Merge commit 'release-0.16.2'

Conflicts:
	Makefile.am
	NEWS
	configure.ac

Makefile.am

@@ -886,6 +886,7 @@ test_run_input_LDADD = $(MPD_LIBS) \
 	$(INPUT_LIBS) \
 	$(GLIB_LIBS)
 test_run_input_SOURCES = test/run_input.c \
+	test/stdbin.h \
 	src/conf.c src/tokenizer.c src/utils.c src/string_util.c\
 	src/tag.c src/tag_pool.c src/tag_save.c \
 	src/fd_util.c \
@@ -933,6 +934,7 @@ test_run_decoder_LDADD = $(MPD_LIBS) \
 	$(INPUT_LIBS) $(DECODER_LIBS) \
 	$(GLIB_LIBS)
 test_run_decoder_SOURCES = test/run_decoder.c \
+	test/stdbin.h \
 	src/conf.c src/tokenizer.c src/utils.c src/string_util.c src/log.c \
 	src/tag.c src/tag_pool.c \
 	src/replay_gain_info.c \
@@ -973,6 +975,7 @@ test_run_filter_LDADD = $(MPD_LIBS) \
 	$(SAMPLERATE_LIBS) \
 	$(GLIB_LIBS)
 test_run_filter_SOURCES = test/run_filter.c \
+	test/stdbin.h \
 	src/filter_plugin.c \
 	src/filter_registry.c \
 	src/conf.c src/tokenizer.c src/utils.c src/string_util.c \
@@ -995,6 +998,7 @@ endif
 if ENABLE_ENCODER
 noinst_PROGRAMS += test/run_encoder
 test_run_encoder_SOURCES = test/run_encoder.c \
+	test/stdbin.h \
 	src/conf.c src/tokenizer.c \
 	src/utils.c src/string_util.c \
 	src/tag.c src/tag_pool.c \
@@ -1002,12 +1006,15 @@ test_run_encoder_SOURCES = test/run_encoder.c \
 	src/audio_format.c \
 	src/audio_parser.c \
 	$(ENCODER_SRC)
+test_run_encoder_CPPFLAGS = $(AM_CPPFLAGS) \
+	$(ENCODER_CFLAGS)
 test_run_encoder_LDADD = $(MPD_LIBS) \
 	$(ENCODER_LIBS) \
 	$(GLIB_LIBS)
 endif
 
 test_software_volume_SOURCES = test/software_volume.c \
+	test/stdbin.h \
 	src/audio_check.c \
 	src/audio_parser.c \
 	src/pcm_volume.c
@@ -1015,6 +1022,7 @@ test_software_volume_LDADD = \
 	$(GLIB_LIBS)
 
 test_run_normalize_SOURCES = test/run_normalize.c \
+	test/stdbin.h \
 	src/audio_check.c \
 	src/audio_parser.c \
 	src/AudioCompress/compress.c
@@ -1052,6 +1060,7 @@ test_run_output_LDADD = $(MPD_LIBS) \
 	$(OUTPUT_LIBS) \
 	$(GLIB_LIBS)
 test_run_output_SOURCES = test/run_output.c \
+	test/stdbin.h \
 	src/conf.c src/tokenizer.c src/utils.c src/string_util.c src/log.c \
 	src/audio_check.c \
 	src/audio_format.c \

NEWS

@@ -15,6 +15,21 @@ ver 0.17 (2011/??/??)
 * state_file: add option "restore_paused"
 
 
+ver 0.16.2 (2011/03/18)
+* configure.ac:
+  - fix bashism in tremor test
+* decoder:
+  - tremor: fix configure test
+  - gme: detect end of song
+* encoder:
+  - vorbis: reset the Ogg stream after flush
+* output:
+  - httpd: fix uninitialized variable
+  - httpd: include sys/socket.h
+  - oss: AFMT_S24_PACKED is little-endian
+  - oss: disable 24 bit playback on FreeBSD
+
+
 ver 0.16.1 (2011/01/09)
 * audio_check: fix parameter in prototype
 * add void casts to suppress "result unused" warnings (clang)
@@ -145,9 +160,13 @@ ver 0.16 (2010/12/11)
 * make single mode 'sticky'
 
