mpd/src/output/pulse_plugin.c

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/* the Music Player Daemon (MPD)
* Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
* This project's homepage is: http://www.musicpd.org
*
* 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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
#include "../output_api.h"
#include <glib.h>
#include <pulse/simple.h>
#include <pulse/error.h>
#define MPD_PULSE_NAME "mpd"
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struct pulse_data {
struct audio_output *ao;
pa_simple *s;
char *server;
char *sink;
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};
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static struct pulse_data *pulse_new_data(void)
{
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struct pulse_data *ret;
ret = g_new(struct pulse_data, 1);
ret->s = NULL;
ret->server = NULL;
ret->sink = NULL;
return ret;
}
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static void pulse_free_data(struct pulse_data *pd)
{
g_free(pd->server);
g_free(pd->sink);
g_free(pd);
}
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static void *
pulse_init(struct audio_output *ao,
G_GNUC_UNUSED const struct audio_format *audio_format,
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ConfigParam *param)
{
BlockParam *server = NULL;
BlockParam *sink = NULL;
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struct pulse_data *pd;
if (param) {
server = getBlockParam(param, "server");
sink = getBlockParam(param, "sink");
}
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pd = pulse_new_data();
pd->ao = ao;
pd->server = server != NULL ? g_strdup(server->value) : NULL;
pd->sink = sink != NULL ? g_strdup(sink->value) : NULL;
return pd;
}
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static void pulse_finish(void *data)
{
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struct pulse_data *pd = data;
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pulse_free_data(pd);
}
static bool pulse_test_default_device(void)
{
pa_simple *s;
pa_sample_spec ss;
int error;
ss.format = PA_SAMPLE_S16NE;
ss.rate = 44100;
ss.channels = 2;
s = pa_simple_new(NULL, MPD_PULSE_NAME, PA_STREAM_PLAYBACK, NULL,
MPD_PULSE_NAME, &ss, NULL, NULL, &error);
if (!s) {
g_message("Cannot connect to default PulseAudio server: %s\n",
pa_strerror(error));
return false;
}
pa_simple_free(s);
return true;
}
static bool
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pulse_open(void *data, struct audio_format *audio_format)
{
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struct pulse_data *pd = data;
pa_sample_spec ss;
int error;
/* MPD doesn't support the other pulseaudio sample formats, so
we just force MPD to send us everything as 16 bit */
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audio_format->bits = 16;
ss.format = PA_SAMPLE_S16NE;
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ss.rate = audio_format->sample_rate;
ss.channels = audio_format->channels;
pd->s = pa_simple_new(pd->server, MPD_PULSE_NAME, PA_STREAM_PLAYBACK,
pd->sink, audio_output_get_name(pd->ao),
&ss, NULL, NULL,
&error);
if (!pd->s) {
g_warning("Cannot connect to server in PulseAudio output "
"\"%s\": %s\n",
audio_output_get_name(pd->ao),
pa_strerror(error));
return false;
}
g_debug("PulseAudio output \"%s\" connected and playing %i bit, %i "
"channel audio at %i Hz\n",
audio_output_get_name(pd->ao),
audio_format->bits,
audio_format->channels, audio_format->sample_rate);
return true;
}
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static void pulse_cancel(void *data)
{
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struct pulse_data *pd = data;
int error;
if (pd->s == NULL)
return;
if (pa_simple_flush(pd->s, &error) < 0)
g_warning("Flush failed in PulseAudio output \"%s\": %s\n",
audio_output_get_name(pd->ao),
pa_strerror(error));
}
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static void pulse_close(void *data)
{
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struct pulse_data *pd = data;
if (pd->s) {
pa_simple_drain(pd->s, NULL);
pa_simple_free(pd->s);
pd->s = NULL;
}
}
static bool
pulse_play(void *data, const char *playChunk, size_t size)
{
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struct pulse_data *pd = data;
int error;
if (pa_simple_write(pd->s, playChunk, size, &error) < 0) {
g_warning("PulseAudio output \"%s\" disconnecting due to "
"write error: %s\n",
audio_output_get_name(pd->ao),
pa_strerror(error));
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pulse_close(pd);
return false;
}
return true;
}
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const struct audio_output_plugin pulse_plugin = {
.name = "pulse",
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.test_default_device = pulse_test_default_device,
.init = pulse_init,
.finish = pulse_finish,
.open = pulse_open,
.play = pulse_play,
.cancel = pulse_cancel,
.close = pulse_close,
};