From 52bf2ff5bd78a475746c06b38bf860962959b299 Mon Sep 17 00:00:00 2001 From: Andrew Date: Fri, 13 Jan 2017 20:05:00 -0800 Subject: [PATCH] build under linux and start ALSA implementation --- Makefile.linux | 2 +- audio_ptt.c | 147 ++++++++++++++++++++++++++++++++++++++++++++++++ audio_ptt_win.c | 25 ++++---- ptt.c | 6 +- 4 files changed, 160 insertions(+), 20 deletions(-) diff --git a/Makefile.linux b/Makefile.linux index be9bd59..5960a29 100644 --- a/Makefile.linux +++ b/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 \ diff --git a/audio_ptt.c b/audio_ptt.c index 9bfa245..ae91e3f 100644 --- a/audio_ptt.c +++ b/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 +#include +#include + +#if USE_ALSA +#include +#else +#include +#ifdef __OpenBSD__ +#include +#else +#include +#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 diff --git a/audio_ptt_win.c b/audio_ptt_win.c index 2b044d7..92c0c50 100644 --- a/audio_ptt_win.c +++ b/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 sample; - int nsamples = save_audio_config_p->adev[a].samples_per_sec / freq; + short* pnData; + short sample; + 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 = 1 * nsamples * sizeof( SHORT ); - } - else { - waveHeader.dwBufferLength = 2 * nsamples * sizeof( SHORT ); - } + waveHeader.dwBufferLength = save_audio_config_p->adev[a].num_channels * 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++ ) { - sample = (SHORT)( (double)SHRT_MAX * sin( ( (double)i / (double)nsamples ) * 2.0 * M_PI ) ); + for( i = 0; i < nSamples; i++ ) { + sample = (short)( (double)SHRT_MAX * sin( ( (double)i / (double)nSamples ) * 2.0 * M_PI ) ); pnData[i] = sample; } } else { - for( i = 0; i < nsamples; i++ ) { - sample = (SHORT)( (double)SHRT_MAX * sin( ( (double)i / (double)nsamples ) * 2.0 * M_PI ) ); + for( i = 0; i < nSamples; i++ ) { + sample = (short)( (double)SHRT_MAX * sin( ( (double)i / (double)nSamples ) * 2.0 * M_PI ) ); if( channel == ADEVFIRSTCHAN( a ) ) { // Stereo, left channel. @@ -191,4 +186,4 @@ unsigned __stdcall ptt_thread (void *arg) return 0; } -#endif \ No newline at end of file +#endif diff --git a/ptt.c b/ptt.c index c948a0b..1b2e34c 100644 --- a/ptt.c +++ b/ptt.c @@ -774,10 +774,8 @@ void ptt_init (struct audio_s *audio_config_p) return; } #else - int e; - - e = pthread_create (&(ptt_tid[j]), NULL, ptt_thread, (void *)(long)ch); - if (e != 0) { + audio_ptt_tid[ch] = start_ptt_thread (audio_config_p, ch); + if (audio_ptt_tid[ch] == 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 );