14 files changed, 287 insertions, 51 deletions

diff --git a/src/decoder/_flac_common.c b/src/decoder/_flac_common.c
index e096750f3..09f7269bd 100644
--- a/src/decoder/_flac_common.c
+++ b/src/decoder/_flac_common.c
@@ -201,9 +201,8 @@ void flac_metadata_common_cb(const FLAC__StreamMetadata * block,
 
 	switch (block->type) {
 	case FLAC__METADATA_TYPE_STREAMINFO:
-		data->audio_format.bits = (int8_t)si->bits_per_sample;
-		data->audio_format.sample_rate = si->sample_rate;
-		data->audio_format.channels = (int8_t)si->channels;
+		audio_format_init(&data->audio_format, si->sample_rate,
+				  si->bits_per_sample, si->channels);
 		data->total_time = ((float)si->total_samples) / (si->sample_rate);
 		break;
 	case FLAC__METADATA_TYPE_VORBIS_COMMENT:
diff --git a/src/decoder/audiofile_plugin.c b/src/decoder/audiofile_plugin.c
index f66d90dc1..b4959f6c2 100644
--- a/src/decoder/audiofile_plugin.c
+++ b/src/decoder/audiofile_plugin.c
@@ -136,11 +136,9 @@ audiofile_stream_decode(struct decoder *decoder, struct input_stream *is)
 	afSetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK,
 	                         AF_SAMPFMT_TWOSCOMP, bits);
 	afGetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits);
-	audio_format.bits = (uint8_t)bits;
-	audio_format.sample_rate =
-	                      (unsigned int)afGetRate(af_fp, AF_DEFAULT_TRACK);
-	audio_format.channels =
-	              (uint8_t)afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK);
+
+	audio_format_init(&audio_format, afGetRate(af_fp, AF_DEFAULT_TRACK),
+			  bits, afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK)); 
 
 	if (!audio_format_valid(&audio_format)) {
 		g_warning("Invalid audio format: %u:%u:%u\n",
diff --git a/src/decoder/faad_plugin.c b/src/decoder/faad_plugin.c
index d0537dd5b..1b8b2b784 100644
--- a/src/decoder/faad_plugin.c
+++ b/src/decoder/faad_plugin.c
@@ -262,11 +262,7 @@ faad_decoder_init(faacDecHandle decoder, struct decoder_buffer *buffer,
 
 	decoder_buffer_consume(buffer, nbytes);
 
-	*audio_format = (struct audio_format){
-		.bits = 16,
-		.channels = channels,
-		.sample_rate = sample_rate,
-	};
+	audio_format_init(audio_format, sample_rate, 16, channels);
 
 	return true;
 }
diff --git a/src/decoder/ffmpeg_plugin.c b/src/decoder/ffmpeg_plugin.c
index abccdf977..e6646f649 100644
--- a/src/decoder/ffmpeg_plugin.c
+++ b/src/decoder/ffmpeg_plugin.c
@@ -267,6 +267,7 @@ ffmpeg_decode_internal(struct ffmpeg_context *ctx)
 	struct audio_format audio_format;
 	enum decoder_command cmd;
 	int total_time;
+	uint8_t bits;
 
 	total_time = 0;
 
@@ -275,13 +276,13 @@ ffmpeg_decode_internal(struct ffmpeg_context *ctx)
 	}
 
 #if LIBAVCODEC_VERSION_INT >= ((51<<16)+(41<<8)+0)
-	audio_format.bits = (uint8_t) av_get_bits_per_sample_format(codec_context->sample_fmt);
+	bits = (uint8_t) av_get_bits_per_sample_format(codec_context->sample_fmt);
 #else
 	/* XXX fixme 16-bit for older ffmpeg (13 Aug 2007) */
-	audio_format.bits = (uint8_t) 16;
+	bits = (uint8_t) 16;
 #endif
-	audio_format.sample_rate = (unsigned int)codec_context->sample_rate;
-	audio_format.channels = codec_context->channels;
+	audio_format_init(&audio_format, codec_context->sample_rate, bits,
+			  codec_context->channels);
 
