mirror of
https://github.com/saitohirga/WSJT-X.git
synced 2024-11-04 08:21:17 -05:00
Update the list of Fortran LUs.
Remove several unused files. Replace use of rfile2 with fortran I/O using access='stream'. git-svn-id: svn+ssh://svn.code.sf.net/p/wsjt/wsjt/branches/map65@604 ab8295b8-cf94-4d9e-aec4-7959e3be5d79
This commit is contained in:
Joe Taylor
parent
18c2577626
commit
47c76fd13b
7
astro.F
7
astro.F
@ -34,8 +34,11 @@ C NB: may want to smooth the Tsky map to 10 degrees or so.
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read(13) nsky
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close(13)
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#else
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call rfile2(fname,nsky,129600,nr)
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if(nr.ne.129600) go to 10
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open(13,file=fname,status='old',access='stream',err=10)
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read(13) nsky
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close(13)
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! call rfile2(fname,nsky,129600,nr)
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! if(nr.ne.129600) go to 10
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#endif
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ltsky=.true.
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first=.false.
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@ -1,13 +1,13 @@
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! Fortran logical units used in WSJT6
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!
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! 10 wave files read from disk
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! 10
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! 11 decoded.txt
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! 12 decoded.ave
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! 13 tsky.dat
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! 14 azel.dat
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! 15
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! 16 c:/wsjt.reg
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! 17 wave files written to disk
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! 16
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! 17 saved *.tf2 files
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! 18 test file to be transmitted (wsjtgen.f90)
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! 19 messages.txt
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! 20 bandmap.txt
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@ -18,7 +18,7 @@
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! 25 meas25.dat
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! 26 tmp26.txt
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! 27 dphi.txt
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! 28 saved *.tf2 files
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! 28
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! 29 debug.txt
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!------------------------------------------------ ftn_init
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subroutine ftn_init
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2
map65.py
2
map65.py
@ -1,4 +1,4 @@
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#---------------------------------------------------------------------- MAP65
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#--------------------------------------------------------------------- MAP65
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# $Date$ $Revision$
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#
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from Tkinter import *
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@ -91,6 +91,7 @@ subroutine recvpkt(iarg)
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rxnoise=10.0*log10(sqave) - 48.0
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kxp=k
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! This may be a bad idea, because of non-reentrant Fortran I/O?
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if(mode.eq.'Measur') then
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npkt=npkt+1
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if(npkt.ge.551) then
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568
start_alsa.c
568
start_alsa.c
@ -1,568 +0,0 @@
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#include <stdio.h>
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#include <stdlib.h>
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#include <pthread.h>
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#include <alsa/asoundlib.h>
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#include <inttypes.h>
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#include <time.h>
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#include <sys/time.h>
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#include "fivehz.h"
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#if 1
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#define ALSA_LOG
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#endif
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#if 0
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#define ALSA_LOG_BUFFERS
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#endif
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#if 0
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#define ALSA_PLAYBACK_LOG
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#define ALSA_CAPTURE_LOG
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#endif
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#define BUFFER_TIME 2000*1000
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typedef struct alsa_driver_s {
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snd_pcm_t *audio_fd;
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int capabilities;
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int open_mode;
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int has_pause_resume;
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int is_paused;
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uint32_t output_sample_rate, input_sample_rate;
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double sample_rate_factor;
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uint32_t num_channels;
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uint32_t bits_per_sample;
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uint32_t bytes_per_frame;
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uint32_t bytes_in_buffer; /* number of bytes writen to audio hardware */
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int16_t *app_buffer_y1;
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int16_t *app_buffer_y2;
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int *app_buffer_offset;
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int app_buffer_length;
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double *Tsec;
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double *tbuf;
