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Subsections

• Samplerate
• Bitrate
• Stereo mode
• Encoding speed
• Input file specification
  • Input from file: filename
  • Piping data into the encoder
  • plain PCM data input
• Output file specification
  • Streaming data out of the encoder

Basics

Samplerate

The encoder changes the sample rate of your audio data to match it to the audio quality of the bitstream produced by the encoder. This process is called  downsampling.

Bitrate

The bitrate of the encoder is linked to the samplerate that the encoded file will have. Usually, the encoder will choose a samplerate that is suited best for encoding at that bitrate. You can override this samplerate using the -esr  switch (see section 2.2.1).

 The bitrate of the bitstream output is selected via the -br  switch. The bitrate is specified in bits/second. The bitrate is the total bitrate for all encoded channels, i.e. if you select -br  112000 and encode in stereo, both channels will be stuffed into one bitstream of 112000 bits/second.

The encoder supports bitrates of 8, 16, 18, 20, 24, 32, 40, 48, 56, 64, 96, 112, 128, 160, 192 and 256 kBit/s. While all of these can be used with mono signals, stereo works from 20 kBit/s on upwards.

Stereo mode

 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.

 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.

Table 2.1 gives you an overview which mode will be used for which bitrate.

Encoding speed

• 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).
• Mismatch between input and output sample rate. If your input and output sample rates differ, the encoder will have to run a resampling filter and thus will be slower. (Integer ratios between input and output sample rate perform slightly better than non-integer ratios, though).
• Time-domain bandlimiting  . The encoder needs to band-limit the signal to compress it. By default, the encoder will use a high-quality time domain filter to do this band-limiting. You can tell it to use a faster filter, possibly sacrificing some quality (see 2.2.2).
• Full huffman search and careful iteration. You can tell the encoder to try hard to do the best encoding possible, at the expense of a factor of up to three in running time (see 2.2.2).

Version V3.0 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.

Input file specification

Input from file: filename

-if  filename

will tell the encoder the filename it reads it input from. If the file is a RIFF/WAVE file or an AIFF/AIFC file, the encoder will automatically adapt to the sound file format. For other formats or plain PCM data, see below.

Piping data into the encoder

-sti 

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).

plain PCM data input

-iff  fileformat

This is a string containing name=value pairs, separated by blanks. Table 2.2 gives a reference which names and values are possible here. 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.

   

Table 2.2: input file format specification

Name	Value(s)	Explanation
sr	any	The rate the PCM signal is sampled at [Hz]
nc	1, 2	The number of channels in the signal
bps	8, 16, 24, 32	The number of bits per sample
little-endian		The signal is little-endian (Intel format)
big-endian		The signal is big-endian (Motorola format)

As an example, -iff "nc=2 sr=44100 bps=16" would be used to read a 44.1 kHz stereo file with 16 bits per sample while -iff "nc=1 sr=8000 bps=8" would tell the encoder that the data is mono, sampled at 8 kHz with 8 bits per sample.

Remember that this feature is only needed for input from files other than RIFF/WAV, AIFF and AIFC.

Output file specification

-of  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.

-l3wav 

tells the encoder to wrap the MPEG Layer-3 file into a Microsoft RIFF/WAVE file.

Streaming data out of the encoder  

-sto 

tells the encoder to write its output into stdout rather than in a file. This only works when the output is a raw Layer-3 bitstream (i.e. it does not work in conjunction with -l3wav ).