to output a consistent number of channels. I've not tried one of those file with TAudioConverter yet.
Currently I either force MeGUI to open audio files via DirectShow if need be, or I open them via a DirectShow script with foobar2000, but TAudioConverter doesn't seem to like opening scripts.

My first TAudioConverter encoding job failed because I tried converting 5.1ch audio to MP3 without setting the downmix to stereo in the filters section. I kind of assumed if I chose MP3 as the output and I was converting 5.1ch audio, the downmixing part would go without saying....

I hate window animations with a passion. I have them all disabled. So I kind of get annoyed when a program doesn't follow the way Windows is configured in that regard and animates anyway. No big deal, but seeing as you asked.....

[hello_hello]

PS While I know TAudioConverter is designed for converting...... given it also has a trim function..... it might be nice to be able to trim the audio without re-encoding it. MeGUI will cut sections of audio then join the remaining bits back together, but it needs to use the "cuts file" it creates when adding cuts to AVISynth scripts. I'm not sure I've come across too many conversion programs which make it nice and easy to do something like "trim "x"amount from the beginning but don't re-encode the rest". Currently I use MP3DirectCut for MP3 but for other formats.... well if there's no other choice I'll sometimes do silly things like mux the audio into an AVI or MKV and then use VirtualDubMod or MKVMergeGUI to split it, then extract the audio again.....

Often I've longed for a program which makes doing the silly things nice and easy..... inserting silence into any audio file or removing bits of it without re-encoding, being able to re-encode multiple audio files as a single encode, or being able to join files without re-encoding etc. Just some thoughts.....

[ozok]

@hello_hello
I guess streching can be done using SoX, I didn't make a search yet but I believe you can change length of audio with it (speedup/speeddown). I can add an option to change it for each file, percentage based. A few users have requested this before but I missed it by the looks of it.
I don't use avs or DirectShow. I guess there was a tool like avs2yuv for avs to wav or to raw audio. I can use it if you think it is worth and said tool is supported or works well in most cases.
You are right about mp3 encoding. It must mixdown channels first if source has >2 channels. It will be done like this in the future.
I guess you can disable animations by disabling skins. But I'll disable them anyways, they don't look fine if you ask me and sometimes they slow down TAC if your system is under pressure.
I thought trimming worked when you selected "Extract Audio" but it looks like it doesn't work. It shouldn't be a problem since both extracting/copying and trimming is done by ffmpeg. "Extract Audio" doesn't touch audio (ffmpeg -acodec copy is used).
One user requested delay function before. What they really asked was that TAC would extract delay value from file name and apply it. Again I guess SoX can handle that. Adding silence or removing a part from audio was supported by SoX, I remember reading in its manual.
Currently when I work on TAC, I focus on proper tagging and cue sheet support. Tagging is almost done; if I'm not mistaken all of the encoders now have tagging support. Cue sheets are coming on well too but I don't have much samples and support at the moment is somewhat limited (no section support etc.).

Thanks for the suggestions and comments.

[ozok]

When you say time-strech, do you mean speeding/slowing audio so that it's duration changes? If so, I think "speed" command from SoX does exactly what you want. Setting speed to 200% makes duration half.

[ozok]

I've made some progress on some of the thing I wanted to implement. I need them to be tested before a release so I'm sharing TAC.exe+TTagger.exe so you can do some testing. Overwrite old exes if you want to try them.
-Tagging is improved. Monkey's audio, TAK, TTA now has tagging support. Also number of tag fileds for aac encoder has increased. It is done by a tagging tool I'm writing called TTagger. I'll release it as a seperate tool and provide information about how to use it because it has a strange way of doing things.
-Cue sheet support. It is limited and might have bugs because I don't have much samples. If you encounter problemsi please feel free to send samples as there is still room for improvement.
-Playback speed change option. You can change playback speed of audio files. This will effect the duration of output file and it is applied after trimming.
-Trimming works for "Extract audio" too now.
https://dl.dropboxusercontent.com/u/9617171/TAudioConverter_32_portable.7z
https://dl.dropboxusercontent.com/u/9617171/TAudioConverter_64_portable.7z

[hello_hello]

Thanks for the response. I haven't had a chance to play with the new version (the real world gets in the way sometimes), but I'll download them and try them out in the next day or so. Cheers!

