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-rw-r--r--src/decoder/_flac_common.c323
-rw-r--r--src/decoder/_flac_common.h168
-rw-r--r--src/decoder/_ogg_common.c49
-rw-r--r--src/decoder/_ogg_common.h31
-rw-r--r--src/decoder/aac_plugin.c602
-rw-r--r--src/decoder/audiofile_plugin.c147
-rw-r--r--src/decoder/ffmpeg_plugin.c419
-rw-r--r--src/decoder/flac_plugin.c459
-rw-r--r--src/decoder/mod_plugin.c278
-rw-r--r--src/decoder/mp3_plugin.c1086
-rw-r--r--src/decoder/mp4_plugin.c423
-rw-r--r--src/decoder/mpc_plugin.c308
-rw-r--r--src/decoder/oggflac_plugin.c355
-rw-r--r--src/decoder/oggvorbis_plugin.c387
-rw-r--r--src/decoder/wavpack_plugin.c574
15 files changed, 5609 insertions, 0 deletions
diff --git a/src/decoder/_flac_common.c b/src/decoder/_flac_common.c
new file mode 100644
index 000000000..db43e0003
--- /dev/null
+++ b/src/decoder/_flac_common.c
@@ -0,0 +1,323 @@
+/* the Music Player Daemon (MPD)
+ * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
+ * This project's homepage is: http://www.musicpd.org
+ *
+ * Common data structures and functions used by FLAC and OggFLAC
+ * (c) 2005 by Eric Wong <normalperson@yhbt.net>
+ *
+ * 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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include "_flac_common.h"
+#include "../log.h"
+
+#include <FLAC/format.h>
+#include <FLAC/metadata.h>
+
+void init_FlacData(FlacData * data, struct decoder * decoder,
+ InputStream * inStream)
+{
+ data->time = 0;
+ data->position = 0;
+ data->bitRate = 0;
+ data->decoder = decoder;
+ data->inStream = inStream;
+ data->replayGainInfo = NULL;
+ data->tag = NULL;
+}
+
+static int flacFindVorbisCommentFloat(const FLAC__StreamMetadata * block,
+ const char *cmnt, float *fl)
+{
+ int offset =
+ FLAC__metadata_object_vorbiscomment_find_entry_from(block, 0, cmnt);
+
+ if (offset >= 0) {
+ size_t pos = strlen(cmnt) + 1; /* 1 is for '=' */
+ int len = block->data.vorbis_comment.comments[offset].length
+ - pos;
+ if (len > 0) {
+ unsigned char tmp;
+ unsigned char *p = &(block->data.vorbis_comment.
+ comments[offset].entry[pos]);
+ tmp = p[len];
+ p[len] = '\0';
+ *fl = (float)atof((char *)p);
+ p[len] = tmp;
+
+ return 1;
+ }
+ }
+
+ return 0;
+}
+
+/* replaygain stuff by AliasMrJones */
+static void flacParseReplayGain(const FLAC__StreamMetadata * block,
+ FlacData * data)
+{
+ int found = 0;
+
+ if (data->replayGainInfo)
+ freeReplayGainInfo(data->replayGainInfo);
+
+ data->replayGainInfo = newReplayGainInfo();
+
+ found |= flacFindVorbisCommentFloat(block, "replaygain_album_gain",
+ &data->replayGainInfo->albumGain);
+ found |= flacFindVorbisCommentFloat(block, "replaygain_album_peak",
+ &data->replayGainInfo->albumPeak);
+ found |= flacFindVorbisCommentFloat(block, "replaygain_track_gain",
+ &data->replayGainInfo->trackGain);
+ found |= flacFindVorbisCommentFloat(block, "replaygain_track_peak",
+ &data->replayGainInfo->trackPeak);
+
+ if (!found) {
+ freeReplayGainInfo(data->replayGainInfo);
+ data->replayGainInfo = NULL;
+ }
+}
+
+/* tracknumber is used in VCs, MPD uses "track" ..., all the other
+ * tag names match */
+static const char *VORBIS_COMMENT_TRACK_KEY = "tracknumber";
+static const char *VORBIS_COMMENT_DISC_KEY = "discnumber";
+
+static unsigned int commentMatchesAddToTag(const
+ FLAC__StreamMetadata_VorbisComment_Entry
+ * entry, unsigned int itemType,
+ struct tag ** tag)
+{
+ const char *str;
+ size_t slen;
+ int vlen;
+
+ switch (itemType) {
+ case TAG_ITEM_TRACK:
+ str = VORBIS_COMMENT_TRACK_KEY;
+ break;
+ case TAG_ITEM_DISC:
+ str = VORBIS_COMMENT_DISC_KEY;
+ break;
+ default:
+ str = mpdTagItemKeys[itemType];
+ }
+ slen = strlen(str);
+ vlen = entry->length - slen - 1;
+
+ if ((vlen > 0) && (0 == strncasecmp(str, (char *)entry->entry, slen))
+ && (*(entry->entry + slen) == '=')) {
+ if (!*tag)
+ *tag = tag_new();
+
+ tag_add_item_n(*tag, itemType,
+ (char *)(entry->entry + slen + 1), vlen);
+
+ return 1;
+ }
+
+ return 0;
+}
+
+struct tag *copyVorbisCommentBlockToMpdTag(const FLAC__StreamMetadata * block,
+ struct tag * tag)
+{
+ unsigned int i, j;
+ FLAC__StreamMetadata_VorbisComment_Entry *comments;
+
+ comments = block->data.vorbis_comment.comments;
+
+ for (i = block->data.vorbis_comment.num_comments; i != 0; --i) {
+ for (j = TAG_NUM_OF_ITEM_TYPES; j--;) {
+ if (commentMatchesAddToTag(comments, j, &tag))
+ break;
+ }
+ comments++;
+ }
+
+ return tag;
+}
+
+void flac_metadata_common_cb(const FLAC__StreamMetadata * block,
+ FlacData * data)
+{
+ const FLAC__StreamMetadata_StreamInfo *si = &(block->data.stream_info);
+
+ 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;
+ data->total_time = ((float)si->total_samples) / (si->sample_rate);
+ break;
+ case FLAC__METADATA_TYPE_VORBIS_COMMENT:
+ flacParseReplayGain(block, data);
+ default:
+ break;
+ }
+}
+
+void flac_error_common_cb(const char *plugin,
+ const FLAC__StreamDecoderErrorStatus status,
+ mpd_unused FlacData * data)
+{
+ if (decoder_get_command(data->decoder) == DECODE_COMMAND_STOP)
+ return;
+
+ switch (status) {
+ case FLAC__STREAM_DECODER_ERROR_STATUS_LOST_SYNC:
+ ERROR("%s lost sync\n", plugin);
+ break;
+ case FLAC__STREAM_DECODER_ERROR_STATUS_BAD_HEADER:
+ ERROR("bad %s header\n", plugin);
+ break;
+ case FLAC__STREAM_DECODER_ERROR_STATUS_FRAME_CRC_MISMATCH:
+ ERROR("%s crc mismatch\n", plugin);
+ break;
+ default:
+ ERROR("unknown %s error\n", plugin);
+ }
+}
+
+static void flac_convert_stereo16(int16_t *dest,
+ const FLAC__int32 * const buf[],
+ unsigned int position, unsigned int end)
+{
+ for (; position < end; ++position) {
+ *dest++ = buf[0][position];
+ *dest++ = buf[1][position];
+ }
+}
+
+static void
+flac_convert_16(int16_t *dest,
+ unsigned int num_channels,
+ const FLAC__int32 * const buf[],
+ unsigned int position, unsigned int end)
+{
+ unsigned int c_chan;
+
+ for (; position < end; ++position)
+ for (c_chan = 0; c_chan < num_channels; c_chan++)
+ *dest++ = buf[c_chan][position];
+}
+
+/**
+ * Note: this function also handles 24 bit files!
+ */
+static void
+flac_convert_32(int32_t *dest,
+ unsigned int num_channels,
+ const FLAC__int32 * const buf[],
+ unsigned int position, unsigned int end)
+{
+ unsigned int c_chan;
+
+ for (; position < end; ++position)
+ for (c_chan = 0; c_chan < num_channels; c_chan++)
+ *dest++ = buf[c_chan][position];
+}
+
+static void
+flac_convert_8(int8_t *dest,
+ unsigned int num_channels,
+ const FLAC__int32 * const buf[],
+ unsigned int position, unsigned int end)
+{
+ unsigned int c_chan;
+
+ for (; position < end; ++position)
+ for (c_chan = 0; c_chan < num_channels; c_chan++)
+ *dest++ = buf[c_chan][position];
+}
+
+static void flac_convert(unsigned char *dest,
+ unsigned int num_channels,
+ unsigned int bytes_per_sample,
+ const FLAC__int32 * const buf[],
+ unsigned int position, unsigned int end)
+{
+ switch (bytes_per_sample) {
+ case 2:
+ if (num_channels == 2)
+ flac_convert_stereo16((int16_t*)dest, buf,
+ position, end);
+ else
+ flac_convert_16((int16_t*)dest, num_channels, buf,
+ position, end);
+ break;
+
+ case 4:
+ flac_convert_32((int32_t*)dest, num_channels, buf,
+ position, end);
+ break;
+
+ case 1:
+ flac_convert_8((int8_t*)dest, num_channels, buf,
+ position, end);
+ break;
+ }
+}
+
+FLAC__StreamDecoderWriteStatus
+flac_common_write(FlacData *data, const FLAC__Frame * frame,
+ const FLAC__int32 *const buf[])
+{
+ unsigned int c_samp;
+ const unsigned int num_channels = frame->header.channels;
+ const unsigned int bytes_per_sample =
+ audio_format_sample_size(&data->audio_format);
+ const unsigned int bytes_per_channel =
+ bytes_per_sample * frame->header.channels;
+ const unsigned int max_samples = FLAC_CHUNK_SIZE / bytes_per_channel;
+ unsigned int num_samples;
+ enum decoder_command cmd;
+
+ if (bytes_per_sample != 1 && bytes_per_sample != 2 &&
+ bytes_per_sample != 4)
+ /* exotic unsupported bit rate */
+ return FLAC__STREAM_DECODER_WRITE_STATUS_ABORT;
+
+ for (c_samp = 0; c_samp < frame->header.blocksize;
+ c_samp += num_samples) {
+ num_samples = frame->header.blocksize - c_samp;
+ if (num_samples > max_samples)
+ num_samples = max_samples;
+
+ flac_convert(data->chunk,
+ num_channels, bytes_per_sample, buf,
+ c_samp, c_samp + num_samples);
+
+ cmd = decoder_data(data->decoder, data->inStream,
+ 1, data->chunk,
+ num_samples * bytes_per_channel,
+ data->time, data->bitRate,
+ data->replayGainInfo);
+ switch (cmd) {
+ case DECODE_COMMAND_NONE:
+ case DECODE_COMMAND_START:
+ break;
+
+ case DECODE_COMMAND_STOP:
+ return
+ FLAC__STREAM_DECODER_WRITE_STATUS_ABORT;
+
+ case DECODE_COMMAND_SEEK:
+ return
+ FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE;
+ }
+ }
+
+ return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE;
+}
diff --git a/src/decoder/_flac_common.h b/src/decoder/_flac_common.h
new file mode 100644
index 000000000..45714b4bd
--- /dev/null
+++ b/src/decoder/_flac_common.h
@@ -0,0 +1,168 @@
+/* the Music Player Daemon (MPD)
+ * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
+ * This project's homepage is: http://www.musicpd.org
+ *
+ * Common data structures and functions used by FLAC and OggFLAC
+ * (c) 2005 by Eric Wong <normalperson@yhbt.net>
+ *
+ * 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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#ifndef _FLAC_COMMON_H
+#define _FLAC_COMMON_H
+
+#include "../decoder_api.h"
+
+#include <FLAC/export.h>
+#if !defined(FLAC_API_VERSION_CURRENT) || FLAC_API_VERSION_CURRENT <= 7
+# include <FLAC/seekable_stream_decoder.h>
+# define flac_decoder FLAC__SeekableStreamDecoder
+# define flac_new() FLAC__seekable_stream_decoder_new()
+
+# define flac_ogg_init(a,b,c,d,e,f,g,h,i,j) (0)
+
+# define flac_get_decode_position(x,y) \
+ FLAC__seekable_stream_decoder_get_decode_position(x,y)
+# define flac_get_state(x) FLAC__seekable_stream_decoder_get_state(x)
+# define flac_process_single(x) FLAC__seekable_stream_decoder_process_single(x)
+# define flac_process_metadata(x) \
+ FLAC__seekable_stream_decoder_process_until_end_of_metadata(x)
+# define flac_seek_absolute(x,y) \
+ FLAC__seekable_stream_decoder_seek_absolute(x,y)
+# define flac_finish(x) FLAC__seekable_stream_decoder_finish(x)
+# define flac_delete(x) FLAC__seekable_stream_decoder_delete(x)
+
+# define flac_decoder_eof FLAC__SEEKABLE_STREAM_DECODER_END_OF_STREAM
+
+typedef unsigned flac_read_status_size_t;
+# define flac_read_status FLAC__SeekableStreamDecoderReadStatus
+# define flac_read_status_continue \
+ FLAC__SEEKABLE_STREAM_DECODER_READ_STATUS_OK
+# define flac_read_status_eof FLAC__SEEKABLE_STREAM_DECODER_READ_STATUS_OK
+# define flac_read_status_abort \
+ FLAC__SEEKABLE_STREAM_DECODER_READ_STATUS_ERROR
+
+# define flac_seek_status FLAC__SeekableStreamDecoderSeekStatus
+# define flac_seek_status_ok FLAC__SEEKABLE_STREAM_DECODER_SEEK_STATUS_OK
+# define flac_seek_status_error FLAC__SEEKABLE_STREAM_DECODER_SEEK_STATUS_ERROR
+
+# define flac_tell_status FLAC__SeekableStreamDecoderTellStatus
+# define flac_tell_status_ok FLAC__SEEKABLE_STREAM_DECODER_TELL_STATUS_OK
+# define flac_tell_status_error \
+ FLAC__SEEKABLE_STREAM_DECODER_TELL_STATUS_ERROR
+# define flac_tell_status_unsupported \
+ FLAC__SEEKABLE_STREAM_DECODER_TELL_STATUS_ERROR
+
+# define flac_length_status FLAC__SeekableStreamDecoderLengthStatus
+# define flac_length_status_ok FLAC__SEEKABLE_STREAM_DECODER_LENGTH_STATUS_OK
+# define flac_length_status_error \
+ FLAC__SEEKABLE_STREAM_DECODER_LENGTH_STATUS_ERROR
+# define flac_length_status_unsupported \
+ FLAC__SEEKABLE_STREAM_DECODER_LENGTH_STATUS_ERROR
+
+# ifdef HAVE_OGGFLAC
+# include <OggFLAC/seekable_stream_decoder.h>
+# endif
+#else /* FLAC_API_VERSION_CURRENT > 7 */
+
+/*
+ * OggFLAC support is handled by our flac_plugin already, and
+ * thus we *can* always have it if libFLAC was compiled with it
+ */
+# include "_ogg_common.h"
+
+# include <FLAC/stream_decoder.h>
+# define flac_decoder FLAC__StreamDecoder
+# define flac_new() FLAC__stream_decoder_new()
+
+# define flac_init(a,b,c,d,e,f,g,h,i,j) \
