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soundmodem/soundcard/alsaio.c
Matt Grice f4c4438e0e First commit
2018-06-17 22:19:22 +10:00

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/*****************************************************************************/
/*
* alsaio.c -- Audio I/O using the ALSA API.
*
* Copyright (C) 1999-2000, 2004
* Thomas Sailer (sailer@ife.ee.ethz.ch)
*
* 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
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*
* Please note that the GPL allows you to use the driver, NOT the radio.
* In order to use the radio, you need a license from the communications
* authority of your country.
*
*/
/*****************************************************************************/
#define _GNU_SOURCE
#define _REENTRANT
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "soundio.h"
#include "audioio.h"
#include <sys/types.h>
#include <fcntl.h>
#include <sys/ioctl.h>
#include <sys/poll.h>
#include <pthread.h>
#include <stdio.h>
#include <string.h>
#include <stdlib.h>
#include <unistd.h>
#include <math.h>
#include <errno.h>
#ifdef HAVE_ALSA
#include <alsa/asoundlib.h>
/* ---------------------------------------------------------------------- */
#define AUDIOIBUFSIZE 4096
struct audioio_unix {
struct audioio audioio;
unsigned int samplerate;
unsigned int capturechannelmode;
snd_pcm_t *playback_handle;
snd_pcm_t *capture_handle;
unsigned int fragsize;
pthread_mutex_t iomutex;
pthread_cond_t iocond;
unsigned int flags;
unsigned int ptr;
u_int16_t ptime;
int16_t ibuf[AUDIOIBUFSIZE];
};
struct modemparams ioparams_alsasoundcard[] = {
{ "device", "ALSA Audio Driver", "Path name of the audio (soundcard) driver", "hw:0,0", MODEMPAR_COMBO,
{ c: { { "hw:0,0", "plughw:0,0", "hw:1,0", "plughw:1,0", "hw:2,0", "plughw:2,0", "hw:3,0", "plughw:3,0" } } } },
{ "halfdup", "Half Duplex", "Force operating the Sound Driver in Half Duplex mode", "0", MODEMPAR_CHECKBUTTON },
{ "capturechannelmode", "Capture Channel", "Capture Channel", "Mono", MODEMPAR_COMBO,
{ c: { { "Mono", "Left", "Right" } } } },
{ NULL, }
};
#define CAP_HALFDUPLEX 0x100
#define FLG_READING 0x1000
#define FLG_HALFDUPLEXTX 0x2000
#define FLG_TERMINATERX 0x4000
#ifndef INFTIM
#define INFTIM (-1)
#endif
/* ---------------------------------------------------------------------- */
static void iorelease(struct audioio *aio);
static void iowrite(struct audioio *aio, const int16_t *samples, unsigned int nr);
static void ioread(struct audioio *aio, int16_t *samples, unsigned int nr, u_int16_t tim);
static u_int16_t iocurtime(struct audioio *aio);
static void iotransmitstart(struct audioio *aio);
static void iotransmitstop(struct audioio *aio);
static void ioterminateread(struct audioio *aio);
/* ---------------------------------------------------------------------- */
static inline int iomodetofmode(unsigned int flags)
{
switch (flags & IO_RDWR) {
default:
case IO_RDONLY:
return O_RDONLY;
case IO_WRONLY:
return O_WRONLY;
case IO_RDWR:
return O_RDWR;
}
}
/* ---------------------------------------------------------------------- */