[ozok]

@hello_hello, did you have a chance to test these exes?

[hello_hello]

Sorry..... I forgot.

I know I won't be able to for a day or so now as I've got to work for a change, but I promise I will sometime over the weekend.

[ozok]

@hello_hello Okay, I'll wait.
BTW, I fixed unicode tags not being written. You can download ttagger.exe from here if you downloaded archives from links above https://dl.dropboxusercontent.com/u/9617171/TTagger.7z
I also moved source code to https://bitbucket.org/ozok/taudioconverter-audio-converter.
Silly of me, you'll need TAC.exe to get unicode tags working:
https://dl.dropboxusercontent.com/u/9617171/TAudioConverter_32_portable.7z
https://dl.dropboxusercontent.com/u/9617171/TAudioConverter_64_portable.7z

[socrates]

-y -i "E:\XXXXXX.dts" -threads 0 -vn -acodec pcm_s24le -f wav "C:\Temp\XXXXXXXXX.wav"
-y -i "C:\Temp\XXXXXXXXX.wav" -b:a 448000 -acodec ac3 "E:\TAC\\XXXXXXXXX.ac3"

ffmpeg version N-53868-gc51654f Copyright (c) 2000-2013 the FFmpeg developers
built on Jun 6 2013 03:00:45 with gcc 4.7.3 (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-libcaca --enable-libfreetype --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-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib
libavutil 52. 35.100 / 52. 35.100
libavcodec 55. 15.100 / 55. 15.100
libavformat 55. 8.102 / 55. 8.102
libavdevice 55. 2.100 / 55. 2.100
libavfilter 3. 75.101 / 3. 75.101
libswscale 2. 3.100 / 2. 3.100
libswresample 0. 17.102 / 0. 17.102
libpostproc 52. 3.100 / 52. 3.100
[dts @ 0402eaa0] max_analyze_duration 5000000 reached at 5002667 microseconds
[dts @ 0402eaa0] Estimating duration from bitrate, this may be inaccurate
Input #0, dts, from 'E:\XXXXXXXXX.dts':
Duration: 01:40:54.82, start: 0.000000, bitrate: 1535 kb/s
Stream #0:0: Audio: dts (DTS), 48000 Hz, 5.1(side), fltp, 1536 kb/s
Output #0, wav, to 'C:\Temp\XXXXXXXXX.wav':
Metadata:
ISFT : Lavf55.8.102
Stream #0:0: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, 5.1(side), s32, 6912 kb/s
Stream mapping:
Stream #0:0 -> #0:0 (dca -> pcm_s24le)
Press [q] to stop, [?] for help
size= 5200164kB time=01:42:43.15 bitrate=6912.0kbits/s
video:0kB audio:5200164kB subtitle:0 global headers:0kB muxing overhead 0.000002%

ffmpeg version N-53868-gc51654f Copyright (c) 2000-2013 the FFmpeg developers
built on Jun 6 2013 03:00:45 with gcc 4.7.3 (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-libcaca --enable-libfreetype --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-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib
libavutil 52. 35.100 / 52. 35.100
libavcodec 55. 15.100 / 55. 15.100
libavformat 55. 8.102 / 55. 8.102
libavdevice 55. 2.100 / 55. 2.100
libavfilter 3. 75.101 / 3. 75.101
libswscale 2. 3.100 / 2. 3.100
libswresample 0. 17.102 / 0. 17.102
libpostproc 52. 3.100 / 52. 3.100
[wav @ 026b79c0] max_analyze_duration 5000000 reached at 5003458 microseconds
Input #0, wav, from 'C:\Temp\XXXXXXXXX.wav':
Metadata:
encoder : Lavf55.8.102
Duration: 00:19:52.13, bitrate: 35734 kb/s
Stream #0:0: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, 5.1(side), s32, 6912 kb/s
Output #0, ac3, to 'E:\TAC\\XXXXXXXXX.ac3':
Metadata:
encoder : Lavf55.8.102
Stream #0:0: Audio: ac3, 48000 Hz, 5.1(side), fltp, 448 kb/s
Stream mapping:
Stream #0:0 -> #0:0 (pcm_s24le -> ac3)
Press [q] to stop, [?] for help
Multiple frames in a packet from stream 0 448.0kbits/s
[pcm_s24le @ 026bb660] Invalid PCM packet, data has size 14 but at least a size of 18 was expected
Error while decoding stream #0:0: Invalid data found when processing input
size= 65196kB time=00:19:52.15 bitrate= 448.0kbits/s
video:0kB audio:65196kB subtitle:0 global headers:0kB muxing overhead 0.000000%

