mirror of
https://github.com/openharmony/third_party_lame.git
synced 2026-07-01 05:53:58 -04:00
Updated USAGE: fixed typos, formatting, some grammar
This commit is contained in:
rjamorim
@@ -7,6 +7,7 @@ For more options, just type:
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Note: The HTML documentation is more complete than this text file.
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=======================================================================
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Constant Bitrate Examples:
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=======================================================================
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@@ -14,10 +15,11 @@ fixed bit rate jstereo 128 kbps encoding:
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% lame -b128 sample.wav sample.mp3
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Use highest quality mode, slowest:
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% lame -q0 -b128 sample.wav sample.mp3
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% lame -q0 -b128 sample.wav sample.mp3
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Use fast encode, low quality (no noise shaping)
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% lame -f -b128 sample.wav sample.mp3
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% lame -f -b128 sample.wav sample.mp3
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=======================================================================
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Variable Bitrate Examples:
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@@ -43,16 +45,16 @@ each frame on the measured quantization error relative to the
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estimated allowed masking. The value 0 is the highest quality, which
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creates bigger files, and the lowest is 9.999, which creates the smallest
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files. Decimal values can be specified, like: 4.51.
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The resulting filesizes depend on the input material. On typical music you
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can expect -V5 resulting in files averaging 132 kbps, -V2 averaging 200 kbps.
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The resulting filesizes depend on the input material. On typical music
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you can expect -V5 resulting in files averaging 132 kbps, -V2 averaging
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200 kbps.
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Variable Bitrate (VBR): (use -V n to adjust quality/filesize)
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% lame -V2 sample.wav sample.mp3
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=======================================================================
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LOW BITRATES
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Low Bitrates
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=======================================================================
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At lower bitrates, (like 24 kbps per channel), it is recommended that
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you use a 16 kHz sampling rate combined with lowpass filtering. LAME,
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@@ -63,26 +65,25 @@ and passband width (--resample, --lowpass and --lowpass-width options).
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=======================================================================
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STREAMING EXAMPLES
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Streaming Example
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=======================================================================
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% cat inputfile | lame [options] - - > output
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=======================================================================
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Scripts are included (in the 'misc' subdirectory)
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to run lame on multiple files:
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bach script: mlame Run "mlame -?" for instructions.
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sh script: auenc Run auenc for instructions
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bash script: mlame Run "mlame -?" for instructions.
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sh script: auenc Run auenc for instructions
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sh script: mugeco.sh
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Pearl script which will re-encode mp3 files and preserve id3 tags:
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lameid3.pl
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Perl script which will re-encode mp3 files and preserve id3 tags:
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lameid3.pl
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Windows scripts:
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lame4dos.bat
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lame4dos.bat
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Lame.vbs (and an HTML frontend: LameGUI.html)
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@@ -95,8 +96,7 @@ By default, LAME accepts a PCM audio sample inside a .WAV container as the
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input file, in 8, 16, 24 and 32 bits integer and in IEEE FLOAT.
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If it is compiled with libsndfile, then it also supports the extra formats
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that the library supports.
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There is also support for RAW PCM data and and piped input
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There is also support for raw PCM data and piped input
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Input options:
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@@ -113,7 +113,7 @@ Input options:
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--nogapout <dir>
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output dir for gapless encoding (must precede --nogap)
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--nogaptags allow the use of VBR tags in gapless encoding
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--out-dir path If no explicit output file is specified, a file will be
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--out-dir path If no explicit output file is specified, a file will be
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written at given path. Ignored when using piped/streamed input
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@@ -127,20 +127,20 @@ Input options for raw PCM:
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--little-endian input is little-endian (default)
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--big-endian input is big-endian
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-a downmix stereo file to mono file for mono encoding.
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Needed with RAW input for the -mm mode to do the downmix.
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Needed with raw input for the -mm mode to do the downmix.
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Resampling and filtering:
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--lowpass Frequency(kHz), lowpass filter cutoff above freq.
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--lowpass Frequency(kHz), lowpass filter cutoff above freq.
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Range [0.001..50]kHz or [50..50000]Hz
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--lowpass-width
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Frequency(kHz), lowpass window width.
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Range [0.001..16]kHz or [16..50000]Hz
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Frequency(kHz), lowpass window width.
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Range [0.001..16]kHz or [16..50000]Hz
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(See further restriction in the detailed explanation)
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--highpass Frequency(kHz), highpass filter cutoff below freq.
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Range [0.001..16]kHz or [16..50000]Hz
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Range [0.001..16]kHz or [16..50000]Hz
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(See further restriction in the detailed explanation)
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--highpass-width
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Frequency(kHz), highpass window width
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Frequency(kHz), highpass window width
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(See further restriction in the detailed explanation)
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--resample n Sampling frequency of output file(kHz)
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Default=automatic depending on settings like bitrate.
