mpd version 0.17.3

Track whether _finish() has been called, and deliver all partial
results then.  Fixes Mantis ticket 0003621.

.gitignore

@@ -67,3 +67,7 @@ test/run_ntp_server
 test/run_resolver
 test/run_tcp_connect
 test/test_pcm
+test/dump_rva2
+test/dump_text_file
+test/test_byte_reverse
+test/test_vorbis_encoder

NEWS

@@ -1,3 +1,16 @@
+ver 0.17.3 (2013/01/06)
+* output:
+  - osx: fix pops during playback
+  - recorder: fix I/O error check
+  - shout: fix memory leak in error handler
+  - recorder, shout: support Ogg packets that span more than one page
+* decoder:
+  - ffmpeg: ignore negative time stamps
+  - ffmpeg: support planar audio
+* playlist:
+  - cue: fix memory leak
+  - cue: fix CUE files with only one track
+
 ver 0.17.2 (2012/09/30)
 * protocol:
   - fix crash in local file check

configure.ac

@@ -1,6 +1,6 @@
 AC_PREREQ(2.60)
 
-AC_INIT(mpd, 0.17.2, musicpd-dev-team@lists.sourceforge.net)
+AC_INIT(mpd, 0.17.3, musicpd-dev-team@lists.sourceforge.net)
 
 VERSION_MAJOR=0
 VERSION_MINOR=17

m4/mpd_auto.m4

@@ -73,7 +73,8 @@ AC_DEFUN([MPD_AUTO_PKG_LIB], [
 			[eval "found_$1=yes"],
 			AC_CHECK_LIB($4, $5,
 				[eval "found_$1=yes $2_LIBS='$6' $2_CFLAGS='$7'"],
-				[eval "found_$1=no"]))
+				[eval "found_$1=no"],
+				[$6]))
 	fi
 
 	MPD_AUTO_RESULT([$1], [$8], [$9])

mpd.service.in

@@ -1,6 +1,6 @@
 [Unit]
 Description=Music Player Daemon
-After=sound.target
+After=network.target sound.target
 
 [Service]
 ExecStart=@prefix@/bin/mpd --no-daemon

src/cmdline.c

@@ -213,12 +213,12 @@ parse_cmdline(int argc, char **argv, struct options *options,
 				if(g_file_test(system_path,
 						G_FILE_TEST_IS_REGULAR)) {
 					ret = config_read_file(system_path,error_r);
+					g_free(system_path);
 					break;
-				}
-				++i;;
+				} else
+					g_free(system_path);
+				++i;
 			}
-			g_free(system_path);
-			g_free(&system_config_dirs);
 		}
 #else /* G_OS_WIN32 */
 		char *path2;

src/cue/cue_parser.c

@@ -58,9 +58,35 @@ struct cue_parser {
 
 	char *filename;
 
-	struct song *current, *previous, *finished;
+	/**
+	 * The song currently being edited.
+	 */
+	struct song *current;
 
+	/**
+	 * The previous song.  It is remembered because its end_time
+	 * will be set to the current song's start time.
+	 */
+	struct song *previous;
+
+	/**
+	 * A song that is completely finished and can be returned to
+	 * the caller via cue_parser_get().
+	 */
+	struct song *finished;
+
+	/**
+	 * Set to true after previous.end_time has been updated to the
+	 * start time of the current song.
+	 */
 	bool last_updated;
+
+	/**
+	 * Tracks whether cue_parser_finish() has been called.  If
+	 * true, then all remaining (partial) results will be
+	 * delivered by cue_parser_get().
+	 */
+	bool end;
 };
 
 struct cue_parser *
@@ -73,6 +99,7 @@ cue_parser_new(void)
 	parser->current = NULL;
 	parser->previous = NULL;
 	parser->finished = NULL;
+	parser->end = false;
 	return parser;
 }
 
@@ -85,6 +112,9 @@ cue_parser_free(struct cue_parser *parser)
 	if (parser->current != NULL)
 		song_free(parser->current);
 
+	if (parser->previous != NULL)
+		song_free(parser->previous);
+
 	if (parser->finished != NULL)
 		song_free(parser->finished);
 
@@ -201,10 +231,32 @@ cue_parse_position(const char *p)
 	return minutes * 60000 + seconds * 1000 + frames * 1000 / 75;
 }
 
