MacOS的强大一方面也在于其继承了众多实用工具,比如今天介绍的这个afconvert。
位置:/usr/bin/afconvert
用法:需要使用命令行操作,在终端cd到bin下即可使用
参数说明:
afconvert [option…] input_file [output_file]
Options may appear before or after the direct arguments. If output_file
is not specified, a name is generated programmatically and the file
is written into the same directory as input_file.
afconvert input_file [-o output_file [option…]]…
Output file options apply to the previous output_file. Other options
may appear anywhere.
General options:
{ -d | –data } data_format[@sample_rate][/format_flags][#frames_per_packet]
[-][BE|LE]{F|[U]I}{8|16|24|32|64} (PCM)
e.g. BEI16 F32@44100
or a data format appropriate to file format (see -hf)
format_flags: hex digits, e.g. ‘80’
Frames per packet can be specified for some encoders, e.g.: samr#12
A format of “0” specifies the same format as the source file,
with packets copied exactly.
A format of “N” specifies the destination format should be the
native format of the lossless encoded source file (alac, FLAC only)
{ -c | –channels } number_of_channels
add/remove channels without regard to order
{ -l | –channellayout } layout_tag
layout_tag: name of a constant from CoreAudioTypes.h
(prefix “kAudioChannelLayoutTag_” may be omitted)
if specified once, applies to output file; if twice, the first
applies to the input file, the second to the output file
{ -b | –bitrate } total_bit_rate_bps
e.g. 256000 will give you roughly:
for stereo source: 128000 bits per channel
for 5.1 source: 51000 bits per channel
(the .1 channel consumes few bits and can be discounted in the
total bit rate calculation)
{ -q | –quality } codec_quality
codec_quality: 0-127
{ -r | –src-quality } src_quality
src_quality (sample rate converter quality): 0-127 (default is 127)
{ –src-complexity } src_complexity
src_complexity (sample rate converter complexity): line, norm, bats
{ -s | –strategy } strategy
bitrate allocation strategy for encoding an audio track
0 for CBR, 1 for ABR, 2 for VBR_constrained, 3 for VBR
–prime-method method
decode priming method (see AudioConverter.h)
–prime-override samples_prime samples_remain
can be used to override the priming information stored in the source
file to the specified values. If -1 is specified for either, the value
in the file is used.
–no-filler
don’t page-align audio data in the output file
–soundcheck-generate
analyze audio, add SoundCheck data to the output file
–media-kind “media kind string”
media kinds are: “Audio Ad”, “Video Ad”
–anchor-loudness
set a single precision floating point value to
indicate the anchor loudness of the content in dB
–generate-hash
generate an SHA-1 hash of the input audio data and add it to the output file.
–codec-manuf codec_manuf
specify the codec with the specified 4-character component manufacturer
code
–dither algorithm
algorithm: 1-2
–mix
enable channel downmixing
{ -u | –userproperty } property value
set an arbitrary AudioConverter property to a given value
property is a four-character code; value can be a signed
32-bit integer or a single precision floating point value.
e.g. ‘-u vbrq ’ sets the sound quality level
(: 0-127)
May not be used in a transcoding situation.
-ud property value
identical to -u except only applies to a decoder. Fails if there is no
decoder.
-ue property value
identical to -u except only applies to an encoder. Fails if there is no
encoder.
Input file options:
–read-track track_index
For input files containing multiple tracks, the index (0..n-1)
of the track to read and convert.
–offset number_of_frames
the starting offset in the input file in frames. (The first frame is
frame zero.)
–soundcheck-read
read SoundCheck data from source file and set it on any destination
file(s) of appropriate filetype (.m4a, .caf).
–copy-hash
copy an SHA-1 hash chunk, if present, from the source file to the output file.
–gapless-before filename
file coming before the current input file of a gapless album
–gapless-after filename
file coming after the current input file of a gapless album
Output file options:
-o filename
specify an (additional) output file.
{ -f | –file } file_format
use -hf for a complete list of supported file/data formats
–condensed-framing field_size_in_bits
specify storage size in bits for externally framed packet sizes.
Supported value is 16 for aac in m4a file format.
Other options:
{ -v | –verbose }
print progress verbosely
{ -t | –tag }
If encoding to CAF, store the source file’s format and name in a user
chunk. If decoding from CAF, use the destination format and filename
found in a user chunk.