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@@ -155,7 +155,7 @@ Operational:
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(decode to raw pcm, native endian format (use -x to swap))
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-m m/s/j/f/a mode selection
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-q n Internal algorithm quality setting 0..9.
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-q n Internal algorithm quality setting 0..9.
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0 = slowest algorithms, but potentially highest quality
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9 = faster algorithms, very poor quality
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Default is 3. Read the differences between VBR and CBR below.
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@@ -184,7 +184,7 @@ Variable Bit Rate (VBR)
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-b n specify a minimum allowed bitrate (8,16,24,...,320)
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-B n specify a maximum allowed bitrate (8,16,24,...,320)
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-F strictly enforce minimum bitrate
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-t disable VBR informational tag
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-t disable VBR informational tag
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--nohist disable display of VBR bitrate histogram
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--abr n specify average bitrate desired
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@@ -228,7 +228,7 @@ ID3 tagging:
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--pad-id3v2-size <num>
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adds version 2 tag, pad with extra <num> bytes
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--genre-list print alphabetically sorted ID3 genre list and exit
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--ignore-tag-errors
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--ignore-tag-errors
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ignore errors in values passed for tags
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Note: A version 2 tag will NOT be added unless one of the input fields
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@@ -245,7 +245,7 @@ Verbosity:
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--verbose Print a lot of useful information
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--version Print License information
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--license Print License information
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--help Shows the common list of switches.
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--help Shows the common list of switches.
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Add id3 or dev to get help for a specified topic
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--usage Shows the common list of switches.
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Add id3 or dev to get help for a specified topic
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@@ -259,13 +259,13 @@ Detailed description of all options in alphabetical order
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=======================================================================
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downmix
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Downmix
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=======================================================================
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-a
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-a
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mix the stereo input file to mono and encode as mono.
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mix the stereo input file to mono and encode as mono.
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This option is only needed in the case of raw PCM stereo input
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This option is only needed in the case of raw PCM stereo input
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(because LAME cannot determine the number of channels in the input file).
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To encode a stereo (RAW) PCM input file as mono, use "lame -m m -a"
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@@ -274,12 +274,12 @@ mono .mp3 file from both mono and stereo input.
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=======================================================================
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average bitrate encoding (aka Safe VBR)
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Average bitrate encoding (aka Safe VBR)
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=======================================================================
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--abr n
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turns on encoding with a targeted average bitrate of n kbps, allowing
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to use frames of different sizes. The allowed range of n is 8...320
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to use frames of different sizes. The allowed range of n is 8...320
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kbps, you can use any integer value within that range.
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@@ -290,11 +290,11 @@ Use version 2 of the ID3 tag standard
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Tells LAME to add the tag information as id3v2. This implies adding both,
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a version 1 and a version 2 tag, if the values fit on a version 1 tag.
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See --id3v1-only and --id3v2-only if you want a more fine-grained control.
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See --id3v1-only and --id3v2-only if you want a more fine-grained control.
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=======================================================================
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bitrate
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Bitrate
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=======================================================================
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-b n
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@@ -316,7 +316,7 @@ a buggy hardware device with strange bitrate constrains.
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=======================================================================
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max bitrate
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Max bitrate
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=======================================================================
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-B n
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@@ -331,6 +331,7 @@ thus need to allow 320 kbps frames to get the same flexability as CBR
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streams. This is useful only if you need to circumvent a buggy hardware
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device with strange bitrate constrains.
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=======================================================================
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Endianess, big.
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=======================================================================
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@@ -339,43 +340,45 @@ Endianess, big.
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This switch tells LAME that the RAW pcm input is encoded in big-endian
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instead of little-endian.
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=======================================================================
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Sample bit with
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=======================================================================
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--bitwidth Sets the bitwidth value
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With RAW pcm input, this switch lets specify the bitwidth of the same
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(8 bits, 16 bits...)
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With RAW pcm input, this switch lets you specify the bitwidth of the same
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(8 bits, 16 bits...)
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=======================================================================
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copyright
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Copyright
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=======================================================================
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-c
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-c
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flag the encoded file as copyrighted
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mark the encoded file as copyrighted
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=======================================================================
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clipping detection
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Clipping detection
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=======================================================================
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--clipdetect
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Enable --replaygain-accurate and print a message whether clipping
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Enable --replaygain-accurate and print a message whether clipping
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occurs and how far in dB the waveform is from full scale.
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This option is not usable if the MP3 decoder was _explicitly_ disabled
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This option is not usable if the MP3 decoder was _explicitly_ disabled
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in the build of LAME.