+/**
+ * Commit the current song.  It will be moved to "previous", so the
+ * next song may soon edit its end time (using the next song's start
+ * time).
+ */
+static void
+cue_parser_commit(struct cue_parser *parser)
+{
+	/* the caller of this library must call cue_parser_get() often
+	   enough */
+	assert(parser->finished == NULL);
+	assert(!parser->end);
+
+	if (parser->current == NULL)
+		return;
+
+	parser->finished = parser->previous;
+	parser->previous = parser->current;
+	parser->current = NULL;
+}
+
 static void
 cue_parser_feed2(struct cue_parser *parser, char *p)
 {
 	assert(parser != NULL);
+	assert(!parser->end);
 	assert(p != NULL);
 
 	const char *command = cue_next_token(&p);
@@ -235,7 +287,7 @@ cue_parser_feed2(struct cue_parser *parser, char *p)
 		else if (parser->state == TRACK)
 			cue_add_tag(parser->current->tag, TAG_TITLE, p);
 	} else if (strcmp(command, "FILE") == 0) {
-		cue_parser_finish(parser);
+		cue_parser_commit(parser);
 
 		const char *filename = cue_next_value(&p);
 		if (filename == NULL)
@@ -258,7 +310,7 @@ cue_parser_feed2(struct cue_parser *parser, char *p)
 	} else if (parser->state == IGNORE_FILE) {
 		return;
 	} else if (strcmp(command, "TRACK") == 0) {
-		cue_parser_finish(parser);
+		cue_parser_commit(parser);
 
 		const char *nr = cue_next_token(&p);
 		if (nr == NULL)
@@ -310,6 +362,7 @@ void
 cue_parser_feed(struct cue_parser *parser, const char *line)
 {
 	assert(parser != NULL);
+	assert(!parser->end);
 	assert(line != NULL);
 
 	char *allocated = g_strdup(line);
@@ -320,12 +373,12 @@ cue_parser_feed(struct cue_parser *parser, const char *line)
 void
 cue_parser_finish(struct cue_parser *parser)
 {
-	if (parser->finished != NULL)
-		song_free(parser->finished);
+	if (parser->end)
+		/* has already been called, ignore */
+		return;
 
-	parser->finished = parser->previous;
-	parser->previous = parser->current;
-	parser->current = NULL;
+	cue_parser_commit(parser);
+	parser->end = true;
 }
 
 struct song *
@@ -333,6 +386,15 @@ cue_parser_get(struct cue_parser *parser)
 {
 	assert(parser != NULL);
 
+	if (parser->finished == NULL && parser->end) {
+		/* cue_parser_finish() has been called already:
+		   deliver all remaining (partial) results */
+		assert(parser->current == NULL);
+
+		parser->finished = parser->previous;
+		parser->previous = NULL;
+	}
+
 	struct song *song = parser->finished;
 	parser->finished = NULL;
 	return song;

src/decoder/ffmpeg_decoder_plugin.c

@@ -234,6 +234,21 @@ time_to_ffmpeg(double t, const AVRational time_base)
 }
 
 #if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(53,25,0)
+
+static void
+copy_interleave_frame2(uint8_t *dest, uint8_t **src,
+		       unsigned nframes, unsigned nchannels,
+		       unsigned sample_size)
+{
+	for (unsigned frame = 0; frame < nframes; ++frame) {
+		for (unsigned channel = 0; channel < nchannels; ++channel) {
+			memcpy(dest, src[channel] + frame * sample_size,
+			       sample_size);
+			dest += sample_size;
+		}
+	}
+}
+
 /**
  * Copy PCM data from a AVFrame to an interleaved buffer.
  */
@@ -254,11 +269,10 @@ copy_interleave_frame(const AVCodecContext *codec_context,
 
 	if (av_sample_fmt_is_planar(codec_context->sample_fmt) &&
 	    codec_context->channels > 1) {
-		for (int i = 0, channels = codec_context->channels;
-		     i < channels; i++) {
-			memcpy(buffer, frame->extended_data[i], plane_size);
-			buffer += plane_size;
-		}
+		copy_interleave_frame2(buffer, frame->extended_data,
+				       frame->nb_samples,
+				       codec_context->channels,
+				       av_get_bytes_per_sample(codec_context->sample_fmt));
 	} else {
 		memcpy(buffer, frame->extended_data[0], data_size);
 	}
@@ -273,7 +287,7 @@ ffmpeg_send_packet(struct decoder *decoder, struct input_stream *is,
 		   AVCodecContext *codec_context,
 		   const AVRational *time_base)
 {
-	if (packet->pts != (int64_t)AV_NOPTS_VALUE)
+	if (packet->pts >= 0 && packet->pts != (int64_t)AV_NOPTS_VALUE)
 		decoder_timestamp(decoder,
 				  time_from_ffmpeg(packet->pts, *time_base));
 
@@ -352,12 +366,20 @@ ffmpeg_send_packet(struct decoder *decoder, struct input_stream *is,
 	return cmd;
 }
 
