Linux-audio-dev

linux-audio-dev@lists.linuxaudio.org

7516 discussions

by krampenschiesser＠freenet.de

Does someone have a very simple sequencer example? Something sending one note in 4/4? I don't get along with the timing of the alsa sequencer. I hope you can help me. Scar -- All the evidence concerning the universe has not yet been collected, so there's still hope.

19 years, 9 months

[linux-audio-dev] liblo lo_message_deserialise?

by Mario Lang

Hi. While playing a little with the liblo lowlevel interface (I am trying to deliver and receive/decode OSC messages via a non-UDP/TCP transport), I found lo_message_serialise. I also had to rip a bit of code from lo_send_internal and friends. However, I cant find a counterpart for decomposing messages, so I guess currently, I would have to rip even more code out of the lib itself. I am wondering, would anyone else benefit from a clean lowlevel API for decoding messages to a lo_message? I am imagining something like lo_message lo_message_from_buffer (char *buf, int size) Any comments? -- CYa, Mario

19 years, 9 months

[linux-audio-dev] Re: ANN: DT-42 DrumToy 0.1.0

by Juhana Sadeharju

Hello. I downloaded the mentioned TR808 samples. The samples are in some odd SGI package file, and so, I quickly wrote an extractor. But to save your time, I placed all at http://www.funet.fi/~kouhia/sgitr808samples.tar.gz The extractor and the original sample package are at http://www.funet.fi/~kouhia/sgitr808.tar.gz Juhana -- http://music.columbia.edu/mailman/listinfo/linux-graphics-dev for developers of open source graphics software

19 years, 9 months

[linux-audio-dev] Live music performing setup using VNC?

by Ferdinand Vesely

Hello! I'm not quite shure, if this belongs here, or to the LAU list, but I am subscribed here, so this is why I post it here. First, let me explain: I'm in an urgent need for a notebook, as I am a university student (from Slovakia) working as a part time PHP programmer and playing in a (prog)rock band. I don't have much money I can spend on that book and I need it now, so I am considering buying a refurbished IBM ThinkPad 600X (PIII/450MHz, 128 or 256 MB RAM, 12 GB HDD). I'm not very demanding, but I would be very glad, if I could use it for performing music on stage. In the beginning maybe only playing a bit with PD and playing samples and loops. I had an idea (this is why I'm writing) to buy later a better desktop PC (with a good audio interface; maybe with a rack-mount case) which would be running some audio apps (Jack, softsynths, sequencer, ...) and could be controlled on stage via MIDI and/or from the notebook using VNC (or NX) or MIDI. Or do you have any suggestion, what could work better with lower latencies? Or is this idea foolish as a whole? I'm also interested if it is worth thinking about Linux HPC clustering in a studio environment (Ardour with MPI?). Thank you! FerD

19 years, 9 months

[linux-audio-dev] one-sample length impulse - HOWTO?

by Andrew Gaydenko

Hi! The aim is to generate audiofile with one/few-sample-length impulse inside silence. Is there suitable software to do the job? Thanks in advance! Andrew

19 years, 9 months

[linux-audio-dev] ALSA USB Driver

by Thomas Vecchione

Ok I apologize if this should go on the user list, wasnt really sure.. I am considering purchasing a PDA that will eventually have the express purpose of recording 24/96 audio, and I was looking at picking up an apogee Mini-me USB. I was wondering how uniform USB audio interfaces were, and the likelihood the Apogee would actually work or not as it is not listed at all in the Alsa wiki. Seablade

19 years, 9 months

[linux-audio-dev] detune

by Hans Fugal

I'm about to write a DSSI/LADSPA plugin that among other things, detunes the signal by up to 15 cents. My understanding is that detuning is accomplished by resampling. If that's the case, what do you do with the time difference? Do you pad/truncate to get the same number of samples you started out with? Wouldn't that introduce undesirables? -- Hans Fugal ; http://hans.fugal.net There's nothing remarkable about it. All one has to do is hit the right keys at the right time and the instrument plays itself. -- Johann Sebastian Bach

19 years, 9 months

[linux-audio-dev] Re: detune

by Bill Schottstaedt

Another method, related to the vocoder but more "real-time" oriented (i.e. sample-by-sample, not fft block processed) uses a bank of Hilbert transforms (for the pitch shift) and a bank of bandpass filters (each harmonic shifted individually to maintain harmonic relations) -- see ssb-bank et al in dsp.scm in the Snd tarball (ssb refers to the single sideband suppressed carrier amplitude modulation generator). sndscm.html has examples.

