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/*
* Copyright (C) 2003-2009 The Music Player Daemon Project
* 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.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#include "../output_api.h"
#include "../mixer_api.h"
#include <glib.h>
#include <pulse/simple.h>
#include <pulse/error.h>
#define MPD_PULSE_NAME "mpd"
struct pulse_data {
const char *name;
struct mixer *mixer;
pa_simple *s;
char *server;
char *sink;
};
/**
* The quark used for GError.domain.
*/
static inline GQuark
pulse_output_quark(void)
{
return g_quark_from_static_string("pulse_output");
}
static struct pulse_data *pulse_new_data(void)
{
struct pulse_data *ret;
ret = g_new(struct pulse_data, 1);
ret->server = NULL;
ret->sink = NULL;
return ret;
}
static void pulse_free_data(struct pulse_data *pd)
{
g_free(pd->server);
g_free(pd->sink);
g_free(pd);
mixer_free(pd->mixer);
}
static void *
pulse_init(G_GNUC_UNUSED const struct audio_format *audio_format,
const struct config_param *param, G_GNUC_UNUSED GError **error)
{
struct pulse_data *pd;
pd = pulse_new_data();
pd->name = config_get_block_string(param, "name", "mpd_pulse");
pd->server = param != NULL
? config_dup_block_string(param, "server", NULL) : NULL;
pd->sink = param != NULL
? config_dup_block_string(param, "sink", NULL) : NULL;
pd->mixer=mixer_new(&pulse_mixer, param);
return pd;
}
static void pulse_finish(void *data)
{
struct pulse_data *pd = data;
pulse_free_data(pd);
}
static struct mixer *
pulse_get_mixer(void *data)
{
struct pulse_data *pd = data;
return pd->mixer;
}
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
pulse_open(void *data, struct audio_format *audio_format, GError **error_r)
{
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 */
audio_format->bits = 16;
ss.format = PA_SAMPLE_S16NE;
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, pd->name,
&ss, NULL, NULL,
&error);
if (!pd->s) {
g_set_error(error_r, pulse_output_quark(), error,
"Cannot connect to PulseAudio server: %s",
pa_strerror(error));
return false;
}
mixer_open(pd->mixer);
return true;
}
static void pulse_cancel(void *data)
{
struct pulse_data *pd = data;
int error;
if (pa_simple_flush(pd->s, &error) < 0)
g_warning("Flush failed in PulseAudio output \"%s\": %s\n",
pd->name, pa_strerror(error));
}
static void pulse_close(void *data)
{
struct pulse_data *pd = data;
pa_simple_drain(pd->s, NULL);
pa_simple_free(pd->s);
}
static size_t
pulse_play(void *data, const void *chunk, size_t size, GError **error_r)
{
struct pulse_data *pd = data;
int error;
if (pa_simple_write(pd->s, chunk, size, &error) < 0) {
g_set_error(error_r, pulse_output_quark(), error,
"%s", pa_strerror(error));
return 0;
}
return size;
}
const struct audio_output_plugin pulse_plugin = {
.name = "pulse",
.test_default_device = pulse_test_default_device,
.init = pulse_init,
.finish = pulse_finish,
.get_mixer = pulse_get_mixer,
.open = pulse_open,
.play = pulse_play,
.cancel = pulse_cancel,
.close = pulse_close,
};
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