 	if (!audio_format_valid(&audio_format)) {
 		g_warning("Invalid audio format: %u:%u:%u\n",
@@ -342,8 +343,9 @@ static void
 ffmpeg_copy_metadata(struct tag *tag, AVMetadata *m,
 		     enum tag_type type, const char *name)
 {
-	AVMetadataTag *mt = av_metadata_get(m, name, NULL, 0);
-	if (mt != NULL)
+	AVMetadataTag *mt = NULL;
+
+	while ((mt = av_metadata_get(m, name, mt, 0)) != NULL)
 		tag_add_item(tag, type, mt->value);
 }
 #endif
@@ -351,13 +353,15 @@ ffmpeg_copy_metadata(struct tag *tag, AVMetadata *m,
 static bool ffmpeg_tag_internal(struct ffmpeg_context *ctx)
 {
 	struct tag *tag = (struct tag *) ctx->tag;
-	const AVFormatContext *f = ctx->format_context;
+	AVFormatContext *f = ctx->format_context;
 
 	tag->time = 0;
 	if (f->duration != (int64_t)AV_NOPTS_VALUE)
 		tag->time = f->duration / AV_TIME_BASE;
 
 #if LIBAVFORMAT_VERSION_INT >= ((52<<16)+(31<<8)+0)
+	av_metadata_conv(f, NULL, f->iformat->metadata_conv);
+
 	ffmpeg_copy_metadata(tag, f->metadata, TAG_ITEM_TITLE, "title");
 	ffmpeg_copy_metadata(tag, f->metadata, TAG_ITEM_ARTIST, "author");
 	ffmpeg_copy_metadata(tag, f->metadata, TAG_ITEM_ALBUM, "album");
diff --git a/src/decoder/flac_plugin.c b/src/decoder/flac_plugin.c
index 1d7a9f868..89a812f52 100644
--- a/src/decoder/flac_plugin.c
+++ b/src/decoder/flac_plugin.c
@@ -300,6 +300,8 @@ flac_cue_tag_load(const char *file)
 	FLAC__uint64 track_time = 0;
 #ifdef HAVE_CUE /* libcue */
 	FLAC__StreamMetadata* vc = FLAC__metadata_object_new(FLAC__METADATA_TYPE_VORBIS_COMMENT);
+	char* cs_filename;
+	FILE* cs_file;
 #endif /* libcue */
 	FLAC__StreamMetadata* si = FLAC__metadata_object_new(FLAC__METADATA_TYPE_STREAMINFO);
 	FLAC__StreamMetadata* cs = FLAC__metadata_object_new(FLAC__METADATA_TYPE_CUESHEET);
@@ -328,6 +330,18 @@ flac_cue_tag_load(const char *file)
 		}
 		FLAC__metadata_object_delete(vc);
 	}
+
+	if (tag == NULL) {
+		cs_filename = g_strconcat(file, ".cue", NULL);
+
+		cs_file = fopen(cs_filename, "rt");
+		g_free(cs_filename);
+
+		if (cs_file != NULL) {
+			tag = cue_tag_file(cs_file, tnum);
+			fclose(cs_file);
+		}
+	}
 #endif /* libcue */
 
 	if (tag == NULL)
diff --git a/src/decoder/mad_plugin.c b/src/decoder/mad_plugin.c
index 1ef7183fa..c5287564f 100644
--- a/src/decoder/mad_plugin.c
+++ b/src/decoder/mad_plugin.c
@@ -1170,13 +1170,6 @@ mp3_read(struct mp3_data *data, struct replay_gain_info **replay_gain_info_r)
 	return ret != DECODE_BREAK;
 }
 
-static void mp3_audio_format(struct mp3_data *data, struct audio_format *af)
-{
-	af->bits = 24;
-	af->sample_rate = (data->frame).header.samplerate;
-	af->channels = MAD_NCHANNELS(&(data->frame).header);
-}
-
 static void
 mp3_decode(struct decoder *decoder, struct input_stream *input_stream)
 {
@@ -1192,7 +1185,8 @@ mp3_decode(struct decoder *decoder, struct input_stream *input_stream)
 		return;
 	}
 
-	mp3_audio_format(&data, &audio_format);
+	audio_format_init(&audio_format, data.frame.header.samplerate, 24,
+			  MAD_NCHANNELS(&data.frame.header));
 
 	decoder_initialized(decoder, &audio_format,
 			    data.input_stream->seekable, data.total_time);
diff --git a/src/decoder/mikmod_plugin.c b/src/decoder/mikmod_plugin.c
index 065c34319..e7b7bfb03 100644
--- a/src/decoder/mikmod_plugin.c
+++ b/src/decoder/mikmod_plugin.c
@@ -175,9 +175,7 @@ mod_decode(struct decoder *decoder, const char *path)
 		return;
 	}
 
