mirror of
https://github.com/cookiengineer/audacity
synced 2025-05-04 09:39:42 +02:00
525 lines
12 KiB
XML
525 lines
12 KiB
XML
<?xml version="1.0" encoding="UTF-8"?>
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<!DOCTYPE refentry PUBLIC "-//OASIS//DTD DocBook XML V4.2//EN" "http://www.oasis-open.org/docbook/xml/4.2/docbookx.dtd">
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<refentry>
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<refmeta>
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<refentrytitle>twolame</refentrytitle>
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<manvolnum>1</manvolnum>
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</refmeta>
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<refnamediv>
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<refname>twolame</refname>
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<refpurpose>an optimised MPEG Audio Layer 2 (MP2) encoder</refpurpose>
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</refnamediv>
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<refsynopsisdiv>
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<simpara><emphasis>twolame</emphasis> [options] <infile> [outfile]</simpara>
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</refsynopsisdiv>
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<refsect1>
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<title>DESCRIPTION</title>
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<simpara>TwoLAME is an optimised MPEG Audio Layer 2 (MP2) encoder based on tooLAME by
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Mike Cheng, which in turn is based upon the ISO dist10 code and portions of
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LAME. Encoding is performed by the libtwolame library backend.</simpara>
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</refsect1>
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<refsect1>
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<title>OPTIONS</title>
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<refsect2>
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<title>Input File</title>
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<simpara>twolame uses libsndfile for reading the input sound file, so
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the input file can be in any format supported by libsndfile.
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To read raw PCM audio from STDIN, then use - as the input filename.</simpara>
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</refsect2>
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<refsect2>
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<title>Output File</title>
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<simpara>If no output filename is specified, then suffix of the input filename
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is automatically changed to .mp2. To write the encoded audio to STDOUT
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then use - as the output filename.</simpara>
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</refsect2>
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<refsect2>
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<title>Input Options</title>
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<variablelist>
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<varlistentry>
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<term>
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-r, --raw-input
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</term>
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<listitem>
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<simpara>
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Specifies that input is raw signed PCM audio.
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If audio is stereo, than audio samples are interleaved
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between the two channels.
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</simpara>
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</listitem>
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</varlistentry>
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<varlistentry>
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<term>
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-x, --byte-swap
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</term>
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<listitem>
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<simpara>
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Force byte-swapping of the input. Endian detection is performed
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automatically by libsndfile, so this option shouldn't
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normally be needed.
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</simpara>
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</listitem>
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</varlistentry>
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<varlistentry>
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<term>
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-s, --samplerate <int>
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</term>
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<listitem>
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<simpara>
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If inputting raw PCM sound, you must specify the sample rate of
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the audio in Hz.
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Valid sample rates: 16000, 22050, 24000, 32000, 44100, 48000Hz.
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Default sample rate is 44100Hz.
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</simpara>
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</listitem>
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</varlistentry>
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<varlistentry>
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<term>
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--samplesize <int>
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</term>
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<listitem>
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<simpara>
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Specifies the sample size (in bits) of the raw PCM audio.
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Valid sample sizes: 8, 16, 24, 32.
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Default sample size is 16-bit.
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</simpara>
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</listitem>
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</varlistentry>
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<varlistentry>
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<term>
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-N, --channels <int>
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</term>
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<listitem>
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<simpara>
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If inputting raw PCM sound, you must specify the number of channels
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in the input audio. Default number of channels is 2.
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</simpara>
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</listitem>
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</varlistentry>
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<varlistentry>
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<term>
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-g, --swap-channels
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</term>
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<listitem>
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<simpara>
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Swap the Left and Right channels of a stereo input file.
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</simpara>
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</listitem>
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</varlistentry>
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<varlistentry>
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<term>
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--scale <float>
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</term>
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<listitem>
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<simpara>
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Scale the input audio prior to encoding.
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All of the input audio is multiplied by specified value.
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Value between 0 and 1 will reduce the audio gain, and a value
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above 1 will increase the gain of the audio.
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</simpara>
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</listitem>
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</varlistentry>
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<varlistentry>
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<term>
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--scale-l <float>
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</term>
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<listitem>
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<simpara>
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Same as --scale, but only affects the left channel.
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</simpara>
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</listitem>
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</varlistentry>
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<varlistentry>
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<term>
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--scale-r <float>
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</term>
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<listitem>
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<simpara>
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Same as --scale, but only affects the right channel.
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</simpara>
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</listitem>
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</varlistentry>
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</variablelist>
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</refsect2>
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<refsect2>
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<title>Output Options</title>
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<variablelist>
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<varlistentry>
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<term>
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-m, --mode <char>
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</term>
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<listitem>
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<simpara>
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Choose the mode of the resulting audio. Default is auto.
