Merge branch 'v0.16.x'

Conflicts:
	src/player_thread.c
	src/playlist_control.c

NEWS

@@ -21,9 +21,16 @@ ver 0.17 (2011/??/??)
 
 ver 0.16.4 (2011/??/??)
 * fix memory leaks
+* don't resume playback when seeking to another song while paused
+* apply follow_inside_symlinks to absolute symlinks
 * decoder:
   - ffmpeg: workaround for semantic API change in recent ffmpeg versions
   - flac: validate the sample rate when scanning the tag
+  - wavpack: obey all decoder commands, stop at CUE track border
+* encoder:
+  - vorbis: don't send end-of-stream on flush
+* output:
+  - alsa: fix SIGFPE when alsa announces a period size of 0
 
 
 ver 0.16.3 (2011/06/04)

src/decoder/wavpack_decoder_plugin.c

@@ -34,9 +34,6 @@
 #undef G_LOG_DOMAIN
 #define G_LOG_DOMAIN "wavpack"
 
-/* pick 1020 since its devisible for 8,16,24, and 32-bit audio */
-#define CHUNK_SIZE		1020
-
 #define ERRORLEN 80
 
 static struct {
@@ -162,8 +159,6 @@ wavpack_decode(struct decoder *decoder, WavpackContext *wpc, bool can_seek)
 	enum sample_format sample_format;
 	struct audio_format audio_format;
 	format_samples_t format_samples;
-	char chunk[CHUNK_SIZE];
-	int samples_requested, samples_got;
 	float total_time;
 	int bytes_per_sample, output_sample_size;
 
@@ -193,12 +188,15 @@ wavpack_decode(struct decoder *decoder, WavpackContext *wpc, bool can_seek)
 	output_sample_size = audio_format_frame_size(&audio_format);
 
 	/* wavpack gives us all kind of samples in a 32-bit space */
-	samples_requested = sizeof(chunk) / (4 * audio_format.channels);
+	int32_t chunk[1024];
+	const uint32_t samples_requested = G_N_ELEMENTS(chunk) /
+		audio_format.channels;
 
 	decoder_initialized(decoder, &audio_format, can_seek, total_time);
 
-	do {
-		if (decoder_get_command(decoder) == DECODE_COMMAND_SEEK) {
+	enum decoder_command cmd = decoder_get_command(decoder);
+	while (cmd != DECODE_COMMAND_STOP) {
+		if (cmd == DECODE_COMMAND_SEEK) {
 			if (can_seek) {
 				unsigned where = decoder_seek_where(decoder) *
 					audio_format.sample_rate;
@@ -213,29 +211,20 @@ wavpack_decode(struct decoder *decoder, WavpackContext *wpc, bool can_seek)
 			}
 		}
 
-		if (decoder_get_command(decoder) == DECODE_COMMAND_STOP) {
+		uint32_t samples_got = WavpackUnpackSamples(wpc, chunk,
+							    samples_requested);
+		if (samples_got == 0)
 			break;
-		}
 
-		samples_got = WavpackUnpackSamples(
-			wpc, (int32_t *)chunk, samples_requested
-		);
-		if (samples_got > 0) {
-			int bitrate = (int)(WavpackGetInstantBitrate(wpc) /
-			              1000 + 0.5);
+		int bitrate = (int)(WavpackGetInstantBitrate(wpc) / 1000 +
+				    0.5);
+		format_samples(bytes_per_sample, chunk,
+			       samples_got * audio_format.channels);
 
-			format_samples(
-				bytes_per_sample, chunk,
-				samples_got * audio_format.channels
-			);
-
-			decoder_data(
-				decoder, NULL, chunk,
-				samples_got * output_sample_size,
-				bitrate
-			);
-		}
-	} while (samples_got > 0);
+		cmd = decoder_data(decoder, NULL, chunk,
+				   samples_got * output_sample_size,
+				   bitrate);
+	}
 }
 
 /**

src/encoder/vorbis_encoder.c

@@ -266,6 +266,15 @@ vorbis_encoder_flush(struct encoder *_encoder, G_GNUC_UNUSED GError **error)
 {
 	struct vorbis_encoder *encoder = (struct vorbis_encoder *)_encoder;
 