static snd_pcm_t *open_alsa(const char *name, snd_pcm_stream_t direction, unsigned int *samplerate, unsigned int *chanmode)
{
snd_pcm_t *pcm_handle;
snd_pcm_hw_params_t *hwparams;
snd_pcm_format_mask_t *fmtmask;
snd_pcm_sw_params_t *swparams;
/* set fragment size so we have approx. 10-20ms wakeup latency */
unsigned int buffer_time = 500000; /* ring buffer length in us */
unsigned int period_time = 15000; /* period time in us */
snd_pcm_format_t samplefmt;
snd_pcm_uframes_t buffer_size;
snd_pcm_uframes_t period_size;
int err, dir;
if (snd_pcm_open(&pcm_handle, name, direction, 0) < 0) {
logprintf(MLOG_ERROR, "alsa: Error opening PCM device %s\n", name);
return NULL;
}
/*
* Set hardware parameter
*/
snd_pcm_hw_params_alloca(&hwparams);
/* choose all parameters */
err = snd_pcm_hw_params_any(pcm_handle, hwparams);
if (err < 0) {
logprintf(MLOG_ERROR, "Broken configuration for capture: no configurations available: %s\n", snd_strerror(err));
goto err;
}
/* set the interleaved read/write format */
err = snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED);
if (err < 0) {
logprintf(MLOG_ERROR, "Access type RW_INTERLEAVED not available for capture: %s\n", snd_strerror(err));
goto err;
}
/* set the sample format */
snd_pcm_format_mask_alloca(&fmtmask);
snd_pcm_format_mask_set(fmtmask, SND_PCM_FORMAT_S16);
err = snd_pcm_hw_params_set_format_mask(pcm_handle, hwparams, fmtmask);
if (err < 0) {
logprintf(MLOG_ERROR, "Sample format S16_NE not available for capture: %s\n", snd_strerror(err));
goto err;
}
/* set the count of channels */
if (chanmode && *chanmode) {
err = snd_pcm_hw_params_set_channels(pcm_handle, hwparams, 2);
if (err < 0) {
logprintf(MLOG_ERROR, "Channels count (2) not available for captures: %s; trying Mono\n", snd_strerror(err));
*chanmode = 0;
}
}
if (!chanmode || !*chanmode) {
err = snd_pcm_hw_params_set_channels(pcm_handle, hwparams, 1);
if (err < 0) {
logprintf(MLOG_ERROR, "Channels count (1) not available for captures: %s\n", snd_strerror(err));
goto err;
}
}
/* set the stream rate */
err = snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, samplerate, 0);
if (err < 0) {
logprintf(MLOG_ERROR, "Rate %iHz not available for capture: %s\n", *samplerate, snd_strerror(err));
goto err;
}
/* set the buffer time */
err = snd_pcm_hw_params_set_buffer_time_near(pcm_handle, hwparams, &buffer_time, &dir);
if (err < 0) {
logprintf(MLOG_ERROR, "Unable to set buffer time %i for capture: %s\n", buffer_time, snd_strerror(err));
goto err;
}
/* set the period time */
err = snd_pcm_hw_params_set_period_time_near(pcm_handle, hwparams, &period_time, &dir);
if (err < 0) {
logprintf(MLOG_ERROR, "Unable to set period time %i for capture: %s\n", period_time, snd_strerror(err));
goto err;
}
/* write the parameters to device */
err = snd_pcm_hw_params(pcm_handle, hwparams);
if (err < 0) {
logprintf(MLOG_ERROR, "Unable to set hw params for capture: %s\n", snd_strerror(err));
goto err;
}
/* read current configuration */
err = snd_pcm_hw_params_get_buffer_size(hwparams, &buffer_size);