+#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(52, 94, 1)
+#define AVSampleFormat SampleFormat
+#endif
+
+G_GNUC_CONST
 static enum sample_format
-ffmpeg_sample_format(G_GNUC_UNUSED const AVCodecContext *codec_context)
+ffmpeg_sample_format(enum AVSampleFormat sample_fmt)
 {
-	switch (codec_context->sample_fmt) {
+	switch (sample_fmt) {
 #if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(52, 94, 1)
 	case AV_SAMPLE_FMT_S16:
+#if LIBAVUTIL_VERSION_INT >= AV_VERSION_INT(51,17,0)
+	case AV_SAMPLE_FMT_S16P:
+#endif
 #else
 	case SAMPLE_FMT_S16:
 #endif
@@ -365,16 +387,30 @@ ffmpeg_sample_format(G_GNUC_UNUSED const AVCodecContext *codec_context)
 
 #if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(52, 94, 1)
 	case AV_SAMPLE_FMT_S32:
+#if LIBAVUTIL_VERSION_INT >= AV_VERSION_INT(51,17,0)
+	case AV_SAMPLE_FMT_S32P:
+#endif
 #else
 	case SAMPLE_FMT_S32:
 #endif
 		return SAMPLE_FORMAT_S32;
 
 	default:
-		g_warning("Unsupported libavcodec SampleFormat value: %d",
-			  codec_context->sample_fmt);
-		return SAMPLE_FORMAT_UNDEFINED;
+		break;
 	}
+
+#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(52, 94, 1)
+	char buffer[64];
+	const char *name = av_get_sample_fmt_string(buffer, sizeof(buffer),
+						    sample_fmt);
+	if (name != NULL)
+		g_warning("Unsupported libavcodec SampleFormat value: %s (%d)",
+			  name, sample_fmt);
+	else
+#endif
+		g_warning("Unsupported libavcodec SampleFormat value: %d",
+			  sample_fmt);
+	return SAMPLE_FORMAT_UNDEFINED;
 }
 
 static AVInputFormat *
@@ -485,11 +521,16 @@ ffmpeg_decode(struct decoder *decoder, struct input_stream *input)
 		return;
 	}
 
+	const enum sample_format sample_format =
+		ffmpeg_sample_format(codec_context->sample_fmt);
+	if (sample_format == SAMPLE_FORMAT_UNDEFINED)
+		return;
+
 	GError *error = NULL;
 	struct audio_format audio_format;
 	if (!audio_format_init_checked(&audio_format,
 				       codec_context->sample_rate,
-				       ffmpeg_sample_format(codec_context),
+				       sample_format,
 				       codec_context->channels, &error)) {
 		g_warning("%s", error->message);
 		g_error_free(error);

src/encoder/vorbis_encoder.c

@@ -65,13 +65,11 @@ static bool
 vorbis_encoder_configure(struct vorbis_encoder *encoder,
 			 const struct config_param *param, GError **error)
 {
-	const char *value;
-	char *endptr;
-
-	value = config_get_block_string(param, "quality", NULL);
+	const char *value = config_get_block_string(param, "quality", NULL);
 	if (value != NULL) {
 		/* a quality was configured (VBR) */
 
+		char *endptr;
 		encoder->quality = g_ascii_strtod(value, &endptr);
 
 		if (*endptr != '\0' || encoder->quality < -1.0 ||
@@ -103,8 +101,9 @@ vorbis_encoder_configure(struct vorbis_encoder *encoder,
 		}
 
 		encoder->quality = -2.0;
-		encoder->bitrate = g_ascii_strtoll(value, &endptr, 10);
 
+		char *endptr;
+		encoder->bitrate = g_ascii_strtoll(value, &endptr, 10);
 		if (*endptr != '\0' || encoder->bitrate <= 0) {
 			g_set_error(error, vorbis_encoder_quark(), 0,
 				    "bitrate at line %i should be a positive integer",
@@ -119,9 +118,7 @@ vorbis_encoder_configure(struct vorbis_encoder *encoder,
 static struct encoder *
 vorbis_encoder_init(const struct config_param *param, GError **error)
 {
-	struct vorbis_encoder *encoder;
-
-	encoder = g_new(struct vorbis_encoder, 1);
+	struct vorbis_encoder *encoder = g_new(struct vorbis_encoder, 1);
 	encoder_struct_init(&encoder->encoder, &vorbis_encoder_plugin);
 
 	/* load configuration from "param" */
@@ -211,14 +208,12 @@ vorbis_encoder_open(struct encoder *_encoder,
 		    GError **error)
 {
 	struct vorbis_encoder *encoder = (struct vorbis_encoder *)_encoder;
-	bool ret;
 
 	audio_format->format = SAMPLE_FORMAT_S16;
 
 	encoder->audio_format = *audio_format;
 
