renamed src/audioOutputs/ to src/output/

Again, no CamelCase in the directory name.

1 files changed, 218 insertions, 0 deletions

diff --git a/src/output/pulse_plugin.c b/src/output/pulse_plugin.c
new file mode 100644
index 000000000..b1ce3d049
--- /dev/null
+++ b/src/output/pulse_plugin.c
@@ -0,0 +1,218 @@
+/* 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"
+
+#ifdef HAVE_PULSE
+
+#include "../utils.h"
+#include "../log.h"
+
+#include <pulse/simple.h>
+#include <pulse/error.h>
+
+#define MPD_PULSE_NAME "mpd"
+#define CONN_ATTEMPT_INTERVAL 60
+
+typedef struct _PulseData {
+	struct audio_output *ao;
+
+	pa_simple *s;
+	char *server;
+	char *sink;
+	int connAttempts;
+	time_t lastAttempt;
+} PulseData;
+
+static PulseData *newPulseData(void)
+{
+	PulseData *ret;
+
+	ret = xmalloc(sizeof(PulseData));
+
+	ret->s = NULL;
+	ret->server = NULL;
+	ret->sink = NULL;
+	ret->connAttempts = 0;
+	ret->lastAttempt = 0;
+
+	return ret;
+}
+
+static void freePulseData(PulseData * pd)
+{
+	if (pd->server)
+		free(pd->server);
+	if (pd->sink)
+		free(pd->sink);
+	free(pd);
+}
+
+static void *pulse_initDriver(struct audio_output *ao,
+			      mpd_unused const struct audio_format *audio_format,
+			      ConfigParam * param)
+{
+	BlockParam *server = NULL;
+	BlockParam *sink = NULL;
+	PulseData *pd;
+
+	if (param) {
+		server = getBlockParam(param, "server");
+		sink = getBlockParam(param, "sink");
+	}
+
+	pd = newPulseData();
+	pd->ao = ao;
+	pd->server = server ? xstrdup(server->value) : NULL;
+	pd->sink = sink ? xstrdup(sink->value) : NULL;
+
+	return pd;
+}
+
+static void pulse_finishDriver(void *data)
+{
+	PulseData *pd = data;
+
+	freePulseData(pd);
+}
+
+static int pulse_testDefault(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) {
+		WARNING("Cannot connect to default PulseAudio server: %s\n",
+			pa_strerror(error));
+		return -1;
+	}
+
+	pa_simple_free(s);
+
+	return 0;
+}
+
+static int pulse_openDevice(void *data,
+			    struct audio_format *audioFormat)
+{
+	PulseData *pd = data;
+	pa_sample_spec ss;
+	time_t t;
+	int error;
+
+	t = time(NULL);
+
+	if (pd->connAttempts != 0 &&
+	    (t - pd->lastAttempt) < CONN_ATTEMPT_INTERVAL)
+		return -1;
+
+	pd->connAttempts++;
+	pd->lastAttempt = t;
+
+	/* MPD doesn't support the other pulseaudio sample formats, so
+	   we just force MPD to send us everything as 16 bit */
+	audioFormat->bits = 16;
+
+	ss.format = PA_SAMPLE_S16NE;
+	ss.rate = audioFormat->sample_rate;
+	ss.channels = audioFormat->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) {
+		ERROR("Cannot connect to server in PulseAudio output "
+		      "\"%s\" (attempt %i): %s\n",
+		      audio_output_get_name(pd->ao),
+		      pd->connAttempts, pa_strerror(error));
+		return -1;
+	}
+
+	pd->connAttempts = 0;
+
+	DEBUG("PulseAudio output \"%s\" connected and playing %i bit, %i "
+	      "channel audio at %i Hz\n",
+	      audio_output_get_name(pd->ao),
+	      audioFormat->bits,
+	      audioFormat->channels, audioFormat->sample_rate);
+
+	return 0;
+}
+
+static void pulse_dropBufferedAudio(void *data)
+{
+	PulseData *pd = data;
+	int error;
+
+	if (pa_simple_flush(pd->s, &error) < 0)
+		WARNING("Flush failed in PulseAudio output \"%s\": %s\n",
+			audio_output_get_name(pd->ao),
+			pa_strerror(error));
+}
+
+static void pulse_closeDevice(void *data)
+{
+	PulseData *pd = data;
+
+	if (pd->s) {
+		pa_simple_drain(pd->s, NULL);
+		pa_simple_free(pd->s);
+	}
+}
+
+static int pulse_playAudio(void *data,
+			   const char *playChunk, size_t size)
+{
+	PulseData *pd = data;
+	int error;
+
+	if (pa_simple_write(pd->s, playChunk, size, &error) < 0) {
+		ERROR("PulseAudio output \"%s\" disconnecting due to write "
+		      "error: %s\n",
+		      audio_output_get_name(pd->ao),
+		      pa_strerror(error));
+		pulse_closeDevice(pd);
+		return -1;
+	}
+
+	return 0;
+}
+
+const struct audio_output_plugin pulsePlugin = {
+	.name = "pulse",
+	.test_default_device = pulse_testDefault,
+	.init = pulse_initDriver,
+	.finish = pulse_finishDriver,
+	.open = pulse_openDevice,
+	.play = pulse_playAudio,
+	.cancel = pulse_dropBufferedAudio,
+	.close = pulse_closeDevice,
+};
+
+#else /* HAVE_PULSE */
+
+DISABLED_AUDIO_OUTPUT_PLUGIN(pulsePlugin)
+#endif /* HAVE_PULSE */