<li><a href="#12d077d5">How can I force LameXP to create ID3 version 2 (ID3v2) tags?</a>
<li><a href="#d83ab3c6">Why does LameXP use LAME v3.99 rather than v3.98?</a><br>
<li><a href="#4213adbc">Can LameXP be used to convert/extract tracks from an Audio CD?</a>
+<li><a href="#434f2578">Why is the maximum normalization level limited to -0.5 dB?</a>
<li><a href="#e7c9ae2b">Why do I get the error 'Executable doesn't support Windows compatibility mode' on startup?</a>
<li><a href="#328b0a18">Why do I get the error 'Executable requires Windows XP or later' on startup?</a>
<li><a href="#3ffe490e">Why do I get the error 'The procedure entry point De/EncodePointer could not be located' on startup?</a>
<a name="29065b86"></a><b>What output formats (encoders) does LameXP support?</b><br>
<br>
Currently the following output formats are supported by LameXP:<br><ul>
-<li>MPEG Audio-Layer III (MP3), using the LAME encoder [built-in]
-<li>Ogg Vorbis, using the OggEnc2/libvorbis encoder [built-in]
-<li>Advanced Audio Coding (AAC), using Nero AAC encoder [available as separate download]
-<li>Free Lossless Audio Codec (FLAC) [built-in]
+<li>MPEG Audio-Layer III (MP3), using the <a href="http://lamexp.sourceforge.net/" target="_blank">LAME</a> encoder [built-in]
+<li>Ogg Vorbis, using the <a href="http://www.xiph.org/vorbis/" target="_blank">OggEnc2/libvorbis</a> encoder with <a href="http://www.geocities.jp/aoyoume/aotuv/" target="_blank">aoTuV</a> [built-in]
+<li>Advanced Audio Coding (AAC), using <a href="http://www.audiocoding.com/nero_aacenc.html" target="_blank">Nero AAC</a> encoder [separate download!]
+<li>ATSC A/52 (aka "AC-3"), using the <a href="http://aften.sourceforge.net/" target="_blank">Aften</a> encoder [built-in]
+<li>Free Lossless Audio Codec (<a href="http://flac.sourceforge.net/" target="_blank">FLAC</a>) [built-in]
<li>Uncompressed PCM / Waveform Audio File (WAV/RIFF)</ul>
<br><br>
<br><br>
+<a name="434f2578"></a><b>Why is the maximum normalization level limited to -0.5 dB?</b><br>
+<br>
+
+When an analogue [audio] signal is converted to the digital domain, the signal is sampled at a fixed rate<br>
+(e.g. 44100 samples per second) and each "sample" value is stored with a fixed number of bits (e.g. 16 or 24<br>
+bits per sample). Consequently [uncompressed] digital audio is represented as a sequence of binary sample<br>
+values. The range of possible sample values is determined by the word size ("bits per sample"). For example<br>
+with a word size of 16 bit, the minimum value is −32768 and the maximum value is 32767 - assuming the values<br>
+are signed. The range of the sample values corresponds to the voltage range of the electrical input signal.<br>
+The maximum digital sample value (i.e. 32767 at 16-Bit) often is referred to as 0dBFS (0dB "full scale").<br>
+<br>
+Performing a Normalization in the digital domain seems straightforward: We simply multiply all sample values<br>
+with the same factor. And we choose this factor in such a way that the highest sample value(s) in the track<br>
+will become exactly 0dBFS after the normalization has been performed. However one needs to be aware that when<br>
+playing back the digital audio track, it needs to be converted back to an analogue signal. The D/A converter<br>
+will convert each sample from its binary representation to the corresponding voltage. Then a "reconstruction"<br>
+filter will be applied in order to recover a continuous signal from these individual voltages. And for the<br>
+reconstructed analogue signal it is possible to have voltages that are higher than the highest digital sample<br>
+in the audio track! This is illustrated in the following image (samples are represented as tiny squares):<br>
+<br>
+<a target="_blank" href="http://img3.imageshack.us/img3/9153/sine0dbfs.png"><img src="http://img3.imageshack.us/img3/9153/sine0dbfs.th.png" border="0"></a><br>
+<br>
+Consequently normalizing the sample values to 0dBFS is NOT a very good idea, as this may very well result in<br>
+a reconstructed analogue signal which exceeds(!) 0dBFS. And, as the analogue parts of the playback equipment<br>
+generally are NOT prepared for +0dBFS voltages, this may cause problems, such as annoying distortions!<br>
+<br>
+The help document of a well-known audio editing software contains the following advice:<br>
+<blockquote>If you’re planning to put normalized audio on CD, you might want to normalize the waveforms to<br>
+no more than 96% [-0.36 dB] as some audio compact disc players have problems accurately reproducing<br>
+bits that have been processed to 100% (maximum) amplitude [0dBFS].</blockquote>
+For details please refer to the following article:<br><ul>
+<li><a href="http://www.audioholics.com/education/audio-formats-technology/issues-with-0dbfs-levels-on-digital-audio-playback-systems" target="_blank">http://www.audioholics.com/...issues-with-0dbfs-levels-on-digital-audio-playback-systems</a></ul>
+
+<br><br>
+
<a name="e7c9ae2b"></a><b>Why do I get the error 'Executable doesn't support Windows compatibility mode' on startup?</b><br>
<br>
LameXP was designed to run on all supported platforms natively (except for Linux/Wine). If you see this error<br>
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