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author | Max Kellermann <max@duempel.org> | 2009-10-20 21:05:11 +0200 |
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committer | Max Kellermann <max@duempel.org> | 2009-10-20 21:05:11 +0200 |
commit | bc4266bef8b4997332de706be28904d8b8f7d593 (patch) | |
tree | d613262e3d780497a412bdf83c928665c554b293 /src/output/pulse_output_plugin.c | |
parent | bc629c8a3e5f90c827cc3ad2d896d2d0737e3d62 (diff) | |
download | mpd-bc4266bef8b4997332de706be28904d8b8f7d593.tar.gz mpd-bc4266bef8b4997332de706be28904d8b8f7d593.tar.xz mpd-bc4266bef8b4997332de706be28904d8b8f7d593.zip |
pulse: renamed source files
Diffstat (limited to 'src/output/pulse_output_plugin.c')
-rw-r--r-- | src/output/pulse_output_plugin.c | 183 |
1 files changed, 183 insertions, 0 deletions
diff --git a/src/output/pulse_output_plugin.c b/src/output/pulse_output_plugin.c new file mode 100644 index 000000000..2557106f9 --- /dev/null +++ b/src/output/pulse_output_plugin.c @@ -0,0 +1,183 @@ +/* + * 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_list.h" + +#include <glib.h> +#include <pulse/simple.h> +#include <pulse/error.h> + +#define MPD_PULSE_NAME "mpd" + +struct pulse_data { + const char *name; + const char *server; + const char *sink; + + pa_simple *s; +}; + +/** + * 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); +} + +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; + + g_setenv("PULSE_PROP_media.role", "music", true); + + pd = pulse_new_data(); + pd->name = config_get_block_string(param, "name", "mpd_pulse"); + pd->server = config_get_block_string(param, "server", NULL); + pd->sink = config_get_block_string(param, "sink", 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, 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; + } + + 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_output_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, + + .mixer_plugin = &pulse_mixer_plugin, +}; |