if (err < 0) {
logprintf(MLOG_ERROR, "Unable to get buffer size for capture: %s\n", snd_strerror(err));
goto err;
}
err = snd_pcm_hw_params_get_period_size(hwparams, &period_size, &dir);
if (err < 0) {
logprintf(MLOG_ERROR, "Unable to get period size for capture: %s\n", snd_strerror(err));
goto err;
}
err = snd_pcm_hw_params_get_rate(hwparams, samplerate, &dir);
if (err < 0) {
logprintf(MLOG_ERROR, "Unable to get the sample rate for capture: %s\n", snd_strerror(err));
goto err;
}
err = snd_pcm_hw_params_get_format(hwparams, &samplefmt);
if (err < 0) {
logprintf(MLOG_ERROR, "Unable to get the sample format for capture: %s\n", snd_strerror(err));
goto err;
}
err = snd_pcm_hw_params_get_sbits(hwparams);
if (err < 0) {
logprintf(MLOG_ERROR, "Unable to get the number of significant bits for capture: %s\n", snd_strerror(err));
goto err;
}
printf("ALSA: Using sample rate %u, sample format %d, significant bits %d, buffer size %u, period size %u\n",
*samplerate, (int)samplefmt, err, (unsigned int)buffer_size, (unsigned int)period_size);
/*
* Set Software Parameters
*/
snd_pcm_sw_params_alloca(&swparams);
/* get the current swparams */
err = snd_pcm_sw_params_current(pcm_handle, swparams);
if (err < 0) {
logprintf(MLOG_ERROR, "Unable to determine current swparams for capture: %s\n", snd_strerror(err));
goto err;
}
/* start the transfer when samples are available */
err = snd_pcm_sw_params_set_start_threshold(pcm_handle, swparams, 64);
if (err < 0) {
logprintf(MLOG_ERROR, "Unable to set start threshold mode for capture: %s\n", snd_strerror(err));
goto err;
}
/* allow the transfer when at least 1 samples can be processed */
err = snd_pcm_sw_params_set_avail_min(pcm_handle, swparams, 1);
if (err < 0) {
logprintf(MLOG_ERROR, "Unable to set avail min for capture: %s\n", snd_strerror(err));
goto err;
}
/* align all transfers to 1 sample */
err = snd_pcm_sw_params_set_xfer_align(pcm_handle, swparams, 1);
if (err < 0) {
logprintf(MLOG_ERROR, "Unable to set transfer align for capture: %s\n", snd_strerror(err));
goto err;
}
/* write the parameters to the capture device */
err = snd_pcm_sw_params(pcm_handle, swparams);
if (err < 0) {
logprintf(MLOG_ERROR, "Unable to set sw params for capture: %s\n", snd_strerror(err));
goto err;
}
if (snd_pcm_prepare(pcm_handle) < 0) {
logprintf(MLOG_ERROR, "Error preparing capture.\n");
goto err;
}
return pcm_handle;
err:
snd_pcm_close(pcm_handle);
return NULL;
}
/* ---------------------------------------------------------------------- */
/*
* Linux ALSA audio
*/
struct audioio *ioopen_alsasoundcard(unsigned int *samplerate, unsigned int flags, const char *params[])
{
const char *audiopath = params[0];
struct audioio_unix *audioio;
unsigned int prate, crate, i;
audioio = calloc(1, sizeof(struct audioio_unix));
if (!audioio)
return NULL;
audioio->audioio.release = iorelease;
if (!audiopath)
audiopath = "hw:0,0";
prate = crate = *samplerate;
audioio->capturechannelmode = 0;
if (params[2] && !strcmp(params[2], ioparams_alsasoundcard[2].u.c.combostr[1]))
audioio->capturechannelmode = 1;
