mpd version 0.15.16

With mono sound, jack_sample_size is smaller than frame_size (4 vs 2
bytes), and "space/jack_sample_size==0".  That means mpd_jack_play()
will return 0, although no error has occurred.

NEWS

@@ -1,3 +1,28 @@
+ver 0.15.16 (2011/03/13)
+* output:
+  - ao: initialize the ao_sample_format struct
+  - jack: fix crash with mono playback
+* encoders:
+  - lame: explicitly configure the output sample rate
+* update: log all file permission problems
+
+
+ver 0.15.15 (2010/11/08)
+* input:
+  - rewind: fix assertion failure
+* output:
+  - shout: artist comes first in stream title
+
+
+ver 0.15.14 (2010/11/06)
+* player_thread: fix assertion failure due to wrong music pipe on seek
+* output_thread: fix assertion failure due to race condition in OPEN
+* input:
+  - rewind: fix double free bug
+* decoders:
+  - mp4ff, ffmpeg: add extension ".m4b" (audio book)
+
+
 ver 0.15.13 (2010/10/10)
 * output_thread: fix race condition after CANCEL command
 * output:

configure.ac

@@ -1,5 +1,5 @@
 AC_PREREQ(2.60)
-AC_INIT(mpd, 0.15.13, musicpd-dev-team@lists.sourceforge.net)
+AC_INIT(mpd, 0.15.16, musicpd-dev-team@lists.sourceforge.net)
 AC_CONFIG_SRCDIR([src/main.c])
 AM_INIT_AUTOMAKE([foreign 1.9 dist-bzip2])
 AM_CONFIG_HEADER(config.h)

m4/faad.m4

@@ -58,7 +58,7 @@ if test x$enable_aac = xyes; then
 	fi
 	if test x$enable_aac = xyes; then
   		AC_MSG_CHECKING(that FAAD2 uses buffer and bufferlen)
-		AC_COMPILE_IFELSE([
+		AC_COMPILE_IFELSE([AC_LANG_SOURCE([
 #include <faad.h>
 
 int main() {
@@ -82,9 +82,9 @@ int main() {
 
 	return 0;
 }
-],[AC_MSG_RESULT(yes);AC_DEFINE(HAVE_FAAD_BUFLEN_FUNCS,1,[Define if FAAD2 uses buflen in function calls])],[AC_MSG_RESULT(no);
+])],[AC_MSG_RESULT(yes);AC_DEFINE(HAVE_FAAD_BUFLEN_FUNCS,1,[Define if FAAD2 uses buflen in function calls])],[AC_MSG_RESULT(no);
 		AC_MSG_CHECKING(that FAAD2 can even be used)
-		AC_COMPILE_IFELSE([
+		AC_COMPILE_IFELSE([AC_LANG_SOURCE([
 #include <faad.h>
 
 int main() {
@@ -113,7 +113,7 @@ int main() {
 
 	return 0;
 }
-],AC_MSG_RESULT(yes),[AC_MSG_RESULT(no);enable_aac=no])
+])],AC_MSG_RESULT(yes),[AC_MSG_RESULT(no);enable_aac=no])
 		])
 	fi
 	if test x$enable_aac = xyes; then
@@ -136,7 +136,7 @@ if test x$enable_aac = xyes; then
 	CPPFLAGS=$CFLAGS
 
 	AC_MSG_CHECKING(for broken libfaad headers)
-	AC_COMPILE_IFELSE([
+	AC_COMPILE_IFELSE([AC_LANG_SOURCE([
 #include <faad.h>
 #include <stddef.h>
 #include <stdint.h>
@@ -148,7 +148,7 @@ int main() {
 	faacDecInit2(NULL, NULL, 0, &sample_rate, &channels);
 	return 0;
 }
-	],
+	])],
 		[AC_MSG_RESULT(correct)],
 		[AC_MSG_RESULT(broken);
 		AC_DEFINE(HAVE_FAAD_LONG, 1, [Define if faad.h uses the broken "unsigned long" pointers])])

