Manual.txt for Version 2.00 of ISO/MPEG Audio Layer 3 software only 
encoder/decoder for PCs.

1. ENCODER V2.00
   =============

 L3ENC.EXE is an ISO/MPEG Layer-3 software only encoder. It takes 
 PCM audio data files as input and delivers Layer-3 coded bit stream 
 files as output. Several options can be selected via command line 
 switches. Usage:
 
   L3ENC <audio_data> <bitstream> [-switch1 [-switch2 [...]]]

 L3ENC.EXE should work on any IBM PC compatible computer with a 386 type 
 CPU or better with external or built in 387 hardware floating point 
 support. A 486DX CPU or better is recommended.

 L3ENC uses the DOS extender GO32.EXE.

 PLEASE NOTE: Non-registered users may use the encoder only with the
 following options:

 bitrate    sampling rate     mode      	format
 256 kbps   44.1  kHz         stereo (mode 0)   MPEG-1
 128 kbps   44.1  kHz         stereo (mode 0)   MPEG-1
  64 kbps   22.05 kHz         stereo (mode 1)   MPEG-2
  32 kbps   22.05 kHz         mono (mode 4)     MPEG-2
 For non-registered users, ancillary data processing is not supported.
 
1.1 PCM audio input file
 The first command line argument specifies the name for the PCM audio
 data file. Version 2.00 of the encoder accepts either raw PCM audio 
 data files, PCM audio data files in RIFF/WAVE format as used by
 Microsoft Windows, PCM audio data files in the sun .au or PCM audio
 data files in the Apple AIFF format.
 The samples must be 16 bit signed integer values.
   
 for raw PCM audio data:
    By default the input file is assumed to contain raw PCM audio data.
    Stereo audio data is input in interleaved format, the first channel
    beeing the left channel.
      <sample #1 channel #1> <s. #1 ch. #2> <s.#2 ch.#1> <s.#2 ch.#2> ...
    Mono audio data has the format
      <sample #1> <sample #2> <sample #3> ....
    Whether the input file is treated as mono or stereo audio data is set
    by the encoding mode parameter (1.3). Default is stereo.

1.2 bitstream output file
 The second command line argument specifies the name for the bitstream 
 output file. The extension of the file name should be .mp3.
 The format of the bit stream is as defined in the
 ISO/MPEG publications IS11172-3 (MPEG-1) and IS13818-3 (MPEG-2).

1.3 encoding mode
 Depending on the setting of the '-mod' switch, the encoder will treat the 
 two input channels as:
	-mod 0  stereo (ms stereo),
	-mod 1  stereo (intensity stereo),
	-mod 2  dual channel
	-mod 3  stereo input -> mono output (L+R)/2 ("downmix")
	-mod 4  mono input

 For bitrates <= 96 kbps, the default is intensity stereo (-mod 1). For
 bitrates >= 112 kbps, the default is ms-stereo (-mod 0). For
 more details about encoding modes, please refer to section 1.11 'Encoding
 Recommendations'
 For stereo, the first channel is the left channel. The second channel is 
 the right channel.
 If input files in SND/WAVE/AIFF format are used, the number of channels
 is detected.

1.4 sampling rate
 Version 2.00 of the encoder can use the following sampling rates:
  o 16000 Hz
  o 22050 Hz
  o 24000 Hz
  o 32000 Hz
  o 44100 Hz
  o 48000 Hz

1.5 effective sampling rate
 For registered users the following effective frequencies are supported:  
  o 16000 Hz
  o 22050 Hz
  o 24000 Hz
  o 32000 Hz
  o 44100 Hz
  o 48000 Hz
 If you apply an effective sampling rate using the -esr switch, a downsampling
 from the sampling rate to that effective sampling rate is done.
 Currently only downsampling /2 and /3 work.