 
-ver 0.15.16 (2010/??/??)
+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)

configure.ac

@@ -660,7 +660,7 @@ fi
 AM_CONDITIONAL(ENABLE_CDIO_PARANOIA, test x$enable_cdio_paranoia = xyes)
 
 dnl ---------------------------------- libogg ---------------------------------
-if test x$with_tremor == xno || test -z $with_tremor; then
+if test x$with_tremor = xno || test -z $with_tremor; then
 	PKG_CHECK_MODULES(OGG, [ogg], enable_ogg=yes, enable_ogg=no)
 fi
 
@@ -959,13 +959,19 @@ if test x$enable_tremor = xyes; then
 	ac_save_LIBS="$LIBS"
 	CFLAGS="$CFLAGS $TREMOR_CFLAGS"
 	LIBS="$LIBS $TREMOR_LIBS"
-	AC_CHECK_LIB(vorbisidec,ov_read,enable_vorbis=yes,enable_vorbis=no;
+	AC_CHECK_LIB(vorbisidec,ov_read,,enable_tremor=no;
 		AC_MSG_WARN([vorbisidec lib needed for ogg support with tremor -- disabling ogg support]))
 	CFLAGS="$ac_save_CFLAGS"
 	LIBS="$ac_save_LIBS"
+fi
 
+if test x$enable_tremor = xyes; then
 	AC_DEFINE(HAVE_TREMOR,1,
 		[Define to use tremor (libvorbisidec) for ogg support])
+	AC_DEFINE(ENABLE_VORBIS_DECODER, 1, [Define for Ogg Vorbis support]),
+else
+	TREMOR_CFLAGS=
+	TREMOR_LIBS=
 fi
 
 AC_SUBST(TREMOR_CFLAGS)
@@ -1005,7 +1011,7 @@ if test x$enable_vorbis = xyes; then
 	fi
 fi
 
-AM_CONDITIONAL(ENABLE_VORBIS_DECODER, test x$enable_vorbis = xyes)
+AM_CONDITIONAL(ENABLE_VORBIS_DECODER, test x$enable_vorbis = xyes || test x$enable_tremor = xyes)
 
 dnl --------------------------------- sidplay ---------------------------------
 found_sidplay=$HAVE_CXX

src/AudioCompress/compress.c

@@ -16,16 +16,16 @@
 struct Compressor {
         //! The compressor's preferences
         struct CompressorConfig prefs;
-        
+
         //! History of the peak values
         int *peaks;
-                
+
         //! History of the gain values
         int *gain;
-        
+
         //! History of clip amounts
         int *clipped;
-        
+
         unsigned int pos;
         unsigned int bufsz;
 };
@@ -41,9 +41,9 @@ struct Compressor *Compressor_new(unsigned int history)
         obj->peaks = obj->gain = obj->clipped = NULL;
 	obj->bufsz = 0;
         obj->pos = 0;
-        
+
         Compressor_setHistory(obj, history);
-        
+
         return obj;
 }
 
@@ -70,7 +70,7 @@ void Compressor_setHistory(struct Compressor *obj, unsigned int history)
 {
 	if (!history)
                 history = BUCKETS;
-        
+
         obj->peaks = resizeArray(obj->peaks, history, obj->bufsz);
         obj->gain = resizeArray(obj->gain, history, obj->bufsz);
         obj->clipped = resizeArray(obj->clipped, history, obj->bufsz);
@@ -82,7 +82,7 @@ struct CompressorConfig *Compressor_getConfig(struct Compressor *obj)
         return &obj->prefs;
 }
 
-void Compressor_Process_int16(struct Compressor *obj, int16_t *audio, 
+void Compressor_Process_int16(struct Compressor *obj, int16_t *audio,
                               unsigned int count)
 {
         struct CompressorConfig *prefs = Compressor_getConfig(obj);
@@ -97,7 +97,7 @@ void Compressor_Process_int16(struct Compressor *obj, int16_t *audio,
         int *clipped = obj->clipped + slot;
         unsigned int ramp = count;
         int delta;
-        
+
 	ap = audio;
 	for (i = 0; i < count; i++)
 	{
@@ -124,15 +124,15 @@ void Compressor_Process_int16(struct Compressor *obj, int16_t *audio,
 
