X-Git-Url: http://git.tuebingen.mpg.de/?a=blobdiff_plain;f=m4%2Flls%2Ffilter_cmd.suite.m4;fp=m4%2Flls%2Ffilter_cmd.suite.m4;h=4dda0bf9dcd70d3dec7b7eafd81b52d45c681ae4;hb=4adde8dae3317fa83b81e7a860c9ed9133e99bb0;hp=0000000000000000000000000000000000000000;hpb=44779382f0699be6b8daeeb25e7e1a2c40116d4b;p=paraslash.git diff --git a/m4/lls/filter_cmd.suite.m4 b/m4/lls/filter_cmd.suite.m4 new file mode 100644 index 00000000..4dda0bf9 --- /dev/null +++ b/m4/lls/filter_cmd.suite.m4 @@ -0,0 +1,197 @@ +[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