+	encoder->flush = true;
+	return true;
+}
+
+static bool
+vorbis_encoder_pre_tag(struct encoder *_encoder, G_GNUC_UNUSED GError **error)
+{
+	struct vorbis_encoder *encoder = (struct vorbis_encoder *)_encoder;
+
 	vorbis_analysis_wrote(&encoder->vd, 0);
 	vorbis_encoder_blockout(encoder);
 
@@ -366,6 +375,7 @@ vorbis_encoder_read(struct encoder *_encoder, void *_dest, size_t length)
 	if (ret == 0 && encoder->flush) {
 		encoder->flush = false;
 		ret = ogg_stream_flush(&encoder->os, &page);
+
 	}
 
 	if (ret == 0)
@@ -398,6 +408,7 @@ const struct encoder_plugin vorbis_encoder_plugin = {
 	.open = vorbis_encoder_open,
 	.close = vorbis_encoder_close,
 	.flush = vorbis_encoder_flush,
+	.pre_tag = vorbis_encoder_pre_tag,
 	.tag = vorbis_encoder_tag,
 	.write = vorbis_encoder_write,
 	.read = vorbis_encoder_read,

src/encoder_plugin.h

@@ -50,6 +50,8 @@ struct encoder_plugin {
 
 	bool (*flush)(struct encoder *encoder, GError **error);
 
+	bool (*pre_tag)(struct encoder *encoder, GError **error);
+
 	bool (*tag)(struct encoder *encoder, const struct tag *tag,
 		    GError **error);
 
@@ -147,9 +149,31 @@ encoder_flush(struct encoder *encoder, GError **error)
 		: true;
 }
 
+/**
+ * Prepare for sending a tag to the encoder.  This is used by some
+ * encoders to flush the previous sub-stream, in preparation to begin
+ * a new one.
+ *
+ * @param encoder the encoder
+ * @param tag the tag object
+ * @param error location to store the error occuring, or NULL to ignore errors.
+ * @return true on success
+ */
+static inline bool
+encoder_pre_tag(struct encoder *encoder, GError **error)
+{
+	/* this method is optional */
+	return encoder->plugin->pre_tag != NULL
+		? encoder->plugin->pre_tag(encoder, error)
+		: true;
+}
+
 /**
  * Sends a tag to the encoder.
  *
+ * Instructions: call encoder_pre_tag(); then obtain flushed data with
+ * encoder_read(); finally call encoder_tag().
+ *
  * @param encoder the encoder
  * @param tag the tag object
  * @param error location to store the error occurring, or NULL to ignore errors.

src/output/alsa_plugin.c

@@ -508,6 +508,14 @@ configure_hw:
 	g_debug("buffer_size=%u period_size=%u",
 		(unsigned)alsa_buffer_size, (unsigned)alsa_period_size);
 
+	if (alsa_period_size == 0)
+		/* this works around a SIGFPE bug that occurred when
+		   an ALSA driver indicated period_size==0; this
+		   caused a division by zero in alsa_play().  By using
+		   the fallback "1", we make sure that this won't
+		   happen again. */
+		alsa_period_size = 1;
+
 	ad->period_frames = alsa_period_size;
 	ad->period_position = 0;
 

src/output/httpd_output_plugin.c

@@ -493,7 +493,8 @@ httpd_output_pause(void *data)
 
 	if (has_clients) {
 		static const char silence[1020];
-		return httpd_output_play(data, silence, sizeof(silence), NULL);
+		return httpd_output_play(data, silence, sizeof(silence),
+					 NULL) > 0;
 	} else {
 		g_usleep(100000);
 		return true;
@@ -522,7 +523,7 @@ httpd_output_tag(void *data, const struct tag *tag)
 
 		/* flush the current stream, and end it */
 
-		encoder_flush(httpd->encoder, NULL);
+		encoder_pre_tag(httpd->encoder, NULL);
 		httpd_output_encoder_to_clients(httpd);
 