m4/mpd_check_cflag.m4

@@ -4,9 +4,9 @@ AC_DEFUN([MPD_CHECK_FLAG],[
     [mpd_check_cflag_$var],[
     save_CFLAGS="$CFLAGS"
     CFLAGS="$CFLAGS $1"
-    AC_COMPILE_IFELSE([
+    AC_COMPILE_IFELSE([AC_LANG_SOURCE([
      int main(void) { return 0; }
-     ], [ eval "mpd_check_cflag_$var=yes"
+     ])], [ eval "mpd_check_cflag_$var=yes"
      ], [ eval "mpd_check_cflag_$var=no" ])
     CFLAGS="$save_CFLAGS"
   ])

src/decoder/ffmpeg_plugin.c

@@ -501,7 +501,9 @@ static const char *const ffmpeg_suffixes[] = {
 	"atrac", "au", "aud", "avi", "avm2", "avs", "bap", "bfi", "c93", "cak",
 	"cin", "cmv", "cpk", "daud", "dct", "divx", "dts", "dv", "dvd", "dxa",
 	"eac3", "film", "flac", "flc", "fli", "fll", "flx", "flv", "g726",
-	"gsm", "gxf", "iss", "m1v", "m2v", "m2t", "m2ts", "m4a", "m4v", "mad",
+	"gsm", "gxf", "iss", "m1v", "m2v", "m2t", "m2ts",
+	"m4a", "m4b", "m4v",
+	"mad",
 	"mj2", "mjpeg", "mjpg", "mka", "mkv", "mlp", "mm", "mmf", "mov", "mp+",
 	"mp1", "mp2", "mp3", "mp4", "mpc", "mpeg", "mpg", "mpga", "mpp", "mpu",
 	"mve", "mvi", "mxf", "nc", "nsv", "nut", "nuv", "oga", "ogm", "ogv",

src/decoder/mp4ff_plugin.c

@@ -425,7 +425,13 @@ mp4_tag_dup(const char *file)
 	return ret;
 }
 
-static const char *const mp4_suffixes[] = { "m4a", "mp4", NULL };
+static const char *const mp4_suffixes[] = {
+	"m4a",
+	"m4b",
+	"mp4",
+	NULL
+};
+
 static const char *const mp4_mime_types[] = { "audio/mp4", "audio/m4a", NULL };
 
 const struct decoder_plugin mp4ff_decoder_plugin = {

src/decoder_control.c

@@ -18,6 +18,7 @@
  */
 
 #include "decoder_control.h"
+#include "pipe.h"
 
 #include <assert.h>
 
@@ -58,22 +59,28 @@ static void dc_command_async(enum decoder_command cmd)
 }
 
 void
-dc_start(struct notify *notify, struct song *song)
+dc_start(struct notify *notify, struct song *song, struct music_pipe *pipe)
 {
-	assert(dc.pipe != NULL);
+	assert(dc.pipe == NULL);
 	assert(song != NULL);
+	assert(pipe != NULL);
+	assert(music_pipe_empty(pipe));
 
 	dc.next_song = song;
+	dc.pipe = pipe;
 	dc_command(notify, DECODE_COMMAND_START);
 }
 
 void
-dc_start_async(struct song *song)
+dc_start_async(struct song *song, struct music_pipe *pipe)
 {
-	assert(dc.pipe != NULL);
+	assert(dc.pipe == NULL);
 	assert(song != NULL);
+	assert(pipe != NULL);
+	assert(music_pipe_empty(pipe));
 
 	dc.next_song = song;
+	dc.pipe = pipe;
 	dc_command_async(DECODE_COMMAND_START);
 }
 

src/decoder_control.h

@@ -118,10 +118,10 @@ void
 dc_command_wait(struct notify *notify);
 
 void
-dc_start(struct notify *notify, struct song *song);
+dc_start(struct notify *notify, struct song *song, struct music_pipe *pipe);
 
 void
-dc_start_async(struct song *song);
+dc_start_async(struct song *song, struct music_pipe *pipe);
 
 void
 dc_stop(struct notify *notify);

src/directory.h

@@ -28,8 +28,8 @@
 
 #define DIRECTORY_DIR		"directory: "
 
-#define DEVICE_INARCHIVE	(unsigned)(-1)
+#define DEVICE_INARCHIVE (dev_t)(-1)
-#define DEVICE_CONTAINER	(unsigned)(-2)
+#define DEVICE_CONTAINER (dev_t)(-2)
 
 struct directory {
 	struct dirvec children;

src/encoder/lame_encoder.c

@@ -169,6 +169,13 @@ lame_encoder_setup(struct lame_encoder *encoder, GError **error)
 		return false;
 	}
 
