Audio Playback
Synopsis
na_play [input file0] [input file1] ... [-h ] [-itype string] [-n int] [-f int] [-ibo string] [-iswap ] [-istype string] [-c string] [-start float] [-end float] [-from int] [-to int] [-p string] [-command string] [-basic ] [-r* ] [-quality string] [-server string] [-audiodevice string] [-scale float] [-v ] [-wait ]
na_play is a general playback program for playing sound files on a variety of platforms and sound cards.
Currently, the following audio devices are supported:
- sunaudio: 8k ulaw direct to
/dev/audio
found on most Sun machines. This is also found under Linux and FreeBSD, and possibly others. This is the default if netaudio
is not supported.
- netaudio: NCD's network transparent audio system (NAS). This allows use of audio devices across a network. NAS has support for, Suns, Linux, FreeBSD, HPs and probably other machines by now.
- sun16audio: This is only available on newer Sun workstations and has been enabled at compile time. This provides 16bit linear PCM at various sample rates.
- linux16audio: This is only available on Linux workstations and has been enabled at compile time. This provides 16bit linear PCM at various sample rates.
- freebsd16audio: This is only available on workstations running FreeBSD and has been enabled at compile time. This provides 16bit linear PCM at various sample rates.
- mplayeraudio: This is only available under Windows NT 4.0 and Windows 95 and has been enabled at compile time. This provides 16bit linear PCM at various sample rates.
- win32audio : This is only available under Windows NT 4.0 and Windows 95 and has been enabled at compile time. This provides 16bit linear PCM at various sample rates, playing the audio directly rather than saving to a file as with mplayeraudio.
- irixaudio: Audio support for SGI's IRIX 6.2.
- Audio_Command: Allows the specification of an arbitrary UNIX command to play the waveform. This won't normally be used with
na_play
as you could just use the command directly but is necessary with some systems using the speech tools.
The default audio is netaudio if it is supported. If not the platform specific auido mode is the default (e.g. sun16audio, linux16audio, freebsd16audio or mplayeraudio). If none of these is supported, sunaudio is the default.
Options
- -h: options help
- -itype: string Input file type (optional). If set to raw, this indicates that the input file does not have a header. While this can be used to specify file types other than raw, this is rarely used for other purposes as the file type of all the existing supported types can be determined automatically from the file's header. If the input file is unheadered, files are assumed to be shorts (16bit). Supported types are nist, est, esps, snd, riff, aiff, audlab, raw, ascii
- -n: int Number of channels in an unheadered input file
- -f: int Sample rate in Hertz for an unheadered input file
- -ibo: string Input byte order in an unheadered input file: possibliities are: MSB , LSB, native or nonnative. Suns, HP, SGI Mips, M68000 are MSB (big endian) Intel, Alpha, DEC Mips, Vax are LSB (little endian)
- -iswap: Swap bytes. (For use on an unheadered input file)
- -istype: string Sample type in an unheadered input file: short, alaw, mulaw, byte, ascii
- -c: string Select a single channel (starts from 0). Waveforms can have multiple channels. This option extracts a single channel for progcessing and discards the rest.
- -start: float Extract sub-wave starting at this time, specified in seconds
- -end: float Extract sub-wave ending at this time, specified in seconds
- -from: int Extract sub-wave starting at this sample point
- -to: int Extract sub-wave ending at this sample point
- -p: string audio device protocol. Ths supported types are sunaudio audio_command linux16audio
- -command: string command to play wave when protocol is audio_command
- -basic: HTML audio/basic format, if unheadered treat as ulaw 8K
- -r*: ESPS compatible way of selecting subrange of file. The options -start, -end, -to and -from are recommended
- -quality: string either [ high | low ] . "high" will ensure that proper resampling is used. "low" means play as fast as possible, with a minimum of processing
- -server: string play sound on machine (when protocol is server-based)
- -audiodevice: string use specified audiodevice if approrpriate for protocol
- -scale: float change the gain (volume) of the signal. 1.0 is default
- -v: verbose. Print file names when playing
- -wait: wait for a key to be pressed between each file