ddc5f1cb3c191df7d472732f76af40645f02ccf6
[paraslash.git] / m4 / lls / filter_cmd.suite.m4
1 [suite filter_cmd]
2 caption = filters
3 [subcommand aacdec]
4 purpose = decode an aac stream
5 [subcommand amp]
6 purpose = amplify (scale) a raw audio stream
7 [option amp]
8 short_opt = a
9 summary = amplification value
10 typestr = number
11 arg_info = required_arg
12 arg_type = uint32
13 default_val = 32
14 [help]
15 The amplification value determines the scaling factor by which the
16 amplitude of the audio stream is multiplied. The formula for the
17 scaling factor is
18
19 factor = 1 + amp / 64.
20
21 For example, an amplification value of zero results in a scaling factor
22 of one while an amplification value of 64 means to double the volume.
23 [/help]
24 [subcommand compress]
25 purpose = dynamically adjust the volume of an audio stream
26 [option blocksize]
27 short_opt = b
28 summary = use blocks of size 2**bits
29 typestr = bits
30 arg_info = required_arg
31 arg_type = uint32
32 default_val = 15
33 [help]
34 Larger blocksize means fewer volume adjustments per time unit.
35 [/help]
36 [option aggressiveness]
37 short_opt = a
38 summary = controls the maximum amount to amplify by
39 typestr = bits
40 arg_info = required_arg
41 arg_type = uint32
42 default_val = 4
43 [option inertia]
44 short_opt = i
45 summary = how much inertia ramping has
46 typestr = bits
47 arg_info = required_arg
48 arg_type = uint32
49 default_val = 6
50 [option target-level]
51 short_opt = t
52 summary = target signal level (0-32767)
53 typestr = level
54 arg_info = required_arg
55 arg_type = uint32
56 default_val = 20000
57 [option damp]
58 short_opt = d
59 summary = if non-zero, scale down after normalizing
60 typestr = bits
61 arg_info = required_arg
62 arg_type = uint32
63 default_val = 0
64 [subcommand fecdec]
65 purpose = decode a (lossy) input stream using forward error correction
66 [subcommand flacdec]
67 purpose = decode a flac stream
68 [subcommand mp3dec]
69 purpose = decode an mp3 stream
70 [option ignore-crc]
71 short_opt = i
72 summary = ignore CRC information in the audio stream
73 [help]
74 This causes frames with CRC errors to be decoded and played
75 anyway. This option is not recommended, but since some encoders
76 have been known to generate bad CRC information, this option is a
77 work-around to play streams from such encoders.
78 [/help]
79 [subcommand oggdec]
80 purpose = decode an ogg/vorbis stream
81 [subcommand opusdec]
82 purpose = decode an ogg/opus stream
83 [subcommand prebuffer]
84 purpose = delay processing of an audio stream
85 [option duration]
86 short_opt = d
87 summary = length of the prebuffer period
88 typestr = milliseconds
89 arg_info = required_arg
90 arg_type = uint32
91 default_val = 200
92 [help]
93 Wait this many milliseconds before letting data go through. The time
94 interval starts when the first data byte is seen in the input queue.
95 [/help]
96 [option size]
97 short_opt = s
98 summary = amount of data to prebuffer
99 typestr = bytes
100 arg_info = required_arg
101 arg_type = uint32
102 default_val = 0
103 [help]
104 Wait until this many data bytes are available in the input queue. The
105 default value of zero means to not prebuffer by size. If both
106 --duration and --size are given and non-zero, the prebuffer filter
107 waits until both conditions are met.
108 [/help]
109 [subcommand resample]
110 purpose = transform raw audio to a different sample rate
111 [option converter]
112 short_opt = C
113 summary = set conversion algorithm
114 typestr = type
115 arg_info = required_arg
116 arg_type = string
117 values = {
118 # RCT: resample conversion type
119 RCT_BEST = "best",
120 RCT_MEDIUM = "medium",
121 RCT_FASTEST = "fastest",
122 RCT_ZERO_ORDER_HOLD = "zero_order_hold",
123 RCT_LINEAR = "linear"
124 }
125 default_val = medium
126 [help]
127 best: This is a bandlimited interpolator derived from the mathematical
128 sinc function and this is the highest quality sinc based converter,
129 providing a worst case Signal-to-Noise Ratio (SNR) of 97 decibels
130 (dB) at a bandwidth of 97%.
131
132 medium: This is another bandlimited interpolator much like the previous
133 one. It has an SNR of 97dB and a bandwidth of 90%. The speed of the
134 conversion is much faster than the previous one.
135
136 fastest: This is the fastest bandlimited interpolator and has an SNR
137 of 97dB and a bandwidth of 80%.
138
139 zero_order_hold: A Zero Order Hold converter (interpolated value
140 is equal to the last value). The quality is poor but the conversion
141 speed is blindlingly fast.
142
143 linear: A linear converter. Again the quality is poor, but the
144 conversion speed is blindingly fast.
145 [/help]
146 [option dest-sample-rate]
147 short_opt = d
148 summary = sample rate to convert to
149 typestr = rate
150 arg_info = required_arg
151 arg_type = uint32
152 default_val = 44100
153 m4_include(channels.m4)
154 m4_include(sample-rate.m4)
155 m4_include(sample-format.m4)
156 [subcommand spxdec]
157 purpose = decode an ogg/speex stream
158 [subcommand sync]
159 purpose = synchronize playback between multiple clients
160 [option buddy]
161 short_opt = b
162 summary = client to synchronize with
163 typestr = url
164 arg_info = required_arg
165 arg_type = string
166 flag multiple
167 [help]
168 This option may be given multiple times, one per buddy. Each value
169 may be given as a host, port pair in either IPv4 or IPv6 form, with
170 port being optional. If no port was specified the listening port (as
171 specified with --port, see below) is used to send the synchronization
172 packet to this buddy.
173 [/help]
174 [option port]
175 short_opt = p
176 summary = UDP port for incoming synchronization packets
177 typestr = portnumber
178 arg_info = required_arg
179 arg_type = uint32
180 default_val = 29900
181 [help]
182 The sync filter expects incoming synchronization packets on this
183 UDP port.
184 [/help]
185 [option timeout]
186 short_opt = t
187 summary = how long to wait for other clients
188 typestr = milliseconds
189 arg_info = required_arg
190 arg_type = uint32
191 default_val = 2000
192 [help]
193 Once the sync filter receives its first chunk of input,
194 a synchronization period of the given number of milliseconds
195 begins. Playback is deferred until a synchronization packet has
196 been received from each defined buddy, or until the end of the
197 period. Buddies which did not send a synchronization packet in time
198 are temporarily disabled and are not waited for during subsequent
199 synchronization periods. They are re-enabled automatically when
200 another synchronization packet arrives.
201 [/help]
202 [subcommand wav]
203 purpose = insert a Microsoft wave header into a raw audio stream
204 [subcommand wmadec]
205 purpose = decode a wma stream