1.6 bitrate
 The bitrate of the bit stream 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 128000' and 'stereo', both 
 channels will be stuffed into one bit stream of 128000 bits/second. 
 Valid bit rates are:
  o   8000 bit/s
  o  16000 bit/s
  o  24000 bit/s
  o  32000 bit/s
  o  40000 bit/s
  o  48000 bit/s
  o  56000 bit/s
  o  64000 bit/s
  o  80000 bit/s
  o  96000 bit/s
  o 112000 bit/s
  o 128000 bit/s
  o 144000 bit/s
  o 160000 bit/s
  o 192000 bit/s
  o 224000 bit/s
  o 256000 bit/s
  o 320000 bit/s

 The default bitrate is 128000 bits/sec.

1.7 crc check
 If '-crc' is asserted, ISO/MPEG1 crc checking is enabled. Without the 'crc' 
 switch, crc checking is disabled.

1.8 swap low and high byte of input samples
 If the '-tfs' option is specified, the low and high bytes of each audio
 data input sample are swapped. Use '-tfs' if you move your PCM audio data
 from little endian to big endian machines (or vice versa).
 This will only work if the input signal is a pcm file.

1.9 ancillary data
  If the '-anc <filename> <rate>' option is specified, in the bitstream the
  named file is inserted as ancillary data. The rate is in bits/frame.

1.10 examples of switch settings
    L3ENC infile.pcm out.mp3 -br 112000 -cr
    L3ENC c:\music\pcm\newage.pcm c:\bitstr\l3\newage.mp3 -mod 2 -br 64000
    L3ENC pop.wav pop.mp3 -esr 22050 -br 96000

1.11 Encoding Recommendations
 Depending on the desired bitrate, the encoding process should be done
 with different parameters.
 'l3enc' supports two versions of Layer-3 bitstreams called MPEG-1 and MPEG-2. 
 The basic difference is the use of different sampling frequencies:
    MPEG-1 Layer 3       sampling frequencies 32, 44.1,  48 kHz
    MPEG-2 Layer 3       sampling frequencies 16, 22.05, 24 kHz
 MPEG-1 supports higher audio bandwidth and is therefore the best 
 choice for high quality audio coding at bitrates >= 96 kbps (stereo) 
 or >= 48 kbps (mono). For bitrates <= 64 kbps (stereo) or <=32 kbps (mono),
 MPEG-2 offers better sound quality compared to MPEG-1.

 l3enc supports downsampling of input files with MPEG-1 sampling frequencies
 to MPEG-2 sampling frequencies using the -esr switch (section 1.5). We
 recommend this for lower bitrates as mentioned above. Input files with
 sampling frequencies <= 24 kHz can of course only be encoded with 
 MPEG-2.

 For coding of stereo files with bitrates <=96 kbps, the use of intensity
 stereo is highly recommended. This is also the default configuration of
 the encoder. Note, however, that the use of intensity stereo will destroy
 information which is needed for sound processing schemes like 
 Dolby Surround. For bitrates >= 112 kbps, intensity stereo is not used by
 default. Since it may improve the audio quality for 112 and 128 kbps,
 you may try its use by overriding the default settings with the -mod switch
 (see section 1.3).

 The following table summarizes the recommendations.

 - Coding of Mono Input

 bitrate     coding recommendation  
---------------------------------------------------------
 <=40 kbps   MPEG-2
 >=48 kbps   MPEG-1
           
 - Coding of Stereo Input

 bitrate     coding           Intensity        Notes
             recommendation   recommendation
                              (also default)
 -------------------------------------------------------------------------
  <=64 kbps  MPEG-2           on
    96 kbps  MPEG-1           on
   112 kbps  MPEG-1           off              intensity may improve quality
   128 kbps  MPEG-1           off              intensity may improve quality
 >=192 kbps  MPEG-1           off


2. DECODER V2.10
   =============

 L3DEC.EXE is an ISO/MPEG Layer 3 software only decoder. It takes 
 Layer 3 bit stream files as input and delivers PCM audio data files 
 as output. A number of options can be selected via command line 
 switches. Usage:

	l3dec <bitstream> <audio_data> [-switch1 [switch2 [...]]]
 
 L3DEC.EXE should work on any IBM PC compatible computer with a 386 CPU 
 or better with external or built in 387 hardware floating point support.

 L3DEC uses the DOS extender GO32.EXE.

