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 = adjust volume after each block of size 2**bits (1-31)
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 (0-10)
40 arg_info = required_arg
44 This controls the maximal gain factor. Zero means to not amplify
45 at all while the value 10 corresponds to maximal gain factor which
46 results in a 4-fold increase in volume.
50 summary = how much inertia ramping has (1-14)
52 arg_info = required_arg
56 Larger values cause smaller volume adjustments.
60 summary = target signal level (0-32767)
62 arg_info = required_arg
66 If the peak of the previous block is less than the target level,
67 volume is increased slightly for the next block. Otherwise it is
68 decreased. The default value is chosen to minimize clipping. There
69 is usually no reason to change it.
73 summary = if non-zero, scale down after normalizing (0-16)
75 arg_info = required_arg
79 This scales down the volume of the audio stream by factor 2**bits.
80 This is mostly useful if another audio application (e.g., a video
81 game) is running in parallel and the relative volume of the audio
85 purpose = decode a (lossy) input stream using forward error correction
87 purpose = decode a flac stream
89 purpose = decode an mp3 stream
92 summary = ignore CRC information in the audio stream
94 This causes frames with CRC errors to be decoded and played
95 anyway. This option is not recommended, but since some encoders
96 have been known to generate bad CRC information, this option is a
97 work-around to play streams from such encoders.
100 purpose = decode an ogg/vorbis stream
102 purpose = decode an ogg/opus stream
103 [subcommand prebuffer]
104 purpose = delay processing of an audio stream
107 summary = length of the prebuffer period
108 typestr = milliseconds
109 arg_info = required_arg
113 Wait this many milliseconds before letting data go through. The time
114 interval starts when the first data byte is seen in the input queue.
118 summary = amount of data to prebuffer
120 arg_info = required_arg
124 Wait until this many data bytes are available in the input queue. The
125 default value of zero means to not prebuffer by size. If both
126 --duration and --size are given and non-zero, the prebuffer filter
127 waits until both conditions are met.
129 [subcommand resample]
130 purpose = transform raw audio to a different sample rate
133 summary = set conversion algorithm
135 arg_info = required_arg
138 # RCT: resample conversion type
140 RCT_MEDIUM = "medium",
141 RCT_FASTEST = "fastest",
142 RCT_ZERO_ORDER_HOLD = "zero_order_hold",
143 RCT_LINEAR = "linear"
147 best: This is a bandlimited interpolator derived from the mathematical
148 sinc function and this is the highest quality sinc based converter,
149 providing a worst case Signal-to-Noise Ratio (SNR) of 97 decibels
150 (dB) at a bandwidth of 97%.
152 medium: This is another bandlimited interpolator much like the previous
153 one. It has an SNR of 97dB and a bandwidth of 90%. The speed of the
154 conversion is much faster than the previous one.
156 fastest: This is the fastest bandlimited interpolator and has an SNR
157 of 97dB and a bandwidth of 80%.
159 zero_order_hold: A Zero Order Hold converter (interpolated value
160 is equal to the last value). The quality is poor but the conversion
161 speed is blindlingly fast.
163 linear: A linear converter. Again the quality is poor, but the
164 conversion speed is blindingly fast.
166 [option dest-sample-rate]
168 summary = sample rate to convert to
170 arg_info = required_arg
173 m4_include(channels.m4)
174 m4_include(sample-rate.m4)
175 m4_include(sample-format.m4)
177 purpose = decode an ogg/speex stream
179 purpose = synchronize playback between multiple clients
182 summary = client to synchronize with
184 arg_info = required_arg
188 This option may be given multiple times, one per buddy. Each value
189 may be given as a host, port pair in either IPv4 or IPv6 form, with
190 port being optional. If no port was specified the listening port (as
191 specified with --port, see below) is used to send the synchronization
192 packet to this buddy.
196 summary = UDP port for incoming synchronization packets
198 arg_info = required_arg
202 The sync filter expects incoming synchronization packets on this
207 summary = how long to wait for other clients
208 typestr = milliseconds
209 arg_info = required_arg
213 Once the sync filter receives its first chunk of input,
214 a synchronization period of the given number of milliseconds
215 begins. Playback is deferred until a synchronization packet has
216 been received from each defined buddy, or until the end of the
217 period. Buddies which did not send a synchronization packet in time
218 are temporarily disabled and are not waited for during subsequent
219 synchronization periods. They are re-enabled automatically when
220 another synchronization packet arrives.
223 purpose = insert a Microsoft wave header into a raw audio stream
225 purpose = decode a wma stream