[ozok]

Did you try setting bit depth to 16bit?
BTW you can do small tests before full encode by trimming.

[socrates]

Thanks ozok, your suggestion worked.

I want to know why the 24bit depth didn't work?

[ozok]

I don't know why. Probably ac3 encoder in ffmpeg expects 16bit input.

[socrates]

New conversion error- This time tried with depth 16bit. File size is trimmed prematurely


-y -i "E:\Amen.mkv" -vn -acodec copy -map 0:1 "E:\TAC\\Amen.m4a"

ffmpeg version N-53868-gc51654f Copyright (c) 2000-2013 the FFmpeg developers
built on Jun 6 2013 03:00:45 with gcc 4.7.3 (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-libcaca --enable-libfreetype --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-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib
libavutil 52. 35.100 / 52. 35.100
libavcodec 55. 15.100 / 55. 15.100
libavformat 55. 8.102 / 55. 8.102
libavdevice 55. 2.100 / 55. 2.100
libavfilter 3. 75.101 / 3. 75.101
libswscale 2. 3.100 / 2. 3.100
libswresample 0. 17.102 / 0. 17.102
libpostproc 52. 3.100 / 52. 3.100
Input #0, matroska,webm, from 'E:\Amen.mkv':
Metadata:
creation_time : 2013-08-02 12:51:37
Duration: 02:37:45.79, start: 0.000000, bitrate: 1246 kb/s
Stream #0:0: Video: h264 (High), yuv420p, 1280x544 [SAR 1:1 DAR 40:17], 23.98 fps, 23.98 tbr, 1k tbn, 47.95 tbc (default)
Stream #0:1: Audio: aac, 48000 Hz, 5.1, fltp (default)
Output #0, ipod, to 'E:\TAC\\Amen.m4a':
Metadata:
encoder : Lavf55.8.102
Stream #0:0: Audio: aac (mp4a / 0x6134706D), 48000 Hz, 5.1 (default)
Stream mapping:
Stream #0:1 -> #0:0 (copy)
Press [q] to stop, [?] for help
size= 544609kB time=02:37:39.12 bitrate= 471.7kbits/s
video:0kB audio:541431kB subtitle:0 global headers:0kB muxing overhead 0.586863%

-y -i "E:\TAC\Amen.m4a" -threads 0 -vn -acodec pcm_s16le -f wav "C:\Temp\Amen.wav"
-y -i "C:\Temp\Amen.wav" -b:a 448000 -acodec ac3 "E:\TAC\\Amen.ac3"

ffmpeg version N-53868-gc51654f Copyright (c) 2000-2013 the FFmpeg developers
built on Jun 6 2013 03:00:45 with gcc 4.7.3 (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-libcaca --enable-libfreetype --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-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib
libavutil 52. 35.100 / 52. 35.100
libavcodec 55. 15.100 / 55. 15.100
libavformat 55. 8.102 / 55. 8.102
libavdevice 55. 2.100 / 55. 2.100
libavfilter 3. 75.101 / 3. 75.101
libswscale 2. 3.100 / 2. 3.100
libswresample 0. 17.102 / 0. 17.102
libpostproc 52. 3.100 / 52. 3.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'E:\TAC\Amen.m4a':
Metadata:
major_brand : M4A
minor_version : 512
compatible_brands: isomiso2
encoder : Lavf55.8.102
Duration: 02:37:39.12, start: 0.009000, bitrate: 471 kb/s
Stream #0:0(und): Audio: aac (mp4a / 0x6134706D), 48000 Hz, 5.1, fltp, 468 kb/s
Metadata:
handler_name : SoundHandler
Output #0, wav, to 'C:\Temp\Amen.wav':
Metadata:
major_brand : M4A
minor_version : 512
compatible_brands: isomiso2
ISFT : Lavf55.8.102
Stream #0:0(und): Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, 5.1, s16, 4608 kb/s
Metadata:
handler_name : SoundHandler
Stream mapping:
Stream #0:0 -> #0:0 (aac -> pcm_s16le)
Press [q] to stop, [?] for help
size= 5320752kB time=02:37:39.11 bitrate=4608.0kbits/s
video:0kB audio:5320752kB subtitle:0 global headers:0kB muxing overhead 0.000002%