-	ret = vorbis_encoder_reinit(encoder, error);
-	if (!ret)
+	if (!vorbis_encoder_reinit(encoder, error))
 		return false;
 
 	vorbis_encoder_send_header(encoder);
@@ -251,11 +246,10 @@ static void
 vorbis_encoder_blockout(struct vorbis_encoder *encoder)
 {
 	while (vorbis_analysis_blockout(&encoder->vd, &encoder->vb) == 1) {
-		ogg_packet packet;
-
 		vorbis_analysis(&encoder->vb, NULL);
 		vorbis_bitrate_addblock(&encoder->vb);
 
+		ogg_packet packet;
 		while (vorbis_bitrate_flushpacket(&encoder->vd, &packet))
 			ogg_stream_packetin(&encoder->os, &packet);
 	}
@@ -344,9 +338,9 @@ vorbis_encoder_write(struct encoder *_encoder,
 		     G_GNUC_UNUSED GError **error)
 {
 	struct vorbis_encoder *encoder = (struct vorbis_encoder *)_encoder;
-	unsigned num_frames;
 
-	num_frames = length / audio_format_frame_size(&encoder->audio_format);
+	unsigned num_frames = length
+		/ audio_format_frame_size(&encoder->audio_format);
 
 	/* this is for only 16-bit audio */
 
@@ -364,12 +358,10 @@ static size_t
 vorbis_encoder_read(struct encoder *_encoder, void *_dest, size_t length)
 {
 	struct vorbis_encoder *encoder = (struct vorbis_encoder *)_encoder;
-	ogg_page page;
-	int ret;
 	unsigned char *dest = _dest;
-	size_t nbytes;
 
-	ret = ogg_stream_pageout(&encoder->os, &page);
+	ogg_page page;
+	int ret = ogg_stream_pageout(&encoder->os, &page);
 	if (ret == 0 && encoder->flush) {
 		encoder->flush = false;
 		ret = ogg_stream_flush(&encoder->os, &page);
@@ -381,7 +373,7 @@ vorbis_encoder_read(struct encoder *_encoder, void *_dest, size_t length)
 
 	assert(page.header_len > 0 || page.body_len > 0);
 
-	nbytes = (size_t)page.header_len + (size_t)page.body_len;
+	size_t nbytes = (size_t)page.header_len + (size_t)page.body_len;
 
 	if (nbytes > length)
 		/* XXX better error handling */

src/encoder_plugin.h

@@ -119,6 +119,10 @@ encoder_finish(struct encoder *encoder)
  * Before you free it, you must call encoder_close().  You may open
  * and close (reuse) one encoder any number of times.
  *
+ * After this function returns successfully and before the first
+ * encoder_write() call, you should invoke encoder_read() to obtain
+ * the file header.
+ *
  * @param encoder the encoder
  * @param audio_format the encoder's input audio format; the plugin
  * may modify the struct to adapt it to its abilities
@@ -291,6 +295,8 @@ encoder_write(struct encoder *encoder, const void *data, size_t length,
 /**
  * Reads encoded data from the encoder.
  *
+ * Call this repeatedly until no more data is returned.
+ *
  * @param encoder the encoder
  * @param dest the destination buffer to copy to
  * @param length the maximum length of the destination buffer

src/output/httpd_output_plugin.c

@@ -114,10 +114,6 @@ httpd_output_init(const struct config_param *param,
 		return NULL;
 	}
 
-	const char *encoder_name, *bind_to_address;
-	const struct encoder_plugin *encoder_plugin;
-	guint port;
-
 	/* read configuration */
 	httpd->name =
 		config_get_block_string(param, "name", "Set name in config");
@@ -126,10 +122,12 @@ httpd_output_init(const struct config_param *param,
 	httpd->website =
 		config_get_block_string(param, "website", "Set website in config");
 
-	port = config_get_block_unsigned(param, "port", 8000);
+	guint port = config_get_block_unsigned(param, "port", 8000);
 
-	encoder_name = config_get_block_string(param, "encoder", "vorbis");
-	encoder_plugin = encoder_plugin_get(encoder_name);
+	const char *encoder_name =
+		config_get_block_string(param, "encoder", "vorbis");
+	const struct encoder_plugin *encoder_plugin =
+		encoder_plugin_get(encoder_name);
 	if (encoder_plugin == NULL) {
 		g_set_error(error, httpd_output_quark(), 0,
 			    "No such encoder: %s", encoder_name);
@@ -144,7 +142,7 @@ httpd_output_init(const struct config_param *param,
 
 	httpd->server_socket = server_socket_new(httpd_listen_in_event, httpd);
 