 If you specify no output file name and use the -sto option, the audio
 data is written to stdout. If you specify -sti, the decoder reads from stdin
 instead of the bitstream file.

2.1 bit stream input file
 The format of the bit stream input file must comply with ISO/IEC
 IS11172-3 or IS 13818-3.
 The decoder will process all valid MPEG1 Layer-3 bit stream data 
 without restrictions to bit rate or sampling frequency.
 It supports also MPEG2 Layer-3 low sampling frequencies.

2.2 PCM audio data output file
 Audio data is output as samples of 16 bit signed integer PCM data. 
 The default format is raw PCM data and can be either one channel or 
 two interleaved channels.
	format of one (mono) channel PCM audio data:
		<sample #1><sample #2>....
	format of two channel (stereo) PCM audio data:
		<spl.#1 ch.#1><spl.#1 ch.#2><sp.#2 ch.#1><spl.#2 ch.#2>...
 If one or two audio channels are used depends on the encoded information in 
 the bit stream. For stereo output data the first channel is the left 
 channel. Information about sampling frequency and number of used channels 
 is displayed at the beginning of the decoding process.

2.3 RIFF/WAVE format
 If selected by the '-wav' switch, audio data is output in RIFF/WAVE format 
 (*.WAV) as used by Microsoft Windows. The audio data itself is still 
 written as 16 bit PCM data as described in 2.2 but it is preceded by a 
 WAVE-header. The WAVE-Header contains information about the number of 
 channels (1 or 2), sampling frequency (32k/44.1k/48k) and used bits per 
 sample (16).

2.4 SND format
 If selected by the '-snd' switch, audio data files are output in
 the SND format used on SUN and NeXT-Workstations.

2.5 AIFF format
 If selected by the '-aif' switch, audio data files are output in
 the AIFF format.

2.6 AIFC format
 If selected by the '-aic' switch, audio data files are output in
 the AIFC format.

2.7 skip frames
 With the '-fb' option you can skip a number of frames in the bit stream 
 before the decoding starts. '-fb nnn' skips the first nnn frames. Each 
 frame contains 1152 samples of audio data. Depending on the used sampling 
 frequency, the duration of a frame is calculated as 24 msec (@ 48kHz), 26.1 
 msec (@ 44.1kHz) or 36 msec (@ 32kHz). 

2.8 decode only nnn frames
 If you want to decode only a certain number of frames, specify the '-fn' 
 option. '-fn xxx' will decode only xxx frames (see also 2.6).

2.9 search again after loss of synchronisation
 Normally the decoding process is stopped, if a loss of synchronisation is 
 detected, i.e. the synch information is incorrect. To enable decoding of 
 partially damaged bit stream files, you may assert the '-sa' option. In 
 this mode the decoding is not stopped and the file is searched for valid 
 synch information until end of file is encountered.

2.10 write audio data as ascii hex 24bit output file
 If the option '-h24 xxx' is specified an (additional) output file with 
 name 'xxx' is opened. PCM Audio data is output as 24 bit ascii hex values
 followed by carriage return and line feed. Accuracy of the output values
 is 24 bit compared to the 16 bits raw output mode. Files output in 
 'h24' format take four times the storage capacity necessary for raw 
 16bit output format.

2.11 ignore error messages
 If errors in the bit stream are detected, the decoding process is normally
 halted. If the '-ign' option is specified, the decoder tries to continue 
 with the decoding process.

2.11 accept free format bitstream
 If the '-ff' option is specified, a free format bitstream is accepted.

2.11 ancillary data
 If the bit stream contains ancillary data (user data integrated into
 the bit stream) the decoder can write this data into an ancillary 
 data file. Use the switch '-a file' to specify the filename for the
 ancillary data. The default alignment of ancillary data is byte
 aligned ('-aba'). You can also use the switch '-afh' for the FhG mode.
 In FhG-mode, ancillary data is framed, beginning with a Sync, a length
 byte and has a trailing checksum.

2.12 write to stdout
 If the '-sto' option is specified, the PCM data output is written to
 stdout.

2.13 read from stdin
 If the '-sti' option is specified, the bitstream input is read from
 stdin.


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