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Subsections
Sample rate is the rate at which the samples are read
from your sound card when you sample. Sample rate is directly linked to audio
bandwidth achievable: A sound file with a sample rate
of 8 kHz does not contain frequencies beyond 4 kHz. This means that
you should always use the highest sample rate that your sound card supports when
you sample a signal.
The main parameter controlling the sound quality is the bitrate
that the encoder runs at. In a nutshell, the higher the bitrate, the better the
quality.
The bitrate of the encoder is linked to the samplerate that the encoded file
will have.
The bitrate of the bitstream output is selected
via the -b switch. The bitrate is specified
in bits/second. The bitrate is the total bitrate for all encoded channels, i.e.
if you select -br 128000 and encode
in stereo, both channels will be stuffed into one bitstream of 128000 bits/second.
The encoder supports bitrates of 32, 40, 48, 56, 64, 96, 112, 128, 160, 192,
224, 256 and 320 kBit/s.
If encoding stereo, the bitrate of the encoder is linked to a stereo mode. MPEG
Layer-3 knows four modes for stereo encoding.
- dual channel
- (also known as dual mono) In this mode, the
encoder treats the two input channels as separate entities, assuming there
is no similarity between the channels. This would be appropriate if you e.g.
have a bilingual signal where one channel contains a german speaker and one
contains an english speaker. The current version of the encoder does not
support dual channel mode.
- stereo
- In this mode, like in dual channel above, the encoder makes no use of potentially
existing correlations between the two input channels. It can, however, negotiate
the bit demand between both channel, i.e. give one channel more bits if the
other contains silence.
- MS stereo
- In this mode, the encoder will make use of a correlation between both channels.
The signal will be matrixed into a sum (»mid«) and difference (»side«)
signal. For quasi-mono signals, this will give a significant gain in encoding
quality.
This mode does not destroy phase information like IS stereo (see below)
and thus can be used to encode DOLBY ProLogic surround signals.
- MS/IS stereo
- In this mode, high-frequency parts of the signal will be downmixed to mono
and transmitted with a direction information (which is basically a pan). This
mode (called »intensity stereo« will
loose phase information and should not be used for high-quality encoding.
Several factors influence the speed of the encoder.
They include:
- Number of channels in the output signal. If your output signal has only
one channel, the encoder will run at twice the speed compared to stereo encoding.
- Output sample rate. If the encoder produces a file at 22.050 kHz (that
is, a file that contains 22050 samples per second), it runs at twice the speed
compared to one that produces twice the number of samples per second (i.e.
produces a 44.1 kHz output).
- Quality of the psychoaccoustic model. You can tell the encoder to use a
dummy psychoaccoustic model, but the quality will be lower.
Version V3.04 of the encoder reaches realtime speed on a Pentium 166 when
encoding at 64 kBit/s, 22,050 kHz, stereo. On a SUN Sparc Ultra-1
(143 MHz) the performance is similar.
The encoder can read AIFF, AIFF-C, WAV/RIFF, and raw PCM data files. While the
first three only work from a file, plain PCM data can be fed into the encoder
via a pipe. This is useful for live encoding (also known as streaming).
- filename
- will tell the encoder the filename it reads it input from. If the file is
a RIFF/WAVE, or an AIFF/AIFC file, the encoder will automatically adapt to
the sound file format. For other formats or plain PCM data, see below.
- -
- tells the encoder to get its input from stdin rather than from a file. This
only works when the input is plain pcm data (see below).
If the encoder gets its input as plain pcm data (or
if it does not recognize the sound format by itself), you need to tell it all
about the structure of the PCM stream, i.e. the samplerate.
- -s samplerate
- The sample rate in the input file. The default is 44.1 kHz.
- -x
- The input file is endian inversed compared to your platform.
For stereo files, the encoder assumes that the PCM data is interleaved and that
the sample for the right channel follows that for the left channel.
As an example, -s 22050 would be used to read a 22.05 kHz.
Remember that this feature is only needed for input from files other than
RIFF/WAV, AIFF and AIFC.
The output is a plain Layer-3 stream wich can be piped into other applications.
This is useful for live streaming.
- filename
- tells the encoder the filename of the file that the encoder will write the
bitstream to. If the file does not exist, it is created; if it does exist,
it will be overwritten.
- -
- tells the encoder to write its output into stdout rather than in a file.