 	//! Determine target gain
 	newGain = (1 << 10)*prefs->target/peakVal;
-        
+
         //! Adjust the gain with inertia from the previous gain value
-        newGain = (curGain*((1 << prefs->smooth) - 1) + newGain) 
+        newGain = (curGain*((1 << prefs->smooth) - 1) + newGain)
                 >> prefs->smooth;
-        
+
         //! Make sure it's no more than the maximum gain value
         if (newGain > (prefs->maxgain << 10))
                 newGain = prefs->maxgain << 10;
-        
+
         //! Make sure it's no less than 1:1
 	if (newGain < (1 << 10))
 		newGain = 1 << 10;
@@ -144,7 +144,7 @@ void Compressor_Process_int16(struct Compressor *obj, int16_t *audio,
                 //! Truncate the ramp time
                 ramp = peakPos;
         }
-        
+
         //! Record the new gain
         obj->gain[slot] = newGain;
 

src/audio_format.h

@@ -22,6 +22,7 @@
 
 #include <stdint.h>
 #include <stdbool.h>
+#include <assert.h>
 
 enum sample_format {
 	SAMPLE_FORMAT_UNDEFINED = 0,
@@ -219,6 +220,9 @@ static inline void
 audio_format_mask_apply(struct audio_format *af,
 			const struct audio_format *mask)
 {
+	assert(audio_format_valid(af));
+	assert(audio_format_mask_valid(mask));
+
 	if (mask->sample_rate != 0)
 		af->sample_rate = mask->sample_rate;
 
@@ -227,6 +231,8 @@ audio_format_mask_apply(struct audio_format *af,
 
 	if (mask->channels != 0)
 		af->channels = mask->channels;
+
+	assert(audio_format_valid(af));
 }
 
 /**

src/audio_parser.c

@@ -192,6 +192,7 @@ audio_format_parse(struct audio_format *dest, const char *src,
 	}
 
 	audio_format_init(dest, rate, sample_format, channels);
+	assert(audio_format_valid(dest));
 
 	return true;
 }

src/command.c

@@ -763,7 +763,7 @@ handle_load(struct client *client, G_GNUC_UNUSED int argc, char *argv[])
 	result = playlist_open_into_queue(argv[1], &g_playlist,
 					  client->player_control, true);
 	if (result != PLAYLIST_RESULT_NO_SUCH_LIST)
-		return result;
+		return print_playlist_result(client, result);
 
 	result = playlist_load_spl(&g_playlist, client->player_control,
 				   argv[1]);

src/decoder/audiofile_decoder_plugin.c

@@ -244,7 +244,7 @@ static const char *const audiofile_suffixes[] = {
 static const char *const audiofile_mime_types[] = {
 	"audio/x-wav",
 	"audio/x-aiff",
-	NULL 
+	NULL
 };
 
 const struct decoder_plugin audiofile_decoder_plugin = {

src/decoder/gme_decoder_plugin.c

@@ -153,6 +153,9 @@ gme_file_decode(struct decoder *decoder, const char *path_fs)
 	if((gme_err = gme_start_track(emu, song_num)) != NULL)
 		g_warning("%s", gme_err);
 
+	if(ti->length > 0)
+		gme_set_fade(emu, ti->length);
+
 	/* play */
 	do {
 		gme_err = gme_play(emu, GME_BUFFER_SAMPLES, buf);

src/encoder/flac_encoder.c

@@ -55,7 +55,7 @@ static bool
 flac_encoder_configure(struct flac_encoder *encoder,
 		const struct config_param *param, G_GNUC_UNUSED GError **error)
 {
-	encoder->compression = config_get_block_unsigned(param, 
+	encoder->compression = config_get_block_unsigned(param,
 						"compression", 5);
 
 	return true;
@@ -218,7 +218,7 @@ flac_encoder_open(struct encoder *_encoder, struct audio_format *audio_format,
 
