Merge commit 'release-0.16.2'

Conflicts:
	Makefile.am
	NEWS
	configure.ac

8 files changed, 43 insertions, 11 deletions

diff --git a/src/output/ao_plugin.c b/src/output/ao_plugin.c
index 21d60eb58..71e06bad9 100644
--- a/src/output/ao_plugin.c
+++ b/src/output/ao_plugin.c
@@ -26,6 +26,9 @@
 #undef G_LOG_DOMAIN
 #define G_LOG_DOMAIN "ao"
 
+/* An ao_sample_format, with all fields set to zero: */
+static const ao_sample_format OUR_AO_FORMAT_INITIALIZER;
+
 static unsigned ao_output_ref;
 
 struct ao_data {
@@ -167,7 +170,7 @@ static bool
 ao_output_open(void *data, struct audio_format *audio_format,
 	       GError **error)
 {
-	ao_sample_format format;
+	ao_sample_format format = OUR_AO_FORMAT_INITIALIZER;
 	struct ao_data *ad = (struct ao_data *)data;
 
 	switch (audio_format->format) {
diff --git a/src/output/httpd_internal.h b/src/output/httpd_internal.h
index d9c764bfd..67396b667 100644
--- a/src/output/httpd_internal.h
+++ b/src/output/httpd_internal.h
@@ -111,7 +111,7 @@ struct httpd_output {
 	char buffer[32768];
 
 	/**
-	 * The maximum and current number of clients connected 
+	 * The maximum and current number of clients connected
 	 * at the same time.
 	 */
 	guint clients_max, clients_cnt;
diff --git a/src/output/httpd_output_plugin.c b/src/output/httpd_output_plugin.c
index bcc27f884..7fde676c5 100644
--- a/src/output/httpd_output_plugin.c
+++ b/src/output/httpd_output_plugin.c
@@ -36,6 +36,7 @@
 #include <errno.h>
 
 #ifdef HAVE_LIBWRAP
+#include <sys/socket.h> /* needed for AF_UNIX */
 #include <tcpd.h>
 #endif
 
@@ -123,6 +124,7 @@ httpd_output_init(G_GNUC_UNUSED const struct audio_format *audio_format,
 
 	/* initialize metadata */
 	httpd->metadata = NULL;
+	httpd->unflushed_input = 0;
 
 	/* initialize encoder */
 
diff --git a/src/output/jack_output_plugin.c b/src/output/jack_output_plugin.c
index 8ea5cb2fd..4df84fd23 100644
--- a/src/output/jack_output_plugin.c
+++ b/src/output/jack_output_plugin.c
@@ -40,7 +40,7 @@ enum {
 	MAX_PORTS = 16,
 };
 
-static const size_t sample_size = sizeof(jack_default_audio_sample_t);
+static const size_t jack_sample_size = sizeof(jack_default_audio_sample_t);
 
 struct jack_data {
 	/**
@@ -103,9 +103,9 @@ mpd_jack_available(const struct jack_data *jd)
 			min = current;
 	}
 
-	assert(min % sample_size == 0);
+	assert(min % jack_sample_size == 0);
 
-	return min / sample_size;
+	return min / jack_sample_size;
 }
 
 static int
@@ -123,7 +123,7 @@ mpd_jack_process(jack_nframes_t nframes, void *arg)
 		const jack_nframes_t available = mpd_jack_available(jd);
 		for (unsigned i = 0; i < jd->audio_format.channels; ++i)
 			jack_ringbuffer_read_advance(jd->ringbuffer[i],
-						     available * sample_size);
+						     available * jack_sample_size);
 
 		/* generate silence while MPD is paused */
 
@@ -144,7 +144,7 @@ mpd_jack_process(jack_nframes_t nframes, void *arg)
 	for (unsigned i = 0; i < jd->audio_format.channels; ++i) {
 		out = jack_port_get_buffer(jd->ports[i], nframes);
 		jack_ringbuffer_read(jd->ringbuffer[i],
-				     (char *)out, available * sample_size);
+				     (char *)out, available * jack_sample_size);
 
 		for (jack_nframes_t f = available; f < nframes; ++f)
 			/* ringbuffer underrun, fill with silence */
@@ -675,7 +675,7 @@ mpd_jack_play(void *data, const void *chunk, size_t size, GError **error_r)
 				space = space1;
 		}
 
-		if (space >= frame_size)
+		if (space >= jack_sample_size)
 			break;
 