+ (FLAC__stream_decoder_init_stream(a,b,c,d,e,f,g,h,i,j) \
+ == FLAC__STREAM_DECODER_INIT_STATUS_OK)
+# define flac_ogg_init(a,b,c,d,e,f,g,h,i,j) \
+ (FLAC__stream_decoder_init_ogg_stream(a,b,c,d,e,f,g,h,i,j) \
+ == FLAC__STREAM_DECODER_INIT_STATUS_OK)
+
+# define flac_get_decode_position(x,y) \
+ FLAC__stream_decoder_get_decode_position(x,y)
+# define flac_get_state(x) FLAC__stream_decoder_get_state(x)
+# define flac_process_single(x) FLAC__stream_decoder_process_single(x)
+# define flac_process_metadata(x) \
+ FLAC__stream_decoder_process_until_end_of_metadata(x)
+# define flac_seek_absolute(x,y) FLAC__stream_decoder_seek_absolute(x,y)
+# define flac_finish(x) FLAC__stream_decoder_finish(x)
+# define flac_delete(x) FLAC__stream_decoder_delete(x)
+
+# define flac_decoder_eof FLAC__STREAM_DECODER_END_OF_STREAM
+
+typedef size_t flac_read_status_size_t;
+# define flac_read_status FLAC__StreamDecoderReadStatus
+# define flac_read_status_continue \
+ FLAC__STREAM_DECODER_READ_STATUS_CONTINUE
+# define flac_read_status_eof FLAC__STREAM_DECODER_READ_STATUS_END_OF_STREAM
+# define flac_read_status_abort FLAC__STREAM_DECODER_READ_STATUS_ABORT
+
+# define flac_seek_status FLAC__StreamDecoderSeekStatus
+# define flac_seek_status_ok FLAC__STREAM_DECODER_SEEK_STATUS_OK
+# define flac_seek_status_error FLAC__STREAM_DECODER_SEEK_STATUS_ERROR
+# define flac_seek_status_unsupported \
+ FLAC__STREAM_DECODER_SEEK_STATUS_UNSUPPORTED
+
+# define flac_tell_status FLAC__StreamDecoderTellStatus
+# define flac_tell_status_ok FLAC__STREAM_DECODER_TELL_STATUS_OK
+# define flac_tell_status_error FLAC__STREAM_DECODER_TELL_STATUS_ERROR
+# define flac_tell_status_unsupported \
+ FLAC__STREAM_DECODER_TELL_STATUS_UNSUPPORTED
+
+# define flac_length_status FLAC__StreamDecoderLengthStatus
+# define flac_length_status_ok FLAC__STREAM_DECODER_LENGTH_STATUS_OK
+# define flac_length_status_error FLAC__STREAM_DECODER_LENGTH_STATUS_ERROR
+# define flac_length_status_unsupported \
+ FLAC__STREAM_DECODER_LENGTH_STATUS_UNSUPPORTED
+
+#endif /* FLAC_API_VERSION_CURRENT >= 7 */
+
+#include <FLAC/metadata.h>
+
+#define FLAC_CHUNK_SIZE 4080
+
+typedef struct {
+ unsigned char chunk[FLAC_CHUNK_SIZE];
+ float time;
+ unsigned int bitRate;
+ struct audio_format audio_format;
+ float total_time;
+ FLAC__uint64 position;
+ struct decoder *decoder;
+ InputStream *inStream;
+ ReplayGainInfo *replayGainInfo;
+ struct tag *tag;
+} FlacData;
+
+/* initializes a given FlacData struct */
+void init_FlacData(FlacData * data, struct decoder * decoder,
+ InputStream * inStream);
+void flac_metadata_common_cb(const FLAC__StreamMetadata * block,
+ FlacData * data);
+void flac_error_common_cb(const char *plugin,
+ FLAC__StreamDecoderErrorStatus status,
+ FlacData * data);
+
+struct tag *copyVorbisCommentBlockToMpdTag(const FLAC__StreamMetadata * block,
+ struct tag *tag);
+
+FLAC__StreamDecoderWriteStatus
+flac_common_write(FlacData *data, const FLAC__Frame * frame,
+ const FLAC__int32 *const buf[]);
+
+#endif /* _FLAC_COMMON_H */
diff --git a/src/decoder/_ogg_common.c b/src/decoder/_ogg_common.c
new file mode 100644
index 000000000..841b2ad3f
--- /dev/null
+++ b/src/decoder/_ogg_common.c
@@ -0,0 +1,49 @@
+/* the Music Player Daemon (MPD)
+ * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
+ * This project's homepage is: http://www.musicpd.org
+ *
+ * Common functions used for Ogg data streams (Ogg-Vorbis and OggFLAC)
+ * (c) 2005 by Eric Wong <normalperson@yhbt.net>
+ *
+ * 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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include "_ogg_common.h"
+#include "_flac_common.h"
+#include "../utils.h"
+
+ogg_stream_type ogg_stream_type_detect(InputStream * inStream)
+{
+ /* oggflac detection based on code in ogg123 and this post
+ * http://lists.xiph.org/pipermail/flac/2004-December/000393.html
+ * ogg123 trunk still doesn't have this patch as of June 2005 */
+ unsigned char buf[41];
+ size_t r;
+
+ seekInputStream(inStream, 0, SEEK_SET);
+
+ r = decoder_read(NULL, inStream, buf, sizeof(buf));
+
+ if (r > 0)
+ seekInputStream(inStream, 0, SEEK_SET);
+
+ if (r >= 32 && memcmp(buf, "OggS", 4) == 0 && (
+ (memcmp(buf+29, "FLAC", 4) == 0
+ && memcmp(buf+37, "fLaC", 4) == 0)
+ || (memcmp(buf+28, "FLAC", 4) == 0)
+ || (memcmp(buf+28, "fLaC", 4) == 0))) {
+ return FLAC;
+ }
+ return VORBIS;
+}
diff --git a/src/decoder/_ogg_common.h b/src/decoder/_ogg_common.h
new file mode 100644
index 000000000..7c9e7b630
--- /dev/null
+++ b/src/decoder/_ogg_common.h
@@ -0,0 +1,31 @@
+/* the Music Player Daemon (MPD)
+ * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
+ * This project's homepage is: http://www.musicpd.org
+ *
+ * Common functions used for Ogg data streams (Ogg-Vorbis and OggFLAC)
+ * (c) 2005 by Eric Wong <normalperson@yhbt.net>
+ *
+ * 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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#ifndef _OGG_COMMON_H
+#define _OGG_COMMON_H
+
+#include "../decoder_api.h"
+
+typedef enum _ogg_stream_type { VORBIS, FLAC } ogg_stream_type;
+
+ogg_stream_type ogg_stream_type_detect(InputStream * inStream);
+
+#endif /* _OGG_COMMON_H */
diff --git a/src/decoder/aac_plugin.c b/src/decoder/aac_plugin.c
new file mode 100644
index 000000000..7842bcc22
--- /dev/null
+++ b/src/decoder/aac_plugin.c
@@ -0,0 +1,602 @@
+/* the Music Player Daemon (MPD)
+ * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
+ * This project's homepage is: 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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include "../decoder_api.h"
+
+#define AAC_MAX_CHANNELS 6
+
+#include "../utils.h"
+#include "../log.h"
+
+#include <assert.h>
+#include <faad.h>
+
+/* all code here is either based on or copied from FAAD2's frontend code */
+typedef struct {
+ struct decoder *decoder;
+ InputStream *inStream;
+ size_t bytesIntoBuffer;
+ size_t bytesConsumed;
+ off_t fileOffset;
+ unsigned char *buffer;
+ int atEof;
+} AacBuffer;
+
+static void aac_buffer_shift(AacBuffer * b, size_t length)
+{
+ assert(length >= b->bytesConsumed);
+ assert(length <= b->bytesConsumed + b->bytesIntoBuffer);
+
+ memmove(b->buffer, b->buffer + length,
+ b->bytesConsumed + b->bytesIntoBuffer - length);
+
+ length -= b->bytesConsumed;
+ b->bytesConsumed = 0;
+ b->bytesIntoBuffer -= length;
+}
+
+static void fillAacBuffer(AacBuffer * b)
+{
+ size_t bread;
+
+ if (b->bytesIntoBuffer >= FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS)
+ /* buffer already full */
+ return;
+
+ aac_buffer_shift(b, b->bytesConsumed);
+
+ if (!b->atEof) {
+ size_t rest = FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS -
+ b->bytesIntoBuffer;
+
+ bread = decoder_read(b->decoder, b->inStream,
+ (void *)(b->buffer + b->bytesIntoBuffer),
+ rest);
+ if (bread == 0 && inputStreamAtEOF(b->inStream))
+ b->atEof = 1;
+ b->bytesIntoBuffer += bread;
+ }
+
+ if ((b->bytesIntoBuffer > 3 && memcmp(b->buffer, "TAG", 3) == 0) ||
+ (b->bytesIntoBuffer > 11 &&
+ memcmp(b->buffer, "LYRICSBEGIN", 11) == 0) ||
+ (b->bytesIntoBuffer > 8 && memcmp(b->buffer, "APETAGEX", 8) == 0))
+ b->bytesIntoBuffer = 0;
+}
+
+static void advanceAacBuffer(AacBuffer * b, size_t bytes)
+{
+ b->fileOffset += bytes;
+ b->bytesConsumed = bytes;
+ b->bytesIntoBuffer -= bytes;
+}
+
+static int adtsSampleRates[] =
+ { 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
+ 16000, 12000, 11025, 8000, 7350, 0, 0, 0
+};
+
+/**
+ * Check whether the buffer head is an AAC frame, and return the frame
+ * length. Returns 0 if it is not a frame.
+ */
+static size_t adts_check_frame(AacBuffer * b)
+{
+ if (b->bytesIntoBuffer <= 7)
+ return 0;
+
+ /* check syncword */
+ if (!((b->buffer[0] == 0xFF) && ((b->buffer[1] & 0xF6) == 0xF0)))
+ return 0;
+
+ return (((unsigned int)b->buffer[3] & 0x3) << 11) |
+ (((unsigned int)b->buffer[4]) << 3) |
+ (b->buffer[5] >> 5);
+}
+
+/**
+ * Find the next AAC frame in the buffer. Returns 0 if no frame is
+ * found or if not enough data is available.
+ */
+static size_t adts_find_frame(AacBuffer * b)
+{
+ const unsigned char *p;
+ size_t frame_length;
+
+ while ((p = memchr(b->buffer, 0xff, b->bytesIntoBuffer)) != NULL) {
+ /* discard data before 0xff */
+ if (p > b->buffer)
+ aac_buffer_shift(b, p - b->buffer);
+
+ if (b->bytesIntoBuffer <= 7)
+ /* not enough data yet */
+ return 0;
+
+ /* is it a frame? */
+ frame_length = adts_check_frame(b);
+ if (frame_length > 0)
+ /* yes, it is */
+ return frame_length;
+
+ /* it's just some random 0xff byte; discard and and
+ continue searching */
+ aac_buffer_shift(b, 1);
+ }
+
+ /* nothing at all; discard the whole buffer */
+ aac_buffer_shift(b, b->bytesIntoBuffer);
+ return 0;
+}
+
+static void adtsParse(AacBuffer * b, float *length)
+{
+ unsigned int frames, frameLength;
+ int sample_rate = 0;
+ float framesPerSec;
+
+ /* Read all frames to ensure correct time and bitrate */
+ for (frames = 0;; frames++) {
+ fillAacBuffer(b);
+
+ frameLength = adts_find_frame(b);
+ if (frameLength > 0) {
+ if (frames == 0) {
+ sample_rate = adtsSampleRates[(b->
+ buffer[2] & 0x3c)
+ >> 2];
+ }
+
+ if (frameLength > b->bytesIntoBuffer)
+ break;
+
+ advanceAacBuffer(b, frameLength);
+ } else
+ break;
+ }
+
+ framesPerSec = (float)sample_rate / 1024.0;
+ if (framesPerSec != 0)
+ *length = (float)frames / framesPerSec;
+}
+
+static void initAacBuffer(AacBuffer * b,
+ struct decoder *decoder, InputStream * inStream)
+{
+ memset(b, 0, sizeof(AacBuffer));
+
+ b->decoder = decoder;
+ b->inStream = inStream;
+
+ b->buffer = xmalloc(FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS);
+ memset(b->buffer, 0, FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS);
+}
+
+static void aac_parse_header(AacBuffer * b, float *length)
+{
+ size_t fileread;
+ size_t tagsize;
+
+ if (length)
+ *length = -1;
+
+ fileread = b->inStream->size;
+
+ fillAacBuffer(b);
+
+ tagsize = 0;
+ if (b->bytesIntoBuffer >= 10 && !memcmp(b->buffer, "ID3", 3)) {
+ tagsize = (b->buffer[6] << 21) | (b->buffer[7] << 14) |
+ (b->buffer[8] << 7) | (b->buffer[9] << 0);
+
+ tagsize += 10;
+ advanceAacBuffer(b, tagsize);
+ fillAacBuffer(b);
+ }
+
+ if (length == NULL)
+ return;
+
+ if (b->bytesIntoBuffer >= 2 &&
+ (b->buffer[0] == 0xFF) && ((b->buffer[1] & 0xF6) == 0xF0)) {
+ adtsParse(b, length);
+ seekInputStream(b->inStream, tagsize, SEEK_SET);
+
+ b->bytesIntoBuffer = 0;
+ b->bytesConsumed = 0;
+ b->fileOffset = tagsize;
+
+ fillAacBuffer(b);
+ } else if (memcmp(b->buffer, "ADIF", 4) == 0) {
+ int bitRate;
+ int skipSize = (b->buffer[4] & 0x80) ? 9 : 0;
+ bitRate =
+ ((unsigned int)(b->
+ buffer[4 +
+ skipSize] & 0x0F) << 19) | ((unsigned
+ int)b->
+ buffer[5
+ +
+ skipSize]
+ << 11) |
+ ((unsigned int)b->
+ buffer[6 + skipSize] << 3) | ((unsigned int)b->buffer[7 +
+ skipSize]
+ & 0xE0);
+
+ if (fileread != 0 && bitRate != 0)
+ *length = fileread * 8.0 / bitRate;
+ else
+ *length = fileread;
+ }
+}
+
+static float getAacFloatTotalTime(char *file)
+{
+ AacBuffer b;
+ float length;
+ faacDecHandle decoder;
+ faacDecConfigurationPtr config;
+ uint32_t sample_rate;
+ unsigned char channels;
+ InputStream inStream;
+ long bread;
+
+ if (openInputStream(&inStream, file) < 0)
+ return -1;
+
+ initAacBuffer(&b, NULL, &inStream);
+ aac_parse_header(&b, &length);
+
+ if (length < 0) {
+ decoder = faacDecOpen();
+
+ config = faacDecGetCurrentConfiguration(decoder);
+ config->outputFormat = FAAD_FMT_16BIT;
+ faacDecSetConfiguration(decoder, config);
+
+ fillAacBuffer(&b);
+#ifdef HAVE_FAAD_BUFLEN_FUNCS
+ bread = faacDecInit(decoder, b.buffer, b.bytesIntoBuffer,
+ &sample_rate, &channels);
+#else
+ bread = faacDecInit(decoder, b.buffer, &sample_rate, &channels);
+#endif
+ if (bread >= 0 && sample_rate > 0 && channels > 0)
+ length = 0;
+
+ faacDecClose(decoder);
+ }
+
+ if (b.buffer)
+ free(b.buffer);
+ closeInputStream(&inStream);
+
+ return length;
+}
+
+static int getAacTotalTime(char *file)
+{
+ int file_time = -1;
+ float length;
+
+ if ((length = getAacFloatTotalTime(file)) >= 0)
+ file_time = length + 0.5;
+