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int *ibuf;
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int *ndsec;
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int *tx_ok;
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int tx_starting;
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int tx_offset;
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int *tr_period;
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int *nwave;
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int *nmode;
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int *transmitting;
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snd_pcm_uframes_t buffer_size;
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snd_pcm_uframes_t period_size;
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int32_t mmap;
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} alsa_driver_t;
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alsa_driver_t alsa_driver_playback;
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alsa_driver_t alsa_driver_capture;
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void *alsa_capture_buffers[2];
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void *alsa_playback_buffers[2];
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static snd_output_t *jcd_out;
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/*
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* open the audio device for writing to
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*/
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static int ao_alsa_open(alsa_driver_t *this_gen, int32_t *input_rate, snd_pcm_stream_t direction ) {
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alsa_driver_t *this = (alsa_driver_t *) this_gen;
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char *pcm_device;
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snd_pcm_hw_params_t *params;
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snd_pcm_sw_params_t *swparams;
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snd_pcm_access_mask_t *mask;
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snd_pcm_uframes_t period_size_min;
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snd_pcm_uframes_t period_size_max;
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snd_pcm_uframes_t buffer_size_min;
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snd_pcm_uframes_t buffer_size_max;
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snd_pcm_format_t format;
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uint32_t buffer_time=BUFFER_TIME;
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snd_pcm_uframes_t buffer_time_to_size;
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int err, dir;
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int open_mode=1; /* NONBLOCK */
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/* int open_mode=0; BLOCK */
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uint32_t rate=*input_rate;
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this->input_sample_rate=*input_rate;
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snd_pcm_hw_params_alloca(¶ms);
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snd_pcm_sw_params_alloca(&swparams);
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err = snd_output_stdio_attach(&jcd_out, stdout, 0);
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this->num_channels = 2;
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if (direction == SND_PCM_STREAM_PLAYBACK) {
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pcm_device="plug:front";
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} else {
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pcm_device="default";
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}
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#ifdef ALSA_LOG
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printf("audio_alsa_out: Audio Device name = %s\n",pcm_device);
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printf("audio_alsa_out: Number of channels = %d\n",this->num_channels);
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#endif
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if (this->audio_fd) {
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printf("audio_alsa_out:Already open...WHY!");
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snd_pcm_close (this->audio_fd);
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this->audio_fd = NULL;
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}
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this->bytes_in_buffer = 0;
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/*
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* open audio device
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*/
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err=snd_pcm_open(&this->audio_fd, pcm_device, direction, open_mode);
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if(err <0 ) {
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printf ("audio_alsa_out: snd_pcm_open() of %s failed: %s\n", pcm_device, snd_strerror(err));
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printf ("audio_alsa_out: >>> check if another program already uses PCM <<<\n");
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return 0;
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}
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/* printf ("audio_alsa_out: snd_pcm_open() opened %s\n", pcm_device); */
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/* We wanted non blocking open but now put it back to normal */
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//snd_pcm_nonblock(this->audio_fd, 0);
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snd_pcm_nonblock(this->audio_fd, 1);
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/*
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* configure audio device
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*/
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err = snd_pcm_hw_params_any(this->audio_fd, params);
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if (err < 0) {
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printf ("audio_alsa_out: broken configuration for this PCM: no configurations available: %s\n",
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snd_strerror(err));
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goto close;
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}
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/* set interleaved access */
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if (this->mmap != 0) {
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mask = alloca(snd_pcm_access_mask_sizeof());
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snd_pcm_access_mask_none(mask);
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snd_pcm_access_mask_set(mask, SND_PCM_ACCESS_MMAP_INTERLEAVED);
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snd_pcm_access_mask_set(mask, SND_PCM_ACCESS_MMAP_NONINTERLEAVED);
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snd_pcm_access_mask_set(mask, SND_PCM_ACCESS_MMAP_COMPLEX);