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See also: --replaygain-accurate
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=======================================================================
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MPEG audio decode capability
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=======================================================================
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--decode
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--decode
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This uses LAME's HIP decoder to decode an MP3 file (layers 1, 2 and 3) to
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a wav file.
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a wav file.
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If -t is used (disable wav header), LAME will output
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raw pcm in native endian format (use -x to swap bytes).
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@@ -385,6 +388,7 @@ in the build of LAME.
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HIP stands for Hip Isn't a Player and is based off of Michael Hipp's mpglib 0.2a
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=======================================================================
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MPEG audio decode capability
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=======================================================================
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@@ -392,65 +396,68 @@ MPEG audio decode capability
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When decoding an mp3 file, LAME automatically corrects for the start delay
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that the encoder had to put into it. This setting lets you specify a different
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delay than LAME's own one, so that it is possible to compensate for the delay
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delay than LAME's own one, so that it is possible to compensate for the delay
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of mp3's generated with other encoders.
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=======================================================================
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de-emphasis
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De-emphasis
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=======================================================================
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-e n/5/c
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-e n/5/c
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n = (none, default)
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5 = 0/15 microseconds
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c = citt j.17
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c = CCITT j.17
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All this does is set a flag in the bitstream. If you have a PCM
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input file where one of the above types of (obsolete) emphasis has
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been applied, you can set this flag in LAME. Then the mp3 decoder
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should de-emphasize the output during playback, although most
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should de-emphasize the output during playback, although most
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decoders ignore this flag.
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A better solution would be to apply the de-emphasis with a standalone
|
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utility before encoding, and then encode without -e.
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||||
|
||||
|
||||
utility before encoding, and then encode without -e.
|
||||
|
||||
|
||||
=======================================================================
|
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strictly enforce VBR minimum bitrate
|
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Strictly enforce VBR minimum bitrate
|
||||
=======================================================================
|
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-F
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|
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strictly enforce VBR minimum bitrate. Without this option, passages of
|
||||
analog silence will be encoded at the minimum bitrate possible (32 or 8,
|
||||
depending on MPEG version).
|
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-F
|
||||
|
||||
strictly enforce VBR minimum bitrate. Without this option, passages of
|
||||
analog silence will be encoded at the minimum bitrate possible (32 or 8,
|
||||
depending on MPEG version).
|
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|
||||
|
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=======================================================================
|
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free format bitstreams
|
||||
Free format bitstreams
|
||||
=======================================================================
|
||||
--freeformat
|
||||
--freeformat
|
||||
|
||||
LAME will produce a fixed bitrate, free format bitstream.
|
||||
User must specify the desired bitrate in kbps, which can
|
||||
be any integer between 8 and 640.
|
||||
LAME will produce a fixed bitrate, free format bitstream. User must
|
||||
specify the desired bitrate in kbps, which can be any integer between
|
||||
8 and 640.
|
||||
|
||||
Not supported by most decoders. Complient decoders (of which there
|
||||
are few) are only required to support up to 320 kbps.
|
||||
are few) are only required to support up to 320 kbps.
|
||||
|
||||
Decoders known to handle free format:
|
||||
|
||||
supports up to
|
||||
supports up to
|
||||
mpg123 640 kbps
|
||||
MAD 640 kbps
|
||||
"lame --decode" 640 kbps
|
||||
l3dec: 310 kbps
|
||||
"lame --decode" 640 kbps
|
||||
l3dec 310 kbps
|
||||
|
||||
|
||||
=======================================================================
|
||||
gain
|
||||
Gain
|
||||
=======================================================================
|
||||
--gain Apply gain in decibels.
|
||||
|
||||
Apply Gain adjustment in decibels, range -20.0 to +12.0. 0dBFS means no amplification.
|
||||
Apply Gain adjustment in decibels, range -20.0 to +12.0. 0dBFS means no
|
||||
amplification.
|
||||
|
||||
|
||||
=======================================================================
|
||||
High pass filter
|
||||
@@ -462,19 +469,21 @@ Enables a highpass filter of the specified frequency when encoding the source.
|
||||
This is usually not required, and the gains are usually minimal. May be useful
|
||||
to remove an interference signal on 50Hz or 60Hz, or a DC offset.
|
||||
(default: disabled)
|
||||
Note: The current implementation has a minimum highpass frequency of
|
||||
Note: The current implementation has a minimum highpass frequency of
|
||||
(67.5/62)% of the sample rate (I.e. 481Hz at 44Khz).
|
||||
|
||||
|
||||
=======================================================================
|
||||
High pass filter
|
||||
=======================================================================
|
||||
--highpass-width width Set the width of the decaying curve.
|
||||
|
||||
Specify the width in Hz of the decaying curve of the highpass.