19 years, 9 months

[linux-audio-dev] hi

by Vaughan Famularo

Hi guys After reading the last few days postings, it is obvious that this list is WAY OVER MY HEAD techwise. You are obviously developers!! Hence the name of the mailing list. Sorry I missed that! The thing is where can I get everyday info about triggering samples or the program to use to do it etc.???? These type of questions are obviously beneath where your heads are operating. So, can someone humble themselves for a moment and give me some ideas regarding the above forementioned?? Thaks in anticipation Vaughan

19 years, 9 months

[linux-audio-dev] timemachine auto recorder patch

by Garett Shulman

Hello, I have created a patch that allows timeachine to automatically start and stop recording based on the audio signal. Recording will start when a sample value exceeds a start threashold. Recording stops when sample values remain below a stop threashold for a period of time. Recording of a new file will then start again when a sample value again exceeds the start threashold. These values as well as enabeling auto recording can all be set at the command line. With this functionallity I would like to be able to launch timemachine backgrounded. Eg: timemachine args & I would also need to be able to send the process SIGINT and SIGTERM signals and have it exit gracefully. I have attempted to add the necessary handlers for this. However if I try to start timemachine backgrounded it just seems to exit and brings jack down with it. If I run it in the foreground and try to killall timemachine it does not exit gracefully but brings jack down as well. In either case I see output from jack like: subgraph starting at time timed out (subgraph_wait_fd = 7, status = 0, state = Triggered) jackd watchdog: timeout - killed jack [1]+ Killed timemachine Any suggestions or ideas are greatly appreciated. Also, the auto recorder functionallity works great asside from the backgrounding and SIGTERM business. Thanks! -Garett PATCH FOLLOWS diff -urN timemachine-0.3.1/src/main.c timemachine-0.3.1.autorecord/src/main.c --- timemachine-0.3.1/src/main.c 2005-09-19 03:41:30.000000000 -0600 +++ timemachine-0.3.1.autorecord/src/main.c 2006-01-14 11:24:13.000000000 -0700 @@ -28,6 +28,10 @@ #include <sndfile.h> #include <gtk/gtk.h> +/* garetts mod here */ +#include <signal.h> +/* end garetts mod */ + #ifdef HAVE_LASH #include <lash/lash.h> @@ -71,6 +75,10 @@ GdkPixbuf *img_on, *img_off, *img_busy; GdkPixbuf *icon_on, *icon_off; +/* garetts mod here */ +void signal_handler(int signal); +/* end garetts mod */ + int main(int argc, char *argv[]) { unsigned int i; @@ -78,13 +86,22 @@ int help = 0; int console = 0; char port_name[32]; + /* garetts mod here */ + int AUTO_RECORD = 0; + signal(SIGINT, signal_handler); + signal(SIGTERM, signal_handler); + float SECONDS_OF_SILENCE_BEFORE_STOP = 5; + float START_THREASHOLD = 0.000005; + float STOP_THREASHOLD = 0; + /* end garetts mod */ + pthread_t dt; #ifdef HAVE_LASH lash_args_t *lash_args = lash_extract_args(&argc, &argv); lash_event_t *event; #endif - while ((opt = getopt(argc, argv, "hic:t:n:p:f:")) != -1) { + while ((opt = getopt(argc, argv, "hic:t:n:p:f:q:w:e:r")) != -1) { switch (opt) { case 'h': help = 1; @@ -111,6 +128,20 @@ case 'f': format_name = optarg; break; + /* garetts mod here */ + case 'q': + SECONDS_OF_SILENCE_BEFORE_STOP = atof(optarg); + break; + case 'w': + START_THREASHOLD = atof(optarg); + break; + case 'e': + STOP_THREASHOLD = atof(optarg); + break; + case 'r': + AUTO_RECORD = 1; + break; + /* end garetts mod */ default: num_ports = 0; break; @@ -132,6 +163,9 @@ fprintf(stderr, "\t-t\tspecify the pre-recording buffer length\n"); fprintf(stderr, "\t-p\tspecify the saved file prefix, may include