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|
[suite filter_cmd]
caption = filters
[subcommand aacdec]
purpose = decode an aac stream
[subcommand amp]
purpose = amplify (scale) a raw audio stream
[option amp]
short_opt = a
summary = amplification value
typestr = number
arg_info = required_arg
arg_type = uint32
default_val = 32
[help]
The amplification value determines the scaling factor by which the
amplitude of the audio stream is multiplied. The formula for the
scaling factor is
factor = 1 + amp / 64.
For example, an amplification value of zero results in a scaling factor
of one while an amplification value of 64 means to double the volume.
[/help]
[subcommand compress]
purpose = dynamically adjust the volume of an audio stream
[option blocksize]
short_opt = b
summary = adjust volume after each block of size 2**bits (1-31)
typestr = bits
arg_info = required_arg
arg_type = uint32
default_val = 15
[help]
Larger blocksize means fewer volume adjustments per time unit.
[/help]
[option aggressiveness]
short_opt = a
summary = controls the maximum amount to amplify by (0-10)
typestr = bits
arg_info = required_arg
arg_type = uint32
default_val = 4
[help]
This controls the maximal gain factor. Zero means to not amplify
at all while the value 10 corresponds to maximal gain factor which
results in a 4-fold increase in volume.
[/help]
[option inertia]
short_opt = i
summary = how much inertia ramping has (1-14)
typestr = bits
arg_info = required_arg
arg_type = uint32
default_val = 6
[help]
Larger values cause smaller volume adjustments.
[/help]
[option target-level]
short_opt = t
summary = target signal level (0-32767)
typestr = level
arg_info = required_arg
arg_type = uint32
default_val = 16384
[help]
If the peak of the previous block is less than the target level,
volume is increased slightly for the next block. Otherwise it is
decreased. The default value is chosen to minimize clipping. There
is usually no reason to change it.
[/help]
[option damp]
short_opt = d
summary = if non-zero, scale down after normalizing (0-16)
typestr = bits
arg_info = required_arg
arg_type = uint32
default_val = 0
[help]
This scales down the volume of the audio stream by factor 2**bits.
This is mostly useful if another audio application (e.g., a video
game) is running in parallel and the relative volume of the audio
stream is too high.
[/help]
[subcommand fecdec]
purpose = decode a (lossy) input stream using forward error correction
[subcommand flacdec]
purpose = decode a flac stream
[subcommand mp3dec]
purpose = decode an mp3 stream
[option ignore-crc]
short_opt = i
summary = ignore CRC information in the audio stream
[help]
This causes frames with CRC errors to be decoded and played
anyway. This option is not recommended, but since some encoders
have been known to generate bad CRC information, this option is a
work-around to play streams from such encoders.
[/help]
[subcommand oggdec]
purpose = decode an ogg/vorbis stream
[subcommand opusdec]
purpose = decode an ogg/opus stream
[subcommand prebuffer]
purpose = delay processing of an audio stream
[option duration]
short_opt = d
summary = length of the prebuffer period
typestr = milliseconds
arg_info = required_arg
arg_type = uint32
default_val = 200
[help]
Wait this many milliseconds before letting data go through. The time
interval starts when the first data byte is seen in the input queue.
[/help]
[option size]
short_opt = s
summary = amount of data to prebuffer
typestr = bytes
arg_info = required_arg
arg_type = uint32
default_val = 0
[help]
Wait until this many data bytes are available in the input queue. The
default value of zero means to not prebuffer by size. If both
--duration and --size are given and non-zero, the prebuffer filter
waits until both conditions are met.
[/help]
[subcommand spxdec]
purpose = decode an ogg/speex stream
[subcommand sync]
purpose = synchronize playback between multiple clients
[option buddy]
short_opt = b
summary = client to synchronize with
typestr = url
arg_info = required_arg
arg_type = string
flag multiple
[help]
This option may be given multiple times, one per buddy. Each value
may be given as a host, port pair in either IPv4 or IPv6 form, with
port being optional. If no port was specified the listening port (as
specified with --port, see below) is used to send the synchronization
packet to this buddy.
[/help]
[option port]
short_opt = p
summary = UDP port for incoming synchronization packets
typestr = portnumber
arg_info = required_arg
arg_type = uint32
default_val = 29900
[help]
The sync filter expects incoming synchronization packets on this
UDP port.
[/help]
[option timeout]
short_opt = t
summary = how long to wait for other clients
typestr = milliseconds
arg_info = required_arg
arg_type = uint32
default_val = 2000
[help]
Once the sync filter receives its first chunk of input,
a synchronization period of the given number of milliseconds
begins. Playback is deferred until a synchronization packet has
been received from each defined buddy, or until the end of the
period. Buddies which did not send a synchronization packet in time
are temporarily disabled and are not waited for during subsequent
synchronization periods. They are re-enabled automatically when
another synchronization packet arrives.
[/help]
[subcommand wav]
purpose = insert a Microsoft wave header into a raw audio stream
[subcommand wmadec]
purpose = decode a wma stream
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