|
||||
Specify the width in Hz of the decaying curve of the highpass.
|
||||
Range [16..50000]Hz
|
||||
The minimum (and default) width is 75% of a band's width (which is 1/64th of
|
||||
the sample rate).
|
||||
Note: See the remark in the --highpass command above.
|
||||
Note: See the remark in the --highpass command above.
|
||||
|
||||
|
||||
=======================================================================
|
||||
ID3 tag modes
|
||||
@@ -494,37 +503,40 @@ be written.
|
||||
--ignore-tag-errors Ignore tag information errors
|
||||
|
||||
Put it before any tag setting.
|
||||
This tells lame to ignore the tag information that sees as erroneous
|
||||
and continue encoding without those. Without this setting, errors are reported
|
||||
and encoding does not start.
|
||||
This tells lame to ignore the tag information it sees as erroneous and
|
||||
continue encoding without those. Without this setting, errors are reported
|
||||
and encoding does not start.
|
||||
|
||||
|
||||
=======================================================================
|
||||
Endianess. little
|
||||
=======================================================================
|
||||
--little-endian Set the byte order to little-endian.
|
||||
|
||||
This switch tells LAME that the RAW pcm input is encoded in little-endian.
|
||||
It is the default setting.
|
||||
This switch tells LAME that the RAW pcm input is encoded in little-endian.
|
||||
It is the default setting.
|
||||
|
||||
|
||||
=======================================================================
|
||||
Low pass filter
|
||||
=======================================================================
|
||||
--lowpass number Use a lowpass filter when encoding
|
||||
|
||||
Enables a lowpass filter of the specified frequency when encoding the source.
|
||||
Enables a lowpass filter of the specified frequency when encoding the source.
|
||||
Range [0.001..50]kHz or [50..50000]Hz
|
||||
|
||||
Using a lowpass filter helps reducing the amount of data to encode. This is
|
||||
important in MP3 due to a limitation in very high frequencies (>16Khz).
|
||||
important in MP3 due to a limitation in very high frequencies (>16Khz).
|
||||
The default value depends on the target bitrate/quality. It is not recommended
|
||||
to change it as a general basis.
|
||||
|
||||
--lowpass-width width Set the width of the decaying curve.
|
||||
|
||||
Specify the width in Hz of the decaying curve of the lowpass.
|
||||
Specify the width in Hz of the decaying curve of the lowpass.
|
||||
Range [0.001..16]kHz or [16..50000]Hz
|
||||
The lowpass is in the center of this curve. The minimum (and default) width
|
||||
is 75% of a band's width (which is 1/64th of the sample rate).
|
||||
is 75% of a band's width (which is 1/64th of the sample rate).
|
||||
|
||||
|
||||
=======================================================================
|
||||
Modes:
|
||||
@@ -535,7 +547,7 @@ Modes:
|
||||
-m s (forced) L/R stereo
|
||||
-m j joint stereo
|
||||
-m f forced mid/side stereo
|
||||
-m d dual (independent) channels dual. Its purpose was meant for dual language
|
||||
-m d dual (independent) channels. Its purpose was meant for dual language
|
||||
streams where only one of them should be decoded.
|
||||
Most decoders just decode them as a stereo stream.
|
||||
-m a Currently, a synonym of m j. (In older releases it selected
|
||||
@@ -546,7 +558,7 @@ for a stereo input file, the two channels will be averaged into a mono
|
||||
signal. (Note: See comments about the -a switch for RAW PCM streams)
|
||||
|
||||
(FORCED) L/R STEREO encodes the left and the right signals independently,
|
||||
and gives more or less bits to each, depending on the currently available.
|
||||
and gives more or less bits to each, depending on the currently available.
|
||||
|
||||
JOINT STEREO is the default mode of encoding.
|
||||
jstereo means the encoder can use (on a frame by frame basis) either
|
||||
@@ -562,9 +574,9 @@ stereo can also sound bad. To determine when to switch to mid/side
|
||||
stereo, LAME uses a much more sophisticated algorithm than that
|
||||
described in the ISO documentation.
|
||||
|
||||
FORCED MID/SIDE STEREO forces all frames to be encoded mid/side stereo. It
|
||||
FORCED MID/SIDE STEREO forces all frames to be encoded mid/side stereo. It
|
||||
should only be used if you are sure every frame of the input file
|
||||
has very little stereo seperation.
|
||||
has very little stereo seperation.
|
||||
|
||||
DUAL CHANNEL mode is similar to encode the left and right as two mono signals.
|
||||
Its purpose was meant for Dual language streams where only one of them should
|
||||
@@ -574,18 +586,17 @@ INTENSITY STEREO
|
||||
Not supported.
