pulse: renamed source files
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183
src/output/pulse_output_plugin.c
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183
src/output/pulse_output_plugin.c
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/*
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* Copyright (C) 2003-2009 The Music Player Daemon Project
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* http://www.musicpd.org
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License along
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* with this program; if not, write to the Free Software Foundation, Inc.,
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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*/
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#include "output_api.h"
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#include "mixer_list.h"
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#include <glib.h>
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#include <pulse/simple.h>
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#include <pulse/error.h>
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#define MPD_PULSE_NAME "mpd"
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struct pulse_data {
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const char *name;
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const char *server;
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const char *sink;
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pa_simple *s;
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};
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/**
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* The quark used for GError.domain.
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*/
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static inline GQuark
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pulse_output_quark(void)
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{
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return g_quark_from_static_string("pulse_output");
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}
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static struct pulse_data *pulse_new_data(void)
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{
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struct pulse_data *ret;
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ret = g_new(struct pulse_data, 1);
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ret->server = NULL;
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ret->sink = NULL;
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return ret;
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}
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static void pulse_free_data(struct pulse_data *pd)
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{
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g_free(pd);
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}
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static void *
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pulse_init(G_GNUC_UNUSED const struct audio_format *audio_format,
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const struct config_param *param, G_GNUC_UNUSED GError **error)
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{
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struct pulse_data *pd;
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g_setenv("PULSE_PROP_media.role", "music", true);
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pd = pulse_new_data();
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pd->name = config_get_block_string(param, "name", "mpd_pulse");
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pd->server = config_get_block_string(param, "server", NULL);
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pd->sink = config_get_block_string(param, "sink", NULL);
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return pd;
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}
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static void pulse_finish(void *data)
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{
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struct pulse_data *pd = data;
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pulse_free_data(pd);
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}
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static bool pulse_test_default_device(void)
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{
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pa_simple *s;
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pa_sample_spec ss;
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int error;
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ss.format = PA_SAMPLE_S16NE;
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ss.rate = 44100;
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ss.channels = 2;
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s = pa_simple_new(NULL, MPD_PULSE_NAME, PA_STREAM_PLAYBACK, NULL,
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MPD_PULSE_NAME, &ss, NULL, NULL, &error);
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if (!s) {
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g_message("Cannot connect to default PulseAudio server: %s\n",
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pa_strerror(error));
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return false;
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}
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pa_simple_free(s);
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return true;
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}
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static bool
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pulse_open(void *data, struct audio_format *audio_format, GError **error_r)
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{
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struct pulse_data *pd = data;
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pa_sample_spec ss;
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int error;
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/* MPD doesn't support the other pulseaudio sample formats, so
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we just force MPD to send us everything as 16 bit */
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audio_format->bits = 16;
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ss.format = PA_SAMPLE_S16NE;
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ss.rate = audio_format->sample_rate;
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ss.channels = audio_format->channels;
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pd->s = pa_simple_new(pd->server, MPD_PULSE_NAME, PA_STREAM_PLAYBACK,
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pd->sink, pd->name,
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&ss, NULL, NULL,
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&error);
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if (!pd->s) {
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g_set_error(error_r, pulse_output_quark(), error,
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"Cannot connect to PulseAudio server: %s",
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pa_strerror(error));
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return false;
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}
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return true;
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}
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static void pulse_cancel(void *data)
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{
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struct pulse_data *pd = data;
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int error;
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if (pa_simple_flush(pd->s, &error) < 0)
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g_warning("Flush failed in PulseAudio output \"%s\": %s\n",
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pd->name, pa_strerror(error));
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}
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static void pulse_close(void *data)
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{
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struct pulse_data *pd = data;
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pa_simple_drain(pd->s, NULL);
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pa_simple_free(pd->s);
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}
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static size_t
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pulse_play(void *data, const void *chunk, size_t size, GError **error_r)
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{
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struct pulse_data *pd = data;
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int error;
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if (pa_simple_write(pd->s, chunk, size, &error) < 0) {
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g_set_error(error_r, pulse_output_quark(), error,
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"%s", pa_strerror(error));
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return 0;
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}
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return size;
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}
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const struct audio_output_plugin pulse_output_plugin = {
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.name = "pulse",
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.test_default_device = pulse_test_default_device,
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.init = pulse_init,
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.finish = pulse_finish,
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.open = pulse_open,
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.play = pulse_play,
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.cancel = pulse_cancel,
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.close = pulse_close,
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.mixer_plugin = &pulse_mixer_plugin,
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};
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