 		/* send the tag to the encoder - which starts a new

src/output/shout_plugin.c

@@ -518,7 +518,7 @@ static void my_shout_set_tag(void *data,
 	if (sd->encoder->plugin->tag != NULL) {
 		/* encoder plugin supports stream tags */
 
-		ret = encoder_flush(sd->encoder, &error);
+		ret = encoder_pre_tag(sd->encoder, &error);
 		if (!ret) {
 			g_warning("%s", error->message);
 			g_error_free(error);

src/output_thread.c

@@ -641,8 +641,13 @@ static gpointer audio_output_task(gpointer arg)
 
 		case AO_COMMAND_CANCEL:
 			ao->chunk = NULL;
-			if (ao->open)
+
+			if (ao->open) {
+				g_mutex_unlock(ao->mutex);
 				ao_plugin_cancel(ao->plugin, ao->data);
+				g_mutex_lock(ao->mutex);
+			}
+
 			ao_command_finished(ao);
 
 			/* the player thread will now clear our music

src/pipe.c

@@ -187,5 +187,8 @@ music_pipe_push(struct music_pipe *mp, struct music_chunk *chunk)
 unsigned
 music_pipe_size(const struct music_pipe *mp)
 {
-	return mp->size;
+	g_mutex_lock(mp->mutex);
+	unsigned size = mp->size;
+	g_mutex_unlock(mp->mutex);
+	return size;
 }

src/pipe.h

@@ -20,6 +20,7 @@
 #ifndef MPD_PIPE_H
 #define MPD_PIPE_H
 
+#include <glib.h>
 #include <stdbool.h>
 
 #ifndef NDEBUG
@@ -38,6 +39,7 @@ struct music_pipe;
 /**
  * Creates a new #music_pipe object.  It is empty.
  */
+G_GNUC_MALLOC
 struct music_pipe *
 music_pipe_new(void);
 
@@ -70,6 +72,7 @@ music_pipe_contains(const struct music_pipe *mp,
  * Returns the first #music_chunk from the pipe.  Returns NULL if the
  * pipe is empty.
  */
+G_GNUC_PURE
 const struct music_chunk *
 music_pipe_peek(const struct music_pipe *mp);
 
@@ -96,9 +99,11 @@ music_pipe_push(struct music_pipe *mp, struct music_chunk *chunk);
 /**
  * Returns the number of chunks currently in this pipe.
  */
+G_GNUC_PURE
 unsigned
 music_pipe_size(const struct music_pipe *mp);
 
+G_GNUC_PURE
 static inline bool
 music_pipe_empty(const struct music_pipe *mp)
 {

src/player_thread.c

@@ -628,7 +628,9 @@ play_chunk(struct player_control *pc,
 		return true;
 	}
 
+	player_lock(pc);
 	pc->bit_rate = chunk->bit_rate;
+	player_unlock(pc);
 
 	/* send the chunk to the audio outputs */
 

src/playlist_control.c

@@ -230,10 +230,12 @@ playlist_seek_song(struct playlist *playlist, struct player_control *pc,
 	playlist->error_count = 0;
 
 	if (!playlist->playing || (unsigned)playlist->current != i) {
-		/* seeking is not within the current song - first
-		   start playing the new song */
+		/* seeking is not within the current song - prepare
+		   song change */
+
+		playlist->playing = true;
+		playlist->current = i;
 
-		playlist_play_order(playlist, pc, i);
 		queued = NULL;
 	}
 

src/update_walk.c

@@ -714,8 +714,14 @@ skip_symlink(const struct directory *directory, const char *utf8_name)
 		return false;
 	}
 
-	if (buffer[0] == '/')
-		return !follow_outside_symlinks;
+	if (g_path_is_absolute(buffer)) {
+		/* if the symlink points to an absolute path, see if
+		   that path is inside the music directory */
+		const char *relative = map_to_relative_path(buffer);
+		return relative > buffer
+			? !follow_inside_symlinks
+			: !follow_outside_symlinks;
+	}
 
 	p = buffer;
 	while (*p == '.') {