|
||||
|
||||
|
||||
|
||||
=======================================================================
|
||||
MP3 input file
|
||||
=======================================================================
|
||||
--mp1input --mp2input --mp3input MPEG layer I, II or III input file
|
||||
--mp1input --mp2input --mp3input MPEG layer I, II or III input file
|
||||
|
||||
Assume the input file is a MP1/2/3 file. LAME will decode the input file
|
||||
before re-encoding it. Since MP3 is a lossy format, this is
|
||||
not recommended in general. But it is useful for creating low bitrate
|
||||
mp3s from high bitrate mp3s. If the filename ends in ".mp3" LAME will assume
|
||||
it is an MP3. For stdin or MP3 files which dont end in .mp3 you need
|
||||
to use this switch.
|
||||
before re-encoding it. Since MP3 is a lossy format, this is not recommended
|
||||
in general. But it is useful for creating low bitrate mp3s from high bitrate
|
||||
mp3s. If the filename ends in ".mp3" LAME will assume it is an MP3. For
|
||||
stdin or MP3 files which dont end in .mp3 you need to use this switch.
|
||||
|
||||
|
||||
=======================================================================
|
||||
No Gap (continuous audio) encoding of multiple files
|
||||
@@ -602,7 +613,7 @@ fade-in/out (as is the case of continuous playback).
|
||||
This setting solves that by using the samples from the next/previous file to
|
||||
compute the encoding.
|
||||
|
||||
--nogapout dir Specify a directory for the output of the files encoded
|
||||
--nogapout dir Specify a directory for the output of the files encoded
|
||||
with --nogap
|
||||
|
||||
This setting should precede --nogap, and is used to specify the alternate
|
||||
@@ -612,10 +623,11 @@ directory.
|
||||
--nogaptags Enables the use of VBR tags with files encoded with --nogap
|
||||
|
||||
Tells LAME to put VBR tags to encoded files if they are encoded in VBR or ABR
|
||||
modes. Else, using the --nogap option doesn't generate it.
|
||||
modes. Else, using the --nogap option doesn't generate it.
|
||||
|
||||
|
||||
=======================================================================
|
||||
disable historgram display
|
||||
Disable historgram display
|
||||
=======================================================================
|
||||
--nohist
|
||||
|
||||
@@ -624,7 +636,7 @@ VBR mp3 files. This will disable that feature.
|
||||
|
||||
|
||||
=======================================================================
|
||||
disable ReplayGain analysis
|
||||
Disable ReplayGain analysis
|
||||
=======================================================================
|
||||
--noreplaygain
|
||||
|
||||
@@ -634,24 +646,24 @@ See also: --replaygain-accurate, --replaygain-fast
|
||||
|
||||
|
||||
=======================================================================
|
||||
non-original
|
||||
Non-original
|
||||
=======================================================================
|
||||
-o
|
||||
-o
|
||||
|
||||
mark the encoded file as a copy
|
||||
|
||||
|
||||
|
||||
=======================================================================
|
||||
CRC error protection
|
||||
=======================================================================
|
||||
-p
|
||||
-p
|
||||
|
||||
Turn on CRC error protection.
|
||||
It will add a cyclic redundancy check (CRC) code in each frame, allowing
|
||||
to detect transmission errors that could occur on the MP3 stream. However,
|
||||
it takes 16 bits per frame that would otherwise be used for encoding, and
|
||||
then will slightly reduce the sound quality.
|
||||
therefore will slightly reduce the sound quality.
|
||||
|
||||
|
||||
=======================================================================
|
||||
ID3 V2 padding
|
||||
@@ -660,6 +672,7 @@ ID3 V2 padding
|
||||
|
||||
Pads the ID3v2 tag with extra 128bytes to allow it to expand.
|
||||
|
||||
|
||||
=======================================================================
|
||||
Preset system.
|
||||
=======================================================================
|
||||
@@ -677,7 +690,7 @@ Preset system.
|
||||
|
||||
Old compatibility settings. Meaningless
|
||||
--preset phone -b 16 -m m
|
||||
--preset phon+ /
|
||||
--preset phon+ /
|
||||
lw / mw-eu -b 24 -m m
|
||||
--preset mw-us -b 40 -m m
|
||||
--preset voice -b 56 -m m
|
||||
@@ -686,19 +699,19 @@ Old compatibility settings. Meaningless
|
||||
--preset cd -b 192
|
||||
--preset studio -b 256
|
||||
|
||||
|
||||
=======================================================================
|
||||
Window and OS/2 process priority control
|
||||
Windows and OS/2 process priority control
|
||||
=======================================================================
|
||||
--priority <type>
|
||||
|
||||
(Windows and OS/2 only)
|
||||
|
||||
Sets the process priority for LAME while running under IBM OS/2.
