<HTML> <BODY> <H1>Listener</H1> First install <A HREF="http://www.mega-nerd.com/libsndfile/">libsndfile</A>!<BR> Then: <PRE> make install </PRE> Then run '<B>setlistener</B>' with as parameter the device to listen to. <PRE> setlistener /dev/dsp </PRE> Then, with the left and right arrow keys, move the '|' until the word "<silence>" appears. No, don't you move the '|' to the maximum right, move it to the right until the "<silence>" just constantly appears. Then: remember the value "Current level: ".<BR> <BR> Next step is creating the file <B>/usr/local/etc/listener.conf</B> Copy the example <B>listener.conf</B> to the location mentioned above and edit it to suit your needs.<BR> For example: <PRE> wav_path = /home/folkert/recordings devname = /dev/dsp detect_level = 800 min_duration = 1 max_duration = 300 rec_silence = 5 sample_rate = 44100 channels = 1 compression = ima-adpcm format = wav from_pipe = 0 exec = /home/folkert/encode_sample prerecord_n_seconds = 3 </PRE> <H2>format</H2> Format sets what kind of outputfile is generated. Possible are: <TABLE BORDER=1> <TR><TD><B>Keyword</B></TD><TD><B>Description</B></TD></TR> <TR><TD>wav</TD><TD>Microsoft WAV</TD></TR> <TR><TD>aiff</TD><TD>Apple/SGI AIFF</TD></TR> <TR><TD>au</TD><TD>SUN/NeXT AU</TD></TR> <TR><TD>raw</TD><TD>RAW PCM data</TD></TR> <TR><TD>svx</TD><TD>Amiga IFF/SVX8/SV16</TD></TR> <TR><TD>nist</TD><TD>Sphere NIST</TD></TR> <TR><TD>voc</TD><TD>VOC files</TD></TR> <TR><TD>ircam</TD><TD>Berkeley/IRCAM/CARL</TD></TR> <TR><TD>w64</TD><TD>Sonic Foundry's 64 bit RIFF/WAV</TD></TR> <TR><TD>mat4</TD><TD>Matlab v4.2/GNU Octave 2.0</TD></TR> <TR><TD>mat5</TD><TD>Matlab v5.0/GNU Octave 2.1</TD></TR> <TR><TD>pvf</TD><TD>Portable Voice Format</TD></TR> <TR><TD>xi</TD><TD>Fasttracker 2 Extended Instrument</TD></TR> <TR><TD>htk</TD><TD>HMM Toolkit format</TD></TR> <TR><TD>sds</TD><TD>Midi sample dump standard</TD></TR> <TR><TD>avr</TD><TD>Audio Visual Research</TD></TR> <TR><TD>wavex</TD><TD>MS WAVE with WAVEFORMATEX</TD></TR> </TABLE> <H2>sample_rate</H2> Sets the samplerate at which the samples are recorded. Be warned: only the rates supported by your soundcard can be entered here!<BR> <H2>rec_silence</H2> When while recording things become silent, don't stop recording immediately but keep recording for 'rec_silence' seconds to see if the sound restarts.<BR> <H2>min_duration</H2> When the sound starts, record at least that many seconds of audio, measured relative to the start of the recoding.<BR> E.g. if the sound only sounds for 3 seconds and you've set this value to 5, still at least 5 seconds of audio will be recorded.<BR> <H2>min_triggers</H2> sets how many samples must be above the threshold (the '<B>detect_level</B>') before recording starts. The samplerate is set to 44.1kHz so if you set min_duration to 4410, the minimum length (duration) of sound before the recording starts is: 0.1s (or: one tenth of a second).<BR> <H2>max_duration</H2> How long a sample may be before a new file is started.<BR> <H2>one_shot</H2> When set to yes/on/1, listener will exit after the first recorded sample.<BR> <H2>prerecord_n_seconds</H2> How many seconds to record <I>before</I> the sound starts.<BR> <H2>channels</H2> Sets the number of channels to record. default is 1 which is mono, 2 is stereo. Warning: some soundcards don't support mono, listener will then exit with an errormessage.<BR> <H2>compression</H2> sets the compressionmethod. Can be: <UL> <LI>u-law <LI>a-law <LI>ima-adpcm <LI>ms-adpcm <LI>gsm-6.10 <LI>G721_32 (32kbit) <LI>G723_24 (24kbit) <LI>G723_40 (40kbit) </UL> If you omit the compression-line completely, no compression is used.<BR> <H2>from_pipe</H2> sets wether listener is reading from a pipe fed by <B>listenersoundpipe</B>. <B>listenersoundpipe</B> lets you split a sound- device in two so that you can check 2 rooms with only one sound- card. You need to create two pipes and start 2 listener processes, each reading from one pipe.<BR> Set this parameter to yes/on/1 for "pipe-mode", no/off/0 for normal mode. <H2>exec</H2> selects the script/program to execute after the sample was recorded. This first (and only) parameter for that script is the filename of the sample.<BR> <H2>on_event_start</H2> selects the script/program to execute before the sample was recorded.<BR> <H2>output_pipe</H2> Instead of writing to a wav-file, listener can also execute an external program which does the compression and/or file-I/O.<BR> e.g.: <PRE> output_pipe=bladeenc -rawstereo -rawsigned -rawchannels=2 STDIN /tmp/my_file.mp3 </PRE> <H2>filter</H2> lets you add one or more filters. these will be used when detecting sounds. beware: sound is written to the wav-files BEFORE the filtering is done. See the directory '<I>filter_lib_example</I>' for an example on how to write a filter for listener.<BR> <H2>amplify</H2> Wether to automatically amplify the sound or not. Use 'on' and 'off' to toggle.<BR> <H2>start_amplify</H2> What amplification to use when the sound starts. This will be adjusted to something more applicable UNLESS 'fixed_amplify' is set to yes.<BR> <H2>max_amplify</H2> Maximum amplification to use.<BR> <H2>fixed_amplify</H2> Wether to ONLY use the amplification factor set with 'start_amplify' (=yes) or let listener do autotuning (=no).<BR> <B>safe_after_filter</B>sets wether to save the audio-data to disk before or after the filtering. Before the filtering gives you the original sound while after gives you the sound after processing with the filter(s).<BR> Enter '<B>on</B>' or '<B>off</B>'.<BR> <BR> The final step is starting the program: <PRE> listener </PRE> and that's it!<BR> <BR> Thanks to Hugo <A HREF="mailto:hugo@homebaze.net">Hugo</A> for adding code that enables you to set some parameters from the commandline.<BR> <BR> If you have any suggestions or anything else, do not hesitate to contact me at the following e-mail address:<BR> <A HREF="folkert@vanheusden.com">folkert@vanheusden.com</A><BR> </BODY> </HTML>