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err = snd_pcm_hw_params_set_access_mask(this->audio_fd, params, mask);
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if (err < 0) {
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printf ( "audio_alsa_out: mmap not availiable, falling back to compatiblity mode\n");
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this->mmap=0;
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err = snd_pcm_hw_params_set_access(this->audio_fd, params,
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SND_PCM_ACCESS_RW_NONINTERLEAVED);
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}
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} else {
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err = snd_pcm_hw_params_set_access(this->audio_fd, params,
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SND_PCM_ACCESS_RW_NONINTERLEAVED);
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}
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if (err < 0) {
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printf ( "audio_alsa_out: access type not available: %s\n", snd_strerror(err));
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goto close;
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}
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/* set the sample format S16 */
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/* ALSA automatically appends _LE or _BE depending on the CPU */
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format = SND_PCM_FORMAT_S16;
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err = snd_pcm_hw_params_set_format(this->audio_fd, params, format );
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if (err < 0) {
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printf ( "audio_alsa_out: sample format non available: %s\n", snd_strerror(err));
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goto close;
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}
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/* set the number of channels */
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err = snd_pcm_hw_params_set_channels(this->audio_fd, params, this->num_channels);
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if (err < 0) {
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printf ( "audio_alsa_out: Cannot set number of channels to %d (err=%d:%s)\n",
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this->num_channels, err, snd_strerror(err));
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goto close;
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}
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#if SND_LIB_VERSION >= 0x010009
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/* Restrict a configuration space to contain only real hardware rates */
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err = snd_pcm_hw_params_set_rate_resample(this->audio_fd, params, 0);
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#endif
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/* set the stream rate [Hz] */
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dir=0;
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err = snd_pcm_hw_params_set_rate_near(this->audio_fd, params, &rate, &dir);
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if (err < 0) {
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printf ( "audio_alsa_out: rate not available: %s\n", snd_strerror(err));
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goto close;
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}
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this->output_sample_rate = (uint32_t)rate;
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if (this->input_sample_rate != this->output_sample_rate) {
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printf ( "audio_alsa_out: audio rate : %d requested, %d provided by device/sec\n",
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this->input_sample_rate, this->output_sample_rate);
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}
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buffer_time_to_size = ( (uint64_t)buffer_time * rate) / 1000000;
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err = snd_pcm_hw_params_get_buffer_size_min(params, &buffer_size_min);
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err = snd_pcm_hw_params_get_buffer_size_max(params, &buffer_size_max);
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dir=0;
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err = snd_pcm_hw_params_get_period_size_min(params, &period_size_min,&dir);
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dir=0;
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err = snd_pcm_hw_params_get_period_size_max(params, &period_size_max,&dir);
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#ifdef ALSA_LOG_BUFFERS
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printf("Buffer size range from %lu to %lu\n",buffer_size_min, buffer_size_max);
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printf("Period size range from %lu to %lu\n",period_size_min, period_size_max);
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printf("Buffer time size %lu\n",buffer_time_to_size);
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#endif
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this->buffer_size = buffer_time_to_size;
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if (buffer_size_max < this->buffer_size)
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this->buffer_size = buffer_size_max;
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if (buffer_size_min > this->buffer_size)
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this->buffer_size = buffer_size_min;
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this->period_size = this->buffer_size/8;
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if (this->period_size > 2048)
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this->period_size = 2048;
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this->buffer_size = this->period_size*8;
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#ifdef ALSA_LOG_BUFFERS
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printf("To choose buffer_size = %ld\n",this->buffer_size);
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printf("To choose period_size = %ld\n",this->period_size);
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#endif
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#if 0
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/* Set period to buffer size ratios at 8 periods to 1 buffer */
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dir=-1;
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periods=8;
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err = snd_pcm_hw_params_set_periods_near(this->audio_fd, params, &periods ,&dir);
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if (err < 0) {
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xprintf (this->class->xine, XINE_VERBOSITY_DEBUG,
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"audio_alsa_out: unable to set any periods: %s\n", snd_strerror(err));
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goto close;