+	if (0 != lame_set_out_samplerate(encoder->gfp,
+					 encoder->audio_format.sample_rate)) {
+		g_set_error(error, lame_encoder_quark(), 0,
+			    "error setting lame out sample rate");
+		return false;
+	}
+
 	if (0 > lame_init_params(encoder->gfp)) {
 		g_set_error(error, lame_encoder_quark(), 0,
 			    "error initializing lame params");

src/input/rewind_input_plugin.c

@@ -83,6 +83,9 @@ copy_attributes(struct input_stream *dest)
 	const struct input_rewind *r = dest->data;
 	const struct input_stream *src = &r->input;
 
+	assert(dest != src);
+	assert(src->mime == NULL || dest->mime != src->mime);
+
 	dest->ready = src->ready;
 	dest->seekable = src->seekable;
 	dest->error = src->error;
@@ -90,8 +93,7 @@ copy_attributes(struct input_stream *dest)
 	dest->offset = src->offset;
 
 	if (src->mime != NULL) {
-		if (dest->mime != NULL)
+		g_free(dest->mime);
-			g_free(dest->mime);
 		dest->mime = g_strdup(src->mime);
 	}
 }
@@ -244,4 +246,5 @@ input_rewind_open(struct input_stream *is)
 	   input stream */
 	is->plugin = &rewind_input_plugin;
 	is->data = c;
+	is->mime = g_strdup(c->input.mime);
 }

src/output/ao_plugin.c

@@ -25,6 +25,9 @@
 #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 {
@@ -166,7 +169,7 @@ 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;
 
 	/* support for 24 bit samples in libao is currently dubious,

src/output/httpd_client.c

@@ -27,6 +27,9 @@
 #include <assert.h>
 #include <string.h>
 
+#undef G_LOG_DOMAIN
+#define G_LOG_DOMAIN "httpd_output"
+
 struct httpd_client {
 	/**
 	 * The httpd output object this client is connected to.

src/output/jack_plugin.c

@@ -36,7 +36,7 @@
 #undef G_LOG_DOMAIN
 #define G_LOG_DOMAIN "jack"
 
-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);
 
 static const char *const port_names[2] = {
 	"left", "right",
@@ -118,14 +118,15 @@ mpd_jack_process(jack_nframes_t nframes, void *arg)
 
 	for (unsigned i = 0; i < G_N_ELEMENTS(jd->ringbuffer); ++i) {
 		available = jack_ringbuffer_read_space(jd->ringbuffer[i]);
-		assert(available % sample_size == 0);
+		assert(available % jack_sample_size == 0);
-		available /= sample_size;
+		available /= jack_sample_size;
 		if (available > nframes)
 			available = nframes;
 
 		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);
 
 		while (available < nframes)
 			/* ringbuffer underrun, fill with silence */
@@ -422,7 +423,7 @@ mpd_jack_play(void *data, const void *chunk, size_t size, GError **error)
 			/* send data symmetrically */
 			space = space1;
 
-		if (space >= frame_size)
+		if (space >= jack_sample_size)
 			break;
 
 		/* XXX do something more intelligent to
@@ -430,7 +431,7 @@ mpd_jack_play(void *data, const void *chunk, size_t size, GError **error)
 		g_usleep(1000);
 	}
 
-	space /= sample_size;
+	space /= jack_sample_size;
 	if (space < size)
 		size = space;
 

src/output/shout_plugin.c

@@ -483,7 +483,7 @@ shout_tag_to_metadata(const struct tag *tag, char *dest, size_t size)
 		}
 	}
 
-	snprintf(dest, size, "%s - %s", title, artist);
+	snprintf(dest, size, "%s - %s", artist, title);
 }
 
 static void my_shout_set_tag(void *data,

src/output_all.c

@@ -266,7 +266,7 @@ audio_output_all_open(const struct audio_format *audio_format,
 	else
 		/* if the pipe hasn't been cleared, the the audio
 		   format must not have changed */
-		assert(music_pipe_size(g_mp) == 0 ||
+		assert(music_pipe_empty(g_mp) ||
 		       audio_format_equals(audio_format,
 					   &input_audio_format));
 