 		/* XXX do something more intelligent to
@@ -683,7 +683,7 @@ mpd_jack_play(void *data, const void *chunk, size_t size, GError **error_r)
 		g_usleep(1000);
 	}
 
-	space /= sample_size;
+	space /= jack_sample_size;
 	if (space < size)
 		size = space;
 
diff --git a/src/output/mvp_plugin.c b/src/output/mvp_plugin.c
index 64fc77016..be4c8dbc0 100644
--- a/src/output/mvp_plugin.c
+++ b/src/output/mvp_plugin.c
@@ -17,7 +17,7 @@
  * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  */
 
-/* 
+/*
  * Media MVP audio output based on code from MVPMC project:
  * http://mvpmc.sourceforge.net/
  */
diff --git a/src/output/oss_plugin.c b/src/output/oss_plugin.c
index bd3ccb774..d7df594d3 100644
--- a/src/output/oss_plugin.c
+++ b/src/output/oss_plugin.c
@@ -41,6 +41,15 @@
 # include <sys/soundcard.h>
 #endif /* !(defined(__OpenBSD__) || defined(__NetBSD__) */
 
+/* We got bug reports from FreeBSD users who said that the two 24 bit
+   formats generate white noise on FreeBSD, but 32 bit works.  This is
+   a workaround until we know what exactly is expected by the kernel
+   audio drivers. */
+#ifndef __linux__
+#undef AFMT_S24_PACKED
+#undef AFMT_S24_NE
+#endif
+
 struct oss_data {
 	int fd;
 	const char *device;
@@ -347,7 +356,7 @@ oss_setup_sample_rate(int fd, struct audio_format *audio_format,
 		case SUCCESS:
 			if (!audio_valid_sample_rate(sample_rate))
 				break;
-		
+
 			audio_format->sample_rate = sample_rate;
 			return true;
 
@@ -461,6 +470,12 @@ oss_setup_sample_format(int fd, struct audio_format *audio_format,
 			break;
 
 		audio_format->format = mpd_format;
+
+#ifdef AFMT_S24_PACKED
+		if (oss_format == AFMT_S24_PACKED)
+			audio_format->reverse_endian =
+				G_BYTE_ORDER != G_LITTLE_ENDIAN;
+#endif
 		return true;
 
 	case ERROR:
@@ -502,6 +517,12 @@ oss_setup_sample_format(int fd, struct audio_format *audio_format,
 				break;
 
 			audio_format->format = mpd_format;
+
+#ifdef AFMT_S24_PACKED
+			if (oss_format == AFMT_S24_PACKED)
+				audio_format->reverse_endian =
+					G_BYTE_ORDER != G_LITTLE_ENDIAN;
+#endif
 			return true;
 
 		case ERROR:
diff --git a/src/output_control.c b/src/output_control.c
index d4bfc241c..8efe08f05 100644
--- a/src/output_control.c
+++ b/src/output_control.c
@@ -139,6 +139,7 @@ audio_output_open(struct audio_output *ao,
 {
 	bool open;
 
+	assert(audio_format_valid(audio_format));
 	assert(mp != NULL);
 
 	if (ao->fail_timer != NULL) {
diff --git a/src/output_thread.c b/src/output_thread.c
index 7d47baf46..21096eb04 100644
--- a/src/output_thread.c
+++ b/src/output_thread.c
@@ -96,6 +96,8 @@ ao_filter_open(struct audio_output *ao,
 	       struct audio_format *audio_format,
 	       GError **error_r)
 {
+	assert(audio_format_valid(audio_format));
+
 	/* the replay_gain filter cannot fail here */
 	if (ao->replay_gain_filter != NULL)
 		filter_open(ao->replay_gain_filter, audio_format, error_r);
@@ -137,6 +139,7 @@ ao_open(struct audio_output *ao)
 	assert(!ao->open);
 	assert(ao->pipe != NULL);
 	assert(ao->chunk == NULL);
+	assert(audio_format_valid(&ao->in_audio_format));
 
 	if (ao->fail_timer != NULL) {
 		/* this can only happen when this
@@ -165,6 +168,8 @@ ao_open(struct audio_output *ao)
 		return;
 	}
 
+	assert(audio_format_valid(filter_audio_format));
+
 	ao->out_audio_format = *filter_audio_format;
 	audio_format_mask_apply(&ao->out_audio_format,
 				&ao->config_audio_format);