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}
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/* set the ring-buffer time [us] (large enough for x us|y samples ...) */
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dir=0;
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err = snd_pcm_hw_params_set_buffer_time_near(this->audio_fd, params, &buffer_time, &dir);
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if (err < 0) {
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xprintf (this->class->xine, XINE_VERBOSITY_DEBUG,
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"audio_alsa_out: buffer time not available: %s\n", snd_strerror(err));
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goto close;
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}
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#endif
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#if 1
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/* set the period time [us] (interrupt every x us|y samples ...) */
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dir=0;
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err = snd_pcm_hw_params_set_period_size_near(this->audio_fd, params, &(this->period_size), &dir);
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if (err < 0) {
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printf ( "audio_alsa_out: period time not available: %s\n", snd_strerror(err));
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goto close;
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}
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#endif
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dir=0;
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err = snd_pcm_hw_params_get_period_size(params, &(this->period_size), &dir);
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dir=0;
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err = snd_pcm_hw_params_set_buffer_size_near(this->audio_fd, params, &(this->buffer_size));
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if (err < 0) {
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printf ( "audio_alsa_out: buffer time not available: %s\n", snd_strerror(err));
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goto close;
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}
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err = snd_pcm_hw_params_get_buffer_size(params, &(this->buffer_size));
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#ifdef ALSA_LOG_BUFFERS
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printf("was set period_size = %ld\n",this->period_size);
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printf("was set buffer_size = %ld\n",this->buffer_size);
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#endif
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if (2*this->period_size > this->buffer_size) {
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printf ( "audio_alsa_out: buffer to small, could not use\n");
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goto close;
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}
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/* write the parameters to device */
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err = snd_pcm_hw_params(this->audio_fd, params);
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if (err < 0) {
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printf ( "audio_alsa_out: pcm hw_params failed: %s\n", snd_strerror(err));
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goto close;
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}
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/* Check for pause/resume support */
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this->has_pause_resume = ( snd_pcm_hw_params_can_pause (params)
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&& snd_pcm_hw_params_can_resume (params) );
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// printf( "audio_alsa_out:open pause_resume=%d\n", this->has_pause_resume);
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this->sample_rate_factor = (double) this->output_sample_rate / (double) this->input_sample_rate;
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this->bytes_per_frame = snd_pcm_frames_to_bytes (this->audio_fd, 1);
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/*
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* audio buffer size handling
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*/
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/* Copy current parameters into swparams */
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err = snd_pcm_sw_params_current(this->audio_fd, swparams);
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if (err < 0) {
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printf ( "audio_alsa_out: Unable to determine current swparams: %s\n", snd_strerror(err));
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goto close;
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}
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/* align all transfers to 1 sample */
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err = snd_pcm_sw_params_set_xfer_align(this->audio_fd, swparams, 1);
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if (err < 0) {
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printf ( "audio_alsa_out: Unable to set transfer alignment: %s\n", snd_strerror(err));
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goto close;
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}
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/* allow the transfer when at least period_size samples can be processed */
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err = snd_pcm_sw_params_set_avail_min(this->audio_fd, swparams, this->period_size);
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if (err < 0) {
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printf ( "audio_alsa_out: Unable to set available min: %s\n", snd_strerror(err));
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goto close;
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}
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if (direction == SND_PCM_STREAM_PLAYBACK) {
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/* start the transfer when the buffer contains at least period_size samples */
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err = snd_pcm_sw_params_set_start_threshold(this->audio_fd, swparams, this->buffer_size);
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} else {
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err = snd_pcm_sw_params_set_start_threshold(this->audio_fd, swparams, -1);
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}
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if (err < 0) {
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printf ( "audio_alsa_out: Unable to set start threshold: %s\n", snd_strerror(err));
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goto close;
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}
|
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|