else if (params[2] && !strcmp(params[2], ioparams_alsasoundcard[2].u.c.combostr[2]))
audioio->capturechannelmode = 2;
/* todo: remove configurations with different rx/tx rates or make different rates work...*/
if (flags & IO_RDONLY) {
audioio->audioio.terminateread = ioterminateread;
audioio->audioio.read = ioread;
audioio->audioio.curtime = iocurtime;
audioio->capture_handle = open_alsa(audiopath, SND_PCM_STREAM_CAPTURE, &crate, &audioio->capturechannelmode);
if (!audioio->capture_handle)
goto err;
}
if (flags & IO_WRONLY) {
audioio->audioio.transmitstart = iotransmitstart;
audioio->audioio.transmitstop = iotransmitstop;
audioio->audioio.write = iowrite;
i = 0;
audioio->playback_handle = open_alsa(audiopath, SND_PCM_STREAM_PLAYBACK, &prate, &i);
if (!audioio->playback_handle)
goto err;
}
audioio->samplerate = prate;
if ((flags & (IO_RDONLY|IO_WRONLY)) == (IO_RDONLY|IO_WRONLY) && abs(prate - crate) > 1) {
logprintf(MLOG_ERROR, "audio: Error, playback/capture sample rates do not match: %u/%u\n", prate, crate);
goto err;
}
pthread_cond_init(&audioio->iocond, NULL);
pthread_mutex_init(&audioio->iomutex, NULL);
audioio->flags = flags & IO_RDWR;
audioio->ptr = audioio->ptime = 0;
logprintf(MLOG_DEBUG, "audio: starting \"%s\"\n", audiopath);
if (params[1] && params[1][0] != '0') {
audioio->flags |= CAP_HALFDUPLEX;
logprintf(MLOG_INFO, "audio: forcing half duplex mode\n");
}
*samplerate = audioio->samplerate;
return &audioio->audioio;
err:
if (audioio->playback_handle)
snd_pcm_close(audioio->playback_handle);
if (audioio->capture_handle)
snd_pcm_close(audioio->capture_handle);
free(audioio);
return NULL;
}
static inline void iotxend(struct audioio_unix *audioio)
{
int err;
err = snd_pcm_drain(audioio->playback_handle);
if (err < 0)
logprintf(MLOG_ERROR, "snd_pcm_drain in iotxend: %s", snd_strerror(err));
if (!(audioio->flags & CAP_HALFDUPLEX))
return;
err = snd_pcm_start(audioio->capture_handle);
if (err < 0 && err != -EBADFD)
logprintf(MLOG_ERROR, "snd_pcm_start in iotxend: %s", snd_strerror(err));
}
static inline void iotxstart(struct audioio_unix *audioio)
{
int err;
if (snd_pcm_prepare(audioio->playback_handle) < 0) {
logprintf(MLOG_ERROR, "Error preparing tx.\n");
}
err = snd_pcm_start(audioio->playback_handle);
if (err < 0)
logprintf(MLOG_ERROR, "snd_pcm_start in iotxstart: %s", snd_strerror(err));
}
/* ---------------------------------------------------------------------- */
static void iorelease(struct audioio *aio)
{
struct audioio_unix *audioio = (struct audioio_unix *)aio;
pthread_cond_destroy(&audioio->iocond);
pthread_mutex_destroy(&audioio->iomutex);
audioio->flags = audioio->ptr = audioio->ptime = 0;
if (audioio->playback_handle)
snd_pcm_close(audioio->playback_handle);
if (audioio->capture_handle)
snd_pcm_close(audioio->capture_handle);
free(audioio);
}
static void iowrite(struct audioio *aio, const int16_t *samples, unsigned int nr)
{
struct audioio_unix *audioio = (struct audioio_unix *)aio;
unsigned char *p = (unsigned char *)samples;
int err;
if (!audioio->playback_handle)
return;
err = snd_pcm_writei(audioio->playback_handle, p, nr);
