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updates to alsa support

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After studying pcm_min.c in the alsa library git repository, I updated
the alsa support to be similar to what I saw there.
This commit is contained in:
William Hubbs 2009-10-08 17:46:39 -05:00
commit dec561324d
1 changed files with 29 additions and 77 deletions
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106
alsa.c
View file

@ -25,21 +25,13 @@
#include <stdio.h>
#include <stdlib.h>
#include <limits.h>
#define ALSA_PCM_NEW_HW_PARAMS_API
#define ALSA_PCM_NEW_SW_PARAMS_API
#include <alsa/asoundlib.h>
#undef ALSA_PCM_NEW_HW_PARAMS_API
#undef ALSA_PCM_NEW_SW_PARAMS_API
#include "espeakup.h"
static pthread_mutex_t audio_mutex = PTHREAD_MUTEX_INITIALIZER;
static snd_pcm_t *handle;
static snd_pcm_hw_params_t *params;
static snd_pcm_status_t *status;
static int dir = 0;
static void lock_audio_mutex(void)
{
@ -51,16 +43,9 @@ static void unlock_audio_mutex(void)
pthread_mutex_unlock(&audio_mutex);
}
static int sound_error(int err, const char *msg)
static int alsa_callback(short *audio, int numsamples, espeak_EVENT * events)
{
fprintf(stderr, "%s: %s\n", msg, snd_strerror(err));
return err;
}
static int alsa_callback(short *audio, int numsamples,
espeak_EVENT * events)
{
int written = 0;
int frames = 0;
int rc = 0;
lock_audio_mutex();
@ -70,17 +55,17 @@ static int alsa_callback(short *audio, int numsamples,
}
while (numsamples > 0 && (!stop_requested && should_run)) {
written = snd_pcm_writei(handle, audio, numsamples);
if (written < 0)
written = snd_pcm_recover(handle, written, 0);
if (written < 0) {
fprintf(stderr, "snd_pcm_writei failed: %s\n", snd_strerror(written));
frames = snd_pcm_writei(handle, audio, numsamples);
if (frames < 0)
frames = snd_pcm_recover(handle, frames, !debug);
if (frames < 0) {
fprintf(stderr, "snd_pcm_writei failed: %s\n", snd_strerror(frames));
break;
}
if (written > 0 && written < numsamples)
printf("Short write (expected %i, wrote %i)\n", numsamples, written);
numsamples -= written;
audio += written;
if (frames > 0 && frames < numsamples && debug)
fprintf(stderr, "Short write (expected %i, wrote %i)\n", numsamples, frames);
numsamples -= frames;
audio += frames;
}
rc = (stop_requested || !should_run);
unlock_audio_mutex();
@ -94,60 +79,27 @@ void select_audio_mode(void)
int init_audio(unsigned int rate)
{
int rc;
int err;
/* Open PCM device for playback. */
rc = snd_pcm_open(&handle, "default", SND_PCM_STREAM_PLAYBACK, 0);
if (rc < 0)
return sound_error(rc, "unable to open pcm device");
err = snd_pcm_open(&handle, "default", SND_PCM_STREAM_PLAYBACK, 0);
if (err < 0) {
fprintf(stderr, "Playback open error: %s\n", snd_strerror(err));
return err;
}
/* Allocate a hardware parameters object. */
rc = snd_pcm_hw_params_malloc(&params);
if (rc < 0)
return sound_error(rc,
"Unable to allocate memory to store audio parameters");
rc = snd_pcm_status_malloc(&status);
if (rc < 0)
return sound_error(rc,
"Unable to allocate memory to store PCM status");
/* Fill it in with default values. */
rc = snd_pcm_hw_params_any(handle, params);
if (rc < 0)
return sound_error(rc,
"Unable to establish defaults for hardware parameters.");
/* Set the desired hardware parameters. */
/* Interleaved mode */
rc = snd_pcm_hw_params_set_access(handle, params,
SND_PCM_ACCESS_RW_INTERLEAVED);
if (rc < 0)
return sound_error(rc, "Error selecting interleaved mode.");
/* Signed 16-bit little-endian format */
rc = snd_pcm_hw_params_set_format(handle, params,
SND_PCM_FORMAT_S16_LE);
if (rc < 0)
return sound_error(rc, "Unable to select signed 16-bit samples");
/* One channel */
rc = snd_pcm_hw_params_set_channels(handle, params, 1);
if (rc < 0)
return sound_error(rc, "Unable to use mono output.");
rc = snd_pcm_hw_params_set_rate_near(handle, params, &rate, &dir);
if (rc < 0)
return sound_error(rc, "Unable to set sample rate");
/* Write the parameters to the driver */
rc = snd_pcm_hw_params(handle, params);
if (rc < 0)
return sound_error(rc, "unable to set hw parameters");
/* Set parameters. */
err = snd_pcm_set_params(handle,
SND_PCM_FORMAT_S16_LE,
SND_PCM_ACCESS_RW_INTERLEAVED,
1,
rate,
1,
0);
if (err < 0) {
fprintf(stderr, "Playback open error: %s\n", snd_strerror(err));
return err;
}
espeak_SetSynthCallback(alsa_callback);
return 0;