ad_alsa.c
/* -*- c-basic-offset: 4; indent-tabs-mode: nil -*- */
/* ====================================================================
* Copyright (c) 1999-2001 Carnegie Mellon University. All rights
* reserved.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
*
* 1. Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
*
* 2. Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in
* the documentation and/or other materials provided with the
* distribution.
*
* This work was supported in part by funding from the Defense Advanced
* Research Projects Agency and the National Science Foundation of the
* United States of America, and the CMU Sphinx Speech Consortium.
*
* THIS SOFTWARE IS PROVIDED BY CARNEGIE MELLON UNIVERSITY ``AS IS'' AND
* ANY EXPRESSED OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO,
* THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR
* PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL CARNEGIE MELLON UNIVERSITY
* NOR ITS EMPLOYEES BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
* LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
* DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
* THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
* (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
* OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*
* ====================================================================
*
*/
/* -*- mode:c; indent-tabs-mode:t; c-basic-offset:4; comment-column:40 -*-
*
* Sphinx II libad (Linux)
* ^^^^^^^^^^^^^^^^^^^^^^^
* $Id: ad_alsa.c,v 1.6 2001/12/11 00:24:48 lenzo Exp $
*
* John G. Dorsey (jd5q+@andrew.cmu.edu)
* Engineering Design Research Center
* Carnegie Mellon University
* ***************************************************************************
*
* REVISION HISTORY
*
* 18-Mar-2006 David Huggins-Daines <dhuggins@cs.cmu.edu>
* Update this to the ALSA 1.0 API.
*
* 12-Dec-2000 David Huggins-Daines <dhd@cepstral.com> at Cepstral LLC
* Make this at least compile with the new ALSA API.
*
* 05-Nov-1999 Sean Levy (snl@stalphonsos.com) at St. Alphonsos, LLC.
* Ported to ALSA so I can actually get working full-duplex.
*
* 09-Aug-1999 Kevin Lenzo (lenzo@cs.cmu.edu) at Cernegie Mellon University.
* Incorporated nickr@cs.cmu.edu's changes (marked below) and
* SPS_EPSILON to allow for sample rates that are "close enough".
*
* 15-Jun-1999 M. K. Ravishankar (rkm@cs.cmu.edu) at Carnegie Mellon Univ.
* Consolidated all ad functions into
* this one file. Added ad_open_sps().
* Other cosmetic changes for consistency (e.g., use of err.h).
*
* 18-May-1999 Kevin Lenzo (lenzo@cs.cmu.edu) added <errno.h>.
*/
#include <fcntl.h>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <alsa/asoundlib.h>
#include <errno.h>
#include <config.h>
#include <unistd.h>
#include "prim_type.h"
#include "ad.h"
#define AUDIO_FORMAT SND_PCM_SFMT_S16_LE /* 16-bit signed, little endian */
#define INPUT_GAIN 85
#define SPS_EPSILON 200
#define DEFAULT_DEVICE "default"
struct ad_rec_s {
snd_pcm_t *dspH;
int32 recording;
int32 sps;
int32 bps;
};
static int
setparams(int32 sps, snd_pcm_t * handle)
{
snd_pcm_hw_params_t *hwparams;
unsigned int out_sps, buffer_time, period_time;
int err;
snd_pcm_hw_params_alloca(&hwparams);
err = snd_pcm_hw_params_any(handle, hwparams);
if (err < 0) {
fprintf(stderr, "Can not configure this PCM device: %s\n",
snd_strerror(err));
return -1;
}
err =
snd_pcm_hw_params_set_access(handle, hwparams,
SND_PCM_ACCESS_RW_INTERLEAVED);
if (err < 0) {
fprintf(stderr,
"Failed to set PCM device to interleaved: %s\n",
snd_strerror(err));
return -1;
}
err =
snd_pcm_hw_params_set_format(handle, hwparams, SND_PCM_FORMAT_S16);
if (err < 0) {
fprintf(stderr,
"Failed to set PCM device to 16-bit signed PCM: %s\n",
snd_strerror(err));
return -1;
}
err = snd_pcm_hw_params_set_channels(handle, hwparams, 1);
if (err < 0) {
fprintf(stderr, "Failed to set PCM device to mono: %s\n",
snd_strerror(err));
return -1;
}
out_sps = sps;
err =
snd_pcm_hw_params_set_rate_near(handle, hwparams, &out_sps, NULL);
if (err < 0) {
fprintf(stderr, "Failed to set sampling rate: %s\n",
snd_strerror(err));
return -1;
}
if (abs(out_sps - sps) > SPS_EPSILON) {
fprintf(stderr,
"Available samping rate %d is too far from requested %d\n",
out_sps, sps);
return -1;
}
/* Set buffer time to the maximum. */