@@ -378,7 +378,7 @@ audio_output_all_check(void)
 	assert(g_mp != NULL);
 
 	while ((chunk = music_pipe_peek(g_mp)) != NULL) {
-		assert(music_pipe_size(g_mp) > 0);
+		assert(!music_pipe_empty(g_mp));
 
 		if (!chunk_is_consumed(chunk))
 			/* at least one output is not finished playing

src/output_control.c

@@ -50,6 +50,20 @@ static void ao_command(struct audio_output *ao, enum audio_output_command cmd)
 	ao_command_wait(ao);
 }
 
+/**
+ * Like ao_command(), but assumes the object is locked by the caller.
+ */
+static void
+ao_command_locked(struct audio_output *ao, enum audio_output_command cmd)
+{
+	assert(ao->command == AO_COMMAND_NONE);
+	ao->command = cmd;
+
+	g_mutex_unlock(ao->mutex);
+	ao_command_wait(ao);
+	g_mutex_lock(ao->mutex);
+}
+
 static void ao_command_async(struct audio_output *ao,
 			     enum audio_output_command cmd)
 {
@@ -58,6 +72,12 @@ static void ao_command_async(struct audio_output *ao,
 	notify_signal(&ao->notify);
 }
 
+static void
+audio_output_close_locked(struct audio_output *ao);
+
+/**
+ * Object must be locked (and unlocked) by the caller.
+ */
 static bool
 audio_output_open(struct audio_output *ao,
 		  const struct audio_format *audio_format,
@@ -84,7 +104,7 @@ audio_output_open(struct audio_output *ao,
 
 			/* we're not using audio_output_cancel() here,
 			   because that function is asynchronous */
-			ao_command(ao, AO_COMMAND_CANCEL);
+			ao_command_locked(ao, AO_COMMAND_CANCEL);
 		}
 
 		return true;
@@ -95,7 +115,7 @@ audio_output_open(struct audio_output *ao,
 
 	if (!ao->config_audio_format) {
 		if (ao->open)
-			audio_output_close(ao);
+			audio_output_close_locked(ao);
 
 		/* no audio format is configured: copy in->out, let
 		   the output's open() method determine the effective
@@ -110,7 +130,7 @@ audio_output_open(struct audio_output *ao,
 
 	open = ao->open;
 	if (!open) {
-		ao_command(ao, AO_COMMAND_OPEN);
+		ao_command_locked(ao, AO_COMMAND_OPEN);
 		open = ao->open;
 	}
 
@@ -127,12 +147,19 @@ audio_output_update(struct audio_output *ao,
 {
 	assert(mp != NULL);
 
+	g_mutex_lock(ao->mutex);
+
 	if (ao->enabled) {
 		if (ao->fail_timer == NULL ||
-		    g_timer_elapsed(ao->fail_timer, NULL) > REOPEN_AFTER)
+		    g_timer_elapsed(ao->fail_timer, NULL) > REOPEN_AFTER) {
-			return audio_output_open(ao, audio_format, mp);
+			bool ret = audio_output_open(ao, audio_format, mp);
+			g_mutex_unlock(ao->mutex);
+			return ret;
+		}
 	} else if (audio_output_is_open(ao))
-		audio_output_close(ao);
+		audio_output_close_locked(ao);
+
+	g_mutex_unlock(ao->mutex);
 
 	return false;
 }
@@ -162,21 +189,33 @@ void audio_output_cancel(struct audio_output *ao)
 	ao_command_async(ao, AO_COMMAND_CANCEL);
 }
 
-void audio_output_close(struct audio_output *ao)
+static void
+audio_output_close_locked(struct audio_output *ao)
 {
+	assert(ao != NULL);
 	assert(!ao->open || ao->fail_timer == NULL);
 
 	if (ao->mixer != NULL)
 		mixer_auto_close(ao->mixer);
 
 	if (ao->open)
-		ao_command(ao, AO_COMMAND_CLOSE);
+		ao_command_locked(ao, AO_COMMAND_CLOSE);
 	else if (ao->fail_timer != NULL) {
 		g_timer_destroy(ao->fail_timer);
 		ao->fail_timer = NULL;
 	}
 }
 
+void audio_output_close(struct audio_output *ao)
+{
+	assert(ao != NULL);
+	assert(!ao->open || ao->fail_timer == NULL);
+
+	g_mutex_lock(ao->mutex);
+	audio_output_close_locked(ao);
+	g_mutex_unlock(ao->mutex);
+}
+
 void audio_output_finish(struct audio_output *ao)
 {
 	audio_output_close(ao);

src/output_internal.h

@@ -131,7 +131,8 @@ struct audio_output {
 	const struct music_pipe *pipe;
 