-	audio_format.bits = 16;
-	audio_format.sample_rate = 44100;
-	audio_format.channels = 2;
+	audio_format_init(&audio_format, 44100, 16, 2);
 
 	secPerByte =
 	    1.0 / ((audio_format.bits * audio_format.channels / 8.0) *
diff --git a/src/decoder/modplug_plugin.c b/src/decoder/modplug_plugin.c
index f636f2fa6..6c375e6a0 100644
--- a/src/decoder/modplug_plugin.c
+++ b/src/decoder/modplug_plugin.c
@@ -121,9 +121,7 @@ mod_decode(struct decoder *decoder, struct input_stream *is)
 		return;
 	}
 
-	audio_format.bits = 16;
-	audio_format.sample_rate = 44100;
-	audio_format.channels = 2;
+	audio_format_init(&audio_format, 44100, 16, 2);
 
 	sec_perbyte =
 	    1.0 / ((audio_format.bits * audio_format.channels / 8.0) *
@@ -186,7 +184,7 @@ static struct tag *mod_tagdup(const char *file)
 		return NULL;
         }
 	ret = tag_new();
-	ret->time = 0;
+	ret->time = ModPlug_GetLength(f) / 1000;
 
 	title = g_strdup(ModPlug_GetName(f));
 	if (title)
diff --git a/src/decoder/mp4ff_plugin.c b/src/decoder/mp4ff_plugin.c
index cf9382904..d2c63f983 100644
--- a/src/decoder/mp4ff_plugin.c
+++ b/src/decoder/mp4ff_plugin.c
@@ -131,11 +131,7 @@ mp4_faad_new(mp4ff_t *mp4fh, int *track_r, struct audio_format *audio_format)
 	}
 
 	*track_r = track;
-	*audio_format = (struct audio_format){
-		.bits = 16,
-		.channels = channels,
-		.sample_rate = sample_rate,
-	};
+	audio_format_init(audio_format, sample_rate, 16, channels);
 
 	if (!audio_format_valid(audio_format)) {
 		g_warning("Invalid audio format: %u:%u:%u\n",
diff --git a/src/decoder/mpcdec_plugin.c b/src/decoder/mpcdec_plugin.c
index 26349f93a..a684da104 100644
--- a/src/decoder/mpcdec_plugin.c
+++ b/src/decoder/mpcdec_plugin.c
@@ -193,9 +193,7 @@ mpcdec_decode(struct decoder *mpd_decoder, struct input_stream *is)
 	mpc_demux_get_info(demux, &info);
 #endif
 
-	audio_format.bits = 24;
-	audio_format.channels = info.channels;
-	audio_format.sample_rate = info.sample_freq;
+	audio_format_init(&audio_format, info.sample_freq, 24, info.channels);
 
 	if (!audio_format_valid(&audio_format)) {
 #ifndef MPC_IS_OLD_API
diff --git a/src/decoder/sidplay_plugin.cxx b/src/decoder/sidplay_plugin.cxx
index c62e6b4b6..54ab746e2 100644
--- a/src/decoder/sidplay_plugin.cxx
+++ b/src/decoder/sidplay_plugin.cxx
@@ -103,9 +103,7 @@ sidplay_file_decode(struct decoder *decoder, const char *path_fs)
 	/* initialize the MPD decoder */
 
 	struct audio_format audio_format;
-	audio_format.sample_rate = 48000;
-	audio_format.bits = 16;
-	audio_format.channels = 2;
+	audio_format_init(&audio_format, 48000, 16, 2);
 
 	decoder_initialized(decoder, &audio_format, false, -1);
 