if (err == -EPIPE) {
if (snd_pcm_prepare(audioio->playback_handle) < 0) {
logprintf(MLOG_ERROR, "Error preparing tx.\n");
}
err = snd_pcm_writei(audioio->playback_handle, p, nr);
}
if (err < 0) {
logprintf(MLOG_ERROR, "audio: snd_pcm_writei: %s\n", snd_strerror(err));
return;
}
if (err < nr) {
logprintf(MLOG_ERROR, "audio: snd_pcm_writei: not enough samples written: %d < %u\n", err, nr);
return;
}
}
static void ioread(struct audioio *aio, int16_t *samples, unsigned int nr, u_int16_t tim)
{
struct audioio_unix *audioio = (struct audioio_unix *)aio;
int16_t ibuf[2*AUDIOIBUFSIZE/8];
int16_t *ip;
unsigned int p;
int i, j;
pthread_mutex_lock(&audioio->iomutex);
while (nr > 0) {
if (audioio->flags & FLG_TERMINATERX) {
pthread_mutex_unlock(&audioio->iomutex);
pthread_exit(NULL);
}
i = (signed)(int16_t)(audioio->ptime - tim);
if (i > AUDIOIBUFSIZE) {
pthread_mutex_unlock(&audioio->iomutex);
i -= AUDIOIBUFSIZE;
if (i > nr)
i = nr;
memset(samples, 0, i * sizeof(samples[0]));
logprintf(MLOG_ERROR, "ioread: request time %u out of time window [%u,%u)\n", tim, audioio->ptime-AUDIOIBUFSIZE, audioio->ptime);
samples += i;
nr -= i;
tim += i;
pthread_mutex_lock(&audioio->iomutex);
continue;
}
if (i > 0) {
p = (AUDIOIBUFSIZE + audioio->ptr - i) % AUDIOIBUFSIZE;
if (i > nr)
i = nr;
if (i > AUDIOIBUFSIZE-p)
i = AUDIOIBUFSIZE-p;
memcpy(samples, &audioio->ibuf[p], i * sizeof(samples[0]));
nr -= i;
samples += i;
tim += i;
continue;
}
if (audioio->flags & (FLG_READING|FLG_HALFDUPLEXTX)) {
pthread_cond_wait(&audioio->iocond, &audioio->iomutex);
continue;
}
audioio->flags |= FLG_READING;
pthread_mutex_unlock(&audioio->iomutex);
if (!audioio->capture_handle)
logerr(MLOG_FATAL, "audio: read: capture handle NULL");
i = snd_pcm_readi(audioio->capture_handle, ibuf, sizeof(ibuf)/sizeof(ibuf[0])/2);
if (i == -EPIPE) {
if (snd_pcm_prepare(audioio->capture_handle) < 0) {
logprintf(MLOG_ERROR, "Error preparing rx.\n");
}
i = snd_pcm_readi(audioio->capture_handle, ibuf, sizeof(ibuf)/sizeof(ibuf[0])/2);
}
if (i < 0)
logprintf(MLOG_FATAL, "audio: snd_pcm_readi: %s", snd_strerror(i));
if (!i) {
logerr(MLOG_ERROR, "audio: snd_pcm_readi returned 0??");
pthread_mutex_lock(&audioio->iomutex);
audioio->flags &= ~FLG_READING;
pthread_cond_broadcast(&audioio->iocond);
continue;
}
p = i;
pthread_mutex_lock(&audioio->iomutex);
audioio->flags &= ~FLG_READING;
ip = ibuf;
if (audioio->capturechannelmode)
ip += audioio->capturechannelmode-1;
for (; p > 0; ) {
i = p;
if (i > AUDIOIBUFSIZE-audioio->ptr)
i = AUDIOIBUFSIZE-audioio->ptr;
if (audioio->capturechannelmode) {
for (j = 0; j < i; j++, ip += 2)
audioio->ibuf[audioio->ptr + j] = *ip;
} else {
memcpy(&audioio->ibuf[audioio->ptr], ip, i * sizeof(audioio->ibuf[0]));
ip += i;
}
audioio->ptr = (audioio->ptr + i) % AUDIOIBUFSIZE;
audioio->ptime += i;
p -= i;
}
pthread_cond_broadcast(&audioio->iocond);
}
pthread_mutex_unlock(&audioio->iomutex);
}
static u_int16_t iocurtime(struct audioio *aio)
{
struct audioio_unix *audioio = (struct audioio_unix *)aio;
u_int16_t res;
int16_t ibuf[2*AUDIOIBUFSIZE/8];