err = snd_pcm_hw_params_get_buffer_time_max(hwparams, &buffer_time, 0);
period_time = buffer_time / 4;
err = snd_pcm_hw_params_set_period_time_near(handle, hwparams,
&period_time, 0);
if (err < 0) {
fprintf(stderr, "Failed to set period time to %u: %s\n",
period_time, snd_strerror(err));
return -1;
}
err = snd_pcm_hw_params_set_buffer_time_near(handle, hwparams,
&buffer_time, 0);
if (err < 0) {
fprintf(stderr, "Failed to set buffer time to %u: %s\n",
buffer_time, snd_strerror(err));
return -1;
}
err = snd_pcm_hw_params(handle, hwparams);
if (err < 0) {
fprintf(stderr, "Failed to set hwparams: %s\n", snd_strerror(err));
return -1;
}
err = snd_pcm_nonblock(handle, 1);
if (err < 0) {
fprintf(stderr, "Failed to set non-blocking mode: %s\n",
snd_strerror(err));
return -1;
}
return 0;
}
ad_rec_t *
ad_open_dev(const char *dev, int32 sps)
{
ad_rec_t *handle;
snd_pcm_t *dspH;
int err;
if (dev == NULL)
dev = DEFAULT_DEVICE;
err = snd_pcm_open(&dspH, dev, SND_PCM_STREAM_CAPTURE, 0);
if (err < 0) {
fprintf(stderr,
"Error opening audio device %s for capture: %s\n",
dev, snd_strerror(err));
return NULL;
}
if (setparams(sps, dspH) < 0) {
return NULL;
}
if ((handle = (ad_rec_t *) calloc(1, sizeof(ad_rec_t))) == NULL) {
fprintf(stderr, "calloc(%d) failed\n", (int)sizeof(ad_rec_t));
abort();
}
handle->dspH = dspH;
handle->recording = 0;
handle->sps = sps;
handle->bps = sizeof(int16);
return (handle);
}
ad_rec_t *
ad_open_sps(int32 sps)
{
return ad_open_dev(DEFAULT_DEVICE, sps);
}
ad_rec_t *
ad_open(void)
{
return ad_open_sps(DEFAULT_SAMPLES_PER_SEC);
}
int32
ad_close(ad_rec_t * handle)
{
if (handle->dspH == NULL)
return AD_ERR_NOT_OPEN;
if (handle->recording) {
if (ad_stop_rec(handle) < 0)
return AD_ERR_GEN;
}
snd_pcm_close(handle->dspH);
free(handle);
return (0);
}
int32
ad_start_rec(ad_rec_t * handle)
{
int err;
if (handle->dspH == NULL)
return AD_ERR_NOT_OPEN;
if (handle->recording)
return AD_ERR_GEN;
err = snd_pcm_prepare(handle->dspH);
if (err < 0) {
fprintf(stderr, "snd_pcm_prepare failed: %s\n", snd_strerror(err));
return AD_ERR_GEN;
}
err = snd_pcm_start(handle->dspH);
if (err < 0) {
fprintf(stderr, "snd_pcm_start failed: %s\n", snd_strerror(err));
return AD_ERR_GEN;
}
handle->recording = 1;
return (0);
}
int32
ad_stop_rec(ad_rec_t * handle)
{
int err;
if (handle->dspH == NULL)
return AD_ERR_NOT_OPEN;
if (!handle->recording)
return AD_ERR_GEN;
err = snd_pcm_drop(handle->dspH);
if (err < 0) {
fprintf(stderr, "snd_pcm_drop failed: %s\n", snd_strerror(err));
return AD_ERR_GEN;
}
handle->recording = 0;
return (0);
}
int32
ad_read(ad_rec_t * handle, int16 * buf, int32 max)
{
int32 length, err;
if (!handle->recording) {
fprintf(stderr, "Recording is stopped, start recording with ad_start_rec\n");
return AD_EOF;
}
length = snd_pcm_readi(handle->dspH, buf, max);
if (length == -EAGAIN) {
length = 0;
}
else if (length == -EPIPE) {
fprintf(stderr, "Input overrun, read calls are too rare (non-fatal)\n");
err = snd_pcm_prepare(handle->dspH);
if (err < 0) {
fprintf(stderr, "Can't recover from underrun: %s\n",
snd_strerror(err));
return AD_ERR_GEN;
}
length = 0;
}
else if (length == -ESTRPIPE) {
fprintf(stderr, "Resuming sound driver (non-fatal)\n");
while ((err = snd_pcm_resume(handle->dspH)) == -EAGAIN)
usleep(10000); /* Wait for the driver to wake up */
if (err < 0) {
err = snd_pcm_prepare(handle->dspH);
if (err < 0) {
fprintf(stderr, "Can't recover from underrun: %s\n",
snd_strerror(err));
return AD_ERR_GEN;
}
}
length = 0;
}
else if (length < 0) {
fprintf(stderr, "Audio read error: %s\n",
snd_strerror(length));
return AD_ERR_GEN;
}
return length;
}
ad.h
generic live audio interface for recording and playback
ad_open_dev
SPHINXBASE_EXPORT ad_rec_t * ad_open_dev(const char *dev, int32 samples_per_sec)
Open a specific audio device for recording.
Definition ad_alsa.c:187
ad_open
SPHINXBASE_EXPORT ad_rec_t * ad_open(void)
Open the default audio device.
Definition ad_alsa.c:228
ad_open_sps
SPHINXBASE_EXPORT ad_rec_t * ad_open_sps(int32 samples_per_sec)
Open the default audio device with a given sampling rate.
Definition ad_alsa.c:222
prim_type.h
Basic type definitions used in Sphinx.
ad_rec_s
Audio recording structure.
Definition ad_alsa.c:90
ad_rec_s::sps
int32 sps
Samples/sec.
Definition ad_alsa.c:93
ad_rec_s::bps
int32 bps
Bytes/sample.
Definition ad_alsa.c:94
ad_rec_t
Audio recording structure.