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if (direction == SND_PCM_STREAM_PLAYBACK) {
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/* never stop the transfer, even on xruns */
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err = snd_pcm_sw_params_set_stop_threshold(this->audio_fd, swparams, this->buffer_size);
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} else {
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err = snd_pcm_sw_params_set_stop_threshold(this->audio_fd, swparams, this->buffer_size);
|
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}
|
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if (err < 0) {
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printf ( "audio_alsa_out: Unable to set stop threshold: %s\n", snd_strerror(err));
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goto close;
|
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}
|
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|
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/* Install swparams into current parameters */
|
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err = snd_pcm_sw_params(this->audio_fd, swparams);
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if (err < 0) {
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printf ( "audio_alsa_out: Unable to set swparams: %s\n", snd_strerror(err));
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goto close;
|
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}
|
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#ifdef ALSA_LOG
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snd_pcm_dump_setup(this->audio_fd, jcd_out);
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snd_pcm_sw_params_dump(swparams, jcd_out);
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#endif
|
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return this->output_sample_rate;
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|
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close:
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snd_pcm_close (this->audio_fd);
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this->audio_fd=NULL;
|
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return 0;
|
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}
|
||||
|
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int16_t zero_buffer[65536];
|
||||
|
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int playback_callback(alsa_driver_t *alsa_driver_playback) {
|
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alsa_driver_t *this = alsa_driver_playback;
|
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int result;
|
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struct timeval tv;
|
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double stime;
|
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int nsec;
|
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int i,n;
|
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static int ic;
|
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snd_pcm_sframes_t delay;
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static short int n2;
|
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int16_t b0[2048];
|
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|
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// printf("playback callback\n");
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snd_pcm_delay(this->audio_fd, &delay);
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gettimeofday(&tv, NULL);
|
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stime = (double) tv.tv_sec + ((double)tv.tv_usec / 1000000.0) +
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*(this->ndsec) * 0.1;
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// stime = stime + ((double)delay / (double)(this->output_sample_rate));
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*(this->Tsec) = stime;
|
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//printf("PLAY:TIME, %lf, %ld, %ld, %d\n", stime, delay, this->output_sample_rate, *this->ndsec);
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if(!(this->tx_starting) && (*(this->tx_ok)) ) {
|
||||
nsec = (int)stime;
|
||||
n = nsec / *(this->tr_period);
|
||||
ic = (int)(stime - *(this->tr_period) * n) * this->output_sample_rate;
|
||||
ic = ic % *(this->nwave);
|
||||
this->tx_offset = ic;
|
||||
}
|
||||
this->tx_starting = *(this->tx_ok);
|
||||
*(this->transmitting) = *(this->tx_ok);
|
||||
if(*(this->tx_ok)) {
|
||||
/*
|
||||
alsa_playback_buffers[0] = this->app_buffer_y1 + this->tx_offset;
|
||||
alsa_playback_buffers[1] = this->app_buffer_y1 + this->tx_offset;
|
||||
*/
|
||||
alsa_playback_buffers[0] = b0;
|
||||
alsa_playback_buffers[1] = b0;
|
||||
for(i=0; i<this->period_size; i++) {
|
||||
n2=this->app_buffer_y1[ic];
|
||||
addnoise_(&n2);
|
||||
b0[i]=n2;
|
||||
ic++;
|
||||
if(ic>=*this->nwave) {
|
||||
if(*this->nmode==2) {
|
||||
*this->tx_ok=0;
|
||||
ic--;
|
||||
}
|
||||
else
|
||||
ic = ic % *this->nwave; //Wrap buffer pointer
|
||||
}
|
||||
}
|
||||
} else {
|
||||
alsa_playback_buffers[0] = zero_buffer;
|
||||
alsa_playback_buffers[1] = zero_buffer;
|
||||
}
|
||||
result = snd_pcm_writen(this->audio_fd, alsa_playback_buffers, this->period_size);
|
||||
this->tx_offset += this->period_size;
|
||||
if (result != this->period_size) {
|
||||
printf("Playback write failed. Expected %lu samples, sent only %d\n", this->period_size, result);
|
||||
#ifdef ALSA_PLAYBACK_LOG
|
||||
snd_pcm_status_t *pcm_stat;
|
||||
snd_pcm_status_alloca(&pcm_stat);
|
||||
snd_pcm_status(this->audio_fd, pcm_stat);
|
||||
snd_pcm_status_dump(pcm_stat, jcd_out);
|
||||
#endif
|
||||
}
|
||||
fivehztx_(); //Call fortran routine
|
||||
return result;
|
||||
}
|
||||
|
||||
int capture_callback(alsa_driver_t *alsa_driver_capture) {
|
||||
alsa_driver_t *this = alsa_driver_capture;
|
||||
int result;
|
||||
struct timeval tv;
|
||||
double stime;
|
||||
int ib;
|
||||
snd_pcm_sframes_t delay;
|
||||
#ifdef ALSA_CAPTURE_LOG
|
||||
printf("capture callback %d samples\n", this->period_size);
|
||||
#endif
|
||||
#ifdef ALSA_CAPTURE_LOG
|
||||
snd_pcm_status_t *pcm_stat;
|
||||
snd_pcm_status_alloca(&pcm_stat);
|
||||
snd_pcm_status(this->audio_fd, pcm_stat);
|
||||
snd_pcm_status_dump(pcm_stat, jcd_out);
|
||||
#endif
|
||||
snd_pcm_delay(this->audio_fd, &delay);
|
||||
gettimeofday(&tv, NULL);
|
||||
stime = (double) tv.tv_sec + ((double)tv.tv_usec / 1000000.0) +
|
||||
*(this->ndsec) * 0.1;
|
||||
// stime = stime - ((double)delay / (double)(this->output_sample_rate));
|
||||
*(this->Tsec) = stime;
|
||||
ib=*(this->ibuf);
|
||||
this->tbuf[ib] = stime;
|
||||
//printf("CAP:TIME, %d, %lf, %ld, %ld, %d\n",ib, stime, delay, this->output_sample_rate, *this->ndsec);
|
||||
ib++;
|
||||
if(ib>=1024)
|
||||
ib = 0;
|
||||
*(this->ibuf) = ib;
|
||||
|
||||
alsa_capture_buffers[0]=this->app_buffer_y1 + *(this->app_buffer_offset);
|
||||
alsa_capture_buffers[1]=this->app_buffer_y2 + *(this->app_buffer_offset);
|
||||
result = snd_pcm_readn(this->audio_fd, alsa_capture_buffers, this->period_size);
|
||||
*(this->app_buffer_offset) += this->period_size;
|
||||
if ( *(this->app_buffer_offset) >= this->app_buffer_length )
|
||||
*(this->app_buffer_offset)=0; /* FIXME: implement proper wrapping */