+ return file_time;
+}
+
+static int aac_stream_decode(struct decoder * mpd_decoder,
+ InputStream *inStream)
+{
+ float file_time;
+ float totalTime = 0;
+ faacDecHandle decoder;
+ faacDecFrameInfo frameInfo;
+ faacDecConfigurationPtr config;
+ long bread;
+ struct audio_format audio_format;
+ uint32_t sample_rate;
+ unsigned char channels;
+ unsigned int sampleCount;
+ char *sampleBuffer;
+ size_t sampleBufferLen;
+ uint16_t bitRate = 0;
+ AacBuffer b;
+ int initialized = 0;
+
+ initAacBuffer(&b, mpd_decoder, inStream);
+
+ decoder = faacDecOpen();
+
+ config = faacDecGetCurrentConfiguration(decoder);
+ config->outputFormat = FAAD_FMT_16BIT;
+#ifdef HAVE_FAACDECCONFIGURATION_DOWNMATRIX
+ config->downMatrix = 1;
+#endif
+#ifdef HAVE_FAACDECCONFIGURATION_DONTUPSAMPLEIMPLICITSBR
+ config->dontUpSampleImplicitSBR = 0;
+#endif
+ faacDecSetConfiguration(decoder, config);
+
+ while (b.bytesIntoBuffer < FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS &&
+ !b.atEof &&
+ decoder_get_command(mpd_decoder) == DECODE_COMMAND_NONE) {
+ fillAacBuffer(&b);
+ adts_find_frame(&b);
+ fillAacBuffer(&b);
+ my_usleep(10000);
+ }
+
+#ifdef HAVE_FAAD_BUFLEN_FUNCS
+ bread = faacDecInit(decoder, b.buffer, b.bytesIntoBuffer,
+ &sample_rate, &channels);
+#else
+ bread = faacDecInit(decoder, b.buffer, &sample_rate, &channels);
+#endif
+ if (bread < 0) {
+ ERROR("Error not a AAC stream.\n");
+ faacDecClose(decoder);
+ if (b.buffer)
+ free(b.buffer);
+ return -1;
+ }
+
+ audio_format.bits = 16;
+
+ file_time = 0.0;
+
+ advanceAacBuffer(&b, bread);
+
+ while (1) {
+ fillAacBuffer(&b);
+ adts_find_frame(&b);
+ fillAacBuffer(&b);
+
+ if (b.bytesIntoBuffer == 0)
+ break;
+
+#ifdef HAVE_FAAD_BUFLEN_FUNCS
+ sampleBuffer = faacDecDecode(decoder, &frameInfo, b.buffer,
+ b.bytesIntoBuffer);
+#else
+ sampleBuffer = faacDecDecode(decoder, &frameInfo, b.buffer);
+#endif
+
+ if (frameInfo.error > 0) {
+ ERROR("error decoding AAC stream\n");
+ ERROR("faad2 error: %s\n",
+ faacDecGetErrorMessage(frameInfo.error));
+ break;
+ }
+#ifdef HAVE_FAACDECFRAMEINFO_SAMPLERATE
+ sample_rate = frameInfo.samplerate;
+#endif
+
+ if (!initialized) {
+ audio_format.channels = frameInfo.channels;
+ audio_format.sample_rate = sample_rate;
+ decoder_initialized(mpd_decoder, &audio_format, totalTime);
+ initialized = 1;
+ }
+
+ advanceAacBuffer(&b, frameInfo.bytesconsumed);
+
+ sampleCount = (unsigned long)(frameInfo.samples);
+
+ if (sampleCount > 0) {
+ bitRate = frameInfo.bytesconsumed * 8.0 *
+ frameInfo.channels * sample_rate /
+ frameInfo.samples / 1000 + 0.5;
+ file_time +=
+ (float)(frameInfo.samples) / frameInfo.channels /
+ sample_rate;
+ }
+
+ sampleBufferLen = sampleCount * 2;
+
+ decoder_data(mpd_decoder, NULL, 0, sampleBuffer,
+ sampleBufferLen, file_time,
+ bitRate, NULL);
+ if (decoder_get_command(mpd_decoder) == DECODE_COMMAND_SEEK) {
+ decoder_seek_error(mpd_decoder);
+ } else if (decoder_get_command(mpd_decoder) == DECODE_COMMAND_STOP)
+ break;
+ }
+
+ decoder_flush(mpd_decoder);
+
+ faacDecClose(decoder);
+ if (b.buffer)
+ free(b.buffer);
+
+ if (!initialized)
+ return -1;
+
+ if (decoder_get_command(mpd_decoder) == DECODE_COMMAND_SEEK) {
+ decoder_seek_error(mpd_decoder);
+ }
+
+ return 0;
+}
+
+
+static int aac_decode(struct decoder * mpd_decoder, char *path)
+{
+ float file_time;
+ float totalTime;
+ faacDecHandle decoder;
+ faacDecFrameInfo frameInfo;
+ faacDecConfigurationPtr config;
+ long bread;
+ struct audio_format audio_format;
+ uint32_t sample_rate;
+ unsigned char channels;
+ unsigned int sampleCount;
+ char *sampleBuffer;
+ size_t sampleBufferLen;
+ /*float * seekTable;
+ long seekTableEnd = -1;
+ int seekPositionFound = 0; */
+ uint16_t bitRate = 0;
+ AacBuffer b;
+ InputStream inStream;
+ int initialized = 0;
+
+ if ((totalTime = getAacFloatTotalTime(path)) < 0)
+ return -1;
+
+ if (openInputStream(&inStream, path) < 0)
+ return -1;
+
+ initAacBuffer(&b, mpd_decoder, &inStream);
+ aac_parse_header(&b, NULL);
+
+ decoder = faacDecOpen();
+
+ config = faacDecGetCurrentConfiguration(decoder);
+ config->outputFormat = FAAD_FMT_16BIT;
+#ifdef HAVE_FAACDECCONFIGURATION_DOWNMATRIX
+ config->downMatrix = 1;
+#endif
+#ifdef HAVE_FAACDECCONFIGURATION_DONTUPSAMPLEIMPLICITSBR
+ config->dontUpSampleImplicitSBR = 0;
+#endif
+ faacDecSetConfiguration(decoder, config);
+
+ fillAacBuffer(&b);
+
+#ifdef HAVE_FAAD_BUFLEN_FUNCS
+ bread = faacDecInit(decoder, b.buffer, b.bytesIntoBuffer,
+ &sample_rate, &channels);
+#else
+ bread = faacDecInit(decoder, b.buffer, &sample_rate, &channels);
+#endif
+ if (bread < 0) {
+ ERROR("Error not a AAC stream.\n");
+ faacDecClose(decoder);
+ if (b.buffer)
+ free(b.buffer);
+ return -1;
+ }
+
+ audio_format.bits = 16;
+
+ file_time = 0.0;
+
+ advanceAacBuffer(&b, bread);
+
+ while (1) {
+ fillAacBuffer(&b);
+
+ if (b.bytesIntoBuffer == 0)
+ break;
+
+#ifdef HAVE_FAAD_BUFLEN_FUNCS
+ sampleBuffer = faacDecDecode(decoder, &frameInfo, b.buffer,
+ b.bytesIntoBuffer);
+#else
+ sampleBuffer = faacDecDecode(decoder, &frameInfo, b.buffer);
+#endif
+
+ if (frameInfo.error > 0) {
+ ERROR("error decoding AAC file: %s\n", path);
+ ERROR("faad2 error: %s\n",
+ faacDecGetErrorMessage(frameInfo.error));
+ break;
+ }
+#ifdef HAVE_FAACDECFRAMEINFO_SAMPLERATE
+ sample_rate = frameInfo.samplerate;
+#endif
+
+ if (!initialized) {
+ audio_format.channels = frameInfo.channels;
+ audio_format.sample_rate = sample_rate;
+ decoder_initialized(mpd_decoder, &audio_format,
+ totalTime);
+ initialized = 1;
+ }
+
+ advanceAacBuffer(&b, frameInfo.bytesconsumed);
+
+ sampleCount = (unsigned long)(frameInfo.samples);
+
+ if (sampleCount > 0) {
+ bitRate = frameInfo.bytesconsumed * 8.0 *
+ frameInfo.channels * sample_rate /
+ frameInfo.samples / 1000 + 0.5;
+ file_time +=
+ (float)(frameInfo.samples) / frameInfo.channels /
+ sample_rate;
+ }
+
+ sampleBufferLen = sampleCount * 2;
+
+ decoder_data(mpd_decoder, NULL, 0, sampleBuffer,
+ sampleBufferLen, file_time,
+ bitRate, NULL);
+ if (decoder_get_command(mpd_decoder) == DECODE_COMMAND_SEEK) {
+ decoder_seek_error(mpd_decoder);
+ } else if (decoder_get_command(mpd_decoder) == DECODE_COMMAND_STOP)
+ break;
+ }
+
+ decoder_flush(mpd_decoder);
+
+ faacDecClose(decoder);
+ if (b.buffer)
+ free(b.buffer);
+
+ if (!initialized)
+ return -1;
+
+ if (decoder_get_command(mpd_decoder) == DECODE_COMMAND_SEEK) {
+ decoder_seek_error(mpd_decoder);
+ }
+
+ return 0;
+}
+
+static struct tag *aacTagDup(char *file)
+{
+ struct tag *ret = NULL;
+ int file_time = getAacTotalTime(file);
+
+ if (file_time >= 0) {
+ if ((ret = tag_id3_load(file)) == NULL)
+ ret = tag_new();
+ ret->time = file_time;
+ } else {
+ DEBUG("aacTagDup: Failed to get total song time from: %s\n",
+ file);
+ }
+
+ return ret;
+}
+
+static const char *aac_suffixes[] = { "aac", NULL };
+static const char *aac_mimeTypes[] = { "audio/aac", "audio/aacp", NULL };
+
+struct decoder_plugin aacPlugin = {
+ .name = "aac",
+ .stream_decode = aac_stream_decode,
+ .file_decode = aac_decode,
+ .tag_dup = aacTagDup,
+ .stream_types = INPUT_PLUGIN_STREAM_FILE | INPUT_PLUGIN_STREAM_URL,
+ .suffixes = aac_suffixes,
+ .mime_types = aac_mimeTypes
+};
diff --git a/src/decoder/audiofile_plugin.c b/src/decoder/audiofile_plugin.c
new file mode 100644
index 000000000..99846e853
--- /dev/null
+++ b/src/decoder/audiofile_plugin.c
@@ -0,0 +1,147 @@
+/* the Music Player Daemon (MPD)
+ * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
+ * This project's homepage is: http://www.musicpd.org
+ *
+ * libaudiofile (wave) support added by Eric Wong <normalperson@yhbt.net>
+ *
+ * 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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include "../decoder_api.h"
+#include "../log.h"
+
+#include <sys/stat.h>
+#include <audiofile.h>
+
+/* pick 1020 since its devisible for 8,16,24, and 32-bit audio */
+#define CHUNK_SIZE 1020
+
+static int getAudiofileTotalTime(char *file)
+{
+ int total_time;
+ AFfilehandle af_fp = afOpenFile(file, "r", NULL);
+ if (af_fp == AF_NULL_FILEHANDLE) {
+ return -1;
+ }
+ total_time = (int)
+ ((double)afGetFrameCount(af_fp, AF_DEFAULT_TRACK)
+ / afGetRate(af_fp, AF_DEFAULT_TRACK));
+ afCloseFile(af_fp);
+ return total_time;
+}
+
+static int audiofile_decode(struct decoder * decoder, char *path)
+{
+ int fs, frame_count;
+ AFfilehandle af_fp;
+ int bits;
+ struct audio_format audio_format;
+ float total_time;
+ uint16_t bitRate;
+ struct stat st;
+ int ret, current = 0;
+ char chunk[CHUNK_SIZE];
+
+ if (stat(path, &st) < 0) {
+ ERROR("failed to stat: %s\n", path);
+ return -1;
+ }
+
+ af_fp = afOpenFile(path, "r", NULL);
+ if (af_fp == AF_NULL_FILEHANDLE) {
+ ERROR("failed to open: %s\n", path);
+ return -1;
+ }
+
+ afSetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK,
+ AF_SAMPFMT_TWOSCOMP, 16);
+ 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);
+
+ frame_count = afGetFrameCount(af_fp, AF_DEFAULT_TRACK);
+
+ total_time = ((float)frame_count / (float)audio_format.sample_rate);
+
+ bitRate = (uint16_t)(st.st_size * 8.0 / total_time / 1000.0 + 0.5);
+
+ if (audio_format.bits != 8 && audio_format.bits != 16) {
+ ERROR("Only 8 and 16-bit files are supported. %s is %i-bit\n",
+ path, audio_format.bits);
+ afCloseFile(af_fp);
+ return -1;
+ }
+
+ fs = (int)afGetVirtualFrameSize(af_fp, AF_DEFAULT_TRACK, 1);
+
+ decoder_initialized(decoder, &audio_format, total_time);
+
+ do {
+ if (decoder_get_command(decoder) == DECODE_COMMAND_SEEK) {
+ decoder_clear(decoder);
+ current = decoder_seek_where(decoder) *
+ audio_format.sample_rate;
+ afSeekFrame(af_fp, AF_DEFAULT_TRACK, current);
+ decoder_command_finished(decoder);
+ }
+
+ ret = afReadFrames(af_fp, AF_DEFAULT_TRACK, chunk,
+ CHUNK_SIZE / fs);
+ if (ret <= 0)
+ break;
+
+ current += ret;
+ decoder_data(decoder, NULL, 1,
+ chunk, ret * fs,
+ (float)current / (float)audio_format.sample_rate,
+ bitRate, NULL);
+ } while (decoder_get_command(decoder) != DECODE_COMMAND_STOP);
+
+ decoder_flush(decoder);
+
+ afCloseFile(af_fp);
+
+ return 0;
+}
+
+static struct tag *audiofileTagDup(char *file)
+{
+ struct tag *ret = NULL;
+ int total_time = getAudiofileTotalTime(file);
+
+ if (total_time >= 0) {
+ if (!ret)
+ ret = tag_new();
+ ret->time = total_time;
+ } else {
+ DEBUG
+ ("audiofileTagDup: Failed to get total song time from: %s\n",
+ file);
+ }
+
+ return ret;
+}
+
+static const char *audiofileSuffixes[] = { "wav", "au", "aiff", "aif", NULL };
+
+struct decoder_plugin audiofilePlugin = {
+ .name = "audiofile",
+ .file_decode = audiofile_decode,
+ .tag_dup = audiofileTagDup,
+ .stream_types = INPUT_PLUGIN_STREAM_FILE,
+ .suffixes = audiofileSuffixes,
+};
diff --git a/src/decoder/ffmpeg_plugin.c b/src/decoder/ffmpeg_plugin.c
new file mode 100644
index 000000000..6455cd1ce
--- /dev/null
+++ b/src/decoder/ffmpeg_plugin.c
@@ -0,0 +1,419 @@
+/* the Music Player Daemon (MPD)
+ * Copyright (C) 2008 Viliam Mateicka <viliam.mateicka@gmail.com>
+ * This project's homepage is: 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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include "../decoder_api.h"
+#include "../log.h"
+#include "../utils.h"
+#include "../log.h"
+
+#include <stdio.h>
+#include <unistd.h>
+#include <stdlib.h>
+#include <string.h>
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <unistd.h>
+
+#ifdef OLD_FFMPEG_INCLUDES
+#include <avcodec.h>
+#include <avformat.h>
+#include <avio.h>
+#else
+#include <libavcodec/avcodec.h>
+#include <libavformat/avformat.h>
+#include <libavformat/avio.h>
+#endif
+
+typedef struct {
+ int audioStream;
+ AVFormatContext *pFormatCtx;
+ AVCodecContext *aCodecCtx;
+ AVCodec *aCodec;
+ struct decoder *decoder;
+ InputStream *input;
+ struct tag *tag;
+} BasePtrs;
+
+typedef struct {
+ /** hack - see url_to_base() */
+ char url[8];
+
+ struct decoder *decoder;
+ InputStream *input;
+} FopsHelper;
+
+/**
+ * Convert a faked mpd:// URL to a FopsHelper structure. This is a
+ * hack because ffmpeg does not provide a nice API for passing a
+ * user-defined pointer to mpdurl_open().