{ –leaks }
run leaks at the end of the conversion
{ –profile }
collect and print performance information
关于file format(也就是支持转换的格式):
Audio file and data formats:
‘3gpp’ = 3GP Audio (.3gp)
data_formats: ‘Qclp’ ‘aac ’ ‘aace’ ‘aacf’ ‘aach’ ‘aacl’
‘aacp’ ‘samr’
‘3gp2’ = 3GPP-2 Audio (.3g2)
data_formats: ‘Qclp’ ‘aac ’ ‘aace’ ‘aacf’ ‘aach’ ‘aacl’
‘aacp’ ‘samr’
‘adts’ = AAC ADTS (.aac, .adts)
data_formats: ‘aac ’ ‘aach’ ‘aacp’
‘ac-3’ = AC3 (.ac3)
data_formats: ‘ac-3’
‘AIFC’ = AIFC (.aifc, .aiff, .aif)
data_formats: I8 BEI16 BEI24 BEI32 BEF32 BEF64 UI8 ‘ulaw’
‘alaw’ ‘MAC3’ ‘MAC6’ ‘ima4’ ‘QDMC’ ‘QDM2’
‘Qclp’ ‘agsm’
‘AIFF’ = AIFF (.aiff, .aif)
data_formats: I8 BEI16 BEI24 BEI32
‘amrf’ = AMR (.amr)
data_formats: ‘samr’ ‘sawb’
‘m4af’ = Apple MPEG-4 Audio (.m4a, .m4r)
data_formats: ‘aac ’ ‘aace’ ‘aacf’ ‘aach’ ‘aacl’ ‘aacp’
‘ac-3’ ‘alac’ ‘ec-3’ ‘paac’ ‘pac3’ ‘qec3’
‘zec3’
‘m4bf’ = Apple MPEG-4 AudioBooks (.m4b)
data_formats: ‘aac ’ ‘aace’ ‘aacf’ ‘aach’ ‘aacl’ ‘aacp’
‘paac’
‘caff’ = CAF (.caf)
data_formats: ‘.mp1’ ‘.mp2’ ‘.mp3’ ‘QDM2’ ‘QDMC’ ‘Qclp’
‘Qclq’ ‘aac ’ ‘aace’ ‘aacf’ ‘aach’ ‘aacl’
‘aacp’ ‘ac-3’ ‘alac’ ‘alaw’ ‘dvi8’ ‘ec-3’
‘ilbc’ ‘ima4’ I8 BEI16 BEI24 BEI32 BEF32
BEF64 LEI16 LEI24 LEI32 LEF32 LEF64 ‘ms\x00\x02’
‘ms\x00\x11’ ‘ms\x001’ ‘paac’ ‘pac3’ ‘pec3’
‘qaac’ ‘qac3’ ‘qach’ ‘qacp’ ‘qec3’ ‘samr’
‘ulaw’ ‘zaac’ ‘zac3’ ‘zach’ ‘zacp’ ‘zec3’
‘ec-3’ = EC3 (.ec3)
data_formats: ‘ec-3’
‘MPG1’ = MPEG Layer 1 (.mp1, .mpeg, .mpa)
data_formats: ‘.mp1’
‘MPG2’ = MPEG Layer 2 (.mp2, .mpeg, .mpa)
data_formats: ‘.mp2’
‘MPG3’ = MPEG Layer 3 (.mp3, .mpeg, .mpa)
data_formats: ‘.mp3’
‘mp4f’ = MPEG-4 Audio (.mp4)
data_formats: ‘aac ’ ‘aace’ ‘aacf’ ‘aach’ ‘aacl’ ‘aacp’
‘ac-3’ ‘ec-3’ ‘qec3’ ‘zec3’
‘NeXT’ = NeXT/Sun (.snd, .au)
data_formats: I8 BEI16 BEI24 BEI32 BEF32 BEF64 ‘ulaw’
‘Sd2f’ = Sound Designer II (.sd2)
data_formats: I8 BEI16 BEI24 BEI32
‘WAVE’ = WAVE (.wav)
data_formats: UI8 LEI16 LEI24 LEI32 LEF32 LEF64 ‘ulaw’
‘alaw’
以上介绍内容均来自help,这么多的参数该如何使用?对于一般人来说,只需要转换音乐,比如我们项目上需要将背景音乐转换成*.caf的格式,那么我们只需关心-f和-d两个参数。
-f指的是file format,也就是需要转换的目标格式,参考上述说明,caf文件的命令是-f caff
-d指的是data format,是数据格式的意思,对照上述说明,caf支持的数据格式很多,为了节省空间,我们使用“I8”,也就是-d I8,理解这些参数的含义需要一定的音乐知识背景,这里不赘述。
关于输出路径,如果不用-o制指定,则默认和输入文件在同一个目录,以下举例:
假设用户桌面上有音频文件music.mp3,路径为/Users/test/Desktop/music.mp3,需要转换成caf格式,采样32,速率44100,输出路径相同,则可以在终端输入:
afconvert -f caff -d LEI32@44100 “/Users/test/Desktop/music.mp3”
这样输出的就是按要求的音频文件了,很简单,也很方便。关于其他的参数,需要专业的音乐制作使用,一般不去动,不然不知道具体用处的话会对输出造成一定影响。如有问题,可留言探讨。