|
||||
#ifdef ALSA_CAPTURE_LOG
|
||||
printf("result=%d\n",result);
|
||||
snd_pcm_status(this->audio_fd, pcm_stat);
|
||||
snd_pcm_status_dump(pcm_stat, jcd_out);
|
||||
#endif
|
||||
fivehz_(); //Call fortran routine
|
||||
return result;
|
||||
}
|
||||
|
||||
int playback_xrun(alsa_driver_t *alsa_driver_playback) {
|
||||
alsa_driver_t *this = alsa_driver_playback;
|
||||
snd_pcm_status_t *pcm_stat;
|
||||
snd_pcm_status_alloca(&pcm_stat);
|
||||
printf("playback xrun\n");
|
||||
snd_pcm_status(this->audio_fd, pcm_stat);
|
||||
snd_pcm_status_dump(pcm_stat, jcd_out);
|
||||
snd_pcm_prepare(this->audio_fd);
|
||||
return 0;
|
||||
}
|
||||
|
||||
int capture_xrun(alsa_driver_t *alsa_driver_capture) {
|
||||
alsa_driver_t *this = alsa_driver_capture;
|
||||
snd_pcm_status_t *pcm_stat;
|
||||
snd_pcm_status_alloca(&pcm_stat);
|
||||
printf("capture xrun\n");
|
||||
snd_pcm_status(this->audio_fd, pcm_stat);
|
||||
snd_pcm_status_dump(pcm_stat, jcd_out);
|
||||
return 0;
|
||||
}
|
||||
|
||||
void ao_alsa_loop(void *iarg) {
|
||||
int playback_nfds;
|
||||
int capture_nfds;
|
||||
struct pollfd *pfd;
|
||||
int nfds;
|
||||
int capture_index;
|
||||
unsigned short playback_revents;
|
||||
unsigned short capture_revents;
|
||||
playback_nfds = snd_pcm_poll_descriptors_count (
|
||||
alsa_driver_playback.audio_fd);
|
||||
capture_nfds = snd_pcm_poll_descriptors_count (
|
||||
alsa_driver_capture.audio_fd);
|
||||
pfd = (struct pollfd *) malloc (sizeof (struct pollfd) *
|
||||
(playback_nfds + capture_nfds));
|
||||
|
||||
nfds=0;
|
||||
snd_pcm_poll_descriptors (alsa_driver_playback.audio_fd,
|
||||
&pfd[0],
|
||||
playback_nfds);
|
||||
nfds += playback_nfds;
|
||||
snd_pcm_poll_descriptors (alsa_driver_capture.audio_fd,
|
||||
&pfd[nfds],
|
||||
capture_nfds);
|
||||
capture_index = nfds;
|
||||
nfds += capture_nfds;
|
||||
while(1) {
|
||||
if (poll (pfd, nfds, 200000) < 0) {
|
||||
printf("poll failed\n");
|
||||
continue;
|
||||
}
|
||||
snd_pcm_poll_descriptors_revents(alsa_driver_playback.audio_fd, &pfd[0], playback_nfds, &playback_revents);
|
||||
snd_pcm_poll_descriptors_revents(alsa_driver_capture.audio_fd, &pfd[capture_index], capture_nfds, &capture_revents);
|
||||
//if ((playback_revents & POLLERR) || ((capture_revents) & POLLERR)) {
|
||||
if (((capture_revents) & POLLERR)) {
|
||||
printf("pollerr\n");
|
||||
capture_xrun(&alsa_driver_capture);
|
||||
return;
|
||||
}
|
||||
if (((playback_revents) & POLLERR)) {
|
||||
printf("pollerr\n");
|
||||
playback_xrun(&alsa_driver_capture);
|
||||
return;
|
||||
}
|
||||
if (playback_revents & POLLOUT) {
|
||||
playback_callback(&alsa_driver_playback);
|
||||
}
|
||||
if (capture_revents & POLLIN) {
|
||||
capture_callback(&alsa_driver_capture);
|
||||
}
|
||||
}
|
||||
|
||||
return;
|
||||
}
|
||||
|
||||
void decode1_(void *iarg);
|
||||
|
||||
int start_threads_(int *ndevin, int *ndevout, short y1[], short y2[],
|
||||
int *nbuflen, int *iwrite, short iwave[],
|
||||
int *nwave, int *nfsample, int *nsamperbuf,
|
||||
int *TRPeriod, int *TxOK, int *ndebug,
|
||||
int *Transmitting, double *Tsec, int *ngo, int *nmode,
|
||||
double tbuf[], int *ibuf, int *ndsec)
|
||||
{
|
||||
pthread_t thread1,thread2;
|
||||
int iret1,iret2;
|
||||
int iarg1 = 1;
|
||||
//int32_t rate=11025;
|
||||
int32_t rate=*nfsample;
|
||||
alsa_driver_capture.app_buffer_y1 = y1;
|
||||
alsa_driver_capture.app_buffer_y2 = y2;
|
||||
alsa_driver_capture.app_buffer_offset = iwrite;
|
||||
alsa_driver_capture.app_buffer_length = *nbuflen;
|
||||
alsa_driver_capture.Tsec = Tsec;
|
||||
alsa_driver_capture.tbuf = tbuf;
|
||||
alsa_driver_capture.ibuf = ibuf;
|
||||
alsa_driver_capture.ndsec = ndsec;
|
||||
alsa_driver_playback.Tsec = Tsec;
|
||||
alsa_driver_playback.app_buffer_y1 = iwave;
|
||||
alsa_driver_playback.tx_ok = TxOK;
|
||||
alsa_driver_playback.tr_period = TRPeriod;
|
||||
alsa_driver_playback.nwave = nwave;
|
||||
alsa_driver_playback.nmode = nmode;
|
||||
alsa_driver_playback.transmitting = Transmitting;
|
||||
alsa_driver_playback.ndsec = ndsec;
|
||||
// printf("start_threads: creating thread for decode1\n");
|
||||
iret1 = pthread_create(&thread1,NULL,(void*)&decode1_,&iarg1);
|
||||
/* Open audio card. */
|
||||
printf("Using ALSA sound.\n");
|
||||
ao_alsa_open(&alsa_driver_playback, &rate, SND_PCM_STREAM_PLAYBACK);
|
||||
ao_alsa_open(&alsa_driver_capture, &rate, SND_PCM_STREAM_CAPTURE);
|
||||
|
||||
/*
|
||||
* Start audio io thread
|
||||
*/
|
||||
iret2 = pthread_create(&thread2, NULL, (void *)&ao_alsa_loop, NULL);
|
||||
snd_pcm_prepare(alsa_driver_capture.audio_fd);
|
||||
snd_pcm_start(alsa_driver_capture.audio_fd);
|
||||
snd_pcm_prepare(alsa_driver_playback.audio_fd);
|
||||
//snd_pcm_start(alsa_driver_playback.audio_fd);
|
||||
return 0;
|
||||
}
|
||||
306
start_oss.c
306
start_oss.c
@ -1,306 +0,0 @@
|
||||
#include <stdio.h>
|
||||
#include <stdlib.h>
|
||||
#include <pthread.h>
|
||||
#include <inttypes.h>
|
||||
#include <time.h>
|
||||
#include <sys/time.h>
|
||||
#include <fcntl.h>
|
||||
#include <sys/soundcard.h>
|
||||
#include <string.h>
|
||||
|
||||
#define AUDIOBUFSIZE 4096
|
||||
#define FRAMESPERBUFFER 1024
|
||||
#define TIMEOUT 1000L /* select time out for audio device */
|
||||
|
||||
/* XXX probably safer to use a local buffer due to the wsjt threaded nature. */
|
||||
static char rcv_buf[AUDIOBUFSIZE];
|
||||
static char tx_buf[AUDIOBUFSIZE];
|
||||
|
||||
#define MAXDSPNAME 16
|
||||
|
||||
extern void decode1_(int *iarg);
|
||||
void oss_loop(int *iarg);
|
||||
|
||||
/*
|
||||
* local state data referencing some gcom common fortran variables as well
|
||||
*/
|
||||
static struct audio_data {
|
||||
int fd_in; /* Audio fd in; used only locally in this function */
|
||||
int fd_out; /* Audio fd out; used only locally in this function */
|
||||
double *Tsec; /* Present time SoundIn,SoundOut */
|
||||
double *tbuf; /* Tsec at time of input callback SoundIn */
|
||||
int *iwrite; /* Write pointer to Rx ring buffer SoundIn */
|
||||
int *ibuf; /* Most recent input buffer# SoundIn */
|
||||
int *TxOK; /* OK to transmit? SoundIn */
|
||||
int *ndebug; /* Write debugging info? GUI */
|
||||
int *ndsec; /* Dsec in units of 0.1 s GUI */
|
||||
int *Transmitting; /* Actually transmitting? SoundOut */
|
||||
int *nwave; /* Number of samples in iwave SoundIn */
|
||||
int *nmode; /* Which WSJT mode? GUI */
|
||||
int *trperiod; /* Tx or Rx period in seconds GUI */
|
||||
int nbuflen;
|
||||
int nfs;
|
||||
int16_t *y1; /* Ring buffer for audio channel 0 SoundIn */
|
||||
int16_t *y2; /* Ring buffer for audio channel 1 SoundIn */
|
||||
short *iwave;
|
||||
}data;
|
||||
|
||||
/*
|
||||
* start_threads()
|
||||
* inputs - ndevin device number for input
|
||||
* - ndevout device number for output
|
||||
* - y1 short int array for channel 0
|
||||
* - y2 short int array for channel 1
|
||||
* - nmax
|
||||
* - iwrite
|
||||
* - iwave
|
||||
* - nwave
|
||||
* - rate
|
||||
* - NSPB
|
||||
* - TRPeriod
|
||||
* - TxOK
|
||||
* - ndebug debug output or not?
|
||||
* - Transmitting
|
||||
* - Tsec
|
||||
* - ngo
|
||||
* - nmode
|
||||
* - tbuf
|
||||
* - ibuf
|
||||
* - ndsec
|
||||
* output - ?
|
||||
* side effects - Called from audio_init.f90 to start audio decode and
|
||||
* OSS thread.