|
||||
This can be very useful to avoid the system becoming slow and/or
|
||||
unresponsive. By setting LAME to run in a lower priority, you leave
|
||||
more time for the system to update basic processing (drawing windows,
|
||||
polling keyboard/mouse, etc). The impact in LAME's performance is
|
||||
minimal if you use priority 0 to 2.
|
||||
Sets the process priority for LAME while running under Windows or IBM OS/2.
|
||||
This can be very useful to avoid the system becoming slow and/or unresponsive.
|
||||
By setting LAME to run in a lower priority, you leave more time for the system
|
||||
to update basic processing (drawing windows, polling keyboard/mouse, etc). The
|
||||
impact in LAME's performance is minimal if you use priority 0 to 2.
|
||||
|
||||
The valid parameters are:
|
||||
|
||||
@@ -706,26 +719,27 @@ The valid parameters are:
|
||||
1 = Medium priority (IDLE, delta = +31)
|
||||
2 = Regular priority (REGULAR, delta = -31)
|
||||
3 = High priority (REGULAR, delta = 0)
|
||||
4 = Maximum priority (REGULAR, delta = +31)
|
||||
4 = Maximum priority (REGULAR, delta = +31)
|
||||
|
||||
Note that if you call '--priority' without a parameter, priority 0 will be
|
||||
assumed.
|
||||
|
||||
|
||||
Note that if you call '--priority' without a parameter, then
|
||||
priority 0 will be assumed.
|
||||
|
||||
=======================================================================
|
||||
algorithm quality selection
|
||||
=======================================================================
|
||||
-q n
|
||||
Algorithm quality selection
|
||||
=======================================================================
|
||||
-q n
|
||||
|
||||
Bitrate is of course the main influence on quality. The higher the bitrate,
|
||||
Bitrate is of course the main influence on quality. The higher the bitrate,
|
||||
the higher the quality. But for a given bitrate, we have a choice of algorithms
|
||||
to determine the best scalefactors and Huffman coding (noise shaping).
|
||||
|
||||
For CBR, ABR and --vbr-old modes, the following table applies
|
||||
For CBR, ABR and --vbr-old modes, the following table applies
|
||||
|
||||
-q 0 Use the best algorithms (Best Huffman coding search, full outer
|
||||
loop, and the highest precision of several parameters).
|
||||
-q 1 to -q 4 Similar to -q 0 without the full outer loop and decreasing
|
||||
precision of parameters the further fromm q0. -q 3 is the default
|
||||
precision of parameters the further from q0. -q 3 is the default
|
||||
-q 5 and -q 6 Same as -q 7, but enables noise shaping and increases subblock
|
||||
gain
|
||||
-q 7 to -q 9 Same as -f. Very fast, OK quality. Psychoacoustics are used for
|
||||
@@ -740,10 +754,11 @@ For the default VBR mode since LAME 3.98, the following table applies
|
||||
-q 7 to -q 9 This level uses a psymodel but does not calculate quantisation
|
||||
noise when encoding: it takes a quick guess.
|
||||
|
||||
|
||||
=======================================================================
|
||||
input file is raw pcm
|
||||
Input file is raw pcm
|
||||
=======================================================================
|
||||
-r
|
||||
-r
|
||||
|
||||
Assume the input file is raw pcm. Sampling rate and mono/stereo/jstereo
|
||||
must be specified on the command line. Without -r, LAME will perform
|
||||
@@ -752,9 +767,8 @@ several fseek()'s on the input file looking for WAV and AIFF headers.
|
||||
Not supported if LAME is compiled to use LIBSNDFILE.
|
||||
|
||||
|
||||
|
||||
=======================================================================
|
||||
slightly more accurate ReplayGain analysis and finding the peak sample
|
||||
Slightly more accurate ReplayGain analysis and finding the peak sample
|
||||
=======================================================================
|
||||
--replaygain-accurate
|
||||
|
||||
@@ -763,45 +777,45 @@ by decoding on the fly the encoded data stream and store it in the file.
|
||||
|
||||
|
||||
ReplayGain analysis does _not_ affect the content of a compressed data
|
||||
stream itself, it is a value stored in the header of a sound file.
|
||||
Information on the purpose of ReplayGain and the algorithms used is
|
||||
stream itself, it is a value stored in the header of a sound file.
|
||||
Information on the purpose of ReplayGain and the algorithms used is
|
||||
available from http://www.replaygain.org/
|
||||
|
||||
By default, LAME performs ReplayGain analysis on the input data (after
|
||||
the user-specified volume scaling). This behaviour might give slightly
|
||||
inaccurate results because the data on the output of a lossy
|
||||
compression/decompression sequence differs from the initial input data.
