Re: [linux-audio-user] Indexing the Read Pointer in /dev/dsp

Quoting Martin McCormick <martin(a)dc.cis.okstate.edu>du>: [snip] Funny, I've written a simple program (derived from the Jack "simple_client") recent to do something similar. What it does (for now, I'm sure I'll be adding more to it) is: 1) Monitor the command-line specified input port for a sound above the squelch level. 2) Apply a configurable DC Bias adjustment [squelch comparison is performed after this] 3) Keep track of the peak samples processed to pass a "scaling value" to normalise the sound data. [This is also post-DC Bias adjustment.] 3a) I'm also looking at various dynamic compression [gain reduction] strategies. 4) Log all data to a temporary .WAV file. Note, it's sample-perfect. The triggering sample is successfully captured. 5) After a command-line specifiable time where the samples are BELOW the squelch level, the recording is "closed" and a file is written thusly: A) A sub-process is spawned to "wash" and LAME-encode (MP3} whilst embedding tags into the resulting mp3 file to note the origination time/date (fully specified with time-zone). The scaling value from the recording program is passed to LAME to pre-normalise the sound prior to encoding. B) Since the sub-process can "take as long as it wants" without disturbing the audio monitor threads, I have also tinkered with filtering, and have found that for my application [recording voice calls from a MERLIN phone system], a 100Hz-4000Hz bandpass works wonderfully. I use 8-32Kbps MP3 VBR settings, which work very well. Speaker intelligibility is flawless. I record at 16 bits, 11,025 Hz, FYI. LAME command line: voxencode is passed three variables: $1 = a fully specified verbose (RFC-style) Date string $2 = Filename prefix $3 = scaling value [1.00 or no parametre means no scaling] -- this isn't represented in the LAME command below]. lame -S --silent --nohist -q 2 -h --vbr-new -b 8 -B 32 \ --tt "Title Prefix $1" --ta "Your Name or Label" \ --tl 'Recording Source' --ty `date +%Y` --tc "Recorded $1" \ --add-id3v2 --pad-id3v2 --tg Speech "$2.wav" "$2.mp3" Is there a reason why you want to store the timestamp into the actual audio stream? There are many formats where you can place such data "out of band". I generate the .wav and .mp3 filenames thusly: "Prefix-YYYY-MM-DD-HH-MM-SS.wav" and .mp3 respectively. My sub-procress spawns to a voxencode script which decouples the post-processing from the main capturing application which is multi-threaded. The capturing app runs via nohup &, so it doesn't need a TTY. If you only want one big file instead of alot of files, I'm not sure how you'd get what you want unless you wrote your own audio player. I would think that having lots of files wouldn't be problematic since most mp3/media players can let you just "play all files in a directory" easily enough if you just want to listen to the whole thing w/o having to do anything in the UI between files. Why do the Dwarves prize mithril above all else? Quality. =MB= -- A focus on Quality.