build under linux and start ALSA implementation

Makefile.linux

@@ -232,7 +232,7 @@ direwolf : direwolf.o config.o recv.o demod.o dsp.o demod_afsk.o demod_9600.o hd
 		hdlc_rec2.o multi_modem.o redecode.o rdq.o rrbb.o dlq.o \
 		fcs_calc.o ax25_pad.o \
 		decode_aprs.o symbols.o server.o kiss.o kissnet.o kiss_frame.o hdlc_send.o fcs_calc.o \
-		gen_tone.o audio.o audio_stats.o digipeater.o pfilter.o dedupe.o tq.o xmit.o morse.o \
+		gen_tone.o audio.o audio_ptt.o audio_stats.o digipeater.o pfilter.o dedupe.o tq.o xmit.o morse.o \
 		ptt.o beacon.o encode_aprs.o latlong.o encode_aprs.o latlong.o textcolor.o \
 		dtmf.o aprs_tt.o tt_user.o tt_text.o igate.o nmea.o serial_port.o log.o telemetry.o \
 		dwgps.o dwgpsnmea.o dwgpsd.o dtime_now.o \

audio_ptt.c

@@ -2,6 +2,7 @@
 //    This file is part of Dire Wolf, an amateur radio packet TNC.
 //
 //    Copyright (C) 2011, 2012, 2013, 2014, 2015  John Langner, WB2OSZ
+//    Copyright (C) 2017  Andrew Walker, VA7YAA
 //
 //    This program is free software: you can redistribute it and/or modify
 //    it under the terms of the GNU General Public License as published by
@@ -18,3 +19,149 @@
 //
 
 
+/*------------------------------------------------------------------
+ *
+ * Module:      audio_ptt.c
+ *
+ * Purpose:   	Interface to audio device commonly called a "sound card" for
+ *		historical reasons.		
+ *
+ *		This version uses the native Windows sound interface.
+ *
+ *---------------------------------------------------------------*/
+
+#if __WIN32__
+#else
+#include <limits.h>
+#include <math.h>
+#include <pthread.h>
+
+#if USE_ALSA
+#include <alsa/asoundlib.h>
+#else
+#include <errno.h>
+#ifdef __OpenBSD__
+#include <soundcard.h>
+#else
+#include <sys/soundcard.h>
+#endif
+#endif
+
+#include "direwolf.h"
+#include "audio.h"
+#include "audio_stats.h"
+#include "textcolor.h"
+#include "ptt.h"
+#include "audio_ptt.h"
+
+#if USE_ALSA
+static int set_alsa_params (int a, snd_pcm_t *handle, struct audio_s *pa, char *name, char *dir);
+//static void alsa_select_device (char *pick_dev, int direction, char *result);
+#else
+static int set_oss_params (int a, int fd, struct audio_s *pa);
+#endif
+
+static struct audio_s          *save_audio_config_p;
+
+static void * ptt_thread (void *arg);
+
+int start_ptt_thread (struct audio_s *pa, int ch)
+{
+    pthread_t tid = 0;
+    int e;
+
+    save_audio_config_p = pa;
+
+    e = pthread_create (&tid, NULL, ptt_thread, (void*)(long)ch);
+    
+    return tid;
+}
+
+static void * ptt_thread (void *arg) 
+{
+  int ch = (int)(long)arg; // channel number.
+  int channel = save_audio_config_p->achan[ch].octrl[OCTYPE_PTT].ptt_channel;
+  int freq = save_audio_config_p->achan[channel].octrl[OCTYPE_PTT].ptt_frequency;
+  int a = ACHAN2ADEV( channel );
+
+  if( save_audio_config_p->adev[a].defined ) {
+#if USE_ALSA
+    snd_pcm_t *handle;
+    int err;
+
+	  err = snd_pcm_open(&handle, save_audio_config_p->adev[a].adevice_out, SND_PCM_STREAM_PLAYBACK, 0);
+	  if (err == 0) {
+		  snd_pcm_sframes_t frames;
+		  snd_pcm_format_t format = SND_PCM_FORMAT_S16_LE;
+
+	    err = snd_pcm_set_params(handle, format, SND_PCM_ACCESS_RW_INTERLEAVED,
+			                  save_audio_config_p->adev[a].num_channels,
+			                  save_audio_config_p->adev[a].samples_per_sec, 1, 500000);
+  	  if (err == 0) {
+		    short* pnData;
+		    short sample;
+		    int nSamples = save_audio_config_p->adev[a].samples_per_sec / 10;
+		    int nBufferLength = save_audio_config_p->adev[a].num_channels * nSamples * sizeof(short);
+		    int i;
+
+		    pnData = (short*)malloc (nBufferLength);
+
+		    if (save_audio_config_p->adev[a].num_channels == 1) {
+			    for (i = 0; i < nSamples; i++) {
+			      sample = (short)( (double)SHRT_MAX * sin( ( (double)i * freq / (double)save_audio_config_p->adev[a].samples_per_sec ) * 2.0 * M_PI ) );
+			      pnData[i] = sample;
+			    }
+		    }
+		    else {
+			    for (i = 0; i < nSamples; i++) {
+			      sample = (short)( (double)SHRT_MAX * sin( ( (double)i * freq / (double)save_audio_config_p->adev[a].samples_per_sec ) * 2.0 * M_PI ) );
+			      if (channel == ADEVFIRSTCHAN( a )) {
+
+			        // Stereo, left channel.
+
+			        pnData[i*2 + 0] = sample;
+			        pnData[i*2 + 1] = 0;
+			      }
+			      else {
+
+			        // Stereo, right channel.
+
+			        pnData[i*2 + 0] = 0;
+			        pnData[i*2 + 1] = sample;
+			      }
+			    }
+		    }       
+		    
+		    //
+		    // ptt_set on
+		    //
+
+		    for (i=0; i<50; i++) {
+			    frames = snd_pcm_writei(handle, pnData, nSamples);
+		    }
+
+		    //
+		    // ptt_set off
+		    //
+
+		    //
+		    // close
+		    //
+
+		    free (pnData);
+		  }
+
+		  snd_pcm_close(handle);
+	  }
+#else
+    int oss_audio_device_fd;
+
+    oss_audio_device_fd = open (save_audio_config_p->adev[a].adevice_out, O_WRONLY);
+    if (oss_audio_device_fd != -1) {
+
+    }
+#endif
+  }
+}
+
+#endif