+ */
+static FopsHelper *url_to_base(const char *url)
+{
+ union {
+ const char *in;
+ FopsHelper *out;
+ } u = { .in = url };
+ return u.out;
+}
+
+static int mpdurl_open(URLContext *h, const char *filename,
+ mpd_unused int flags)
+{
+ FopsHelper *base = url_to_base(filename);
+ h->priv_data = base;
+ h->is_streamed = (base->input->seekable ? 0 : 1);
+ return 0;
+}
+
+static int mpdurl_read(URLContext *h, unsigned char *buf, int size)
+{
+ int ret;
+ FopsHelper *base = (FopsHelper *) h->priv_data;
+ while (1) {
+ ret = readFromInputStream(base->input, (void *)buf, size);
+ if (ret == 0) {
+ DEBUG("ret 0\n");
+ if (inputStreamAtEOF(base->input) ||
+ (base->decoder &&
+ decoder_get_command(base->decoder) != DECODE_COMMAND_NONE)) {
+ DEBUG("eof stream\n");
+ return ret;
+ } else {
+ my_usleep(10000);
+ }
+ } else {
+ break;
+ }
+ }
+ return ret;
+}
+
+static int64_t mpdurl_seek(URLContext *h, int64_t pos, int whence)
+{
+ FopsHelper *base = (FopsHelper *) h->priv_data;
+ if (whence != AVSEEK_SIZE) { //only ftell
+ (void) seekInputStream(base->input, pos, whence);
+ }
+ return base->input->offset;
+}
+
+static int mpdurl_close(URLContext *h)
+{
+ FopsHelper *base = (FopsHelper *) h->priv_data;
+ if (base && base->input->seekable) {
+ (void) seekInputStream(base->input, 0, SEEK_SET);
+ }
+ h->priv_data = 0;
+ return 0;
+}
+
+static URLProtocol mpdurl_fileops = {
+ .name = "mpd",
+ .url_open = mpdurl_open,
+ .url_read = mpdurl_read,
+ .url_seek = mpdurl_seek,
+ .url_close = mpdurl_close,
+};
+
+static int ffmpeg_init(void)
+{
+ av_register_all();
+ register_protocol(&mpdurl_fileops);
+ return 0;
+}
+
+static int ffmpeg_helper(InputStream *input, int (*callback)(BasePtrs *ptrs),
+ BasePtrs *ptrs)
+{
+ AVFormatContext *pFormatCtx;
+ AVCodecContext *aCodecCtx;
+ AVCodec *aCodec;
+ int ret, audioStream;
+ unsigned i;
+ FopsHelper fopshelp = {
+ .url = "mpd://X", /* only the mpd:// prefix matters */
+ };
+
+ fopshelp.input = input;
+ if (ptrs && ptrs->decoder) {
+ fopshelp.decoder = ptrs->decoder; //are we in decoding loop ?
+ } else {
+ fopshelp.decoder = NULL;
+ }
+
+ //ffmpeg works with ours "fileops" helper
+ if (av_open_input_file(&pFormatCtx, fopshelp.url, NULL, 0, NULL)!=0) {
+ ERROR("Open failed!\n");
+ return -1;
+ }
+
+ if (av_find_stream_info(pFormatCtx)<0) {
+ ERROR("Couldn't find stream info!\n");
+ return -1;
+ }
+
+ audioStream = -1;
+ for(i=0; i<pFormatCtx->nb_streams; i++) {
+ if (pFormatCtx->streams[i]->codec->codec_type==CODEC_TYPE_AUDIO &&
+ audioStream < 0) {
+ audioStream=i;
+ }
+ }
+
+ if(audioStream==-1) {
+ ERROR("No audio stream inside!\n");
+ return -1;
+ }
+
+ aCodecCtx = pFormatCtx->streams[audioStream]->codec;
+ aCodec = avcodec_find_decoder(aCodecCtx->codec_id);
+
+ if (!aCodec) {
+ ERROR("Unsupported audio codec!\n");
+ return -1;
+ }
+
+ if (avcodec_open(aCodecCtx, aCodec)<0) {
+ ERROR("Could not open codec!\n");
+ return -1;
+ }
+
+ if (callback) {
+ ptrs->audioStream = audioStream;
+ ptrs->pFormatCtx = pFormatCtx;
+ ptrs->aCodecCtx = aCodecCtx;
+ ptrs->aCodec = aCodec;
+
+ ret = (*callback)( ptrs );
+ } else {
+ ret = 0;
+ DEBUG("playable\n");
+ }
+
+ avcodec_close(aCodecCtx);
+ av_close_input_file(pFormatCtx);
+
+ return ret;
+}
+
+static bool ffmpeg_try_decode(InputStream *input)
+{
+ int ret;
+ if (input->seekable) {
+ ret = ffmpeg_helper(input, NULL, NULL);
+ } else {
+ ret = 0;
+ }
+ return (ret == -1 ? 0 : 1);
+}
+
+static int ffmpeg_decode_internal(BasePtrs *base)
+{
+ struct decoder *decoder = base->decoder;
+ AVCodecContext *aCodecCtx = base->aCodecCtx;
+ AVFormatContext *pFormatCtx = base->pFormatCtx;
+ AVPacket packet;
+ int len, audio_size;
+ int position;
+ struct audio_format audio_format;
+ int current, total_time;
+ uint8_t audio_buf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 2];
+
+ total_time = 0;
+
+ DEBUG("decoder_start\n");
+
+ if (aCodecCtx->channels > 2) {
+ aCodecCtx->channels = 2;
+ }
+
+ audio_format.bits = (uint8_t)16;
+ audio_format.sample_rate = (unsigned int)aCodecCtx->sample_rate;
+ audio_format.channels = aCodecCtx->channels;
+
+ // frame_count = afGetFrameCount(af_fp, AF_DEFAULT_TRACK);
+ // total_time = ((float)frame_count / (float)audio_format.sample_rate);
+
+ //there is some problem with this on some demux (mp3 at least)
+ if (pFormatCtx->duration != (int)AV_NOPTS_VALUE) {
+ total_time = pFormatCtx->duration / AV_TIME_BASE;
+ }
+
+ DEBUG("ffmpeg sample rate: %dHz %d channels\n",
+ aCodecCtx->sample_rate, aCodecCtx->channels);
+
+ decoder_initialized(decoder, &audio_format, total_time);
+
+ position = 0;
+
+ DEBUG("duration:%d (%d secs)\n", (int) pFormatCtx->duration,
+ (int) total_time);
+
+ do {
+
+ if (decoder_get_command(decoder) == DECODE_COMMAND_SEEK) {
+
+ DEBUG("seek\n");
+ decoder_clear(decoder);
+ current = decoder_seek_where(decoder) * AV_TIME_BASE;
+
+ if (av_seek_frame(pFormatCtx, -1, current , 0) < 0) {
+ WARNING("seek to %d failed\n", current);
+ }
+
+ decoder_command_finished(decoder);
+ }
+
+ if (av_read_frame(pFormatCtx, &packet) >= 0) {
+ if(packet.stream_index == base->audioStream) {
+
+ position = av_rescale_q(packet.pts, pFormatCtx->streams[base->audioStream]->time_base,
+ (AVRational){1, 1});
+
+ audio_size = sizeof(audio_buf);
+ len = avcodec_decode_audio2(aCodecCtx,
+ (int16_t *)audio_buf,
+ &audio_size,
+ packet.data,
+ packet.size);
+
+ if(len >= 0) {
+ if(audio_size >= 0) {
+ // DEBUG("sending data %d/%d\n", audio_size, len);
+
+ decoder_data(decoder, NULL, 1,
+ audio_buf, audio_size,
+ position, //(float)current / (float)audio_format.sample_rate,
+ aCodecCtx->bit_rate / 1000, NULL);
+
+ }
+ } else {
+ WARNING("skiping frame!\n");
+ }
+ }
+ av_free_packet(&packet);
+ } else {
+ //end of file
+ break;
+ }
+ } while (decoder_get_command(decoder) != DECODE_COMMAND_STOP);
+
+ decoder_flush(decoder);
+
+ DEBUG("decoder finish\n");
+
+ return 0;
+}
+
+static int ffmpeg_decode(struct decoder *decoder, InputStream *input)
+{
+ BasePtrs base;
+ int ret;
+
+ DEBUG("decode start\n");
+
+ base.input = input;
+ base.decoder = decoder;
+
+ ret = ffmpeg_helper(input, ffmpeg_decode_internal, &base);
+
+ DEBUG("decode finish\n");
+
+ return ret;
+}
+
+static int ffmpeg_tag_internal(BasePtrs *base)
+{
+ struct tag *tag = (struct tag *) base->tag;
+
+ if (base->pFormatCtx->duration != (int)AV_NOPTS_VALUE) {
+ tag->time = base->pFormatCtx->duration / AV_TIME_BASE;
+ } else {
+ tag->time = 0;
+ }
+ return 0;
+}
+
+//no tag reading in ffmpeg, check if playable
+static struct tag *ffmpeg_tag(char *file)
+{
+ InputStream input;
+ BasePtrs base;
+ int ret;
+ struct tag *tag = NULL;
+
+ if (openInputStream(&input, file) < 0) {
+ ERROR("failed to open %s\n", file);
+ return NULL;
+ }
+
+ tag = tag_new();
+
+ base.tag = tag;
+ ret = ffmpeg_helper(&input, ffmpeg_tag_internal, &base);
+
+ if (ret != 0) {
+ free(tag);
+ tag = NULL;
+ }
+
+ closeInputStream(&input);
+
+ return tag;
+}
+
+/**
+ * ffmpeg can decode almost everything from open codecs
+ * and also some of propietary codecs
+ * its hard to tell what can ffmpeg decode
+ * we can later put this into configure script
+ * to be sure ffmpeg is used to handle
+ * only that files
+ */
+
+static const char *ffmpeg_Suffixes[] = {
+ "wma", "asf", "wmv", "mpeg", "mpg", "avi", "vob", "mov", "qt", "swf", "rm", "swf",
+ "mp1", "mp2", "mp3", "mp4", "m4a", "flac", "ogg", "wav", "au", "aiff", "aif", "ac3", "aac", "mpc",
+ NULL
+};
+
+//not sure if this is correct...
+static const char *ffmpeg_Mimetypes[] = {
+ "video/x-ms-asf",
+ "audio/x-ms-wma",
+ "audio/x-ms-wax",
+ "video/x-ms-wmv",
+ "video/x-ms-wvx",
+ "video/x-ms-wm",
+ "video/x-ms-wmx",
+ "application/x-ms-wmz",
+ "application/x-ms-wmd",
+ "audio/mpeg",
+ NULL
+};
+
+struct decoder_plugin ffmpegPlugin = {
+ .name = "ffmpeg",
+ .init = ffmpeg_init,
+ .try_decode = ffmpeg_try_decode,
+ .stream_decode = ffmpeg_decode,
+ .tag_dup = ffmpeg_tag,
+ .stream_types = INPUT_PLUGIN_STREAM_URL | INPUT_PLUGIN_STREAM_FILE,
+ .suffixes = ffmpeg_Suffixes,
+ .mime_types = ffmpeg_Mimetypes
+};
diff --git a/src/decoder/flac_plugin.c b/src/decoder/flac_plugin.c
new file mode 100644
index 000000000..7b9fce27d
--- /dev/null
+++ b/src/decoder/flac_plugin.c
@@ -0,0 +1,459 @@
+/* the Music Player Daemon (MPD)
+ * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
+ * This project's homepage is: 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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include "_flac_common.h"
+#include "../utils.h"
+#include "../log.h"
+
+#include <assert.h>
+
+/* this code was based on flac123, from flac-tools */
+
+static flac_read_status flacRead(mpd_unused const flac_decoder * flacDec,
+ FLAC__byte buf[],
+ flac_read_status_size_t *bytes,
+ void *fdata)
+{
+ FlacData *data = (FlacData *) fdata;
+ size_t r;
+
+ r = decoder_read(data->decoder, data->inStream, (void *)buf, *bytes);
+ *bytes = r;
+
+ if (r == 0) {
+ if (decoder_get_command(data->decoder) != DECODE_COMMAND_NONE ||
+ inputStreamAtEOF(data->inStream))
+ return flac_read_status_eof;
+ else
+ return flac_read_status_abort;
+ }
+
+ return flac_read_status_continue;
+}
+
+static flac_seek_status flacSeek(mpd_unused const flac_decoder * flacDec,
+ FLAC__uint64 offset,
+ void *fdata)
+{
+ FlacData *data = (FlacData *) fdata;
+
+ if (seekInputStream(data->inStream, offset, SEEK_SET) < 0) {
+ return flac_seek_status_error;
+ }
+
+ return flac_seek_status_ok;
+}
+
+static flac_tell_status flacTell(mpd_unused const flac_decoder * flacDec,
+ FLAC__uint64 * offset,
+ void *fdata)
+{
+ FlacData *data = (FlacData *) fdata;
+
+ *offset = (long)(data->inStream->offset);
+
+ return flac_tell_status_ok;
+}
+
+static flac_length_status flacLength(mpd_unused const flac_decoder * flacDec,
+ FLAC__uint64 * length,
+ void *fdata)
+{
+ FlacData *data = (FlacData *) fdata;
+
+ *length = (size_t) (data->inStream->size);
+
+ return flac_length_status_ok;
+}
+
+static FLAC__bool flacEOF(mpd_unused const flac_decoder * flacDec, void *fdata)
+{
+ FlacData *data = (FlacData *) fdata;
+
+ return (decoder_get_command(data->decoder) != DECODE_COMMAND_NONE &&
+ decoder_get_command(data->decoder) != DECODE_COMMAND_SEEK) ||
+ inputStreamAtEOF(data->inStream);
+}
+
+static void flacError(mpd_unused const flac_decoder *dec,
+ FLAC__StreamDecoderErrorStatus status, void *fdata)
+{
+ flac_error_common_cb("flac", status, (FlacData *) fdata);
+}
+
+#if !defined(FLAC_API_VERSION_CURRENT) || FLAC_API_VERSION_CURRENT <= 7
+static void flacPrintErroredState(FLAC__SeekableStreamDecoderState state)
+{
+ const char *str = ""; /* "" to silence compiler warning */
+ switch (state) {
+ case FLAC__SEEKABLE_STREAM_DECODER_OK:
+ case FLAC__SEEKABLE_STREAM_DECODER_SEEKING:
+ case FLAC__SEEKABLE_STREAM_DECODER_END_OF_STREAM:
+ return;
+ case FLAC__SEEKABLE_STREAM_DECODER_MEMORY_ALLOCATION_ERROR:
+ str = "allocation error";
+ break;
+ case FLAC__SEEKABLE_STREAM_DECODER_READ_ERROR:
+ str = "read error";
+ break;
+ case FLAC__SEEKABLE_STREAM_DECODER_SEEK_ERROR:
+ str = "seek error";
+ break;
+ case FLAC__SEEKABLE_STREAM_DECODER_STREAM_DECODER_ERROR:
+ str = "seekable stream error";
+ break;
+ case FLAC__SEEKABLE_STREAM_DECODER_ALREADY_INITIALIZED:
+ str = "decoder already initialized";
+ break;
+ case FLAC__SEEKABLE_STREAM_DECODER_INVALID_CALLBACK:
+ str = "invalid callback";
+ break;
+ case FLAC__SEEKABLE_STREAM_DECODER_UNINITIALIZED:
+ str = "decoder uninitialized";
+ }
+ ERROR("flac %s\n", str);
+}
+
+static int flac_init(FLAC__SeekableStreamDecoder *dec,
+ FLAC__SeekableStreamDecoderReadCallback read_cb,
+ FLAC__SeekableStreamDecoderSeekCallback seek_cb,
+ FLAC__SeekableStreamDecoderTellCallback tell_cb,
+ FLAC__SeekableStreamDecoderLengthCallback length_cb,
+ FLAC__SeekableStreamDecoderEofCallback eof_cb,
+ FLAC__SeekableStreamDecoderWriteCallback write_cb,
+ FLAC__SeekableStreamDecoderMetadataCallback metadata_cb,
+ FLAC__SeekableStreamDecoderErrorCallback error_cb,
+ void *data)
+{
+ int s = 1;
+ s &= FLAC__seekable_stream_decoder_set_read_callback(dec, read_cb);
+ s &= FLAC__seekable_stream_decoder_set_seek_callback(dec, seek_cb);
+ s &= FLAC__seekable_stream_decoder_set_tell_callback(dec, tell_cb);
+ s &= FLAC__seekable_stream_decoder_set_length_callback(dec, length_cb);
+ s &= FLAC__seekable_stream_decoder_set_eof_callback(dec, eof_cb);
+ s &= FLAC__seekable_stream_decoder_set_write_callback(dec, write_cb);
+ s &= FLAC__seekable_stream_decoder_set_metadata_callback(dec,
+ metadata_cb);
+ s &= FLAC__seekable_stream_decoder_set_metadata_respond(dec,
+ FLAC__METADATA_TYPE_VORBIS_COMMENT);
+ s &= FLAC__seekable_stream_decoder_set_error_callback(dec, error_cb);
+ s &= FLAC__seekable_stream_decoder_set_client_data(dec, data);
+ if (!s || (FLAC__seekable_stream_decoder_init(dec) !=
+ FLAC__SEEKABLE_STREAM_DECODER_OK))
+ return 0;
+ return 1;
+}
+#else /* FLAC_API_VERSION_CURRENT >= 7 */
+static void flacPrintErroredState(FLAC__StreamDecoderState state)
+{
+ const char *str = ""; /* "" to silence compiler warning */
+ switch (state) {
+ case FLAC__STREAM_DECODER_SEARCH_FOR_METADATA:
+ case FLAC__STREAM_DECODER_READ_METADATA:
+ case FLAC__STREAM_DECODER_SEARCH_FOR_FRAME_SYNC:
+ case FLAC__STREAM_DECODER_READ_FRAME:
+ case FLAC__STREAM_DECODER_END_OF_STREAM:
+ return;
+ case FLAC__STREAM_DECODER_OGG_ERROR:
+ str = "error in the Ogg layer";
+ break;
+ case FLAC__STREAM_DECODER_SEEK_ERROR:
+ str = "seek error";
+ break;
+ case FLAC__STREAM_DECODER_ABORTED:
+ str = "decoder aborted by read";
+ break;
+ case FLAC__STREAM_DECODER_MEMORY_ALLOCATION_ERROR:
+ str = "allocation error";
+ break;
+ case FLAC__STREAM_DECODER_UNINITIALIZED:
+ str = "decoder uninitialized";
+ }
+ ERROR("flac %s\n", str);
+}
+#endif /* FLAC_API_VERSION_CURRENT >= 7 */
+
+static void flacMetadata(mpd_unused const flac_decoder * dec,
+ const FLAC__StreamMetadata * block, void *vdata)
+{
+ flac_metadata_common_cb(block, (FlacData *) vdata);
+}
+
+static FLAC__StreamDecoderWriteStatus flacWrite(const flac_decoder *dec,
+ const FLAC__Frame * frame,
+ const FLAC__int32 * const buf[],
+ void *vdata)
+{
+ FLAC__uint32 samples = frame->header.blocksize;
+ FlacData *data = (FlacData *) vdata;
+ float timeChange;
+ FLAC__uint64 newPosition = 0;
+
+ timeChange = ((float)samples) / frame->header.sample_rate;
+ data->time += timeChange;
+
+ flac_get_decode_position(dec, &newPosition);
+ if (data->position && newPosition >= data->position) {
+ assert(timeChange >= 0);
+
+ data->bitRate =
+ ((newPosition - data->position) * 8.0 / timeChange)
+ / 1000 + 0.5;
+ }
+ data->position = newPosition;
+
+ return flac_common_write(data, frame, buf);
+}
+
+static struct tag *flacMetadataDup(char *file, int *vorbisCommentFound)
+{
+ struct tag *ret = NULL;
+ FLAC__Metadata_SimpleIterator *it;
+ FLAC__StreamMetadata *block = NULL;
+
+ *vorbisCommentFound = 0;
+
+ it = FLAC__metadata_simple_iterator_new();
+ if (!FLAC__metadata_simple_iterator_init(it, file, 1, 0)) {
+ const char *err;
+ FLAC_API FLAC__Metadata_SimpleIteratorStatus s;
+
+ s = FLAC__metadata_simple_iterator_status(it);
+
+ switch (s) { /* slightly more human-friendly messages: */
+ case FLAC__METADATA_SIMPLE_ITERATOR_STATUS_ILLEGAL_INPUT:
+ err = "illegal input";
+ break;
+ case FLAC__METADATA_SIMPLE_ITERATOR_STATUS_ERROR_OPENING_FILE:
+ err = "error opening file";
+ break;
+ case FLAC__METADATA_SIMPLE_ITERATOR_STATUS_NOT_A_FLAC_FILE:
+ err = "not a FLAC file";
+ break;
+ default:
+ err = FLAC__Metadata_SimpleIteratorStatusString[s];
+ }
+ DEBUG("flacMetadataDup: Reading '%s' "
+ "metadata gave the following error: %s\n",
+ file, err);
+ FLAC__metadata_simple_iterator_delete(it);
+ return ret;
+ }
+
+ do {
+ block = FLAC__metadata_simple_iterator_get_block(it);
+ if (!block)
+ break;
+ if (block->type == FLAC__METADATA_TYPE_VORBIS_COMMENT) {
+ ret = copyVorbisCommentBlockToMpdTag(block, ret);
+
+ if (ret)
+ *vorbisCommentFound = 1;
+ } else if (block->type == FLAC__METADATA_TYPE_STREAMINFO) {
+ if (!ret)
+ ret = tag_new();
+ ret->time = ((float)block->data.stream_info.
