4 purpose = decode an aac stream
6 purpose = amplify (scale) a raw audio stream
9 summary = amplification value
11 arg_info = required_arg
15 The amplification value determines the scaling factor by which the
16 amplitude of the audio stream is multiplied. The formula for the
19 factor = 1 + amp / 64.
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.
25 purpose = dynamically adjust the volume of an audio stream
28 summary = use blocks of size 2**bits
30 arg_info = required_arg
34 Larger blocksize means fewer volume adjustments per time unit.
36 [option aggressiveness]
38 summary = controls the maximum amount to amplify by
40 arg_info = required_arg
45 summary = how much inertia ramping has
47 arg_info = required_arg
52 summary = target signal level (0-32768)
54 arg_info = required_arg
59 summary = if non-zero, scale down after normalizing
61 arg_info = required_arg
65 purpose = decode a (lossy) input stream using forward error correction
67 purpose = decode a flac stream
69 purpose = decode an mp3 stream
72 summary = ignore CRC information in the audio stream
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.
80 purpose = decode an ogg/vorbis stream
82 purpose = decode an ogg/opus stream
83 [subcommand prebuffer]
84 purpose = delay processing of an audio stream
87 summary = length of the prebuffer period
88 typestr = milliseconds
89 arg_info = required_arg
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.
98 summary = amount of data to prebuffer
100 arg_info = required_arg
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.
109 [subcommand resample]
110 purpose = transform raw audio to a different sample rate
113 summary = set conversion algorithm
115 arg_info = required_arg
118 # RCT: resample conversion type
120 RCT_MEDIUM = "medium",
121 RCT_FASTEST = "fastest",
122 RCT_ZERO_ORDER_HOLD = "zero_order_hold",
123 RCT_LINEAR = "linear"
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%.
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.
136 fastest: This is the fastest bandlimited interpolator and has an SNR
137 of 97dB and a bandwidth of 80%.
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.
143 linear: A linear converter. Again the quality is poor, but the
144 conversion speed is blindingly fast.
146 [option dest-sample-rate]
148 summary = sample rate to convert to
150 arg_info = required_arg
153 m4_include(channels.m4)
154 m4_include(sample-rate.m4)
155 m4_include(sample-format.m4)
157 purpose = decode an ogg/speex stream
159 purpose = synchronize playback between multiple clients
162 summary = client to synchronize with
164 arg_info = required_arg
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.
176 summary = UDP port for incoming synchronization packets
178 arg_info = required_arg
182 The sync filter expects incoming synchronization packets on this
187 summary = how long to wait for other clients
188 typestr = milliseconds
189 arg_info = required_arg
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.
203 purpose = insert a Microsoft wave header into a raw audio stream
205 purpose = decode a wma stream