|
||||
the user-specified volume scaling). This behaviour might give slightly
|
||||
inaccurate results because the data on the output of a lossy
|
||||
compression/decompression sequence differs from the initial input data.
|
||||
When --replaygain-accurate is specified the mp3 stream gets decoded on
|
||||
the fly and the analysis is performed on the decoded data stream.
|
||||
the fly and the analysis is performed on the decoded data stream.
|
||||
Although theoretically this method gives more accurate results, it has
|
||||
several disadvantages:
|
||||
* tests have shown that the difference between the ReplayGain values
|
||||
computed on the input data and decoded data is usually no greater
|
||||
than 0.5dB, although the minimum volume difference the human ear
|
||||
* tests have shown that the difference between the ReplayGain values
|
||||
computed on the input data and decoded data is usually no greater
|
||||
than 0.5dB, although the minimum volume difference the human ear
|
||||
can perceive is about 1.0dB
|
||||
* decoding on the fly significantly slows down the encoding process
|
||||
The apparent advantage is that:
|
||||
* with --replaygain-accurate the peak sample is determined and
|
||||
* with --replaygain-accurate the peak sample is determined and
|
||||
stored in the file. The knowledge of the peak sample can be useful
|
||||
to decoders (players) to prevent a negative effect called 'clipping'
|
||||
that introduces distortion into sound.
|
||||
|
||||
|
||||
Only the "Radio" ReplayGain value is computed. It is stored in the LAME tag.
|
||||
The analysis is performed with the reference volume equal to 89dB.
|
||||
Note: the reference volume has been changed from 83dB on transition
|
||||
from version 3.95 to 3.95.1.
|
||||
|
||||
This option is not usable if the MP3 decoder was _explicitly_ disabled
|
||||
in the build of LAME. (Note: if LAME is compiled without the MP3 decoder,
|
||||
ReplayGain analysis is performed on the input data after user-specified
|
||||
Only the "Radio" ReplayGain value is computed. It is stored in the LAME tag.
|
||||
The analysis is performed with the reference volume equal to 89dB.
|
||||
Note: the reference volume has been changed from 83dB on transition from
|
||||
version 3.95 to 3.95.1.
|
||||
|
||||
This option is not usable if the MP3 decoder was _explicitly_ disabled
|
||||
in the build of LAME. (Note: if LAME is compiled without the MP3 decoder,
|
||||
ReplayGain analysis is performed on the input data after user-specified
|
||||
volume scaling).
|
||||
|
||||
See also: --replaygain-fast, --noreplaygain, --clipdetect
|
||||
|
||||
|
||||
=======================================================================
|
||||
fast ReplayGain analysis
|
||||
Fast ReplayGain analysis
|
||||
=======================================================================
|
||||
--replaygain-fast
|
||||
|
||||
@@ -809,12 +823,12 @@ Compute "Radio" ReplayGain of the input data stream after user-specified
|
||||
volume scaling and/or resampling.
|
||||
|
||||
ReplayGain analysis does _not_ affect the content of a compressed data
|
||||
stream itself, it is a value stored in the header of a sound file.
|
||||
Information on the purpose of ReplayGain and the algorithms used is
|
||||
stream itself, it is a value stored in the header of a sound file.
|
||||
Information on the purpose of ReplayGain and the algorithms used is
|
||||
available from http://www.replaygain.org/
|
||||
|
||||
Only the "Radio" ReplayGain value is computed. It is stored in the LAME tag.
|
||||
The analysis is performed with the reference volume equal to 89dB.
|
||||
Only the "Radio" ReplayGain value is computed. It is stored in the LAME tag.
|
||||
The analysis is performed with the reference volume equal to 89dB.
|
||||
Note: the reference volume has been changed from 83dB on transition
|
||||
from version 3.95 to 3.95.1.
|
||||
|
||||
@@ -823,18 +837,17 @@ This switch is enabled by default.
|
||||
See also: --replaygain-accurate, --noreplaygain
|
||||
|
||||
|
||||
|
||||
=======================================================================
|
||||
output sampling frequency in kHz
|
||||
Output sampling frequency in kHz
|
||||
=======================================================================
|
||||
--resample n
|
||||
|
||||
where n = 8, 11.025, 12, 16, 22.05, 24, 32, 44.1, 48. The values as Hz
|
||||
instead of Khz are also supported. Else, the value will be ignored.
|
||||
|
||||
Output sampling frequency. Resample the input if necessary.
|
||||
Output sampling frequency. Resample the input if necessary.