|
||||
*/
|
||||
|
||||
int
|
||||
start_threads_(int *ndevin, int *ndevout, short y1[], short y2[],
|
||||
int *nbuflen, int *iwrite, short iwave[],
|
||||
int *nwave, int *nfsample, int *nsamperbuf,
|
||||
int *TRPeriod, int *TxOK, int *ndebug,
|
||||
int *Transmitting, double *Tsec, int *ngo, int *nmode,
|
||||
double tbuf[], int *ibuf, int *ndsec,
|
||||
char *PttPort, char *devin_name, char *devout_name)
|
||||
{
|
||||
pthread_t thread1,thread2;
|
||||
int iret1,iret2;
|
||||
int iarg1 = 1,iarg2 = 2;
|
||||
int32_t rate=*nfsample;
|
||||
int samplesize;
|
||||
int format;
|
||||
int channels;
|
||||
double dnfs;
|
||||
int i;
|
||||
char *p;
|
||||
|
||||
/* Remove space if present */
|
||||
p = strchr(devin_name, ' ');
|
||||
if(p != NULL)
|
||||
*p = '\0';
|
||||
|
||||
p = strchr(devout_name, ' ');
|
||||
if(p != NULL)
|
||||
*p = '\0';
|
||||
|
||||
data.fd_in = open(devin_name, O_RDONLY, 0);
|
||||
|
||||
if(data.fd_in < 0) {
|
||||
fprintf(stderr, "Cannot open %s for input.\n", devin_name);
|
||||
return (-1);
|
||||
}
|
||||
|
||||
if (*devout_name == '\0') {
|
||||
close(data.fd_in);
|
||||
data.fd_in = open(devin_name, O_RDWR, 0);
|
||||
|
||||
if(data.fd_in < 0) {
|
||||
fprintf(stderr, "Cannot open %s for input.\n", devin_name);
|
||||
return (-1);
|
||||
}
|
||||
|
||||
data.fd_out = data.fd_in;
|
||||
if(ioctl(data.fd_in, SNDCTL_DSP_SETDUPLEX, 0) < 0) {
|
||||
fprintf(stderr, "Cannot use %s for full duplex.\n", devin_name);
|
||||
return(-1);
|
||||
}
|
||||
} else {
|
||||
data.fd_out = open(devout_name, O_WRONLY, 0);
|
||||
|
||||
if(data.fd_out < 0) {
|
||||
fprintf(stderr, "Cannot open %s for output.\n", devout_name);
|
||||
return (-1);
|
||||
}
|
||||
}
|
||||
|
||||
data.Tsec = Tsec;
|
||||
data.tbuf = tbuf;
|
||||
data.iwrite = iwrite;
|
||||
data.ibuf = ibuf;
|
||||
data.TxOK = TxOK;
|
||||
data.ndebug = ndebug;
|
||||
data.ndsec = ndsec;
|
||||
data.Transmitting = Transmitting;
|
||||
data.y1 = y1;
|
||||
data.y2 = y2;
|
||||
data.nbuflen = *nbuflen;
|
||||
data.nmode = nmode;
|
||||
data.nwave = nwave;
|
||||
data.iwave = iwave;
|
||||
data.nfs = *nfsample;
|
||||
data.trperiod = TRPeriod;
|
||||
|
||||
dnfs=(double)*nfsample;
|
||||
|
||||
channels = 2;
|
||||
if(ioctl (data.fd_in, SNDCTL_DSP_CHANNELS, &channels) == -1) {
|
||||
fprintf (stderr, "Unable to set 2 channels for input.\n");
|
||||
return (-1);
|
||||
}
|
||||
|
||||
if(channels != 2) {
|
||||
fprintf (stderr, "Unable to set 2 channels.\n");
|
||||
return (-1);
|
||||
}
|
||||
|
||||
format = AFMT_S16_NE;
|
||||
if(ioctl (data.fd_in, SNDCTL_DSP_SETFMT, &format) == -1) {
|
||||
fprintf (stderr, "Unable to set format for input.\n");
|
||||
return (-1);
|
||||
}
|
||||
|
||||
if(ioctl (data.fd_in, SNDCTL_DSP_SPEED, &rate) == -1) {
|
||||
fprintf (stderr, "Unable to set rate for input\n");
|
||||
return (-1);
|
||||
}
|
||||
|
||||
printf("Audio OSS streams running normally.\n");
|
||||
printf("******************************************************************\n");
|
||||
printf("Opened %s for input.\n", devin_name);
|
||||
if (*devout_name != '\0')
|
||||
printf("Opened %s for output.\n", devout_name);
|
||||
else
|
||||
printf("Opened %s for output.\n", devin_name);
|
||||
printf("Rate set = %d\n", rate);
|
||||
|
||||
// printf("start_threads: creating thread for oss_loop\n");
|
||||
iret1 = pthread_create(&thread1, NULL,
|
||||
(void *(*)(void *))oss_loop, &iarg1);
|
||||
// printf("start_threads: creating thread for decode1_\n");
|
||||
iret2 = pthread_create(&thread2, NULL,
|
||||
(void *(*)(void *))decode1_,&iarg2);
|
||||
}
|
||||
|
||||
/*
|
||||
* oss_loop
|
||||
*
|
||||
* inputs - int pointer NOT USED
