--- /dev/null
+[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 = use blocks of size 2**bits
+ 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
+ typestr = bits
+ arg_info = required_arg
+ arg_type = uint32
+ default_val = 4
+ [option inertia]
+ short_opt = i
+ summary = how much inertia ramping has
+ typestr = bits
+ arg_info = required_arg
+ arg_type = uint32
+ default_val = 6
+ [option target-level]
+ short_opt = t
+ summary = target signal level (0-32768)
+ typestr = level
+ arg_info = required_arg
+ arg_type = uint32
+ default_val = 20000
+ [option damp]
+ short_opt = d
+ summary = if non-zero, scale down after normalizing
+ typestr = bits
+ arg_info = required_arg
+ arg_type = uint32
+ default_val = 0
+[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 resample]
+ purpose = transform raw audio to a different sample rate
+ [option converter]
+ short_opt = C
+ summary = set conversion algorithm
+ typestr = type
+ arg_info = required_arg
+ arg_type = string
+ default_val = medium
+ [help]
+ best: This is a bandlimited interpolator derived from the mathematical
+ sinc function and this is the highest quality sinc based converter,
+ providing a worst case Signal-to-Noise Ratio (SNR) of 97 decibels
+ (dB) at a bandwidth of 97%.
+
+ medium: This is another bandlimited interpolator much like the previous
+ one. It has an SNR of 97dB and a bandwidth of 90%. The speed of the
+ conversion is much faster than the previous one.
+
+ fastest: This is the fastest bandlimited interpolator and has an SNR
+ of 97dB and a bandwidth of 80%.
+
+ zero_order_hold: A Zero Order Hold converter (interpolated value
+ is equal to the last value). The quality is poor but the conversion
+ speed is blindlingly fast.
+
+ linear: A linear converter. Again the quality is poor, but the
+ conversion speed is blindingly fast.
+ [/help]
+ [option dest-sample-rate]
+ short_opt = d
+ summary = sample rate to convert to
+ typestr = rate
+ arg_info = required_arg
+ arg_type = uint32
+ default_val = 44100
+ m4_include(channels.m4)
+ m4_include(sample-rate.m4)
+ m4_include(sample-format.m4)
+[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
projects / paraslash.git / blobdiff