|
||||
|
||||
If not specified, LAME may sometimes resample automatically
|
||||
If not specified, LAME may sometimes resample automatically
|
||||
when faced with extreme compression conditions (like encoding
|
||||
a 44.1 kHz input file at 32 kbps). To disable this automatic
|
||||
resampling, you have to use --resamle to set the output samplerate
|
||||
@@ -842,9 +855,8 @@ equal to the input samplerate. In that case, LAME will not
|
||||
perform any extra computations.
|
||||
|
||||
|
||||
|
||||
=======================================================================
|
||||
sampling frequency in kHz (for input RAW PCM)
|
||||
Sampling frequency in kHz (for input RAW PCM)
|
||||
=======================================================================
|
||||
-s n
|
||||
|
||||
@@ -858,32 +870,33 @@ supported MP3 samplerates if necessary.
|
||||
|
||||
|
||||
=======================================================================
|
||||
silent operation
|
||||
Silent operation
|
||||
=======================================================================
|
||||
-S
|
||||
|
||||
don't print progress report
|
||||
|
||||
|
||||
=======================================================================
|
||||
scale
|
||||
Scale
|
||||
=======================================================================
|
||||
--scale <arg>
|
||||
|
||||
Scales input by <arg>. This just multiplies the PCM data
|
||||
(after it has been converted to floating point) by <arg>.
|
||||
(after it has been converted to floating point) by <arg>.
|
||||
|
||||
<arg> > 1: increase volume
|
||||
<arg> = 1: no effect
|
||||
<arg> < 1: reduce volume
|
||||
|
||||
Use with care, since most MP3 decoders will truncate data
|
||||
which decodes to values greater than 32768.
|
||||
which decodes to values greater than 32768.
|
||||
|
||||
|
||||
=======================================================================
|
||||
strict ISO complience
|
||||
Strict ISO complience
|
||||
=======================================================================
|
||||
--strictly-enforce-ISO
|
||||
--strictly-enforce-ISO
|
||||
|
||||
With this option, LAME will enforce the 7680 bit limitation on
|
||||
total frame size. This results in many wasted bits for
|
||||
@@ -891,20 +904,19 @@ high bitrate encodings.
|
||||
|
||||
|
||||
=======================================================================
|
||||
disable VBR tag
|
||||
Disable VBR tag/WAV header
|
||||
=======================================================================
|
||||
-t
|
||||
-t
|
||||
This setting has two different uses:
|
||||
|
||||
When encoding to VBR, this setting disables writing the VBR Tag (also
|
||||
known as XING tag). This tag is embedded by default in the frame 0 of
|
||||
MP3 file. It lets VBR aware players to correctly seek and compute playing
|
||||
known as XING tag). This tag is embedded by default in the frame 0 of
|
||||
MP3 file. It allows VBR aware players to accurately seek and compute playing
|
||||
times in such files.
|
||||
|
||||
When decoding MP3 to WAV using --decode, this flag will disable writing the
|
||||
WAV header. The output will be raw pcm, native endian format. Use -x to swap
|
||||
bytes.
|
||||
|
||||
bytes.
|
||||
|
||||
|
||||
=======================================================================
|
||||
@@ -925,23 +937,25 @@ Setting Average bitrate (kbps)
|
||||
5 130
|
||||
|
||||
Using -V 7 or higher (lower quality) is not recommended.
|
||||
ABR usually produces better results.
|
||||
ABR usually produces better results.
|
||||
|
||||
|
||||
=======================================================================
|
||||
swapbytes
|
||||
Swap bytes
|
||||
=======================================================================
|
||||
-x
|
||||
|
||||
swap bytes in the input file (and output file when using --decode).
|
||||
For sorting out little endian/big endian type problems. If your
|
||||
encodings sound like static, try this first.
|
||||
For sorting out little endian/big endian type problems. If your encodings
|
||||
sound like static, try this first.
|
||||
|
||||
|
||||
=======================================================================
|
||||
swapbytes
|
||||
Ignore scalefactor band 21
|
||||
=======================================================================
|
||||
-Y Ignore noise in sbf21, like CBR mode does
|
||||
|
||||
Allows -V2, -V1 and -V0 to not encode the highest frequencies accurately,
|
||||
Allows -V2, -V1 and -V0 to not encode the highest frequencies accurately,
|
||||
if doing so causes disproportional increases in bitrate.
|
||||
This is the same that CBR and ABR modes do.
|
||||
|
||||
@@ -950,5 +964,5 @@ band, an encoder needs to increase the precision in all the bands (not only in
|
||||
this one).
|
||||
The consequence is an increase of bitrate (+60kbps in some cases) compared to
|
||||
not keeping that precision. Generally, this band should allow for distortions,
|
||||
so using this switch shouldn't cause harm.
|
||||
so using this switch shouldn't cause harm.
|
||||
|
||||
|
||||
Reference in New Issue
Block a user