int16_t *ip;
unsigned int p;
int i, j, r;
pthread_mutex_lock(&audioio->iomutex);
for (;;) {
if (audioio->flags & (FLG_READING|FLG_HALFDUPLEXTX))
break;
audioio->flags |= FLG_READING;
pthread_mutex_unlock(&audioio->iomutex);
if (!audioio->capture_handle)
logprintf(MLOG_FATAL, "audio: read: capture handle NULL");
r = snd_pcm_nonblock(audioio->capture_handle, 1);
if (r < 0)
logprintf(MLOG_FATAL, "audio: snd_pcm_nonblock: %s", snd_strerror(r));
i = snd_pcm_readi(audioio->capture_handle, ibuf, sizeof(ibuf)/sizeof(ibuf[0])/2);
r = snd_pcm_nonblock(audioio->capture_handle, 0);
if (r < 0)
logprintf(MLOG_FATAL, "audio: snd_pcm_nonblock: %s", snd_strerror(r));
if (!i || i == -EAGAIN) {
pthread_mutex_lock(&audioio->iomutex);
audioio->flags &= ~FLG_READING;
pthread_cond_broadcast(&audioio->iocond);
break;
}
if (i < 0)
logprintf(MLOG_FATAL, "audio: snd_pcm_readi: %s", snd_strerror(i));
p = i;
pthread_mutex_lock(&audioio->iomutex);
audioio->flags &= ~FLG_READING;
ip = ibuf;
if (audioio->capturechannelmode)
ip += audioio->capturechannelmode-1;
for (; p > 0; ) {
i = p;
if (i > AUDIOIBUFSIZE-audioio->ptr)
i = AUDIOIBUFSIZE-audioio->ptr;
if (audioio->capturechannelmode) {
for (j = 0; j < i; j++, ip += 2)
audioio->ibuf[audioio->ptr + j] = *ip;
} else {
memcpy(&audioio->ibuf[audioio->ptr], ip, i * sizeof(audioio->ibuf[0]));
ip += i;
}
audioio->ptr = (audioio->ptr + i) % AUDIOIBUFSIZE;
audioio->ptime += i;
p -= i;
}
pthread_cond_broadcast(&audioio->iocond);
}
res = audioio->ptime;
pthread_mutex_unlock(&audioio->iomutex);
return res;
}
static void iotransmitstart(struct audioio *aio)
{
struct audioio_unix *audioio = (struct audioio_unix *)aio;
if (audioio->flags & CAP_HALFDUPLEX) {
pthread_mutex_lock(&audioio->iomutex);
audioio->flags |= FLG_HALFDUPLEXTX;
while (audioio->flags & FLG_READING)
pthread_cond_wait(&audioio->iocond, &audioio->iomutex);
pthread_mutex_unlock(&audioio->iomutex);
}
iotxstart(audioio);
}
static void iotransmitstop(struct audioio *aio)
{
struct audioio_unix *audioio = (struct audioio_unix *)aio;
#if 0
short sbuf[256];
unsigned int i, j;
/* add 20ms tail */
i = audioio->samplerate / 50;
memset(sbuf, 0, sizeof(sbuf));
while (i > 0) {
j = sizeof(sbuf)/sizeof(sbuf[0]);
if (j > i)
j = i;
iowrite(audioio, sbuf, j);
i -= j;
}
#endif
iotxend(audioio);
if (audioio->flags & CAP_HALFDUPLEX) {
pthread_mutex_lock(&audioio->iomutex);
audioio->flags &= ~FLG_HALFDUPLEXTX;
pthread_cond_broadcast(&audioio->iocond);
pthread_mutex_unlock(&audioio->iomutex);
}
}
static void ioterminateread(struct audioio *aio)
{
struct audioio_unix *audioio = (struct audioio_unix *)aio;
pthread_mutex_lock(&audioio->iomutex);
audioio->flags |= FLG_TERMINATERX;
pthread_mutex_unlock(&audioio->iomutex);
pthread_cond_broadcast(&audioio->iocond);
}
/* ---------------------------------------------------------------------- */
#else
struct audioio *ioopen_alsasoundcard(unsigned int *samplerate, unsigned int flags, const char *params[])
{
return NULL;
}
#endif
void ioinit_alsasoundcard(void)
{
}
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