|
||||
* output - none
|
||||
* side effects -
|
||||
*/
|
||||
|
||||
void
|
||||
oss_loop(int *iarg)
|
||||
{
|
||||
fd_set readfds, writefds;
|
||||
int nfds = 0;
|
||||
struct timeval timeout = {0, 0};
|
||||
struct timeval tv;
|
||||
int nread;
|
||||
unsigned int i;
|
||||
static int n=0;
|
||||
static int n2=0;
|
||||
static int ia=0;
|
||||
static int ib=0;
|
||||
static int ic=0;
|
||||
static int16_t *in;
|
||||
static int16_t *wptr;
|
||||
static int TxOKz=0;
|
||||
static int ncall=0;
|
||||
static int nsec=0;
|
||||
static double stime;
|
||||
|
||||
for (;;) {
|
||||
FD_ZERO(&readfds );
|
||||
FD_ZERO(&writefds );
|
||||
FD_SET(data.fd_in, &readfds);
|
||||
FD_SET(data.fd_out, &writefds);
|
||||
|
||||
timeout.tv_usec = TIMEOUT;
|
||||
if(select(FD_SETSIZE, &readfds, &writefds, NULL, &timeout) > 0) {
|
||||
if(FD_ISSET(data.fd_in, &readfds)) {
|
||||
nread = read (data.fd_in, rcv_buf, AUDIOBUFSIZE);
|
||||
if(nread <= 0) {
|
||||
fprintf(stderr, "Read error %d\n", nread);
|
||||
return;
|
||||
}
|
||||
if(nread == AUDIOBUFSIZE) {
|
||||
/* Get System time */
|
||||
gettimeofday(&tv, NULL);
|
||||
stime = (double) tv.tv_sec + ((double)tv.tv_usec / 1000000.0) +
|
||||
*(data.ndsec) * 0.1;
|
||||
*(data.Tsec) = stime;
|
||||
|
||||
ncall++;
|
||||
|
||||
/* increment buffer pointers only if data available */
|
||||
ia=*(data.iwrite);
|
||||
ib=*(data.ibuf);
|
||||
data.tbuf[ib-1] = stime; /* convert to c index to store */
|
||||
ib++;
|
||||
if(ib>FRAMESPERBUFFER)
|
||||
ib=1;
|
||||
*(data.ibuf) = ib;
|
||||
in = (int16_t *)rcv_buf; /* XXX */
|
||||
for(i=0; i<FRAMESPERBUFFER; i++) {
|
||||
data.y1[ia] = (*in++);
|
||||
data.y2[ia] = (*in++);
|
||||
ia++;
|
||||
}
|
||||
|
||||
if(ia >= data.nbuflen)
|
||||
ia=0; //Wrap buffer pointer if necessary
|
||||
*(data.iwrite) = ia; /* Save buffer pointer */
|
||||
fivehz_(); /* Call fortran routine */
|
||||
}
|
||||
}
|
||||
if(FD_ISSET(data.fd_in, &writefds)) {
|
||||
/* Get System time */
|
||||
gettimeofday(&tv, NULL);
|
||||
stime = (double) tv.tv_sec + ((double)tv.tv_usec / 1000000.0) +
|
||||
*(data.ndsec) * 0.1;
|
||||
*(data.Tsec) = stime;
|
||||
|
||||
if(*(data.TxOK) && (!TxOKz)) {
|
||||
nsec = (int)stime;
|
||||
n = nsec / *(data.trperiod);
|
||||
ic = (int)(stime - *(data.trperiod) * n) * data.nfs;
|
||||
ic = ic % *(data.nwave);
|
||||
}
|
||||
|
||||
TxOKz = *(data.TxOK);
|
||||
*(data.Transmitting) = *(data.TxOK);
|
||||
wptr = (int16_t *)tx_buf; /* XXX */
|
||||
if(*(data.TxOK)) {
|
||||
for(i=0 ; i<FRAMESPERBUFFER; i++) {
|
||||
n2 = data.iwave[ic];
|
||||
addnoise_(&n2);
|
||||
*wptr++ = n2; /* left */
|
||||
*wptr++ = n2; /* right */
|
||||
ic++;
|
||||
if(ic >= *(data.nwave)) {
|
||||
ic = ic % *(data.nwave); /* Wrap buffer pointer if necessary */
|
||||
if(*(data.nmode) == 2)
|
||||
*(data.TxOK) = 0;
|
||||
}
|
||||
}
|
||||
} else {
|
||||
memset(tx_buf, 0, AUDIOBUFSIZE);
|
||||
}
|
||||
if(write(data.fd_out, tx_buf, AUDIOBUFSIZE) < 0) {
|
||||
fprintf(stderr, "Can't write to soundcard.\n");
|
||||
return;
|
||||
}
|
||||
fivehztx_(); /* Call fortran routine */
|
||||
}
|
||||
}
|
||||
}
|
||||
}
|
||||
19
wfile3.c
19
wfile3.c
@ -1,19 +0,0 @@
|
||||
#include <stdio.h>
|
||||
|
||||
void wfile3_(char *outfile, char buf[], int *nbytes0)
|
||||
{
|
||||
int n,nbytes;
|
||||
static int first=1;
|
||||
static FILE *fp=NULL;
|
||||
|
||||
nbytes=*nbytes0;
|
||||
fp = fopen(outfile,"wb");
|
||||
if(fp == NULL) {
|
||||
printf("Cannot create %s\n",outfile);
|
||||
exit(0);
|
||||
}
|
||||
|
||||
n=fwrite(buf,1,nbytes,fp);
|
||||
fclose(fp);
|
||||
return(n);
|
||||
}
|
||||
Loading…
Reference in New Issue
Block a user