+ total_samples) /
+ block->data.stream_info.sample_rate + 0.5;
+ }
+ FLAC__metadata_object_delete(block);
+ } while (FLAC__metadata_simple_iterator_next(it));
+
+ FLAC__metadata_simple_iterator_delete(it);
+ return ret;
+}
+
+static struct tag *flacTagDup(char *file)
+{
+ struct tag *ret = NULL;
+ int foundVorbisComment = 0;
+
+ ret = flacMetadataDup(file, &foundVorbisComment);
+ if (!ret) {
+ DEBUG("flacTagDup: Failed to grab information from: %s\n",
+ file);
+ return NULL;
+ }
+ if (!foundVorbisComment) {
+ struct tag *temp = tag_id3_load(file);
+ if (temp) {
+ temp->time = ret->time;
+ tag_free(ret);
+ ret = temp;
+ }
+ }
+
+ return ret;
+}
+
+static int flac_decode_internal(struct decoder * decoder,
+ InputStream * inStream, int is_ogg)
+{
+ flac_decoder *flacDec;
+ FlacData data;
+ const char *err = NULL;
+
+ if (!(flacDec = flac_new()))
+ return -1;
+ init_FlacData(&data, decoder, inStream);
+
+#if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7
+ if(!FLAC__stream_decoder_set_metadata_respond(flacDec, FLAC__METADATA_TYPE_VORBIS_COMMENT))
+ {
+ DEBUG(__FILE__": Failed to set metadata respond\n");
+ }
+#endif
+
+
+ if (is_ogg) {
+ if (!flac_ogg_init(flacDec, flacRead, flacSeek, flacTell,
+ flacLength, flacEOF, flacWrite, flacMetadata,
+ flacError, (void *)&data)) {
+ err = "doing Ogg init()";
+ goto fail;
+ }
+ } else {
+ if (!flac_init(flacDec, flacRead, flacSeek, flacTell,
+ flacLength, flacEOF, flacWrite, flacMetadata,
+ flacError, (void *)&data)) {
+ err = "doing init()";
+ goto fail;
+ }
+ if (!flac_process_metadata(flacDec)) {
+ err = "problem reading metadata";
+ goto fail;
+ }
+ }
+
+ decoder_initialized(decoder, &data.audio_format, data.total_time);
+
+ while (1) {
+ if (!flac_process_single(flacDec))
+ break;
+ if (decoder_get_command(decoder) == DECODE_COMMAND_SEEK) {
+ FLAC__uint64 sampleToSeek = decoder_seek_where(decoder) *
+ data.audio_format.sample_rate + 0.5;
+ if (flac_seek_absolute(flacDec, sampleToSeek)) {
+ decoder_clear(decoder);
+ data.time = ((float)sampleToSeek) /
+ data.audio_format.sample_rate;
+ data.position = 0;
+ decoder_command_finished(decoder);
+ } else
+ decoder_seek_error(decoder);
+ } else if (flac_get_state(flacDec) == flac_decoder_eof)
+ break;
+ }
+ if (decoder_get_command(decoder) != DECODE_COMMAND_STOP) {
+ flacPrintErroredState(flac_get_state(flacDec));
+ flac_finish(flacDec);
+ }
+
+fail:
+ if (data.replayGainInfo)
+ freeReplayGainInfo(data.replayGainInfo);
+
+ if (flacDec)
+ flac_delete(flacDec);
+
+ if (err) {
+ ERROR("flac %s\n", err);
+ return -1;
+ }
+ return 0;
+}
+
+static int flac_decode(struct decoder * decoder, InputStream * inStream)
+{
+ return flac_decode_internal(decoder, inStream, 0);
+}
+
+#if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7 && \
+ !defined(HAVE_OGGFLAC)
+static struct tag *oggflac_tag_dup(char *file)
+{
+ struct tag *ret = NULL;
+ FLAC__Metadata_Iterator *it;
+ FLAC__StreamMetadata *block;
+ FLAC__Metadata_Chain *chain = FLAC__metadata_chain_new();
+
+ if (!(FLAC__metadata_chain_read_ogg(chain, file)))
+ goto out;
+ it = FLAC__metadata_iterator_new();
+ FLAC__metadata_iterator_init(it, chain);
+ do {
+ if (!(block = FLAC__metadata_iterator_get_block(it)))
+ break;
+ if (block->type == FLAC__METADATA_TYPE_VORBIS_COMMENT) {
+ ret = copyVorbisCommentBlockToMpdTag(block, ret);
+ } else if (block->type == FLAC__METADATA_TYPE_STREAMINFO) {
+ if (!ret)
+ ret = tag_new();
+ ret->time = ((float)block->data.stream_info.
+ total_samples) /
+ block->data.stream_info.sample_rate + 0.5;
+ }
+ } while (FLAC__metadata_iterator_next(it));
+ FLAC__metadata_iterator_delete(it);
+out:
+ FLAC__metadata_chain_delete(chain);
+ return ret;
+}
+
+static int oggflac_decode(struct decoder *decoder, InputStream * inStream)
+{
+ return flac_decode_internal(decoder, inStream, 1);
+}
+
+static bool oggflac_try_decode(InputStream * inStream)
+{
+ return FLAC_API_SUPPORTS_OGG_FLAC &&
+ ogg_stream_type_detect(inStream) == FLAC;
+}
+
+static const char *oggflac_suffixes[] = { "ogg", "oga", NULL };
+static const char *oggflac_mime_types[] = { "audio/x-flac+ogg",
+ "application/ogg",
+ "application/x-ogg",
+ NULL };
+
+struct decoder_plugin oggflacPlugin = {
+ .name = "oggflac",
+ .try_decode = oggflac_try_decode,
+ .stream_decode = oggflac_decode,
+ .tag_dup = oggflac_tag_dup,
+ .stream_types = INPUT_PLUGIN_STREAM_URL | INPUT_PLUGIN_STREAM_FILE,
+ .suffixes = oggflac_suffixes,
+ .mime_types = oggflac_mime_types
+};
+
+#endif /* FLAC_API_VERSION_CURRENT >= 7 */
+
+static const char *flacSuffixes[] = { "flac", NULL };
+static const char *flac_mime_types[] = { "audio/x-flac",
+ "application/x-flac",
+ NULL };
+
+struct decoder_plugin flacPlugin = {
+ .name = "flac",
+ .stream_decode = flac_decode,
+ .tag_dup = flacTagDup,
+ .stream_types = INPUT_PLUGIN_STREAM_URL | INPUT_PLUGIN_STREAM_FILE,
+ .suffixes = flacSuffixes,
+ .mime_types = flac_mime_types
+};
diff --git a/src/decoder/mod_plugin.c b/src/decoder/mod_plugin.c
new file mode 100644
index 000000000..5916a24ab
--- /dev/null
+++ b/src/decoder/mod_plugin.c
@@ -0,0 +1,278 @@
+/* the Music Player Daemon (MPD)
+ * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
+ * This project's homepage is: 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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include "../decoder_api.h"
+#include "../utils.h"
+#include "../log.h"
+
+#include <mikmod.h>
+
+/* this is largely copied from alsaplayer */
+
+#define MIKMOD_FRAME_SIZE 4096
+
+static BOOL mod_mpd_Init(void)
+{
+ return VC_Init();
+}
+
+static void mod_mpd_Exit(void)
+{
+ VC_Exit();
+}
+
+static void mod_mpd_Update(void)
+{
+}
+
+static BOOL mod_mpd_IsThere(void)
+{
+ return 1;
+}
+
+static char drv_name[] = "MPD";
+static char drv_version[] = "MPD Output Driver v0.1";
+
+#if (LIBMIKMOD_VERSION > 0x030106)
+static char drv_alias[] = "mpd";
+#endif
+
+static MDRIVER drv_mpd = {
+ NULL,
+ drv_name,
+ drv_version,
+ 0,
+ 255,
+#if (LIBMIKMOD_VERSION > 0x030106)
+ drv_alias,
+#if (LIBMIKMOD_VERSION >= 0x030200)
+ NULL, /* CmdLineHelp */
+#endif
+ NULL, /* CommandLine */
+#endif
+ mod_mpd_IsThere,
+ VC_SampleLoad,
+ VC_SampleUnload,
+ VC_SampleSpace,
+ VC_SampleLength,
+ mod_mpd_Init,
+ mod_mpd_Exit,
+ NULL,
+ VC_SetNumVoices,
+ VC_PlayStart,
+ VC_PlayStop,
+ mod_mpd_Update,
+ NULL,
+ VC_VoiceSetVolume,
+ VC_VoiceGetVolume,
+ VC_VoiceSetFrequency,
+ VC_VoiceGetFrequency,
+ VC_VoiceSetPanning,
+ VC_VoiceGetPanning,
+ VC_VoicePlay,
+ VC_VoiceStop,
+ VC_VoiceStopped,
+ VC_VoiceGetPosition,
+ VC_VoiceRealVolume
+};
+
+static int mod_mikModInitiated;
+static int mod_mikModInitError;
+
+static int mod_initMikMod(void)
+{
+ static char params[] = "";
+
+ if (mod_mikModInitError)
+ return -1;
+
+ if (!mod_mikModInitiated) {
+ mod_mikModInitiated = 1;
+
+ md_device = 0;
+ md_reverb = 0;
+
+ MikMod_RegisterDriver(&drv_mpd);
+ MikMod_RegisterAllLoaders();
+ }
+
+ md_pansep = 64;
+ md_mixfreq = 44100;
+ md_mode = (DMODE_SOFT_MUSIC | DMODE_INTERP | DMODE_STEREO |
+ DMODE_16BITS);
+
+ if (MikMod_Init(params)) {
+ ERROR("Could not init MikMod: %s\n",
+ MikMod_strerror(MikMod_errno));
+ mod_mikModInitError = 1;
+ return -1;
+ }
+
+ return 0;
+}
+
+static void mod_finishMikMod(void)
+{
+ MikMod_Exit();
+}
+
+typedef struct _mod_Data {
+ MODULE *moduleHandle;
+ SBYTE *audio_buffer;
+} mod_Data;
+
+static mod_Data *mod_open(char *path)
+{
+ MODULE *moduleHandle;
+ mod_Data *data;
+
+ if (!(moduleHandle = Player_Load(path, 128, 0)))
+ return NULL;
+
+ /* Prevent module from looping forever */
+ moduleHandle->loop = 0;
+
+ data = xmalloc(sizeof(mod_Data));
+
+ data->audio_buffer = xmalloc(MIKMOD_FRAME_SIZE);
+ data->moduleHandle = moduleHandle;
+
+ Player_Start(data->moduleHandle);
+
+ return data;
+}
+
+static void mod_close(mod_Data * data)
+{
+ Player_Stop();
+ Player_Free(data->moduleHandle);
+ free(data->audio_buffer);
+ free(data);
+}
+
+static int mod_decode(struct decoder * decoder, char *path)
+{
+ mod_Data *data;
+ struct audio_format audio_format;
+ float total_time = 0.0;
+ int ret;
+ float secPerByte;
+
+ if (mod_initMikMod() < 0)
+ return -1;
+
+ if (!(data = mod_open(path))) {
+ ERROR("failed to open mod: %s\n", path);
+ MikMod_Exit();
+ return -1;
+ }
+
+ audio_format.bits = 16;
+ audio_format.sample_rate = 44100;
+ audio_format.channels = 2;
+
+ secPerByte =
+ 1.0 / ((audio_format.bits * audio_format.channels / 8.0) *
+ (float)audio_format.sample_rate);
+
+ decoder_initialized(decoder, &audio_format, 0);
+
+ while (1) {
+ if (decoder_get_command(decoder) == DECODE_COMMAND_SEEK) {
+ decoder_seek_error(decoder);
+ }
+
+ if (decoder_get_command(decoder) == DECODE_COMMAND_STOP)
+ break;
+
+ if (!Player_Active())
+ break;
+
+ ret = VC_WriteBytes(data->audio_buffer, MIKMOD_FRAME_SIZE);
+ total_time += ret * secPerByte;
+ decoder_data(decoder, NULL, 0,
+ (char *)data->audio_buffer, ret,
+ total_time, 0, NULL);
+ }
+
+ decoder_flush(decoder);
+
+ mod_close(data);
+
+ MikMod_Exit();
+
+ return 0;
+}
+
+static struct tag *modTagDup(char *file)
+{
+ struct tag *ret = NULL;
+ MODULE *moduleHandle;
+ char *title;
+
+ if (mod_initMikMod() < 0) {
+ DEBUG("modTagDup: Failed to initialize MikMod\n");
+ return NULL;
+ }
+
+ if (!(moduleHandle = Player_Load(file, 128, 0))) {
+ DEBUG("modTagDup: Failed to open file: %s\n", file);
+ MikMod_Exit();
+ return NULL;
+
+ }
+ Player_Free(moduleHandle);
+
+ ret = tag_new();
+
+ ret->time = 0;
+ title = xstrdup(Player_LoadTitle(file));
+ if (title)
+ tag_add_item(ret, TAG_ITEM_TITLE, title);
+
+ MikMod_Exit();
+
+ return ret;
+}
+
+static const char *modSuffixes[] = { "amf",
+ "dsm",
+ "far",
+ "gdm",
+ "imf",
