/* 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" struct pulse_data { struct audio_output *ao; pa_simple *s; char *server; char *sink; }; static struct pulse_data *pulse_new_data(void) { struct pulse_data *ret; ret = g_new(struct pulse_data, 1); ret->s = NULL; 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); } static void * pulse_init(struct audio_output *ao, mpd_unused const struct audio_format *audio_format, ConfigParam *param) { BlockParam *server = NULL; BlockParam *sink = NULL; struct pulse_data *pd; if (param) { server = getBlockParam(param, "server"); sink = getBlockParam(param, "sink"); } 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; } static void pulse_finish(void *data) { struct pulse_data *pd = data; 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 pulse_open(void *data, struct audio_format *audio_format) { 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, 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; } static void pulse_cancel(void *data) { 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)); } static void pulse_close(void *data) { 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) { 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)); pulse_close(pd); return false; } return true; } const struct audio_output_plugin pulse_plugin = { .name = "pulse", .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, };