 	/**
-	 * This mutex protects #open, #chunk and #chunk_finished.
+	 * This mutex protects #open, #fail_timer, #chunk and
+	 * #chunk_finished.
 	 */
 	GMutex *mutex;
 

src/output_thread.c

@@ -105,7 +105,12 @@ ao_play_chunk(struct audio_output *ao, const struct music_chunk *chunk)
 
 			/* don't automatically reopen this device for
 			   10 seconds */
+			g_mutex_lock(ao->mutex);
+
+			assert(ao->fail_timer == NULL);
 			ao->fail_timer = g_timer_new();
+
+			g_mutex_unlock(ao->mutex);
 			return false;
 		}
 
@@ -192,10 +197,18 @@ static gpointer audio_output_task(gpointer arg)
 
 		case AO_COMMAND_OPEN:
 			assert(!ao->open);
-			assert(ao->fail_timer == NULL);
 			assert(ao->pipe != NULL);
 			assert(ao->chunk == NULL);
 
+			if (ao->fail_timer != NULL) {
+				/* this can only happen when this
+				   output thread fails while
+				   audio_output_open() is run in the
+				   player thread */
+				g_timer_destroy(ao->fail_timer);
+				ao->fail_timer = NULL;
+			}
+
 			error = NULL;
 			ret = ao_plugin_open(ao->plugin, ao->data,
 					     &ao->out_audio_format,

src/pipe.h

@@ -20,9 +20,9 @@
 #ifndef MPD_PIPE_H
 #define MPD_PIPE_H
 
-#ifndef NDEBUG
 #include <stdbool.h>
 
+#ifndef NDEBUG
 struct audio_format;
 #endif
 
@@ -99,4 +99,10 @@ music_pipe_push(struct music_pipe *mp, struct music_chunk *chunk);
 unsigned
 music_pipe_size(const struct music_pipe *mp);
 
+static inline bool
+music_pipe_empty(const struct music_pipe *mp)
+{
+	return music_pipe_size(mp) == 0;
+}
+
 #endif

src/player_thread.c

@@ -118,6 +118,33 @@ static void player_command_finished(void)
 	notify_signal(&main_notify);
 }
 
+/**
+ * Is the decoder still busy on the same song as the player?
+ *
+ * Note: this function does not check if the decoder is already
+ * finished.
+ */
+static bool
+player_dc_at_current_song(const struct player *player)
+{
+	assert(player != NULL);
+	assert(player->pipe != NULL);
+
+	return dc.pipe == player->pipe;
+}
+
+/**
+ * Has the decoder already begun decoding the next song?
+ *
+ * Note: this function does not check if the decoder is already
+ * finished.
+ */
+static bool
+player_dc_at_next_song(const struct player *player)
+{
+	return dc.pipe != NULL && !player_dc_at_current_song(player);
+}
+
 /**
  * Stop the decoder and clears (and frees) its music pipe.
  */
@@ -297,10 +324,9 @@ static bool player_seek_decoder(struct player *player)
 		/* clear music chunks which might still reside in the
 		   pipe */
 		music_pipe_clear(player->pipe, player_buffer);
-		dc.pipe = player->pipe;
 
 		/* re-start the decoder */
-		dc_start_async(pc.next_song);
+		dc_start_async(pc.next_song, player->pipe);
 		ret = player_wait_for_decoder(player);
 		if (!ret) {
 			/* decoder failure */
@@ -308,6 +334,14 @@ static bool player_seek_decoder(struct player *player)
 			return false;
 		}
 	} else {
+		if (!player_dc_at_current_song(player)) {
+			/* the decoder is already decoding the "next" song,
+			   but it is the same song file; exchange the pipe */
+			music_pipe_clear(player->pipe, player_buffer);
+			music_pipe_free(player->pipe);
+			player->pipe = dc.pipe;
+		}
+
 		pc.next_song = NULL;
 		player->queued = false;
 	}
@@ -364,7 +398,7 @@ static void player_process_command(struct player *player)
 	case PLAYER_COMMAND_QUEUE:
 		assert(pc.next_song != NULL);
 		assert(!player->queued);
-		assert(dc.pipe == NULL || dc.pipe == player->pipe);
+		assert(!player_dc_at_next_song(player));
 
 		player->queued = true;
 		player_command_finished();
@@ -409,7 +443,7 @@ static void player_process_command(struct player *player)
 			return;
 		}
 