diff --git a/src/decoder/sndfile_decoder_plugin.c b/src/decoder/sndfile_decoder_plugin.c
new file mode 100644
index 000000000..4cc64459f
--- /dev/null
+++ b/src/decoder/sndfile_decoder_plugin.c
@@ -0,0 +1,244 @@
+/*
+ * Copyright (C) 2003-2009 The Music Player Daemon Project
+ * http://www.musicpd.org
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with this program; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include "decoder_api.h"
+
+#include <sndfile.h>
+
+#undef G_LOG_DOMAIN
+#define G_LOG_DOMAIN "sndfile"
+
+static sf_count_t
+sndfile_vio_get_filelen(void *user_data)
+{
+	const struct input_stream *is = user_data;
+
+	return is->size;
+}
+
+static sf_count_t
+sndfile_vio_seek(sf_count_t offset, int whence, void *user_data)
+{
+	struct input_stream *is = user_data;
+	bool success;
+
+	success = input_stream_seek(is, offset, whence);
+	if (!success)
+		return -1;
+
+	return is->offset;
+}
+
+static sf_count_t
+sndfile_vio_read(void *ptr, sf_count_t count, void *user_data)
+{
+	struct input_stream *is = user_data;
+	size_t nbytes;
+
+	nbytes = input_stream_read(is, ptr, count);
+	if (nbytes == 0 && is->error != 0)
+		return -1;
+
+	return nbytes;
+}
+
+static sf_count_t
+sndfile_vio_write(G_GNUC_UNUSED const void *ptr,
+		  G_GNUC_UNUSED sf_count_t count,
+		  G_GNUC_UNUSED void *user_data)
+{
+	/* no writing! */
+	return -1;
+}
+
+static sf_count_t
+sndfile_vio_tell(void *user_data)
+{
+	const struct input_stream *is = user_data;
+
+	return is->offset;
+}
+
+/**
+ * This SF_VIRTUAL_IO implementation wraps MPD's #input_stream to a
+ * libsndfile stream.
+ */
+static SF_VIRTUAL_IO vio = {
+	.get_filelen = sndfile_vio_get_filelen,
+	.seek = sndfile_vio_seek,
+	.read = sndfile_vio_read,
+	.write = sndfile_vio_write,
+	.tell = sndfile_vio_tell,
+};
+
+/**
+ * Converts a frame number to a timestamp (in seconds).
+ */
+static float
+frame_to_time(sf_count_t frame, const struct audio_format *audio_format)
+{
+	return (float)frame / (float)audio_format->sample_rate;
+}
+
+/**
+ * Converts a timestamp (in seconds) to a frame number.
+ */
+static sf_count_t
+time_to_frame(float t, const struct audio_format *audio_format)
+{
+	return (sf_count_t)(t * audio_format->sample_rate);
+}
+
+static void
+sndfile_stream_decode(struct decoder *decoder, struct input_stream *is)
+{
+	SNDFILE *sf;
+	SF_INFO info;
+	struct audio_format audio_format;
+	size_t frame_size;
+	sf_count_t read_frames, num_frames, position = 0;
+	int buffer[4096];
+	enum decoder_command cmd;
+
+	info.format = 0;
+
+	sf = sf_open_virtual(&vio, SFM_READ, &info, is);
+	if (sf == NULL) {
+		g_warning("sf_open_virtual() failed");
+		return;
+	}
+
+	/* for now, always read 32 bit samples.  Later, we could lower
+	   MPD's CPU usage by reading 16 bit samples with
+	   sf_readf_short() on low-quality source files. */
+	audio_format_init(&audio_format, info.samplerate, 32, info.channels);
+
+	if (!audio_format_valid(&audio_format)) {
+		g_warning("invalid audio format");
+		return;
+	}
+
+	decoder_initialized(decoder, &audio_format, info.seekable,
+			    frame_to_time(info.frames, &audio_format));
+
+	frame_size = audio_format_frame_size(&audio_format);
+	read_frames = sizeof(buffer) / frame_size;
+
+	do {
+		num_frames = sf_readf_int(sf, buffer, read_frames);
+		if (num_frames <= 0)
+			break;
+
+		cmd = decoder_data(decoder, is,
+				   buffer, num_frames * frame_size,
+				   frame_to_time(position, &audio_format),
+				   0, NULL);
+		if (cmd == DECODE_COMMAND_SEEK) {
+			sf_count_t c =
+				time_to_frame(decoder_seek_where(decoder),
+					      &audio_format);
+			c = sf_seek(sf, c, SEEK_SET);
+			if (c < 0)
+				decoder_seek_error(decoder);
+			else
+				decoder_command_finished(decoder);
+			cmd = DECODE_COMMAND_NONE;
+		}
+	} while (cmd == DECODE_COMMAND_NONE);
+
+	sf_close(sf);
+}
+
+static struct tag *
+sndfile_tag_dup(const char *path_fs)
+{
+	SNDFILE *sf;
+	SF_INFO info;
+	struct tag *tag;
+	const char *p;
+
+	info.format = 0;
+
+	sf = sf_open(path_fs, SFM_READ, &info);
+	if (sf == NULL)
+		return NULL;
+
+	if (!audio_valid_sample_rate(info.samplerate)) {
+		sf_close(sf);
+		g_warning("Invalid sample rate in %s\n", path_fs);
+		return NULL;
+	}
+
+	tag = tag_new();
+	tag->time = info.frames / info.samplerate;
+
+	p = sf_get_string(sf, SF_STR_TITLE);
+	if (p != NULL)
+		tag_add_item(tag, TAG_ITEM_TITLE, p);
+
+	p = sf_get_string(sf, SF_STR_ARTIST);
+	if (p != NULL)
+		tag_add_item(tag, TAG_ITEM_ARTIST, p);
+
+	p = sf_get_string(sf, SF_STR_DATE);
+	if (p != NULL)
+		tag_add_item(tag, TAG_ITEM_DATE, p);
+
+	sf_close(sf);
+
+	return tag;
+}
+
+static const char *const sndfile_suffixes[] = {
+	"wav", "aiff", "aif", /* Microsoft / SGI / Apple */
+	"au", "snd", /* Sun / DEC / NeXT */
+	"paf", /* Paris Audio File */
+	"iff", "svx", /* Commodore Amiga IFF / SVX */
+	"sf", /* IRCAM */
+	"voc", /* Creative */
+	"w64", /* Soundforge */
+	"pvf", /* Portable Voice Format */
+	"xi", /* Fasttracker */
+	"htk", /* HMM Tool Kit */
+	"caf", /* Apple */
+	"sd2", /* Sound Designer II */
+
+	/* libsndfile also supports FLAC and Ogg Vorbis, but only by
+	   linking with libFLAC and libvorbis - we can do better, we
+	   have native plugins for these libraries */
+
+	NULL
+};
+
+static const char *const sndfile_mime_types[] = {
+	"audio/x-wav",
+	"audio/x-aiff",
+
+	/* what are the MIME types of the other supported formats? */
+
+	NULL
+};
+
+const struct decoder_plugin sndfile_decoder_plugin = {
+	.name = "sndfile",
+	.stream_decode = sndfile_stream_decode,
+	.tag_dup = sndfile_tag_dup,
+	.suffixes = sndfile_suffixes,
+	.mime_types = sndfile_mime_types,
+};
diff --git a/src/decoder/vorbis_plugin.c b/src/decoder/vorbis_plugin.c
index d4f81e91f..bab1d57ec 100644
--- a/src/decoder/vorbis_plugin.c
+++ b/src/decoder/vorbis_plugin.c
@@ -324,8 +324,7 @@ vorbis_stream_decode(struct decoder *decoder,
 			vorbis_info *vi = ov_info(&vf, -1);
 			struct replay_gain_info *new_rgi;
 
