mirror of https://github.com/wb2osz/direwolf.git
build under linux and start ALSA implementation
This commit is contained in:
Andrew
parent
aa5176e7b4
commit
52bf2ff5bd
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@ -232,7 +232,7 @@ direwolf : direwolf.o config.o recv.o demod.o dsp.o demod_afsk.o demod_9600.o hd
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hdlc_rec2.o multi_modem.o redecode.o rdq.o rrbb.o dlq.o \
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fcs_calc.o ax25_pad.o \
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decode_aprs.o symbols.o server.o kiss.o kissnet.o kiss_frame.o hdlc_send.o fcs_calc.o \
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gen_tone.o audio.o audio_stats.o digipeater.o pfilter.o dedupe.o tq.o xmit.o morse.o \
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gen_tone.o audio.o audio_ptt.o audio_stats.o digipeater.o pfilter.o dedupe.o tq.o xmit.o morse.o \
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ptt.o beacon.o encode_aprs.o latlong.o encode_aprs.o latlong.o textcolor.o \
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dtmf.o aprs_tt.o tt_user.o tt_text.o igate.o nmea.o serial_port.o log.o telemetry.o \
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dwgps.o dwgpsnmea.o dwgpsd.o dtime_now.o \
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147
audio_ptt.c
147
audio_ptt.c
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@ -2,6 +2,7 @@
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// This file is part of Dire Wolf, an amateur radio packet TNC.
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//
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// Copyright (C) 2011, 2012, 2013, 2014, 2015 John Langner, WB2OSZ
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// Copyright (C) 2017 Andrew Walker, VA7YAA
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//
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// This program is free software: you can redistribute it and/or modify
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// it under the terms of the GNU General Public License as published by
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@ -18,3 +19,149 @@
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//
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/*------------------------------------------------------------------
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*
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* Module: audio_ptt.c
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*
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* Purpose: Interface to audio device commonly called a "sound card" for
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* historical reasons.
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*
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* This version uses the native Windows sound interface.
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*
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*---------------------------------------------------------------*/
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#if __WIN32__
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#else
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#include <limits.h>
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#include <math.h>
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#include <pthread.h>
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#if USE_ALSA
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#include <alsa/asoundlib.h>
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#else
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#include <errno.h>
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#ifdef __OpenBSD__
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#include <soundcard.h>
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#else
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#include <sys/soundcard.h>
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#endif
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#endif
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#include "direwolf.h"
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#include "audio.h"
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#include "audio_stats.h"
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#include "textcolor.h"
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#include "ptt.h"
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#include "audio_ptt.h"
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#if USE_ALSA
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static int set_alsa_params (int a, snd_pcm_t *handle, struct audio_s *pa, char *name, char *dir);
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//static void alsa_select_device (char *pick_dev, int direction, char *result);
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#else
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static int set_oss_params (int a, int fd, struct audio_s *pa);
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#endif
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static struct audio_s *save_audio_config_p;
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static void * ptt_thread (void *arg);
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int start_ptt_thread (struct audio_s *pa, int ch)
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{
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pthread_t tid = 0;
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int e;
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save_audio_config_p = pa;
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e = pthread_create (&tid, NULL, ptt_thread, (void*)(long)ch);
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return tid;
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}
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static void * ptt_thread (void *arg)
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{
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int ch = (int)(long)arg; // channel number.
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int channel = save_audio_config_p->achan[ch].octrl[OCTYPE_PTT].ptt_channel;
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int freq = save_audio_config_p->achan[channel].octrl[OCTYPE_PTT].ptt_frequency;
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int a = ACHAN2ADEV( channel );
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if( save_audio_config_p->adev[a].defined ) {
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#if USE_ALSA
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snd_pcm_t *handle;
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int err;
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err = snd_pcm_open(&handle, save_audio_config_p->adev[a].adevice_out, SND_PCM_STREAM_PLAYBACK, 0);
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if (err == 0) {
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snd_pcm_sframes_t frames;
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snd_pcm_format_t format = SND_PCM_FORMAT_S16_LE;
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err = snd_pcm_set_params(handle, format, SND_PCM_ACCESS_RW_INTERLEAVED,
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save_audio_config_p->adev[a].num_channels,
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save_audio_config_p->adev[a].samples_per_sec, 1, 500000);
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if (err == 0) {
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short* pnData;
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short sample;
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int nSamples = save_audio_config_p->adev[a].samples_per_sec / 10;
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int nBufferLength = save_audio_config_p->adev[a].num_channels * nSamples * sizeof(short);
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int i;
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pnData = (short*)malloc (nBufferLength);
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if (save_audio_config_p->adev[a].num_channels == 1) {
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for (i = 0; i < nSamples; i++) {
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sample = (short)( (double)SHRT_MAX * sin( ( (double)i * freq / (double)save_audio_config_p->adev[a].samples_per_sec ) * 2.0 * M_PI ) );
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pnData[i] = sample;
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}
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}
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else {
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for (i = 0; i < nSamples; i++) {
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sample = (short)( (double)SHRT_MAX * sin( ( (double)i * freq / (double)save_audio_config_p->adev[a].samples_per_sec ) * 2.0 * M_PI ) );
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if (channel == ADEVFIRSTCHAN( a )) {
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// Stereo, left channel.