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</simpara>
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<itemizedlist>
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<listitem>
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<simpara>
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"a" auto - choose mode automatically based on the input
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</simpara>
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</listitem>
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<listitem>
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<simpara>
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"s" stereo
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</simpara>
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</listitem>
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<listitem>
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<simpara>
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"d" dual channel
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</simpara>
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</listitem>
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<listitem>
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<simpara>
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"j" joint stereo
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</simpara>
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</listitem>
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<listitem>
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<simpara>
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"m" mono
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</simpara>
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</listitem>
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</itemizedlist>
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</listitem>
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</varlistentry>
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<varlistentry>
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<term>
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-a, --downmix
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</term>
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<listitem>
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<simpara>
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If the input file is stereo then, downmix the left and right
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input channels into a single mono channel.
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</simpara>
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</listitem>
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</varlistentry>
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<varlistentry>
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<term>
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-b, --bitrate <int>
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</term>
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<listitem>
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<simpara>
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Sets the total bitrate (in kbps) for the output file.
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The default bitrate depends on the number of
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input channels and samplerate.
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</simpara>
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<literallayout class="monospaced">------------------------------
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Sample Rate Mono Stereo
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------------------------------
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48000 96 192
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44100 96 192
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32000 80 160
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24000 48 96
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22050 48 96
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16000 32 64
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------------------------------</literallayout>
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</listitem>
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</varlistentry>
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<varlistentry>
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<term>
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-P, --psyc-mode <int>
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</term>
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<listitem>
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<simpara>
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Choose the psycho-acoustic model to use (-1 to 4).
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Model number -1 is turns off psycho-acoustic modelling and
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uses fixed default values instead.
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Please see the file <emphasis>psycho</emphasis> for a full description of
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each of the models available.
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Default model is 3.
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</simpara>
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</listitem>
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</varlistentry>
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<varlistentry>
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<term>
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-v, --vbr
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</term>
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<listitem>
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<simpara>
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Enable VBR mode. See <emphasis>vbr</emphasis> documentation file for details.
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Default VBR level is 5.0.
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</simpara>
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</listitem>
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</varlistentry>
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<varlistentry>
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<term>
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-v, --vbr-level <float>
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</term>
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<listitem>
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<simpara>
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Enable VBR mode and set quality level.
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The higher the number the better the quality.
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Maximum range is -50 to 50 but useful range is -10 to 10.
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See <emphasis>vbr</emphasis> documentation file for details.
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</simpara>
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</listitem>
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</varlistentry>
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<varlistentry>
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<term>
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-l, --ath <float>
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</term>
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<listitem>
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<simpara>
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Set the ATH level. Default level is 0.0.
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</simpara>
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</listitem>
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</varlistentry>
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<varlistentry>
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<term>
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-q, --quick <int>
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</term>
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<listitem>
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<simpara>
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Enable quick mode. Only re-calculate psycho-acoustic
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model every specified number of frames.
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</simpara>
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</listitem>
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</varlistentry>
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<varlistentry>
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<term>
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-S, --single-frame
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</term>
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<listitem>
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<simpara>
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Enables single frame mode: only a single frame of MPEG audio
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is output and then the program terminates.
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</simpara>
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</listitem>
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</varlistentry>
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</variablelist>
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</refsect2>
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<refsect2>
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<title>Miscellaneous Options</title>
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<variablelist>
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<varlistentry>
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<term>
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-c, --copyright
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</term>
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<listitem>
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<simpara>
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Turn on Copyright flag in output bitstream.
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</simpara>
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</listitem>
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</varlistentry>
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<varlistentry>
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<term>
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-o, --non-original
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</term>
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<listitem>
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<simpara>
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Turn off Original flag in output bitstream.
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</simpara>
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</listitem>
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</varlistentry>
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<varlistentry>
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<term>
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--original
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</term>
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<listitem>
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<simpara>
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Turn on Original flag in output bitstream.
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</simpara>
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</listitem>
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</varlistentry>
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<varlistentry>
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<term>
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-p, --protect
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</term>
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<listitem>
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<simpara>
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Enable CRC error protection in output bitstream.
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An extra 16-bit checksum is added to frames.
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</simpara>
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</listitem>
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</varlistentry>
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<varlistentry>
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<term>
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-d, --padding
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</term>
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<listitem>
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<simpara>
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Turn on padding in output bitstream.
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</simpara>
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</listitem>
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</varlistentry>
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<varlistentry>
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<term>
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-R, --reserve <int>
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</term>
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<listitem>
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<simpara>
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Reserve specified number of bits in the each from of the
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output bitstream.
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</simpara>
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</listitem>
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</varlistentry>
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<varlistentry>
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<term>
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-e, --deemphasis <char>
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</term>
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<listitem>
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<simpara>
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Set the de-emphasis type (n/c/5). Default is none.