-	bind_to_address =
+	const char *bind_to_address =
 		config_get_block_string(param, "bind_to_address", NULL);
 	bool success = bind_to_address != NULL &&
 		strcmp(bind_to_address, "any") != 0
@@ -275,8 +273,6 @@ httpd_listen_in_event(int fd, const struct sockaddr *address,
 static struct page *
 httpd_output_read_page(struct httpd_output *httpd)
 {
-	size_t size = 0, nbytes;
-
 	if (httpd->unflushed_input >= 65536) {
 		/* we have fed a lot of input into the encoder, but it
 		   didn't give anything back yet - flush now to avoid
@@ -285,9 +281,11 @@ httpd_output_read_page(struct httpd_output *httpd)
 		httpd->unflushed_input = 0;
 	}
 
+	size_t size = 0;
 	do {
-		nbytes = encoder_read(httpd->encoder, httpd->buffer + size,
-				      sizeof(httpd->buffer) - size);
+		size_t nbytes = encoder_read(httpd->encoder,
+					     httpd->buffer + size,
+					     sizeof(httpd->buffer) - size);
 		if (nbytes == 0)
 			break;
 
@@ -307,10 +305,7 @@ httpd_output_encoder_open(struct httpd_output *httpd,
 			  struct audio_format *audio_format,
 			  GError **error)
 {
-	bool success;
-
-	success = encoder_open(httpd->encoder, audio_format, error);
-	if (!success)
+	if (!encoder_open(httpd->encoder, audio_format, error))
 		return false;
 
 	/* we have to remember the encoder header, i.e. the first
@@ -344,14 +339,12 @@ httpd_output_open(struct audio_output *ao, struct audio_format *audio_format,
 		  GError **error)
 {
 	struct httpd_output *httpd = (struct httpd_output *)ao;
-	bool success;
 
 	g_mutex_lock(httpd->mutex);
 
 	/* open the encoder */
 
-	success = httpd_output_encoder_open(httpd, audio_format, error);
-	if (!success) {
+	if (!httpd_output_encoder_open(httpd, audio_format, error)) {
 		g_mutex_unlock(httpd->mutex);
 		return false;
 	}
@@ -495,10 +488,7 @@ static bool
 httpd_output_encode_and_play(struct httpd_output *httpd,
 			     const void *chunk, size_t size, GError **error)
 {
-	bool success;
-
-	success = encoder_write(httpd->encoder, chunk, size, error);
-	if (!success)
+	if (!encoder_write(httpd->encoder, chunk, size, error))
 		return false;
 
 	httpd->unflushed_input += size;
@@ -510,16 +500,12 @@ httpd_output_encode_and_play(struct httpd_output *httpd,
 
 static size_t
 httpd_output_play(struct audio_output *ao, const void *chunk, size_t size,
-		  GError **error)
+		  GError **error_r)
 {
 	struct httpd_output *httpd = (struct httpd_output *)ao;
 
 	if (httpd_output_lock_has_clients(httpd)) {
-		bool success;
-
-		success = httpd_output_encode_and_play(httpd, chunk, size,
-						       error);
-		if (!success)
+		if (!httpd_output_encode_and_play(httpd, chunk, size, error_r))
 			return 0;
 	}
 
@@ -562,7 +548,6 @@ httpd_output_tag(struct audio_output *ao, const struct tag *tag)
 
 	if (httpd->encoder->plugin->tag != NULL) {
 		/* embed encoder tags */
-		struct page *page;
 
 		/* flush the current stream, and end it */
 
@@ -578,7 +563,7 @@ httpd_output_tag(struct audio_output *ao, const struct tag *tag)
 		   used as the new "header" page, which is sent to all
 		   new clients */
 
-		page = httpd_output_read_page(httpd);
+		struct page *page = httpd_output_read_page(httpd);
 		if (page != NULL) {
 			if (httpd->header != NULL)
 				page_unref(httpd->header);

src/output/osx_output_plugin.c

@@ -228,9 +228,13 @@ osx_render(void *vdata,
 	g_cond_signal(od->condition);
 	g_mutex_unlock(od->mutex);
 
-	if (nbytes < buffer_size)
-		memset((unsigned char*)buffer->mData + nbytes, 0,
-		       buffer_size - nbytes);
+	buffer->mDataByteSize = nbytes;
+
+	unsigned i;
+	for (i = 1; i < buffer_list->mNumberBuffers; ++i) {
+		buffer = &buffer_list->mBuffers[i];
+		buffer->mDataByteSize = 0;
+	}
 
 	return 0;
 }

src/output/recorder_output_plugin.c

@@ -77,13 +77,12 @@ recorder_output_init(const struct config_param *param, GError **error_r)
 		return NULL;
 	}
 