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pnData[i*2 + 0] = sample;
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pnData[i*2 + 1] = 0;
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}
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else {
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// Stereo, right channel.
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pnData[i*2 + 0] = 0;
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pnData[i*2 + 1] = sample;
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}
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}
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}
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//
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// ptt_set on
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//
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for (i=0; i<50; i++) {
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frames = snd_pcm_writei(handle, pnData, nSamples);
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}
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//
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// ptt_set off
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//
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//
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// close
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//
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free (pnData);
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}
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snd_pcm_close(handle);
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}
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#else
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int oss_audio_device_fd;
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oss_audio_device_fd = open (save_audio_config_p->adev[a].adevice_out, O_WRONLY);
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if (oss_audio_device_fd != -1) {
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}
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#endif
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}
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}
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#endif
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@ -96,33 +96,28 @@ unsigned __stdcall ptt_thread (void *arg)
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err = waveOutOpen ( &hWaveOut, atoi( save_audio_config_p->adev[a].adevice_out ), &wf, (DWORD_PTR)NULL, 0, CALLBACK_NULL );
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if( err == MMSYSERR_NOERROR ) {
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WAVEHDR waveHeader;
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SHORT* pnData;
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SHORT sample;
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int nsamples = save_audio_config_p->adev[a].samples_per_sec / freq;
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short* pnData;
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short sample;
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int nSamples = save_audio_config_p->adev[a].samples_per_sec / freq;
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int i;
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if( save_audio_config_p->adev[a].num_channels == 1 ) {
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waveHeader.dwBufferLength = 1 * nsamples * sizeof( SHORT );
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}
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else {
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waveHeader.dwBufferLength = 2 * nsamples * sizeof( SHORT );
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}
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waveHeader.dwBufferLength = save_audio_config_p->adev[a].num_channels * nSamples * sizeof( short );
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waveHeader.lpData = malloc( waveHeader.dwBufferLength );
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waveHeader.dwUser = 0;
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waveHeader.dwFlags = WHDR_BEGINLOOP | WHDR_ENDLOOP;
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waveHeader.dwLoops = 0xFFFF;
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pnData = (SHORT*)waveHeader.lpData;
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pnData = (short*)waveHeader.lpData;
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if( save_audio_config_p->adev[a].num_channels == 1 ) {
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for( i = 0; i < nsamples; i++ ) {
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sample = (SHORT)( (double)SHRT_MAX * sin( ( (double)i / (double)nsamples ) * 2.0 * M_PI ) );
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for( i = 0; i < nSamples; i++ ) {
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sample = (short)( (double)SHRT_MAX * sin( ( (double)i / (double)nSamples ) * 2.0 * M_PI ) );
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pnData[i] = sample;
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}
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}
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else {
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for( i = 0; i < nsamples; i++ ) {
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sample = (SHORT)( (double)SHRT_MAX * sin( ( (double)i / (double)nsamples ) * 2.0 * M_PI ) );
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for( i = 0; i < nSamples; i++ ) {
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sample = (short)( (double)SHRT_MAX * sin( ( (double)i / (double)nSamples ) * 2.0 * M_PI ) );
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if( channel == ADEVFIRSTCHAN( a ) ) {
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// Stereo, left channel.
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6
ptt.c
6
ptt.c
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@ -774,10 +774,8 @@ void ptt_init (struct audio_s *audio_config_p)
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return;
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}
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#else
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int e;
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e = pthread_create (&(ptt_tid[j]), NULL, ptt_thread, (void *)(long)ch);
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if (e != 0) {
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audio_ptt_tid[ch] = start_ptt_thread (audio_config_p, ch);
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if (audio_ptt_tid[ch] == 0) {
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text_color_set(DW_COLOR_ERROR);
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dw_printf ("Could not create audio_ptt thread on channel %d for PTT of channel %d.\n",
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audio_config_p->achan[ch].octrl[OCTYPE_PTT].ptt_channel, ch );
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