 		if (init_status != FLAC__STREAM_ENCODER_OK) {
 			g_set_error(error, flac_encoder_quark(), 0,
-			    "failed to initialize encoder: %s\n", 
+			    "failed to initialize encoder: %s\n",
 			    FLAC__StreamEncoderStateString[init_status]);
 			flac_encoder_close(_encoder);
 			return false;
@@ -234,7 +234,7 @@ flac_encoder_open(struct encoder *_encoder, struct audio_format *audio_format,
 
 		if(init_status != FLAC__STREAM_ENCODER_INIT_STATUS_OK) {
 			g_set_error(error, flac_encoder_quark(), 0,
-			    "failed to initialize encoder: %s\n", 
+			    "failed to initialize encoder: %s\n",
 			    FLAC__StreamEncoderInitStatusString[init_status]);
 			flac_encoder_close(_encoder);
 			return false;

src/encoder/vorbis_encoder.c

@@ -276,6 +276,8 @@ vorbis_encoder_flush(struct encoder *_encoder, G_GNUC_UNUSED GError **error)
 	vorbis_analysis_init(&encoder->vd, &encoder->vi);
 	vorbis_block_init(&encoder->vd, &encoder->vb);
 
+	ogg_stream_reset(&encoder->os);
+
 	encoder->flush = true;
 	return true;
 }

src/encoder/wave_encoder.c

@@ -58,7 +58,7 @@ wave_encoder_quark(void)
 }
 
 static void
-fill_wave_header(struct wave_header *header, int channels, int bits, 
+fill_wave_header(struct wave_header *header, int channels, int bits,
 		int freq, int block_size)
 {
 	int data_size = 0x0FFFFFFF;
@@ -142,7 +142,7 @@ wave_encoder_open(struct encoder *_encoder,
 	buffer = pcm_buffer_get(&encoder->buffer, sizeof(struct wave_header) );
 
 	/* create PCM wave header in initial buffer */
-	fill_wave_header((struct wave_header *) buffer, 
+	fill_wave_header((struct wave_header *) buffer,
 			audio_format->channels,
 			 encoder->bits,
 			audio_format->sample_rate,

src/mixer/winmm_mixer_plugin.c

@@ -58,11 +58,11 @@ winmm_mixer_init(void *ao, G_GNUC_UNUSED const struct config_param *param,
 		 G_GNUC_UNUSED GError **error_r)
 {
 	assert(ao != NULL);
-	
+
 	struct winmm_mixer *wm = g_new(struct winmm_mixer, 1);
 	mixer_init(&wm->base, &winmm_mixer_plugin);
 	wm->output = (struct winmm_output *) ao;
-	
+
 	return &wm->base;
 }
 
@@ -79,13 +79,13 @@ winmm_mixer_get_volume(struct mixer *mixer, GError **error_r)
 	DWORD volume;
 	HWAVEOUT handle = winmm_output_get_handle(wm->output);
 	MMRESULT result = waveOutGetVolume(handle, &volume);
-	
+
 	if (result != MMSYSERR_NOERROR) {
 		g_set_error(error_r, 0, winmm_mixer_quark(),
 			    "Failed to get winmm volume");
 		return -1;
 	}
-	
+
 	return winmm_volume_decode(volume);
 }
 
@@ -102,7 +102,7 @@ winmm_mixer_set_volume(struct mixer *mixer, unsigned volume, GError **error_r)
 			    "Failed to set winmm volume");
 		return false;
 	}
-	
+
 	return true;
 }
 

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) {

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;

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 */
 

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;
 

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/
  */

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:

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) {

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);

src/pcm_byteswap.c

@@ -49,7 +49,7 @@ const int16_t *pcm_byteswap_16(struct pcm_buffer *buffer,
 
 static inline uint32_t swab32(uint32_t x)
 {
-	return (x << 24) | 
+	return (x << 24) |
 		((x & 0xff00) << 8) |
 		((x & 0xff0000) >> 8) |
 		(x >> 24);

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
 

src/update_walk.c

@@ -300,6 +300,9 @@ 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;
 }
@@ -316,6 +319,9 @@ 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;
 }
@@ -557,6 +563,7 @@ directory_child_access(const struct directory *directory,
 	/* access() is useless on WIN32 */
 	(void)directory;
 	(void)name;
+	(void)mode;
 	return true;
 #else
 	char *path = map_directory_child_fs(directory, name);