-	const char *encoder_name;
-	const struct encoder_plugin *encoder_plugin;
-
 	/* read configuration */
 
-	encoder_name = config_get_block_string(param, "encoder", "vorbis");
-	encoder_plugin = encoder_plugin_get(encoder_name);
+	const char *encoder_name =
+		config_get_block_string(param, "encoder", "vorbis");
+	const struct encoder_plugin *encoder_plugin =
+		encoder_plugin_get(encoder_name);
 	if (encoder_plugin == NULL) {
 		g_set_error(error_r, recorder_output_quark(), 0,
 			    "No such encoder: %s", encoder_name);
@@ -121,33 +120,22 @@ recorder_output_finish(struct audio_output *ao)
 	g_free(recorder);
 }
 
-/**
- * Writes pending data from the encoder to the output file.
- */
 static bool
-recorder_output_encoder_to_file(struct recorder_output *recorder,
-			      GError **error_r)
+recorder_write_to_file(struct recorder_output *recorder,
+		       const void *_data, size_t length,
+		       GError **error_r)
 {
-	size_t size = 0, position, nbytes;
+	assert(length > 0);
 
-	assert(recorder->fd >= 0);
+	const int fd = recorder->fd;
 
-	/* read from the encoder */
+	const uint8_t *data = (const uint8_t *)_data, *end = data + length;
 
-	size = encoder_read(recorder->encoder, recorder->buffer,
-			    sizeof(recorder->buffer));
-	if (size == 0)
-		return true;
-
-	/* write everything into the file */
-
-	position = 0;
 	while (true) {
-		nbytes = write(recorder->fd, recorder->buffer + position,
-			       size - position);
+		ssize_t nbytes = write(fd, data, end - data);
 		if (nbytes > 0) {
-			position += (size_t)nbytes;
-			if (position >= size)
+			data += nbytes;
+			if (data == end)
 				return true;
 		} else if (nbytes == 0) {
 			/* shouldn't happen for files */
@@ -163,13 +151,37 @@ recorder_output_encoder_to_file(struct recorder_output *recorder,
 	}
 }
 
+/**
+ * Writes pending data from the encoder to the output file.
+ */
+static bool
+recorder_output_encoder_to_file(struct recorder_output *recorder,
+				GError **error_r)
+{
+	assert(recorder->fd >= 0);
+
+	while (true) {
+		/* read from the encoder */
+
+		size_t size = encoder_read(recorder->encoder, recorder->buffer,
+					   sizeof(recorder->buffer));
+		if (size == 0)
+			return true;
+
+		/* write everything into the file */
+
+		if (!recorder_write_to_file(recorder, recorder->buffer, size,
+					    error_r))
+			return false;
+	}
+}
+
 static bool
 recorder_output_open(struct audio_output *ao,
 		     struct audio_format *audio_format,
 		     GError **error_r)
 {
 	struct recorder_output *recorder = (struct recorder_output *)ao;
-	bool success;
 
 	/* create the output file */
 
@@ -185,8 +197,14 @@ recorder_output_open(struct audio_output *ao,
 
 	/* open the encoder */
 
-	success = encoder_open(recorder->encoder, audio_format, error_r);
-	if (!success) {
+	if (!encoder_open(recorder->encoder, audio_format, error_r)) {
+		close(recorder->fd);
+		unlink(recorder->path);
+		return false;
+	}
+
+	if (!recorder_output_encoder_to_file(recorder, error_r)) {
+		encoder_close(recorder->encoder);
 		close(recorder->fd);
 		unlink(recorder->path);
 		return false;

src/output/shout_output_plugin.c

@@ -36,11 +36,6 @@
 
 #define DEFAULT_CONN_TIMEOUT  2
 
-struct shout_buffer {
-	unsigned char data[32768];
-	size_t len;
-};
-
 struct shout_data {
 	struct audio_output base;
 
@@ -54,7 +49,7 @@ struct shout_data {
 
 	int timeout;
 
-	struct shout_buffer buf;
+	uint8_t buffer[32768];
 };
 
 static int shout_init_count;
@@ -114,24 +109,7 @@ static struct audio_output *
 my_shout_init_driver(const struct config_param *param,
 		     GError **error)
 {
-	struct shout_data *sd;
-	char *test;
-	unsigned port;
-	char *host;
-	char *mount;
-	char *passwd;
-	const char *encoding;
-	const struct encoder_plugin *encoder_plugin;
-	unsigned shout_format;
-	unsigned protocol;
-	const char *user;
-	char *name;
-	const char *value;
-	const struct block_param *block_param;
-	int public;
-
-	sd = new_shout_data();
-
+	struct shout_data *sd = new_shout_data();
 	if (!ao_base_init(&sd->base, &shout_output_plugin, param, error)) {
 		free_shout_data(sd);
 		return NULL;
@@ -152,13 +130,14 @@ my_shout_init_driver(const struct config_param *param,
 
 	shout_init_count++;
 
