Today's GOTD is Audiozilla... I personally don't need it, but for those that do I'd recommend trying the free LameXP [portable version also avail.] 1st, partly because of Audiozilla's installation impact, & partly because LameXP's done well for me. http://www.videohelp.com/tools/LameXP
I also wanted to take a moment to warn about something that's often ignored when talking about or doing audio conversions -- quality loss when you change chiefly the sample rate, but also the bit depth of an audio file. Further down I very briefly note how I deal with AC3 conversions to .wav or .w64 so they can be optionally edited/trimmed & re-encoded properly at usually DVD-spec bit rate. It took me some time to come up with a solution that didn't automatically change the levels etc. -- not only do the available apps I've tried change those levels, they also tend to do a poorer job of it than I can do elsewhere, & don't allow for any adjustment or fine tuning. I don't know if the part on AC3 conversion will interest anyone -- I included it mostly because it took me so long to come up with the method, & if I save one person that hassle it will have been worth it. :)
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There's a somewhat hidden problem with audio file conversions that will cause quality loss when you're converting or changing the sampling rate or frequency. Yes, if you're going from 44.1 to 22 you expect the resolution is going to be cut in 1/2, & like converting from .wav to MP3 itself, you expect to lose some quality. But what happens when you change from 48 to 44.1, or 1/2 that to the common 22? The answer is you'll lose more quality than you'd expect. The ideal solution is to avoid that sort of conversion when it's at all possible. Consider that most sound cards or chips operate at 44.1 or 48 internally [most work at 48 nowadays I think], so you can have the same problems with hardware, e.g. with recording, with a hardware [DSP] effect, or with playback. You also want to avoid [at near all costs] jumping between sample rates, e.g. you don't want your audio software or hardware converting a file at 48,000 Hz to 44,100 Hz internally every step of the way as you're working with that audio file -- then those smaller quality losses can really add up. [Note: you can also wind up with errors causing quality loss when you convert from one bit depth to another, but this isn't nearly as common because most of the time non-pros are going to only be working with 16 bit audio.]
Analog audio, what you'd record on tape, is continuous sound -- when it's digitized, snapshots of that same sound are taken most commonly at 44,100, 48,000, or recently 96,000 Hz [times a second]. If you convert audio sampled at 44,100 to 22,050, 1/2 of those snapshots are thrown away, but there's no common denominator between 44,100 [or 22,050] & 48,000 -- there's no easy "divide (or multiply) by this number" for those conversions. Open an audio file that's 48,000 Hz in an editor & zoom all the way in on the waveform, & if the editor goes that far, you'll see what the audio looks like at 48,000 points for every second... What does the audio look or sound like between those 48,000 points? We don't know, because there's no data. If you took that second worth of audio & marked out 44,100 points, most of them would fall where there isn't any data. To go from 48 to 44.1 [&/or the reverse] requires a bit of guesswork, & some apps &/or hardware can guess better than others. Guessing on top of guessing can lead small errors to become large ones, so when you're unsure where a audio file will be played, consider leaving it as is & letting the player hardware up or downsample it.
If/when you're reducing the bit depth [i.e. the amount of data recorded/stored], this article from the Audacity wiki might help [ http://wiki.audacityteam.org/wiki/Dither ], but as above I don't know that all that many people are going to be working with 24 bit audio -- I think most will stick with 16 bit start to finish.
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Converting AC3 &/or DTS audio can be more of a problem because rather than just dealing with stored audio data, you're also working with data on how loud each track should be -- loads of software tends to automatically alter those levels as well as the dynamic range. The dynamic range is the difference between the maximum & minimum volume levels, & it's commonly altered [usually reduced] because with a high range you have to turn the volume up to hear the quietest parts, which means the loud stuff will wake the neighbors -- or damage your hearing if you're wearing headphones. When I record something broadcast via digital OTA [Over The Air], the AC3 is usually non-standard, often a bit mangled compared to what you'd put on a DVD or Blu-Ray disc, plus you can't really edit AC3 without converting to wav format anyway, so I always run through this process or routine to get as close to the original .wav files as I can...
The tools I use are Avisynth [ http://www.videohelp.com/tools/Avisynth or avisynth.org ], the Avisynth nicaudio plugin [ avisynth.org or http://nicaudio.codeplex.com/ ], Virtualdub [ http://www.videohelp.com/tools/Virtualdub or virtualdub.org ]. I use the following script [included 1 for DTS to show the option] in a plain text file named with the .avs file name extension, then open that .avs file in Virtualdub -- Amplify is optional, used as needed to prevent clipping in the output file [instead of .85 you can use .95, .60, whatever you need]. This page shows how you can detect clipping in Audacity for an example [ http://manual.audacityteam.org/man/Audacity_Waveform ]. For multi-channel I save to .w64.
LoadPlugin("[path to] \AviSynth 2.5\plugins\NicAudio.dll")
NicDTSSource("filename.dts", DRC=1)
Amplify(0.85)
Soundout()
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OR
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LoadPlugin("[path to] \AviSynth 2.5\plugins\NicAudio.dll")
NicAC3Source("filename.ac3", DRC=1)
Amplify(0.85)
Soundout()