path\n"); fprintf(stderr, "\t-f\tspecify the saved file format\n"); + fprintf(stderr, "\t-q\tspecify the seconds of silence before stop\n"); + fprintf(stderr, "\t-w\tspecify the start threashold\n"); + fprintf(stderr, "\t-e\tspecify the stop threashold\n"); fprintf(stderr, "\n"); fprintf(stderr, "\tchannels must be in the range 1-8, default %d\n", DEFAULT_NUM_PORTS); @@ -164,7 +198,10 @@ } DEBUG(1, "registering as %s\n", client_name); - process_init(buf_length); + /* garetts mod here + * process_init(buf_length) + */ + process_init(buf_length, SECONDS_OF_SILENCE_BEFORE_STOP, START_THREASHOLD, STOP_THREASHOLD, AUTO_RECORD); #ifdef HAVE_LASH lash_client = lash_init (lash_args, "TimeMachine", @@ -287,6 +324,11 @@ exit(0); } +void signal_handler(int iSignal) { + recording_stop(); + cleanup(); +} + #ifdef HAVE_LASH gboolean idle_cb(gpointer data) { diff -urN timemachine-0.3.1/src/threads.c timemachine-0.3.1.autorecord/src/threads.c --- timemachine-0.3.1/src/threads.c 2005-07-18 08:03:06.000000000 -0600 +++ timemachine-0.3.1.autorecord/src/threads.c 2006-01-14 10:58:18.000000000 -0700 @@ -19,7 +19,7 @@ #include <string.h> #include <stdio.h> #include <unistd.h> -#include <time.h> +#include <sys/time.h> #include <sndfile.h> #include <jack/jack.h> #include <gtk/gtk.h> @@ -46,6 +46,15 @@ static unsigned int disk_read_pos = 0; static unsigned int disk_write_pos = 0; +/* garetts mod here */ +static float seconds_of_silence_before_stop = 0; +static float start_threashold = 0; +static float stop_threashold = 0; +static float seconds_of_silence = 0; +static int sample_rate = 0; +static int autorecord = 0; +/* end garetts mod */ + /* Peak data for meters */ static volatile float peak[MAX_PORTS]; @@ -69,6 +78,32 @@ fprintf(stderr, "bad buffer!\n"); break; } + + /* garetts mod here */ + if (autorecord) { + if (rec) { + for (i = 0; i < nframes; i++) { + if (fabsf(in[i]) <= stop_threashold) { + seconds_of_silence = seconds_of_silence + 1.0/sample_rate; + } + else { + seconds_of_silence = 0; + } + if (seconds_of_silence > seconds_of_silence_before_stop) { + recording_stop(); + } + } + } + else { + for (i = 0; i < nframes; i++) { + if (fabsf(in[i]) > start_threashold) { + recording_start(); + break; + } + } + } + } + /* end garetts mod */ for (i = 0; i < nframes; i++) { if (fabsf(in[i]) > peak[port]) { @@ -202,7 +236,10 @@ return 0; } -void process_init(unsigned int time) +/* garetts mod here + * void process_init(unsigned int time) + */ +void process_init(unsigned int time, float secs_of_silence_before_stop, float strt_threashold, float stp_threashold, int use_autorecord) { unsigned int port; @@ -220,6 +257,14 @@ pre_size = time * jack_get_sample_rate(client); pre_time = time; + /* garetts mod here */ + seconds_of_silence_before_stop = secs_of_silence_before_stop; + start_threashold = strt_threashold; + stop_threashold = stp_threashold; + sample_rate = jack_get_sample_rate(client); + autorecord = use_autorecord; + /* end garetts mod */ + for (port = 0; port < num_ports; port++) { pre_buffer[port] = calloc(pre_size, sizeof(float)); disk_buffer[port] = calloc(DISK_SIZE, sizeof(float)); diff -urN timemachine-0.3.1/src/threads.h timemachine-0.3.1.autorecord/src/threads.h --- timemachine-0.3.1/src/threads.h 2005-07-18 08:03:06.000000000 -0600 +++ timemachine-0.3.1.autorecord/src/threads.h 2006-01-14 10:56:37.000000000 -0700 @@ -9,7 +9,10 @@ int process(jack_nframes_t nframes, void *arg); -void process_init(unsigned int time); +/* garetts mod here + * void process_init(unsigned int time) + */ +void process_init(unsigned int time, float secs_of_silence_before_stop, float strt_threashold, float stp_threashold, int use_autorecord); int writer_thread(void *d);

19 years, 9 months

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Linux-audio-dev