+	const struct block_param *block_param;
 	check_block_param("host");
-	host = block_param->value;
+	char *host = block_param->value;
 
 	check_block_param("mount");
-	mount = block_param->value;
+	char *mount = block_param->value;
 
-	port = config_get_block_unsigned(param, "port", 0);
+	unsigned port = config_get_block_unsigned(param, "port", 0);
 	if (port == 0) {
 		g_set_error(error, shout_output_quark(), 0,
 			    "shout port must be configured");
@@ -166,17 +145,18 @@ my_shout_init_driver(const struct config_param *param,
 	}
 
 	check_block_param("password");
-	passwd = block_param->value;
+	const char *passwd = block_param->value;
 
 	check_block_param("name");
-	name = block_param->value;
+	const char *name = block_param->value;
 
-	public = config_get_block_bool(param, "public", false);
+	bool public = config_get_block_bool(param, "public", false);
 
-	user = config_get_block_string(param, "user", "source");
+	const char *user = config_get_block_string(param, "user", "source");
 
-	value = config_get_block_string(param, "quality", NULL);
+	const char *value = config_get_block_string(param, "quality", NULL);
 	if (value != NULL) {
+		char *test;
 		sd->quality = strtod(value, &test);
 
 		if (*test != '\0' || sd->quality < -1.0 || sd->quality > 10.0) {
@@ -201,6 +181,7 @@ my_shout_init_driver(const struct config_param *param,
 			goto failure;
 		}
 
+		char *test;
 		sd->bitrate = strtol(value, &test, 10);
 
 		if (*test != '\0' || sd->bitrate <= 0) {
@@ -210,8 +191,10 @@ my_shout_init_driver(const struct config_param *param,
 		}
 	}
 
-	encoding = config_get_block_string(param, "encoding", "ogg");
-	encoder_plugin = shout_encoder_plugin_get(encoding);
+	const char *encoding = config_get_block_string(param, "encoding",
+						       "ogg");
+	const struct encoder_plugin *encoder_plugin =
+		shout_encoder_plugin_get(encoding);
 	if (encoder_plugin == NULL) {
 		g_set_error(error, shout_output_quark(), 0,
 			    "couldn't find shout encoder plugin \"%s\"",
@@ -223,11 +206,13 @@ my_shout_init_driver(const struct config_param *param,
 	if (sd->encoder == NULL)
 		goto failure;
 
+	unsigned shout_format;
 	if (strcmp(encoding, "mp3") == 0 || strcmp(encoding, "lame") == 0)
 		shout_format = SHOUT_FORMAT_MP3;
 	else
 		shout_format = SHOUT_FORMAT_OGG;
 
+	unsigned protocol;
 	value = config_get_block_string(param, "protocol", NULL);
 	if (value != NULL) {
 		if (0 == strcmp(value, "shoutcast") &&
@@ -355,26 +340,24 @@ handle_shout_error(struct shout_data *sd, int err, GError **error)
 static bool
 write_page(struct shout_data *sd, GError **error)
 {
-	int err;
-
 	assert(sd->encoder != NULL);
 
-	sd->buf.len = encoder_read(sd->encoder,
-				   sd->buf.data, sizeof(sd->buf.data));
-	if (sd->buf.len == 0)
-		return true;
+	while (true) {
+		size_t nbytes = encoder_read(sd->encoder,
+					     sd->buffer, sizeof(sd->buffer));
+		if (nbytes == 0)
+			return true;
 
-	err = shout_send(sd->shout_conn, sd->buf.data, sd->buf.len);
-	if (!handle_shout_error(sd, err, error))
-		return false;
+		int err = shout_send(sd->shout_conn, sd->buffer, nbytes);
+		if (!handle_shout_error(sd, err, error))
+			return false;
+	}
 
 	return true;
 }
 
 static void close_shout_conn(struct shout_data * sd)
 {
-	sd->buf.len = 0;
-
 	if (sd->encoder != NULL) {
 		if (encoder_end(sd->encoder, NULL))
 			write_page(sd, NULL);
@@ -425,10 +408,7 @@ my_shout_close_device(struct audio_output *ao)
 static bool
 shout_connect(struct shout_data *sd, GError **error)
 {
-	int state;
-
-	state = shout_open(sd->shout_conn);
-	switch (state) {
+	switch (shout_open(sd->shout_conn)) {
 	case SHOUTERR_SUCCESS:
 	case SHOUTERR_CONNECTED:
 		return true;
@@ -448,17 +428,17 @@ my_shout_open_device(struct audio_output *ao, struct audio_format *audio_format,
 		     GError **error)
 {
 	struct shout_data *sd = (struct shout_data *)ao;
-	bool ret;
 