+ "it",
+ "med",
+ "mod",
+ "mtm",
+ "s3m",
+ "stm",
+ "stx",
+ "ult",
+ "uni",
+ "xm",
+ NULL
+};
+
+struct decoder_plugin modPlugin = {
+ .name = "mod",
+ .finish = mod_finishMikMod,
+ .file_decode = mod_decode,
+ .tag_dup = modTagDup,
+ .stream_types = INPUT_PLUGIN_STREAM_FILE,
+ .suffixes = modSuffixes,
+};
diff --git a/src/decoder/mp3_plugin.c b/src/decoder/mp3_plugin.c
new file mode 100644
index 000000000..a0de30ba7
--- /dev/null
+++ b/src/decoder/mp3_plugin.c
@@ -0,0 +1,1086 @@
+/* the Music Player Daemon (MPD)
+ * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
+ * This project's homepage is: 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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include "../decoder_api.h"
+#include "../log.h"
+#include "../utils.h"
+#include "../conf.h"
+
+#include <mad.h>
+
+#ifdef HAVE_ID3TAG
+#include <id3tag.h>
+#endif
+
+#define FRAMES_CUSHION 2000
+
+#define READ_BUFFER_SIZE 40960
+
+enum mp3_action {
+ DECODE_SKIP = -3,
+ DECODE_BREAK = -2,
+ DECODE_CONT = -1,
+ DECODE_OK = 0
+};
+
+enum muteframe {
+ MUTEFRAME_NONE,
+ MUTEFRAME_SKIP,
+ MUTEFRAME_SEEK
+};
+
+/* the number of samples of silence the decoder inserts at start */
+#define DECODERDELAY 529
+
+#define DEFAULT_GAPLESS_MP3_PLAYBACK 1
+
+static int gaplessPlaybackEnabled;
+
+static inline int32_t
+mad_fixed_to_24_sample(mad_fixed_t sample)
+{
+ enum {
+ bits = 24,
+ MIN = -MAD_F_ONE,
+ MAX = MAD_F_ONE - 1
+ };
+
+ /* round */
+ sample = sample + (1L << (MAD_F_FRACBITS - bits));
+
+ /* clip */
+ if (sample > MAX)
+ sample = MAX;
+ else if (sample < MIN)
+ sample = MIN;
+
+ /* quantize */
+ return sample >> (MAD_F_FRACBITS + 1 - bits);
+}
+
+static void
+mad_fixed_to_24_buffer(int32_t *dest, const struct mad_synth *synth,
+ unsigned int start, unsigned int end,
+ unsigned int num_channels)
+{
+ unsigned int i, c;
+
+ for (i = start; i < end; ++i) {
+ for (c = 0; c < num_channels; ++c)
+ *dest++ = mad_fixed_to_24_sample(synth->pcm.samples[c][i]);
+ }
+}
+
+/* end of stolen stuff from mpg321 */
+
+static int mp3_plugin_init(void)
+{
+ gaplessPlaybackEnabled = getBoolConfigParam(CONF_GAPLESS_MP3_PLAYBACK,
+ 1);
+ if (gaplessPlaybackEnabled == CONF_BOOL_UNSET)
+ gaplessPlaybackEnabled = DEFAULT_GAPLESS_MP3_PLAYBACK;
+ return 1;
+}
+
+/* decoder stuff is based on madlld */
+
+#define MP3_DATA_OUTPUT_BUFFER_SIZE 2048
+
+typedef struct _mp3DecodeData {
+ struct mad_stream stream;
+ struct mad_frame frame;
+ struct mad_synth synth;
+ mad_timer_t timer;
+ unsigned char readBuffer[READ_BUFFER_SIZE];
+ int32_t outputBuffer[MP3_DATA_OUTPUT_BUFFER_SIZE];
+ float totalTime;
+ float elapsedTime;
+ enum muteframe muteFrame;
+ long *frameOffset;
+ mad_timer_t *times;
+ unsigned long highestFrame;
+ unsigned long maxFrames;
+ unsigned long currentFrame;
+ unsigned int dropFramesAtStart;
+ unsigned int dropFramesAtEnd;
+ unsigned int dropSamplesAtStart;
+ unsigned int dropSamplesAtEnd;
+ int foundXing;
+ int foundFirstFrame;
+ int decodedFirstFrame;
+ unsigned long bitRate;
+ struct decoder *decoder;
+ InputStream *inStream;
+ enum mad_layer layer;
+} mp3DecodeData;
+
+static void initMp3DecodeData(mp3DecodeData * data, struct decoder *decoder,
+ InputStream * inStream)
+{
+ data->muteFrame = MUTEFRAME_NONE;
+ data->highestFrame = 0;
+ data->maxFrames = 0;
+ data->frameOffset = NULL;
+ data->times = NULL;
+ data->currentFrame = 0;
+ data->dropFramesAtStart = 0;
+ data->dropFramesAtEnd = 0;
+ data->dropSamplesAtStart = 0;
+ data->dropSamplesAtEnd = 0;
+ data->foundXing = 0;
+ data->foundFirstFrame = 0;
+ data->decodedFirstFrame = 0;
+ data->decoder = decoder;
+ data->inStream = inStream;
+ data->layer = 0;
+
+ mad_stream_init(&data->stream);
+ mad_stream_options(&data->stream, MAD_OPTION_IGNORECRC);
+ mad_frame_init(&data->frame);
+ mad_synth_init(&data->synth);
+ mad_timer_reset(&data->timer);
+}
+
+static int seekMp3InputBuffer(mp3DecodeData * data, long offset)
+{
+ if (seekInputStream(data->inStream, offset, SEEK_SET) < 0) {
+ return -1;
+ }
+
+ mad_stream_buffer(&data->stream, data->readBuffer, 0);
+ (data->stream).error = 0;
+
+ return 0;
+}
+
+static int fillMp3InputBuffer(mp3DecodeData * data)
+{
+ size_t readSize;
+ size_t remaining;
+ size_t readed;
+ unsigned char *readStart;
+
+ if ((data->stream).next_frame != NULL) {
+ remaining = (data->stream).bufend - (data->stream).next_frame;
+ memmove(data->readBuffer, (data->stream).next_frame, remaining);
+ readStart = (data->readBuffer) + remaining;
+ readSize = READ_BUFFER_SIZE - remaining;
+ } else {
+ readSize = READ_BUFFER_SIZE;
+ readStart = data->readBuffer, remaining = 0;
+ }
+
+ /* we've exhausted the read buffer, so give up!, these potential
+ * mp3 frames are way too big, and thus unlikely to be mp3 frames */
+ if (readSize == 0)
+ return -1;
+
+ readed = decoder_read(data->decoder, data->inStream,
+ readStart, readSize);
+ if (readed == 0)
+ return -1;
+
+ mad_stream_buffer(&data->stream, data->readBuffer, readed + remaining);
+ (data->stream).error = 0;
+
+ return 0;
+}
+
+#ifdef HAVE_ID3TAG
+static ReplayGainInfo *parseId3ReplayGainInfo(struct id3_tag *tag)
+{
+ int i;
+ char *key;
+ char *value;
+ struct id3_frame *frame;
+ int found = 0;
+ ReplayGainInfo *replayGainInfo;
+
+ replayGainInfo = newReplayGainInfo();
+
+ for (i = 0; (frame = id3_tag_findframe(tag, "TXXX", i)); i++) {
+ if (frame->nfields < 3)
+ continue;
+
+ key = (char *)
+ id3_ucs4_latin1duplicate(id3_field_getstring
+ (&frame->fields[1]));
+ value = (char *)
+ id3_ucs4_latin1duplicate(id3_field_getstring
+ (&frame->fields[2]));
+
+ if (strcasecmp(key, "replaygain_track_gain") == 0) {
+ replayGainInfo->trackGain = atof(value);
+ found = 1;
+ } else if (strcasecmp(key, "replaygain_album_gain") == 0) {
+ replayGainInfo->albumGain = atof(value);
+ found = 1;
+ } else if (strcasecmp(key, "replaygain_track_peak") == 0) {
+ replayGainInfo->trackPeak = atof(value);
+ found = 1;
+ } else if (strcasecmp(key, "replaygain_album_peak") == 0) {
+ replayGainInfo->albumPeak = atof(value);
+ found = 1;
+ }
+
+ free(key);
+ free(value);
+ }
+
+ if (found)
+ return replayGainInfo;
+ freeReplayGainInfo(replayGainInfo);
+ return NULL;
+}
+#endif
+
+#ifdef HAVE_ID3TAG
+static void mp3_parseId3Tag(mp3DecodeData * data, size_t tagsize,
+ struct tag ** mpdTag, ReplayGainInfo ** replayGainInfo)
+{
+ struct id3_tag *id3Tag = NULL;
+ id3_length_t count;
+ id3_byte_t const *id3_data;
+ id3_byte_t *allocated = NULL;
+ struct tag *tmpMpdTag;
+ ReplayGainInfo *tmpReplayGainInfo;
+
+ count = data->stream.bufend - data->stream.this_frame;
+
+ if (tagsize <= count) {
+ id3_data = data->stream.this_frame;
+ mad_stream_skip(&(data->stream), tagsize);
+ } else {
+ allocated = xmalloc(tagsize);
+ if (!allocated)
+ goto fail;
+
+ memcpy(allocated, data->stream.this_frame, count);
+ mad_stream_skip(&(data->stream), count);
+
+ while (count < tagsize) {
+ size_t len;
+
+ len = decoder_read(data->decoder, data->inStream,
+ allocated + count, tagsize - count);
+ if (len == 0)
+ break;
+ else
+ count += len;
+ }
+
+ if (count != tagsize) {
+ DEBUG("mp3_decode: error parsing ID3 tag\n");
+ goto fail;
+ }
+
+ id3_data = allocated;
+ }
+
+ id3Tag = id3_tag_parse(id3_data, tagsize);
+ if (!id3Tag)
+ goto fail;
+
+ if (mpdTag) {
+ tmpMpdTag = tag_id3_import(id3Tag);
+ if (tmpMpdTag) {
+ if (*mpdTag)
+ tag_free(*mpdTag);
+ *mpdTag = tmpMpdTag;
+ }
+ }
+
+ if (replayGainInfo) {
+ tmpReplayGainInfo = parseId3ReplayGainInfo(id3Tag);
+ if (tmpReplayGainInfo) {
+ if (*replayGainInfo)
+ freeReplayGainInfo(*replayGainInfo);
+ *replayGainInfo = tmpReplayGainInfo;
+ }
+ }
+
+ id3_tag_delete(id3Tag);
+fail:
+ if (allocated)
+ free(allocated);
+}
+#endif
+
+static enum mp3_action
+decodeNextFrameHeader(mp3DecodeData * data, struct tag ** tag,
+ ReplayGainInfo ** replayGainInfo)
+{
+ enum mad_layer layer;
+
+ if ((data->stream).buffer == NULL
+ || (data->stream).error == MAD_ERROR_BUFLEN) {
+ if (fillMp3InputBuffer(data) < 0) {
+ return DECODE_BREAK;
+ }
+ }
+ if (mad_header_decode(&data->frame.header, &data->stream)) {
+#ifdef HAVE_ID3TAG
+ if ((data->stream).error == MAD_ERROR_LOSTSYNC &&
+ (data->stream).this_frame) {
+ signed long tagsize = id3_tag_query((data->stream).
+ this_frame,
+ (data->stream).
+ bufend -
+ (data->stream).
+ this_frame);
+
+ if (tagsize > 0) {
+ if (tag && !(*tag)) {
+ mp3_parseId3Tag(data, (size_t)tagsize,
+ tag, replayGainInfo);
+ } else {
+ mad_stream_skip(&(data->stream),
+ tagsize);
+ }
+ return DECODE_CONT;
+ }
+ }
+#endif
+ if (MAD_RECOVERABLE((data->stream).error)) {
+ return DECODE_SKIP;
+ } else {
+ if ((data->stream).error == MAD_ERROR_BUFLEN)
+ return DECODE_CONT;
+ else {
+ ERROR("unrecoverable frame level error "
+ "(%s).\n",
+ mad_stream_errorstr(&data->stream));
+ return DECODE_BREAK;
+ }
+ }
+ }
+
+ layer = data->frame.header.layer;
+ if (!data->layer) {
+ if (layer != MAD_LAYER_II && layer != MAD_LAYER_III) {
+ /* Only layer 2 and 3 have been tested to work */
+ return DECODE_SKIP;
+ }
+ data->layer = layer;
+ } else if (layer != data->layer) {
+ /* Don't decode frames with a different layer than the first */
+ return DECODE_SKIP;
+ }
+
+ return DECODE_OK;
+}
+
+static enum mp3_action
+decodeNextFrame(mp3DecodeData * data)
+{
+ if ((data->stream).buffer == NULL
+ || (data->stream).error == MAD_ERROR_BUFLEN) {
+ if (fillMp3InputBuffer(data) < 0) {
+ return DECODE_BREAK;
+ }
+ }
+ if (mad_frame_decode(&data->frame, &data->stream)) {
+#ifdef HAVE_ID3TAG
+ if ((data->stream).error == MAD_ERROR_LOSTSYNC) {
+ signed long tagsize = id3_tag_query((data->stream).
+ this_frame,
+ (data->stream).
+ bufend -
+ (data->stream).
