-
What output formats (encoders) does LameXP support?
- MPEG Audio-Layer III (MP3), using the LAME encoder [built-in] -
- Ogg Vorbis, using the OggEnc2/libvorbis encoder [built-in] -
- Advanced Audio Coding (AAC), using Nero AAC encoder [available as separate download] -
- Free Lossless Audio Codec (FLAC) [built-in] +
- MPEG Audio-Layer III (MP3), using the LAME encoder [built-in] +
- Ogg Vorbis, using the OggEnc2/libvorbis encoder with aoTuV [built-in] +
- Advanced Audio Coding (AAC), using Nero AAC encoder [separate download!] +
- ATSC A/52 (aka "AC-3"), using the Aften encoder [built-in] +
- Free Lossless Audio Codec (FLAC) [built-in]
- Uncompressed PCM / Waveform Audio File (WAV/RIFF)
Currently the following output formats are supported by LameXP:
-
-
@@ -465,6 +467,41 @@ the '.cda' files directly is destined to fail (as the '.cda' files do NOT actual
+Why is the maximum normalization level limited to -0.5 dB?
+
+ +When an analogue [audio] signal is converted to the digital domain, the signal is sampled at a fixed rate
+(e.g. 44100 samples per second) and each "sample" value is stored with a fixed number of bits (e.g. 16 or 24
+bits per sample). Consequently [uncompressed] digital audio is represented as a sequence of binary sample
+values. The range of possible sample values is determined by the word size ("bits per sample"). For example
+with a word size of 16 bit, the minimum value is −32768 and the maximum value is 32767 - assuming the values
+are signed. The range of the sample values corresponds to the voltage range of the electrical input signal.
+The maximum digital sample value (i.e. 32767 at 16-Bit) often is referred to as 0dBFS (0dB "full scale").
+
+Performing a Normalization in the digital domain seems straightforward: We simply multiply all sample values
+with the same factor. And we choose this factor in such a way that the highest sample value(s) in the track
+will become exactly 0dBFS after the normalization has been performed. However one needs to be aware that when
+playing back the digital audio track, it needs to be converted back to an analogue signal. The D/A converter
+will convert each sample from its binary representation to the corresponding voltage. Then a "reconstruction"
+filter will be applied in order to recover a continuous signal from these individual voltages. And for the
+reconstructed analogue signal it is possible to have voltages that are higher than the highest digital sample
+in the audio track! This is illustrated in the following image (samples are represented as tiny squares):
+
+
+
+Consequently normalizing the sample values to 0dBFS is NOT a very good idea, as this may very well result in
+a reconstructed analogue signal which exceeds(!) 0dBFS. And, as the analogue parts of the playback equipment
+generally are NOT prepared for +0dBFS voltages, this may cause problems, such as annoying distortions!
+
+The help document of a well-known audio editing software contains the following advice:
+
If you’re planning to put normalized audio on CD, you might want to normalize the waveforms to+For details please refer to the following article:
+no more than 96% [-0.36 dB] as some audio compact disc players have problems accurately reproducing
+bits that have been processed to 100% (maximum) amplitude [0dBFS].
+ +
+ Why do I get the error 'Executable doesn't support Windows compatibility mode' on startup?
LameXP was designed to run on all supported platforms natively (except for Linux/Wine). If you see this error
diff --git a/src/Config.h b/src/Config.h index 2c1d69d0..55339014 100644 --- a/src/Config.h +++ b/src/Config.h @@ -30,7 +30,7 @@ #define VER_LAMEXP_MINOR_LO 2 #define VER_LAMEXP_TYPE Alpha #define VER_LAMEXP_PATCH 13 -#define VER_LAMEXP_BUILD 492 +#define VER_LAMEXP_BUILD 493 /////////////////////////////////////////////////////////////////////////////// // Tools versions