-			audio_format.channels = vi->channels;
-			audio_format.sample_rate = vi->rate;
+			audio_format_init(&audio_format, vi->rate, 16, vi->channels);
 
 			if (!audio_format_valid(&audio_format)) {
 				g_warning("Invalid audio format: %u:%u:%u\n",
diff --git a/src/decoder/wavpack_plugin.c b/src/decoder/wavpack_plugin.c
index 821536fb5..f3d701144 100644
--- a/src/decoder/wavpack_plugin.c
+++ b/src/decoder/wavpack_plugin.c
@@ -145,9 +145,9 @@ wavpack_decode(struct decoder *decoder, WavpackContext *wpc, bool can_seek,
 	int bytes_per_sample, output_sample_size;
 	int position;
 
-	audio_format.sample_rate = WavpackGetSampleRate(wpc);
-	audio_format.channels = WavpackGetReducedChannels(wpc);
-	audio_format.bits = WavpackGetBitsPerSample(wpc);
+	audio_format_init(&audio_format, WavpackGetSampleRate(wpc),
+			  WavpackGetBitsPerSample(wpc), 
+			  WavpackGetReducedChannels(wpc));
 
 	/* round bitwidth to 8-bit units */
 	audio_format.bits = (audio_format.bits + 7) & (~7);