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</simpara>
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</listitem>
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</varlistentry>
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<varlistentry>
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<term>
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-E, --energy
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</term>
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<listitem>
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<simpara>
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Turn on energy level extensions.
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</simpara>
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</listitem>
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</varlistentry>
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</variablelist>
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</refsect2>
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<refsect2>
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<title>Verbosity Options</title>
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<variablelist>
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<varlistentry>
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<term>
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-t, --talkativity <int>
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</term>
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<listitem>
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<simpara>
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Set the amount of information to be displayed on stderr (0 to 10).
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Default is 2.
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</simpara>
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</listitem>
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</varlistentry>
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<varlistentry>
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<term>
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--quiet
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</term>
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<listitem>
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<simpara>
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Don't send any messages to stderr, unless there is an error.
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(Same as --talkativity=0)
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</simpara>
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</listitem>
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</varlistentry>
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<varlistentry>
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<term>
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--brief
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</term>
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<listitem>
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<simpara>
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Only display a minimal number of messages while encoding.
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This setting is quieter than the default talkativity setting.
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(Same as --talkativity=1)
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</simpara>
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</listitem>
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</varlistentry>
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<varlistentry>
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<term>
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--verbose
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</term>
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<listitem>
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<simpara>
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Display an increased number of messages on stderr.
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This setting is useful to diagnose problems.
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(Same as --talkativity=4)
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</simpara>
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</listitem>
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</varlistentry>
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</variablelist>
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</refsect2>
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</refsect1>
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<refsect1>
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<title>Return Codes</title>
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<simpara>If encoding completes successfully, then twolame will return 0.
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However if encoding is not successful, then it will return one of the following codes.</simpara>
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<itemizedlist>
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<listitem>
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<simpara>
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1 (No encoding performed)
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</simpara>
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</listitem>
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<listitem>
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<simpara>
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2 (Error opening input file)
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</simpara>
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</listitem>
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<listitem>
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<simpara>
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4 (Error opening output file)
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</simpara>
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</listitem>
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<listitem>
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<simpara>
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6 (Error allocating memory)
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</simpara>
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</listitem>
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<listitem>
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<simpara>
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8 (Error in chosen encoding parameters)
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</simpara>
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</listitem>
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<listitem>
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<simpara>
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10 (Error reading input audio)
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</simpara>
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</listitem>
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<listitem>
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<simpara>
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12 (Error occured while encoding)
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</simpara>
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</listitem>
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<listitem>
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<simpara>
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14 (Error writing output audio)
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</simpara>
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</listitem>
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</itemizedlist>
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</refsect1>
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<refsect1>
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<title>EXAMPLES</title>
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<simpara>This will encode sound.wav to sound.mp2 using the default constant bitrate of 192 kbps
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and using the default psycho-acoustic model (model 3):</simpara>
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<literallayout class="monospaced">twolame sound.wav</literallayout>
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<simpara>Constant bitrate of 160kbps and joint stereo encoding, saved to file sound_160.mp2:</simpara>
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<literallayout class="monospaced">twolame -b 160 -m j sound.aiff sound_160.mp2</literallayout>
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<simpara>Encode sound.wav to newfile.mp2 using psycho-acoustic model 2 and encoding
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with variable bitrate:</simpara>
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<literallayout class="monospaced">twolame -P 2 -v sound.wav newfile.mp2</literallayout>
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<simpara>Same as example above, except that the negative value of the "-V" argument
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means that the lower bitrates will be favoured over the higher ones:</simpara>
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<literallayout class="monospaced">twolame -P 2 -V -5 sound.wav newfile.mp2</literallayout>
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<simpara>Resample audio file using sox and pipe straight through twolame:</simpara>
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<literallayout class="monospaced">sox sound_11025.aiff -t raw -r 16000 | twolame -r -s 16000 - - > out.mp2</literallayout>
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</refsect1>
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<refsect1>
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<title>AUTHORS</title>
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<simpara>The twolame frontend was (re)written by Nicholas J Humfrey.
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The libtwolame library is based on toolame by Mike Cheng.
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For a full list of authors, please see the AUTHORS file.</simpara>
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</refsect1>
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<refsect1>
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<title>RESOURCES</title>
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<simpara>TwoLAME web site: http://www.twolame.org/</simpara>
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</refsect1>
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<refsect1>
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<title>SEE ALSO</title>
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<simpara>lame(1), mpg123(1), madplay(1), sox(1)</simpara>
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</refsect1>
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<refsect1>
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<title>COPYING</title>
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<simpara>Copyright © 2004-2006 The TwoLAME Project. Free use of this software is
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granted under the terms of the GNU Lesser General Public License (LGPL).</simpara>
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</refsect1>
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</refentry>
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