audio_ptt_win.c

@@ -96,33 +96,28 @@ unsigned __stdcall ptt_thread (void *arg)
         err = waveOutOpen ( &hWaveOut, atoi( save_audio_config_p->adev[a].adevice_out ), &wf, (DWORD_PTR)NULL, 0, CALLBACK_NULL );
         if( err == MMSYSERR_NOERROR ) {
             WAVEHDR waveHeader;
-            SHORT* pnData;
+            short* pnData;
-            SHORT sample;
+            short sample;
-            int nsamples = save_audio_config_p->adev[a].samples_per_sec / freq;
+            int nSamples = save_audio_config_p->adev[a].samples_per_sec / freq;
             int i;
 
-            if( save_audio_config_p->adev[a].num_channels == 1 ) {
+            waveHeader.dwBufferLength = save_audio_config_p->adev[a].num_channels * nSamples * sizeof( short );
-                waveHeader.dwBufferLength = 1 * nsamples * sizeof( SHORT );
-            }
-            else {
-                waveHeader.dwBufferLength = 2 * nsamples * sizeof( SHORT );
-            }
             waveHeader.lpData = malloc( waveHeader.dwBufferLength );
             waveHeader.dwUser = 0;
             waveHeader.dwFlags = WHDR_BEGINLOOP | WHDR_ENDLOOP;
             waveHeader.dwLoops = 0xFFFF;
 
-            pnData = (SHORT*)waveHeader.lpData;
+            pnData = (short*)waveHeader.lpData;
 
             if( save_audio_config_p->adev[a].num_channels == 1 ) {
-                for( i = 0; i < nsamples; i++ ) {
+                for( i = 0; i < nSamples; i++ ) {
-                    sample = (SHORT)( (double)SHRT_MAX * sin( ( (double)i / (double)nsamples ) * 2.0 * M_PI ) );
+                    sample = (short)( (double)SHRT_MAX * sin( ( (double)i / (double)nSamples ) * 2.0 * M_PI ) );
                     pnData[i] = sample;
                 }
             }
             else {
-                for( i = 0; i < nsamples; i++ ) {
+                for( i = 0; i < nSamples; i++ ) {
-                    sample = (SHORT)( (double)SHRT_MAX * sin( ( (double)i / (double)nsamples ) * 2.0 * M_PI ) );
+                    sample = (short)( (double)SHRT_MAX * sin( ( (double)i / (double)nSamples ) * 2.0 * M_PI ) );
                     if( channel == ADEVFIRSTCHAN( a ) ) {
 
                         // Stereo, left channel.

ptt.c

@@ -774,10 +774,8 @@ void ptt_init (struct audio_s *audio_config_p)
             return;
           }
 #else
-          int e;
+	  audio_ptt_tid[ch] = start_ptt_thread (audio_config_p, ch);
-
+          if (audio_ptt_tid[ch] == 0) {
-          e = pthread_create (&(ptt_tid[j]), NULL, ptt_thread, (void *)(long)ch);
-          if (e != 0) {
             text_color_set(DW_COLOR_ERROR);
             dw_printf ("Could not create audio_ptt thread on channel %d for PTT of channel %d.\n",
                 audio_config_p->achan[ch].octrl[OCTYPE_PTT].ptt_channel, ch );