49a50,61 > # use ALSA Sequencer MIDI interface > CFLAGS+= -DHAVE_ASEQ > LIBS+= -lasound > > # JACKify > CFLAGS+= -DHAVE_JACK > LIBS+= -ljack > > # Do not use libsamplerate (experimental). > #CFLAGS+= -DHAVE_SRC > #LIBS+= -lsamplerate > 1060a1061,1194 > > > #ifdef HAVE_ASEQ > #include > // getpid() > #include > #include > > snd_seq_t *seq= NULL; > int aseq_stop=0; // only modify in main thread. > > void aseq_close(void) { > if(!seq) return; > printf("closing alsa midi..."); > snd_seq_close(seq); > seq=NULL; > } > > #if 0 > inline void midi_close(void) { > if(!seq) return; > aseq_stop=1; > sleep(1); > //pthread_join(t_midi,NULL); > aseq_close(); > } > #endif > > int aseq_open(char *port_name) { > int err=0; > snd_seq_addr_t port; > char seq_name[32]; > snprintf(seq_name,32,"beatrix-%i",(int) getpid()); > > if (seq) return (-1); > > if ((err = snd_seq_open(&seq, "default", SND_SEQ_OPEN_INPUT, 0)) <0 ) { > fprintf(stderr,"cannot open alsa sequencer: %s\n", snd_strerror(err)); > seq=NULL; > return(-1); > } > > if ((err = snd_seq_set_client_name(seq, seq_name)) <0 ) { > fprintf(stderr,"cannot set client name: %s\n", snd_strerror(err)); > aseq_close(); > return(-1); > } > > if ((err = snd_seq_create_simple_port(seq, "MTC in", > SND_SEQ_PORT_CAP_WRITE | SND_SEQ_PORT_CAP_SUBS_WRITE, > SND_SEQ_PORT_TYPE_MIDI_GENERIC | SND_SEQ_PORT_TYPE_APPLICATION)) < 0) { > fprintf(stderr,"cannot create port: %s\n", snd_strerror(err)); > aseq_close(); > return(-1); > } > > if (port_name) { > err = snd_seq_parse_address(seq, &port, port_name); > if (err < 0) { > fprintf(stderr,"Cannot find port %s - %s\n", port_name, snd_strerror(err)); > } > err = snd_seq_connect_from(seq, 0, port.client, port.port); > if (err < 0) { > fprintf(stderr,"Cannot connect from port %d:%d - %s\n", port.client, port.port, snd_strerror(err)); > } > } > > snd_seq_nonblock(seq, 1); > > /* Load the tables */ > > loadControllerTable (); > loadKeyTableA (); > loadKeyTableB (); > loadKeyTableC (); > loadStatusTable (); > > return(0); > } > > void process_seq_event(const snd_seq_event_t *ev) { > // see "snd_seq_event_type" file:///usr/share/doc/libasound2-doc/html/group___seq_events.html > switch(ev->type) { > case SND_SEQ_EVENT_NOTEON: > //printf("DEBUG KEY on %i %i\n",ev->data.note.channel, ev->data.note.note); > keyTable = (unsigned char *) statusTable[MIDI_NOTEON|ev->data.note.channel].handback; > oscKeyOn (keyTable[ev->data.note.note]); > break; > case SND_SEQ_EVENT_NOTEOFF: > //printf("DEBUG KEY off %i %i\n",ev->data.note.channel, ev->data.note.note); > keyTable = (unsigned char *) statusTable[MIDI_NOTEOFF|ev->data.note.channel].handback; > oscKeyOff (keyTable[ev->data.note.note]); > break; > case SND_SEQ_EVENT_PGMCHANGE: > printf("DEBUG Programchange %i\n",ev->data.control.value); > installProgram(ev->data.control.value); > break; > case SND_SEQ_EVENT_CONTROLLER: > printf("DEBUG Controller [%i] %i -> %i\n",ev->data.control.channel, ev->data.control.param, ev->data.control.value); > ctrlvec = (void (**) (unsigned char)) statusTable[MIDI_CTL_CHANGE|ev->data.control.channel].handback; > (ctrlvec[ev->data.control.param])(ev->data.control.value); > break; > default: > printf("unhandled midi event: %i\n", ev->type); > assert(1); > break; > } > } > > void *aseq_run(void *arg) { > int err; > int npfds = 0; > struct pollfd *pfds; > > npfds = snd_seq_poll_descriptors_count(seq, POLLIN); > pfds = alloca(sizeof(*pfds) * npfds); > for (;;) { > snd_seq_poll_descriptors(seq, pfds, npfds, POLLIN); > if (poll(pfds, npfds, 1) < 0) break; > do { > snd_seq_event_t *event; > err = snd_seq_event_input(seq, &event); > if (err < 0) break; > if (event) process_seq_event(event); > } while (err > 0); > if (aseq_stop) break; > } > pthread_exit(NULL); > return (NULL); > } > > #endif > > /* vi:set ts=8 sts=2 sw=2: */ 44a45,49 > #ifdef HAVE_ASEQ > extern void *aseq_run(void *arg); > extern int aseq_open(char *port_name); > #endif > 122a123,289 > #ifdef HAVE_JACK > #include > > jack_client_t *j_client = NULL; > jack_port_t **j_output_port; > jack_default_audio_sample_t **j_output_bufferptrs; // > jack_default_audio_sample_t bufJ [2][BUFFER_SIZE_SAMPLES]; > > jack_nframes_t jack_splrate=44100; > jack_nframes_t jack_fragsize=128; > > > /* when jack shuts down... */ > void jack_shutdown_callback(void *arg) { > fprintf(stderr,"jack server shut us down.\n"); > jack_deactivate(j_client); > j_client=NULL; > } > > int jack_srate_callback(jack_nframes_t nframes, void *arg) { > jack_splrate= nframes; > return(0); > } > > int jack_bufsiz_callback(jack_nframes_t nframes, void *arg) { > jack_fragsize= nframes; > return(0); > } > #ifdef HAVE_SRC > #include > #include > #endif > > int jack_audio_callback (jack_nframes_t nframes, void *arg) { > jack_default_audio_sample_t **out = j_output_bufferptrs; > int i,s; > jack_nframes_t my_tot_latency = 0; > > for (i=0;i jack_nframes_t my_latency = jack_port_get_total_latency(j_client,j_output_port[i]); > if (my_latency > my_tot_latency) my_tot_latency = my_latency; > //printf("DEBUG: c=%i pl=%i tl=%i\n",i,jack_port_get_latency(j_output_port[i]) ,jack_port_get_total_latency(j_client,j_output_port[i])); > out[i] = jack_port_get_buffer (j_output_port[i], nframes); > //memset(out[i],0, sizeof (jack_default_audio_sample_t) * nframes); > } > > static unsigned char * obuf; > static int boffset = BUFFER_SIZE_SAMPLES; > > float m_fResampleRatio = (float) jack_splrate / (float)SampleRateD; > #ifndef HAVE_SRC > static int info =0; > #if 1 > if (!info) { > info=1; > printf ("JACK: integer resampling factor: %f\nJACK: suggest: osc.tuning=%f\n", > floorf(m_fResampleRatio+.5), > 440.0/m_fResampleRatio*floorf(m_fResampleRatio+.5)); > } > m_fResampleRatio = floorf(m_fResampleRatio+.5); > # endif > #endif > > jack_nframes_t out_off = 0; > > while (out_off < nframes) { > int remain_src = BUFFER_SIZE_SAMPLES - boffset; > int remain_dst = nframes - out_off; > if (remain_src <= 0) { > boffset = 0; remain_src+=BUFFER_SIZE_SAMPLES; > // generate sound > oscGenerateFragment (bufA, BUFFER_SIZE_SAMPLES); > obuf = preamp (bufA, bufB, BUFFER_SIZE_SAMPLES); > obuf = reverb (obuf, bufB, BUFFER_SIZE_SAMPLES); > whirlProc (obuf, bufC, BUFFER_SIZE_SAMPLES); > > // convert interlaced channels > float nfactor = 1.0 / ((float) 0x8000); > for (s=0;s for (i=0;i //bufJ[i][s]= (((jack_default_audio_sample_t)(((signed short*)bufA)[s]))/32768.0); > //bufJ[i][s]= ((((signed short*)obuf)[s]))/32768.0; > bufJ[i][s]= ((jack_default_audio_sample_t)(((signed short*)bufC)[audio_channels*s+i]))*nfactor; > } > > jack_nframes_t nread = (jack_nframes_t) ceilf((float)remain_dst / m_fResampleRatio); > if (nread > remain_src) nread = remain_src; > jack_nframes_t nwrite = (jack_nframes_t) floorf((float)nread * m_fResampleRatio); > //printf("DEBUG %i %i %f\n", nread, nwrite, m_fResampleRatio); > #ifdef HAVE_SRC > SRC_DATA src_data; > src_data.input_frames = nread; > src_data.output_frames = nwrite; > src_data.end_of_input = 0; > src_data.src_ratio = m_fResampleRatio; > src_data.input_frames_used = 0; > src_data.output_frames_gen = 0; > > for (i=0;i< audio_channels; i++ ) { > src_data.data_in = &(bufJ[i][boffset]); > src_data.data_out = &(out[i][out_off]); > src_simple (&src_data, SRC_SINC_FASTEST, 1) ; > //src_simple (&src_data, SRC_SINC_MEDIUM_QUALITY, 1) ; > //src_simple (&src_data, SRC_ZERO_ORDER_HOLD, 1) ; > //src_simple (&src_data, SRC_LINEAR, 1) ; > //src_simple (&src_data, SRC_SINC_BEST_QUALITY, 1) ; > } > out_off+=src_data.output_frames_gen; > boffset+=src_data.input_frames_used; > #else > assert (m_fResampleRatio == 2.0 || m_fResampleRatio==1.0 || m_fResampleRatio==4.0 || m_fResampleRatio==3.0); > int r,stride = (int) rintf(m_fResampleRatio); > > for (i=0;i< audio_channels; i++ ) { > for (s=0;s for (r=0;r out[i][out_off+r+stride*s] = bufJ[i][boffset+s]; > } > } > out_off+=stride*nread; > boffset+=nread; > #endif > } > return(0); > } > > int open_jack(void) { > int i; > i = 0; > do { > char jackid[16]; > snprintf(jackid,16,"beatrix-%i",i); > j_client = jack_client_new (jackid); > } while (j_client == 0 && i++<16); > > if (!j_client) { > fprintf(stderr, "could not connect to jack.\n"); > return(1); > } > audio_channels=audio_channels_requested; > audio_format = audio_format_requested; > > jack_on_shutdown (j_client, jack_shutdown_callback, NULL); > jack_set_process_callback(j_client,jack_audio_callback,NULL); > jack_set_sample_rate_callback (j_client, jack_srate_callback, NULL); > jack_set_buffer_size_callback (j_client, jack_bufsiz_callback, NULL); > > j_output_port= calloc(audio_channels,sizeof(jack_port_t*)); > j_output_bufferptrs = calloc(audio_channels,sizeof(jack_default_audio_sample_t*)); > > for (i=0;i char channelid[16]; > snprintf(channelid,16,"output-%i",i); > j_output_port[i] = jack_port_register (j_client, channelid, JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0); > if (!j_output_port[i]) { > fprintf(stderr, "no more jack ports available.\n"); > jack_client_close (j_client); > return(1); > } > } > jack_srate_callback(jack_get_sample_rate(j_client),NULL); // init j_srate > jack_bufsiz_callback(jack_get_buffer_size(j_client),NULL); // init j_bufsiz & alloc > > return(0); > } > #endif > 129a297,303 > #ifdef HAVE_JACK > if (open_jack()) { > perror ("could not connect to JACK."); > exit(1); > } > #else /* NO JACK - OSS */ > 211a386 > #endif /* NO JACK = OSS */ 408a584,585 > printf ("-R set samplerate for synth. default 22050.\n"); > printf (" jack-sr must be a multiple of this.\n"); 455a633,635 > else if (!strcmp (av, "-R")) { > state = 2; > } 479a660,665 > else if (state == 2) { > SampleRateI = atoi (av); > SampleRateD = (double) SampleRateI; > printf(" beatrix internal samplerate: %.1f SPS\n",SampleRateD); > state = 0; > } 541a728,731 > # ifdef HAVE_ASEQ > if (!aseq_open(NULL)) > k= pthread_create(&t_midi, NULL, aseq_run, NULL); > # else 542a733 > # endif 595a787,792 > #ifdef HAVE_JACK > jack_activate(j_client); > while (j_client) > sleep (1); // jack callback is doing this: > return(0); > #else 629a827 > #endif 636a835,836 > > /* vi:set ts=8 sts=2 sw=2: */