+ this_frame);
+ if (tagsize > 0) {
+ mad_stream_skip(&(data->stream), tagsize);
+ return DECODE_CONT;
+ }
+ }
+#endif
+ if (MAD_RECOVERABLE((data->stream).error)) {
+ return DECODE_SKIP;
+ } else {
+ if ((data->stream).error == MAD_ERROR_BUFLEN)
+ return DECODE_CONT;
+ else {
+ ERROR("unrecoverable frame level error "
+ "(%s).\n",
+ mad_stream_errorstr(&data->stream));
+ return DECODE_BREAK;
+ }
+ }
+ }
+
+ return DECODE_OK;
+}
+
+/* xing stuff stolen from alsaplayer, and heavily modified by jat */
+#define XI_MAGIC (('X' << 8) | 'i')
+#define NG_MAGIC (('n' << 8) | 'g')
+#define IN_MAGIC (('I' << 8) | 'n')
+#define FO_MAGIC (('f' << 8) | 'o')
+
+enum xing_magic {
+ XING_MAGIC_XING, /* VBR */
+ XING_MAGIC_INFO /* CBR */
+};
+
+struct xing {
+ long flags; /* valid fields (see below) */
+ unsigned long frames; /* total number of frames */
+ unsigned long bytes; /* total number of bytes */
+ unsigned char toc[100]; /* 100-point seek table */
+ long scale; /* VBR quality */
+ enum xing_magic magic; /* header magic */
+};
+
+enum {
+ XING_FRAMES = 0x00000001L,
+ XING_BYTES = 0x00000002L,
+ XING_TOC = 0x00000004L,
+ XING_SCALE = 0x00000008L
+};
+
+struct version {
+ unsigned major;
+ unsigned minor;
+};
+
+struct lame {
+ char encoder[10]; /* 9 byte encoder name/version ("LAME3.97b") */
+ struct version version; /* struct containing just the version */
+ float peak; /* replaygain peak */
+ float trackGain; /* replaygain track gain */
+ float albumGain; /* replaygain album gain */
+ int encoderDelay; /* # of added samples at start of mp3 */
+ int encoderPadding; /* # of added samples at end of mp3 */
+ int crc; /* CRC of the first 190 bytes of this frame */
+};
+
+static int parse_xing(struct xing *xing, struct mad_bitptr *ptr, int *oldbitlen)
+{
+ unsigned long bits;
+ int bitlen;
+ int bitsleft;
+ int i;
+
+ bitlen = *oldbitlen;
+
+ if (bitlen < 16) goto fail;
+ bits = mad_bit_read(ptr, 16);
+ bitlen -= 16;
+
+ if (bits == XI_MAGIC) {
+ if (bitlen < 16) goto fail;
+ if (mad_bit_read(ptr, 16) != NG_MAGIC) goto fail;
+ bitlen -= 16;
+ xing->magic = XING_MAGIC_XING;
+ } else if (bits == IN_MAGIC) {
+ if (bitlen < 16) goto fail;
+ if (mad_bit_read(ptr, 16) != FO_MAGIC) goto fail;
+ bitlen -= 16;
+ xing->magic = XING_MAGIC_INFO;
+ }
+ else if (bits == NG_MAGIC) xing->magic = XING_MAGIC_XING;
+ else if (bits == FO_MAGIC) xing->magic = XING_MAGIC_INFO;
+ else goto fail;
+
+ if (bitlen < 32) goto fail;
+ xing->flags = mad_bit_read(ptr, 32);
+ bitlen -= 32;
+
+ if (xing->flags & XING_FRAMES) {
+ if (bitlen < 32) goto fail;
+ xing->frames = mad_bit_read(ptr, 32);
+ bitlen -= 32;
+ }
+
+ if (xing->flags & XING_BYTES) {
+ if (bitlen < 32) goto fail;
+ xing->bytes = mad_bit_read(ptr, 32);
+ bitlen -= 32;
+ }
+
+ if (xing->flags & XING_TOC) {
+ if (bitlen < 800) goto fail;
+ for (i = 0; i < 100; ++i) xing->toc[i] = mad_bit_read(ptr, 8);
+ bitlen -= 800;
+ }
+
+ if (xing->flags & XING_SCALE) {
+ if (bitlen < 32) goto fail;
+ xing->scale = mad_bit_read(ptr, 32);
+ bitlen -= 32;
+ }
+
+ /* Make sure we consume no less than 120 bytes (960 bits) in hopes that
+ * the LAME tag is found there, and not right after the Xing header */
+ bitsleft = 960 - ((*oldbitlen) - bitlen);
+ if (bitsleft < 0) goto fail;
+ else if (bitsleft > 0) {
+ mad_bit_read(ptr, bitsleft);
+ bitlen -= bitsleft;
+ }
+
+ *oldbitlen = bitlen;
+
+ return 1;
+fail:
+ xing->flags = 0;
+ return 0;
+}
+
+static int parse_lame(struct lame *lame, struct mad_bitptr *ptr, int *bitlen)
+{
+ int adj = 0;
+ int name;
+ int orig;
+ int sign;
+ int gain;
+ int i;
+
+ /* Unlike the xing header, the lame tag has a fixed length. Fail if
+ * not all 36 bytes (288 bits) are there. */
+ if (*bitlen < 288)
+ return 0;
+
+ for (i = 0; i < 9; i++)
+ lame->encoder[i] = (char)mad_bit_read(ptr, 8);
+ lame->encoder[9] = '\0';
+
+ *bitlen -= 72;
+
+ /* This is technically incorrect, since the encoder might not be lame.
+ * But there's no other way to determine if this is a lame tag, and we
+ * wouldn't want to go reading a tag that's not there. */
+ if (prefixcmp(lame->encoder, "LAME"))
+ return 0;
+
+ if (sscanf(lame->encoder+4, "%u.%u",
+ &lame->version.major, &lame->version.minor) != 2)
+ return 0;
+
+ DEBUG("detected LAME version %i.%i (\"%s\")\n",
+ lame->version.major, lame->version.minor, lame->encoder);
+
+ /* The reference volume was changed from the 83dB used in the
+ * ReplayGain spec to 89dB in lame 3.95.1. Bump the gain for older
+ * versions, since everyone else uses 89dB instead of 83dB.
+ * Unfortunately, lame didn't differentiate between 3.95 and 3.95.1, so
+ * it's impossible to make the proper adjustment for 3.95.
+ * Fortunately, 3.95 was only out for about a day before 3.95.1 was
+ * released. -- tmz */
+ if (lame->version.major < 3 ||
+ (lame->version.major == 3 && lame->version.minor < 95))
+ adj = 6;
+
+ mad_bit_read(ptr, 16);
+
+ lame->peak = mad_f_todouble(mad_bit_read(ptr, 32) << 5); /* peak */
+ DEBUG("LAME peak found: %f\n", lame->peak);
+
+ lame->trackGain = 0;
+ name = mad_bit_read(ptr, 3); /* gain name */
+ orig = mad_bit_read(ptr, 3); /* gain originator */
+ sign = mad_bit_read(ptr, 1); /* sign bit */
+ gain = mad_bit_read(ptr, 9); /* gain*10 */
+ if (gain && name == 1 && orig != 0) {
+ lame->trackGain = ((sign ? -gain : gain) / 10.0) + adj;
+ DEBUG("LAME track gain found: %f\n", lame->trackGain);
+ }
+
+ /* tmz reports that this isn't currently written by any version of lame
+ * (as of 3.97). Since we have no way of testing it, don't use it.
+ * Wouldn't want to go blowing someone's ears just because we read it
+ * wrong. :P -- jat */
+ lame->albumGain = 0;
+#if 0
+ name = mad_bit_read(ptr, 3); /* gain name */
+ orig = mad_bit_read(ptr, 3); /* gain originator */
+ sign = mad_bit_read(ptr, 1); /* sign bit */
+ gain = mad_bit_read(ptr, 9); /* gain*10 */
+ if (gain && name == 2 && orig != 0) {
+ lame->albumGain = ((sign ? -gain : gain) / 10.0) + adj;
+ DEBUG("LAME album gain found: %f\n", lame->trackGain);
+ }
+#else
+ mad_bit_read(ptr, 16);
+#endif
+
+ mad_bit_read(ptr, 16);
+
+ lame->encoderDelay = mad_bit_read(ptr, 12);
+ lame->encoderPadding = mad_bit_read(ptr, 12);
+
+ DEBUG("encoder delay is %i, encoder padding is %i\n",
+ lame->encoderDelay, lame->encoderPadding);
+
+ mad_bit_read(ptr, 80);
+
+ lame->crc = mad_bit_read(ptr, 16);
+
+ *bitlen -= 216;
+
+ return 1;
+}
+
+static int decodeFirstFrame(mp3DecodeData * data,
+ struct tag ** tag, ReplayGainInfo ** replayGainInfo)
+{
+ struct decoder *decoder = data->decoder;
+ struct xing xing;
+ struct lame lame;
+ struct mad_bitptr ptr;
+ int bitlen;
+ int ret;
+
+ /* stfu gcc */
+ memset(&xing, 0, sizeof(struct xing));
+ xing.flags = 0;
+
+ while (1) {
+ while ((ret = decodeNextFrameHeader(data, tag, replayGainInfo)) == DECODE_CONT &&
+ (!decoder || decoder_get_command(decoder) == DECODE_COMMAND_NONE));
+ if (ret == DECODE_BREAK ||
+ (decoder && decoder_get_command(decoder) != DECODE_COMMAND_NONE))
+ return -1;
+ if (ret == DECODE_SKIP) continue;
+
+ while ((ret = decodeNextFrame(data)) == DECODE_CONT &&
+ (!decoder || decoder_get_command(decoder) == DECODE_COMMAND_NONE));
+ if (ret == DECODE_BREAK ||
+ (decoder && decoder_get_command(decoder) != DECODE_COMMAND_NONE))
+ return -1;
+ if (ret == DECODE_OK) break;
+ }
+
+ ptr = data->stream.anc_ptr;
+ bitlen = data->stream.anc_bitlen;
+
+ /*
+ * Attempt to calulcate the length of the song from filesize
+ */
+ {
+ size_t offset = data->inStream->offset;
+ mad_timer_t duration = data->frame.header.duration;
+ float frameTime = ((float)mad_timer_count(duration,
+ MAD_UNITS_MILLISECONDS)) / 1000;
+
+ if (data->stream.this_frame != NULL)
+ offset -= data->stream.bufend - data->stream.this_frame;
+ else
+ offset -= data->stream.bufend - data->stream.buffer;
+
+ if (data->inStream->size >= offset) {
+ data->totalTime = ((data->inStream->size - offset) *
+ 8.0) / (data->frame).header.bitrate;
+ data->maxFrames = data->totalTime / frameTime +
+ FRAMES_CUSHION;
+ } else {
+ data->maxFrames = FRAMES_CUSHION;
+ data->totalTime = 0;
+ }
+ }
+ /*
+ * if an xing tag exists, use that!
+ */
+ if (parse_xing(&xing, &ptr, &bitlen)) {
+ data->foundXing = 1;
+ data->muteFrame = MUTEFRAME_SKIP;
+
+ if ((xing.flags & XING_FRAMES) && xing.frames) {
+ mad_timer_t duration = data->frame.header.duration;
+ mad_timer_multiply(&duration, xing.frames);
+ data->totalTime = ((float)mad_timer_count(duration, MAD_UNITS_MILLISECONDS)) / 1000;
+ data->maxFrames = xing.frames;
+ }
+
+ if (parse_lame(&lame, &ptr, &bitlen)) {
+ if (gaplessPlaybackEnabled &&
+ data->inStream->seekable) {
+ data->dropSamplesAtStart = lame.encoderDelay +
+ DECODERDELAY;
+ data->dropSamplesAtEnd = lame.encoderPadding;
+ }
+
+ /* Album gain isn't currently used. See comment in
+ * parse_lame() for details. -- jat */
+ if (replayGainInfo && !*replayGainInfo &&
+ lame.trackGain) {
+ *replayGainInfo = newReplayGainInfo();
+ (*replayGainInfo)->trackGain = lame.trackGain;
+ (*replayGainInfo)->trackPeak = lame.peak;
+ }
+ }
+ }
+
+ if (!data->maxFrames) return -1;
+
+ if (data->maxFrames > 8 * 1024 * 1024) {
+ ERROR("mp3 file header indicates too many frames: %lu",
+ data->maxFrames);
+ return -1;
+ }
+
+ data->frameOffset = xmalloc(sizeof(long) * data->maxFrames);
+ data->times = xmalloc(sizeof(mad_timer_t) * data->maxFrames);
+
+ return 0;
+}
+
+static void mp3DecodeDataFinalize(mp3DecodeData * data)
+{
+ mad_synth_finish(&data->synth);
+ mad_frame_finish(&data->frame);
+ mad_stream_finish(&data->stream);
+
+ if (data->frameOffset) free(data->frameOffset);
+ if (data->times) free(data->times);
+}
+
+/* this is primarily used for getting total time for tags */
+static int getMp3TotalTime(char *file)
+{
+ InputStream inStream;
+ mp3DecodeData data;
+ int ret;
+
+ if (openInputStream(&inStream, file) < 0)
+ return -1;
+ initMp3DecodeData(&data, NULL, &inStream);
+ if (decodeFirstFrame(&data, NULL, NULL) < 0)
+ ret = -1;
+ else
+ ret = data.totalTime + 0.5;
+ mp3DecodeDataFinalize(&data);
+ closeInputStream(&inStream);
+
+ return ret;
+}
+
+static int openMp3FromInputStream(InputStream * inStream, mp3DecodeData * data,
+ struct decoder * decoder, struct tag ** tag,
+ ReplayGainInfo ** replayGainInfo)
+{
+ initMp3DecodeData(data, decoder, inStream);
+ *tag = NULL;
+ if (decodeFirstFrame(data, tag, replayGainInfo) < 0) {
+ mp3DecodeDataFinalize(data);
+ if (tag && *tag)
+ tag_free(*tag);
+ return -1;
+ }
+
+ return 0;
+}
+
+static enum mp3_action
+mp3Read(mp3DecodeData * data, ReplayGainInfo ** replayGainInfo)
+{
+ struct decoder *decoder = data->decoder;
+ unsigned int pcm_length, max_samples;
+ unsigned int i;
+ int ret;
+ int skip;
+
+ if (data->currentFrame >= data->highestFrame) {
+ mad_timer_add(&data->timer, (data->frame).header.duration);
+ data->bitRate = (data->frame).header.bitrate;
+ if (data->currentFrame >= data->maxFrames) {
+ data->currentFrame = data->maxFrames - 1;
+ } else {
+ data->highestFrame++;
+ }
+ data->frameOffset[data->currentFrame] = data->inStream->offset;
+ if (data->stream.this_frame != NULL) {
+ data->frameOffset[data->currentFrame] -=
+ data->stream.bufend - data->stream.this_frame;
+ } else {
+ data->frameOffset[data->currentFrame] -=
+ data->stream.bufend - data->stream.buffer;
+ }
+ data->times[data->currentFrame] = data->timer;
+ } else {
+ data->timer = data->times[data->currentFrame];
+ }
+ data->currentFrame++;
+ data->elapsedTime =
+ ((float)mad_timer_count(data->timer, MAD_UNITS_MILLISECONDS)) /
+ 1000;
+
+ switch (data->muteFrame) {
+ case MUTEFRAME_SKIP:
+ data->muteFrame = MUTEFRAME_NONE;
+ break;
+ case MUTEFRAME_SEEK:
+ if (decoder_seek_where(decoder) <= data->elapsedTime) {
+ decoder_clear(decoder);
+ data->muteFrame = MUTEFRAME_NONE;
+ decoder_command_finished(decoder);
+ }
+ break;
+ case MUTEFRAME_NONE:
+ mad_synth_frame(&data->synth, &data->frame);
+
+ if (!data->foundFirstFrame) {
+ unsigned int samplesPerFrame = (data->synth).pcm.length;
+ data->dropFramesAtStart = data->dropSamplesAtStart / samplesPerFrame;
+ data->dropFramesAtEnd = data->dropSamplesAtEnd / samplesPerFrame;
+ data->dropSamplesAtStart = data->dropSamplesAtStart % samplesPerFrame;
+ data->dropSamplesAtEnd = data->dropSamplesAtEnd % samplesPerFrame;
