afconvert in iPhone reference library
Can you give me some information about using afconvert in iPhone to convert file formats? Or let me know some links that give me basic information o开发者_Go百科n afconvert. I want to know the commands used - what do -f, -d , -c etc. stands for in:
afconvert -f aac -d mp3 [input] [output]
Where do I mention the source data format, file format and destination data format, file formats in the above command?
afconvert -h
at the Terminal prompt produces, in 10.6.6:
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.
{ -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)
--no-filler
don't page-align audio data in the output file
--soundcheck-generate
analyze audio, add SoundCheck data to the output file
{ -u | --userproperty } property value
set an arbitrary AudioConverter property to a given value
property is a four-character code; value is signed 32-bit integer.
A maximum of 8 properties may be set.
e.g. '-u vbrq <sound_quality>' sets the sound quality level
(<sound_quality>: 0-127)
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).
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
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
Help options:
{ -hf | --help-formats }
print a list of supported file/data formats
{ -h | --help }
print this help
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