-	ret = shout_connect(sd, error);
-	if (!ret)
+	if (!shout_connect(sd, error))
 		return false;
 
-	sd->buf.len = 0;
+	if (!encoder_open(sd->encoder, audio_format, error)) {
+		shout_close(sd->shout_conn);
+		return false;
+	}
 
-	ret = encoder_open(sd->encoder, audio_format, error) &&
-		write_page(sd, error);
-	if (!ret) {
+	if (!write_page(sd, error)) {
+		encoder_close(sd->encoder);
 		shout_close(sd->shout_conn);
 		return false;
 	}
@@ -528,32 +508,27 @@ static void my_shout_set_tag(struct audio_output *ao,
 			     const struct tag *tag)
 {
 	struct shout_data *sd = (struct shout_data *)ao;
-	bool ret;
 	GError *error = NULL;
 
 	if (sd->encoder->plugin->tag != NULL) {
 		/* encoder plugin supports stream tags */
 
-		ret = encoder_pre_tag(sd->encoder, &error);
-		if (!ret) {
+		if (!encoder_pre_tag(sd->encoder, &error)) {
 			g_warning("%s", error->message);
 			g_error_free(error);
 			return;
 		}
 
-		ret = write_page(sd, NULL);
-		if (!ret)
+		if (!write_page(sd, NULL))
 			return;
 
-		ret = encoder_tag(sd->encoder, tag, &error);
-		if (!ret) {
+		if (!encoder_tag(sd->encoder, tag, &error)) {
 			g_warning("%s", error->message);
 			g_error_free(error);
 		}
 	} else {
 		/* no stream tag support: fall back to icy-metadata */
 		char song[1024];
-
 		shout_tag_to_metadata(tag, song, sizeof(song));
 
 		shout_metadata_add(sd->shout_meta, "song", song);

src/playlist_save.c

@@ -137,7 +137,8 @@ playlist_load_spl(struct playlist *playlist, struct player_control *pc,
 					*p = '/';
 				p++;
 			}
-			if ((playlist_append_uri(playlist, pc, temp, NULL)) != PLAYLIST_RESULT_SUCCESS) {
+			if ((playlist_append_uri(playlist, pc, temp2,
+						 NULL)) != PLAYLIST_RESULT_SUCCESS) {
 				g_warning("can't add file \"%s\"", temp2);
 			}
 			g_free(temp2);

src/playlist_song.c

@@ -23,6 +23,7 @@
 #include "mapper.h"
 #include "song.h"
 #include "uri.h"
+#include "path.h"
 #include "ls.h"
 #include "tag.h"
 
@@ -62,8 +63,14 @@ apply_song_metadata(struct song *dest, const struct song *src)
 		if (path_fs == NULL)
 			return dest;
 
-		tmp = song_file_new(path_fs, NULL);
-		g_free(path_fs);
+		char *path_utf8 = fs_charset_to_utf8(path_fs);
+		if (path_utf8 != NULL)
+			g_free(path_fs);
+		else
+			path_utf8 = path_fs;
+
+		tmp = song_file_new(path_utf8, NULL);
+		g_free(path_utf8);
 
 		merge_song_metadata(tmp, dest, src);
 	} else {

test/dump_playlist.c

@@ -161,7 +161,7 @@ int main(int argc, char **argv)
 	config_global_init();
 	success = config_read_file(argv[1], &error);
 	if (!success) {
-		g_printerr("%s:", error->message);
+		g_printerr("%s\n", error->message);
 		g_error_free(error);
 		return 1;
 	}

test/run_encoder.c

@@ -99,14 +99,15 @@ int main(int argc, char **argv)
 		}
 	}
 
-	ret = encoder_open(encoder, &audio_format, &error);
-	if (encoder == NULL) {
+	if (!encoder_open(encoder, &audio_format, &error)) {
 		g_printerr("Failed to open encoder: %s\n",
 			   error->message);
 		g_error_free(error);
 		return 1;
 	}
 
+	encoder_to_stdout(encoder);
+
 	/* do it */
 
 	while ((nbytes = read(0, buffer, sizeof(buffer))) > 0) {

test/test_vorbis_encoder.c

@@ -67,6 +67,8 @@ main(G_GNUC_UNUSED int argc, G_GNUC_UNUSED char **argv)
 	success = encoder_open(encoder, &audio_format, NULL);
 	assert(success);
 
+	encoder_to_stdout(encoder);
+
 	/* write a block of data */
 
 	success = encoder_write(encoder, zero, sizeof(zero), NULL);