ffmpeg version N-53868-gc51654f Copyright (c) 2000-2013 the FFmpeg developers
built on Jun 6 2013 03:00:45 with gcc 4.7.3 (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-libcaca --enable-libfreetype --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-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib
libavutil 52. 35.100 / 52. 35.100
libavcodec 55. 15.100 / 55. 15.100
libavformat 55. 8.102 / 55. 8.102
libavdevice 55. 2.100 / 55. 2.100
libavfilter 3. 75.101 / 3. 75.101
libswscale 2. 3.100 / 2. 3.100
libswresample 0. 17.102 / 0. 17.102
libpostproc 52. 3.100 / 52. 3.100
[wav @ 02597920] max_analyze_duration 5000000 reached at 5001333 microseconds
Input #0, wav, from 'C:\Temp\Amen.wav':
Metadata:
encoder : Lavf55.8.102
Duration: 00:33:22.57, bitrate: 21765 kb/s
Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, 5.1, s16, 4608 kb/s
Output #0, ac3, to 'E:\TAC\\Amen.ac3':
Metadata:
encoder : Lavf55.8.102
Stream #0:0: Audio: ac3, 48000 Hz, 5.1, fltp, 448 kb/s
Stream mapping:
Stream #0:0 -> #0:0 (pcm_s16le -> ac3)
Press [q] to stop, [?] for help
Multiple frames in a packet from stream 0 448.0kbits/s
[pcm_s16le @ 0259b680] Invalid PCM packet, data has size 8 but at least a size of 12 was expected
Error while decoding stream #0:0: Invalid data found when processing input
size= 109517kB time=00:33:22.58 bitrate= 448.0kbits/s
video:0kB audio:109517kB subtitle:0 global headers:0kB muxing overhead 0.000000%

[ozok]

Can you encode to flac or some other lossless codec?

[ozok]

Beta build 2609 changes:

added: TTagger can now write tags to Vorbis and Opus
added: Default "Disc No" and "Disc Count" tag values are 1
added: If encoder exits with an exit code other than 0, it'll be added to log
added: Re-written cue sheet parser
added: An option to go to bug report page in bitbucket project page
added: GUID values will be used instead of numbers for temp files
added: Exit codes to TTagger
added: TTagger will now report tag writing outcome properly
added: Min/Max bitrate options for Ogg Vorbis
added: A log showing compression percentages
added: Overwrite option will be shown in summary list
added: Extra columns in file list and an option to hide/show them
added: Progress list item adding is done according to file overwrite option
added: Trimming is now millisecond based
added: New progress bars that work better when skinning is disabled
added: Info will be added to log if file's length was shorter/longer than threshold
added: Custom tag usage will be shown in main window summary
added: Pressing esc key will close windows
added: Default skin is "AutumnSky"
fixed: File check failed if file doesn't exist (huh?)
fixed: Artwork image size wasn't written for some tag types
fixed: A bug that caused wrong encoder to be selected for some cases
fixed: Renaming was done twice if encoding was successful
fixed: "Overwrite if exists" didn't work because older files were not deleted
fixed: Clicking on a cue sheet item in file list would show full file duration
fixed: Cue sheet items in progress list wasn't shown properly
updated: all tag libraries

There might be things I forgot to add here, you may refer to changelog.txt for further info.


Downloads: https://bitbucket.org/ozok/taudioconverter-audio-converter/downloads