I’m about to convert my MP3 Collection to OGG. But before I start I would like to automatically normalize each ogg file to a standard volume while or before the encoding process from mp3 to ogg. Some of my MP3 files are too quiet, the others are too loud… I would like a an Option like: Normalize to 98 % if Audio > 98 or < 98.
Is there any good Encoding Software, supporting a working normalization function while using the latest OGG Vorbis Encoder (Xiph.Org libVorbis I 20050304)?
Maybe I can give you a rather good option. I don’t know if it contain the latest OGG libraries, but you can use “Nero Wave editor”.
Simply open a mp3 file, then go on menu “Volume” and choose “Normalize”.
Then go on menu “File” and select “Save as”. On the window choose the output type (ogg in your case), set options for ogg, and then save.
Thank you for your advice… But I’m looking for kinda Batch-Software because I don’t want to open every file by myself… I would like a professional Tool, I can configure before, load the files I’d like to convert and then go to bed
I do because I tested it with many of my MP3 files and I can’t notice a difference! All my MP3’s are 192kbits or even higher. I can’t notice a difference between 192kbits MP3 and 96kbits OGG VBR or even 320kbits MP3 and 96kbits OGG!!!
Now I can convert all my MP3s (with different volumes) to one standard (latest OGG Library) without losing hearable Quality!!! And I save half the Space! Even if I should decide that i’ve made a mistake later… I can restore my MP3’s because i backed them up on DVD…
You really should give it a try! 96kbits VBR (NBR)!!! The files are so tiny, but the sound kicks ass!
Given that I can buy a new 200GB HD for $150 and TY DVD’s cost just 60c, and my car stereo, portable Mp3 player & DVD players play MP3, it’s not worth my time converting a 6MB MP3 to a 4MB OggV
Not that I’m disagreeing with the fact that OGGV has higher compression ratio/sounds better than MP3 from the same source.
I will however point out that while you can convert a 256kbps MP3 to a 96Kbps ogg, you are losing alot of quality. You need to convert to a comparable bitrate, aka 256Kbps->196Kbps. Also, Ogg is ALWAYS variable bitrate, which can be a bastard for A/V synchronisation…
This is a list of features in Audacity, the free audio editor. For more information on how to use these features, go to the help pages.
Audacity can record live audio through a microphone or mixer, or digitize recordings from cassette tapes, vinyl records, or minidiscs. With some sound cards, it can also capture streaming audio.
* Record from microphone, line input, or other sources.
* Dub over existing tracks to create multi-track recordings.
* Record up to 16 channels at once (requires multi-channel hardware).
* Level meters can monitor volume levels before, during, and after recording.
Import and Export
Import sound files, edit them, and combine them with other files or new recordings. Export your recordings in several common file formats.
* Import and export WAV, AIFF, AU, and Ogg Vorbis files.
* Import MPEG audio (including MP2 and MP3 files) with libmad.
* Export MP3s with the optional LAME encoder library.
* Create WAV or AIFF files suitable for burning to CD.
* Import and export all file formats supported by libsndfile.
* Open raw (headerless) audio files using the â€œImport Rawâ€ command.
* Note: Audacity does not currently support WMA, AAC, or most other proprietary or restricted file formats.
* Easy editing with Cut, Copy, Paste, and Delete.
* Use unlimited Undo (and Redo) to go back any number of steps.
* Very fast editing of large files.
* Edit and mix an unlimited number of tracks.
* Use the Drawing tool to alter individual sample points.
* Fade the volume up or down smoothly with the Envelope tool.
* Change the pitch without altering the tempo, or vice-versa.
* Remove static, hiss, hum, or other constant background noises.
* Alter frequencies with Equalization, FFT Filter, and Bass Boost effects.
* Adjust volumes with Compressor, Amplify, and Normalize effects.
* Other built-in effects include:
* Record and edit 16-bit, 24-bit, and 32-bit (floating point) samples.
* Record at up to 96 KHz.
* Sample rates and formats are converted using high-quality resampling and dithering.
* Mix tracks with different sample rates or formats, and Audacity will convert them automatically in realtime.
* Add new effects with LADSPA plugins.
* Audacity includes some sample plugins by Steve Harris.
* Load VST plugins for Windows and Mac, with the optional VST Enabler.
* Write new effects with the built-in Nyquist programming language.
* Spectrogram mode for visualizing frequencies.
* â€œPlot Spectrumâ€ command for detailed frequency analysis.
Free and Cross-Platform
* Licensed under the GNU General Public License (GPL).
* Runs on Mac OS X, Windows, and GNU/Linux.
since Audacity uses an external encoder for MP3, i believe it’s unable to edit in real-time meaning it needs to convert either to wav or it’s own format first…it only edits uncompressed files in real-time…
you sure there’s quality loss in MP3>WAV>MP3? i thought it’s only lossy->lossy that results in a loss of quality…
I’m a quality freak wid big speakers and big amp. MP3 at 192 is scratchy in the high range and the tweeters just output trash sometimes (depends on encoder too). Since space is not a problem, 256 or higher is my preference.
Any lossy compression format results in an approximate reconstruction of the original. A great lossy format results in an inaudible difference
Ogg is a great lossy format. Mp3 is a good lossy format.
Mp3->Wav->Normalised Wav->Mp3 =
Lossed info -> Changed info -> More lost info.
ie: an approximation of a normalised approximation.
Of course, the second compression will result in (hopefully) significantly less detail than the original compression.
Also keep in mind that compressing between different compressed formats means that they throw away different information (aka different harmonics / frequency cut-offs/ phase / amplitude differences / etc). Compressing a file in one format, then another means alot more info is thrown away, than compressing->recompressing to the same format.