-		if (dc.pipe != NULL && dc.pipe != player->pipe)
+		if (player_dc_at_next_song(player))
 			/* the decoder is already decoding the song -
 			   stop it and reset the position */
 			player_dc_stop(player);
@@ -505,7 +539,7 @@ play_next_chunk(struct player *player)
 		return true;
 
 	if (player->xfade == XFADE_ENABLED &&
-	    dc.pipe != NULL && dc.pipe != player->pipe &&
+	    player_dc_at_next_song(player) &&
 	    (cross_fade_position = music_pipe_size(player->pipe))
 	    <= player->cross_fade_chunks) {
 		/* perform cross fade */
@@ -638,8 +672,7 @@ static void do_play(void)
 	player.pipe = music_pipe_new();
 
 	dc.buffer = player_buffer;
-	dc.pipe = player.pipe;
+	dc_start(&pc.notify, pc.next_song, player.pipe);
-	dc_start(&pc.notify, pc.next_song);
 	if (!player_wait_for_decoder(&player)) {
 		player_dc_stop(&player);
 		player_command_finished();
@@ -706,14 +739,15 @@ static void do_play(void)
 			/* the decoder has finished the current song;
 			   make it decode the next song */
 			assert(pc.next_song != NULL);
-			assert(dc.pipe == NULL || dc.pipe == player.pipe);
+			assert(!player_dc_at_next_song(&player));
+
+			dc.pipe = NULL;
 
 			player.queued = false;
-			dc.pipe = music_pipe_new();
+			dc_start_async(pc.next_song, music_pipe_new());
-			dc_start_async(pc.next_song);
 		}
 
-		if (dc.pipe != NULL && dc.pipe != player.pipe &&
+		if (player_dc_at_next_song(&player) &&
 		    player.xfade == XFADE_UNKNOWN &&
 		    !decoder_is_starting()) {
 			/* enable cross fading in this song?  if yes,
@@ -736,7 +770,7 @@ static void do_play(void)
 
 		if (player.paused)
 			notify_wait(&pc.notify);
-		else if (music_pipe_size(player.pipe) > 0) {
+		else if (!music_pipe_empty(player.pipe)) {
 			/* at least one music chunk is ready - send it
 			   to the audio output */
 
@@ -748,7 +782,7 @@ static void do_play(void)
 
 			/* XXX synchronize in a better way */
 			g_usleep(10000);
-		} else if (dc.pipe != NULL && dc.pipe != player.pipe) {
+		} else if (player_dc_at_next_song(&player)) {
 			/* at the beginning of a new song */
 
 			if (!player_song_border(&player))
@@ -757,7 +791,7 @@ static void do_play(void)
 			/* check the size of the pipe again, because
 			   the decoder thread may have added something
 			   since we last checked */
-			if (music_pipe_size(player.pipe) == 0)
+			if (music_pipe_empty(player.pipe))
 				break;
 		} else {
 			/* the decoder is too busy and hasn't provided

src/update.c

@@ -254,6 +254,9 @@ stat_directory(const struct directory *directory, struct stat *st)
 	if (path_fs == NULL)
 		return -1;
 	ret = stat(path_fs, st);
+	if (ret < 0)
+		g_warning("Failed to stat %s: %s", path_fs, g_strerror(errno));
+
 	g_free(path_fs);
 	return ret;
 }
@@ -270,6 +273,9 @@ stat_directory_child(const struct directory *parent, const char *name,
 		return -1;
 
 	ret = stat(path_fs, st);
+	if (ret < 0)
+		g_warning("Failed to stat %s: %s", path_fs, g_strerror(errno));
+
 	g_free(path_fs);
 	return ret;
 }

src/zeroconf-bonjour.c

@@ -62,7 +62,7 @@ void init_zeroconf_osx(const char *serviceName)
 	DNSServiceErrorType error = DNSServiceRegister(&dnsReference,
 						       0, 0, serviceName,
 						       SERVICE_TYPE, NULL, NULL,
-						       htons(listen_port), 0,
+						       g_htons(listen_port), 0,
 						       NULL,
 						       dnsRegisterCallback,
 						       NULL);