+ data->foundFirstFrame = 1;
+ }
+
+ if (data->dropFramesAtStart > 0) {
+ data->dropFramesAtStart--;
+ break;
+ } else if ((data->dropFramesAtEnd > 0) &&
+ (data->currentFrame == (data->maxFrames + 1 - data->dropFramesAtEnd))) {
+ /* stop decoding, effectively dropping all remaining
+ * frames */
+ return DECODE_BREAK;
+ }
+
+ if (data->inStream->metaTitle) {
+ struct tag *tag = tag_new();
+ if (data->inStream->metaName) {
+ tag_add_item(tag, TAG_ITEM_NAME,
+ data->inStream->metaName);
+ }
+ tag_add_item(tag, TAG_ITEM_TITLE,
+ data->inStream->metaTitle);
+ free(data->inStream->metaTitle);
+ data->inStream->metaTitle = NULL;
+ tag_free(tag);
+ }
+
+ if (!data->decodedFirstFrame) {
+ i = data->dropSamplesAtStart;
+ data->decodedFirstFrame = 1;
+ } else
+ i = 0;
+
+ pcm_length = data->synth.pcm.length;
+ if (data->dropSamplesAtEnd &&
+ (data->currentFrame == data->maxFrames - data->dropFramesAtEnd)) {
+ if (data->dropSamplesAtEnd >= pcm_length)
+ pcm_length = 0;
+ else
+ pcm_length -= data->dropSamplesAtEnd;
+ }
+
+ max_samples = sizeof(data->outputBuffer) /
+ sizeof(data->outputBuffer[0]) /
+ MAD_NCHANNELS(&(data->frame).header);
+
+ while (i < pcm_length) {
+ enum decoder_command cmd;
+ unsigned int num_samples = pcm_length - i;
+ if (num_samples > max_samples)
+ num_samples = max_samples;
+
+ i += num_samples;
+
+ mad_fixed_to_24_buffer(data->outputBuffer,
+ &data->synth,
+ i - num_samples, i,
+ MAD_NCHANNELS(&(data->frame).header));
+ num_samples *= MAD_NCHANNELS(&(data->frame).header);
+
+ cmd = decoder_data(decoder, data->inStream,
+ data->inStream->seekable,
+ data->outputBuffer,
+ sizeof(data->outputBuffer[0]) * num_samples,
+ data->elapsedTime,
+ data->bitRate / 1000,
+ (replayGainInfo != NULL) ? *replayGainInfo : NULL);
+ if (cmd == DECODE_COMMAND_STOP)
+ return DECODE_BREAK;
+ }
+
+ if (data->dropSamplesAtEnd &&
+ (data->currentFrame == data->maxFrames - data->dropFramesAtEnd))
+ /* stop decoding, effectively dropping
+ * all remaining samples */
+ return DECODE_BREAK;
+
+ if (decoder_get_command(decoder) == DECODE_COMMAND_SEEK &&
+ data->inStream->seekable) {
+ unsigned long j = 0;
+ data->muteFrame = MUTEFRAME_SEEK;
+ while (j < data->highestFrame &&
+ decoder_seek_where(decoder) >
+ ((float)mad_timer_count(data->times[j],
+ MAD_UNITS_MILLISECONDS))
+ / 1000) {
+ j++;
+ }
+ if (j < data->highestFrame) {
+ if (seekMp3InputBuffer(data,
+ data->frameOffset[j]) ==
+ 0) {
+ decoder_clear(decoder);
+ data->currentFrame = j;
+ decoder_command_finished(decoder);
+ } else
+ decoder_seek_error(decoder);
+ data->muteFrame = MUTEFRAME_NONE;
+ }
+ } else if (decoder_get_command(decoder) == DECODE_COMMAND_SEEK &&
+ !data->inStream->seekable) {
+ decoder_seek_error(decoder);
+ }
+ }
+
+ while (1) {
+ skip = 0;
+ while ((ret =
+ decodeNextFrameHeader(data, NULL,
+ replayGainInfo)) == DECODE_CONT
+ && decoder_get_command(decoder) == DECODE_COMMAND_NONE) ;
+ if (ret == DECODE_BREAK || decoder_get_command(decoder) != DECODE_COMMAND_NONE)
+ break;
+ else if (ret == DECODE_SKIP)
+ skip = 1;
+ if (data->muteFrame == MUTEFRAME_NONE) {
+ while ((ret = decodeNextFrame(data)) == DECODE_CONT &&
+ decoder_get_command(decoder) == DECODE_COMMAND_NONE) ;
+ if (ret == DECODE_BREAK ||
+ decoder_get_command(decoder) != DECODE_COMMAND_NONE)
+ break;
+ }
+ if (!skip && ret == DECODE_OK)
+ break;
+ }
+
+ switch (decoder_get_command(decoder)) {
+ case DECODE_COMMAND_NONE:
+ case DECODE_COMMAND_START:
+ break;
+
+ case DECODE_COMMAND_STOP:
+ return DECODE_BREAK;
+
+ case DECODE_COMMAND_SEEK:
+ return DECODE_CONT;
+ }
+
+ return ret;
+}
+
+static void initAudioFormatFromMp3DecodeData(mp3DecodeData * data,
+ struct audio_format * af)
+{
+ af->bits = 24;
+ af->sample_rate = (data->frame).header.samplerate;
+ af->channels = MAD_NCHANNELS(&(data->frame).header);
+}
+
+static int mp3_decode(struct decoder * decoder, InputStream * inStream)
+{
+ mp3DecodeData data;
+ struct tag *tag = NULL;
+ ReplayGainInfo *replayGainInfo = NULL;
+ struct audio_format audio_format;
+
+ if (openMp3FromInputStream(inStream, &data, decoder,
+ &tag, &replayGainInfo) < 0) {
+ if (decoder_get_command(decoder) == DECODE_COMMAND_NONE) {
+ ERROR
+ ("Input does not appear to be a mp3 bit stream.\n");
+ return -1;
+ }
+ return 0;
+ }
+
+ initAudioFormatFromMp3DecodeData(&data, &audio_format);
+
+ if (inStream->metaTitle) {
+ if (tag)
+ tag_free(tag);
+ tag = tag_new();
+ tag_add_item(tag, TAG_ITEM_TITLE, inStream->metaTitle);
+ free(inStream->metaTitle);
+ inStream->metaTitle = NULL;
+ if (inStream->metaName) {
+ tag_add_item(tag, TAG_ITEM_NAME, inStream->metaName);
+ }
+ tag_free(tag);
+ } else if (tag) {
+ if (inStream->metaName) {
+ tag_clear_items_by_type(tag, TAG_ITEM_NAME);
+ tag_add_item(tag, TAG_ITEM_NAME, inStream->metaName);
+ }
+ tag_free(tag);
+ } else if (inStream->metaName) {
+ tag = tag_new();
+ if (inStream->metaName) {
+ tag_add_item(tag, TAG_ITEM_NAME, inStream->metaName);
+ }
+ tag_free(tag);
+ }
+
+ decoder_initialized(decoder, &audio_format, data.totalTime);
+
+ while (mp3Read(&data, &replayGainInfo) != DECODE_BREAK) ;
+
+ if (replayGainInfo)
+ freeReplayGainInfo(replayGainInfo);
+
+ if (decoder_get_command(decoder) == DECODE_COMMAND_SEEK &&
+ data.muteFrame == MUTEFRAME_SEEK) {
+ decoder_clear(decoder);
+ decoder_command_finished(decoder);
+ }
+
+ decoder_flush(decoder);
+ mp3DecodeDataFinalize(&data);
+
+ return 0;
+}
+
+static struct tag *mp3_tagDup(char *file)
+{
+ struct tag *ret = NULL;
+ int total_time;
+
+ ret = tag_id3_load(file);
+
+ total_time = getMp3TotalTime(file);
+
+ if (total_time >= 0) {
+ if (!ret)
+ ret = tag_new();
+ ret->time = total_time;
+ } else {
+ DEBUG("mp3_tagDup: Failed to get total song time from: %s\n",
+ file);
+ }
+
+ return ret;
+}
+
+static const char *mp3_suffixes[] = { "mp3", "mp2", NULL };
+static const char *mp3_mimeTypes[] = { "audio/mpeg", NULL };
+
+struct decoder_plugin mp3Plugin = {
+ .name = "mp3",
+ .init = mp3_plugin_init,
+ .stream_decode = mp3_decode,
+ .tag_dup = mp3_tagDup,
+ .stream_types = INPUT_PLUGIN_STREAM_FILE | INPUT_PLUGIN_STREAM_URL,
+ .suffixes = mp3_suffixes,
+ .mime_types = mp3_mimeTypes
+};
diff --git a/src/decoder/mp4_plugin.c b/src/decoder/mp4_plugin.c
new file mode 100644
index 000000000..4a613744e
--- /dev/null
+++ b/src/decoder/mp4_plugin.c
@@ -0,0 +1,423 @@
+/* the Music Player Daemon (MPD)
+ * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
+ * This project's homepage is: 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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include "../decoder_api.h"
+#include "../utils.h"
+#include "../log.h"
+
+#include "mp4ff.h"
+
+#include <limits.h>
+#include <faad.h>
+/* all code here is either based on or copied from FAAD2's frontend code */
+
+static int mp4_getAACTrack(mp4ff_t * infile)
+{
+ /* find AAC track */
+ int i, rc;
+ int numTracks = mp4ff_total_tracks(infile);
+
+ for (i = 0; i < numTracks; i++) {
+ unsigned char *buff = NULL;
+ unsigned int buff_size = 0;
+#ifdef HAVE_MP4AUDIOSPECIFICCONFIG
+ mp4AudioSpecificConfig mp4ASC;
+#else
+ unsigned long dummy1_32;
+ unsigned char dummy2_8, dummy3_8, dummy4_8, dummy5_8, dummy6_8,
+ dummy7_8, dummy8_8;
+#endif
+
+ mp4ff_get_decoder_config(infile, i, &buff, &buff_size);
+
+ if (buff) {
+#ifdef HAVE_MP4AUDIOSPECIFICCONFIG
+ rc = AudioSpecificConfig(buff, buff_size, &mp4ASC);
+#else
+ rc = AudioSpecificConfig(buff, &dummy1_32, &dummy2_8,
+ &dummy3_8, &dummy4_8,
+ &dummy5_8, &dummy6_8,
+ &dummy7_8, &dummy8_8);
+#endif
+ free(buff);
+ if (rc < 0)
+ continue;
+ return i;
+ }
+ }
+
+ /* can't decode this */
+ return -1;
+}
+
+static uint32_t mp4_inputStreamReadCallback(void *inStream, void *buffer,
+ uint32_t length)
+{
+ return readFromInputStream((InputStream *) inStream, buffer, length);
+}
+
+static uint32_t mp4_inputStreamSeekCallback(void *inStream, uint64_t position)
+{
+ return seekInputStream((InputStream *) inStream, position, SEEK_SET);
+}
+
+static int mp4_decode(struct decoder * mpd_decoder, InputStream * inStream)
+{
+ mp4ff_t *mp4fh;
+ mp4ff_callback_t *mp4cb;
+ int32_t track;
+ float file_time, total_time;
+ int32_t scale;
+ faacDecHandle decoder;
+ faacDecFrameInfo frameInfo;
+ faacDecConfigurationPtr config;
+ struct audio_format audio_format;
+ unsigned char *mp4Buffer;
+ unsigned int mp4BufferSize;
+ uint32_t sample_rate;
+ unsigned char channels;
+ long sampleId;
+ long numSamples;
+ long dur;
+ unsigned int sampleCount;
+ char *sampleBuffer;
+ size_t sampleBufferLen;
+ unsigned int initial = 1;
+ float *seekTable;
+ long seekTableEnd = -1;
+ bool seekPositionFound = false;
+ long offset;
+ uint16_t bitRate = 0;
+ bool seeking = false;
+ double seek_where = 0;
+ bool initialized = false;
+
+ mp4cb = xmalloc(sizeof(mp4ff_callback_t));
+ mp4cb->read = mp4_inputStreamReadCallback;
+ mp4cb->seek = mp4_inputStreamSeekCallback;
+ mp4cb->user_data = inStream;
+
+ mp4fh = mp4ff_open_read(mp4cb);
+ if (!mp4fh) {
+ ERROR("Input does not appear to be a mp4 stream.\n");
+ free(mp4cb);
+ return -1;
+ }
+
+ track = mp4_getAACTrack(mp4fh);
+ if (track < 0) {
+ ERROR("No AAC track found in mp4 stream.\n");
+ mp4ff_close(mp4fh);
+ free(mp4cb);
+ return -1;
+ }
+
+ decoder = faacDecOpen();
+
+ config = faacDecGetCurrentConfiguration(decoder);
+ config->outputFormat = FAAD_FMT_16BIT;
+#ifdef HAVE_FAACDECCONFIGURATION_DOWNMATRIX
+ config->downMatrix = 1;
+#endif
+#ifdef HAVE_FAACDECCONFIGURATION_DONTUPSAMPLEIMPLICITSBR
+ config->dontUpSampleImplicitSBR = 0;
+#endif
+ faacDecSetConfiguration(decoder, config);
+
+ audio_format.bits = 16;
+
+ mp4Buffer = NULL;
+ mp4BufferSize = 0;
+ mp4ff_get_decoder_config(mp4fh, track, &mp4Buffer, &mp4BufferSize);
+
+ if (faacDecInit2
+ (decoder, mp4Buffer, mp4BufferSize, &sample_rate, &channels) < 0) {
+ ERROR("Error not a AAC stream.\n");
+ faacDecClose(decoder);
+ mp4ff_close(mp4fh);
+ free(mp4cb);
+ return -1;
+ }
+
+ audio_format.sample_rate = sample_rate;
+ audio_format.channels = channels;
+ file_time = mp4ff_get_track_duration_use_offsets(mp4fh, track);
+ scale = mp4ff_time_scale(mp4fh, track);
+
+ if (mp4Buffer)
+ free(mp4Buffer);
+
+ if (scale < 0) {
+ ERROR("Error getting audio format of mp4 AAC track.\n");
+ faacDecClose(decoder);
+ mp4ff_close(mp4fh);
+ free(mp4cb);
+ return -1;
+ }
+ total_time = ((float)file_time) / scale;
+
+ numSamples = mp4ff_num_samples(mp4fh, track);
+ if (numSamples > (long)(INT_MAX / sizeof(float))) {
+ ERROR("Integer overflow.\n");
+ faacDecClose(decoder);
+ mp4ff_close(mp4fh);
+ free(mp4cb);
+ return -1;
+ }
+
+ file_time = 0.0;
+
+ seekTable = xmalloc(sizeof(float) * numSamples);
+
+ for (sampleId = 0; sampleId < numSamples; sampleId++) {
+ if (decoder_get_command(mpd_decoder) == DECODE_COMMAND_SEEK) {
+ seeking = true;
+ seek_where = decoder_seek_where(mpd_decoder);
+ }
+
+ if (seeking && seekTableEnd > 1 &&
+ seekTable[seekTableEnd] >= seek_where) {
+ int i = 2;
+ while (seekTable[i] < seek_where)
+ i++;
+ sampleId = i - 1;
+ file_time = seekTable[sampleId];
+ }
+
+ dur = mp4ff_get_sample_duration(mp4fh, track, sampleId);
+ offset = mp4ff_get_sample_offset(mp4fh, track, sampleId);
+
+ if (sampleId > seekTableEnd) {
+ seekTable[sampleId] = file_time;
+ seekTableEnd = sampleId;
+ }
+
+ if (sampleId == 0)
+ dur = 0;
+ if (offset > dur)
+ dur = 0;
+ else
+ dur -= offset;
+ file_time += ((float)dur) / scale;
+
+ if (seeking && file_time > seek_where)
+ seekPositionFound = true;
+
+ if (seeking && seekPositionFound) {
+ seekPositionFound = false;
+ decoder_clear(mpd_decoder);
+ seeking = 0;
+ decoder_command_finished(mpd_decoder);
+ }
+
+ if (seeking)
+ continue;
+
+ if (mp4ff_read_sample(mp4fh, track, sampleId, &mp4Buffer,
+ &mp4BufferSize) == 0)
+ break;
+
+#ifdef HAVE_FAAD_BUFLEN_FUNCS
+ sampleBuffer = faacDecDecode(decoder, &frameInfo, mp4Buffer,
+ mp4BufferSize);
+#else
+ sampleBuffer = faacDecDecode(decoder, &frameInfo, mp4Buffer);
+#endif
+
+ if (mp4Buffer)
+ free(mp4Buffer);
+ if (frameInfo.error > 0) {
+ ERROR("faad2 error: %s\n",
+ faacDecGetErrorMessage(frameInfo.error));
+ break;
+ }
+
+ if (!initialized) {
+ channels = frameInfo.channels;
+#ifdef HAVE_FAACDECFRAMEINFO_SAMPLERATE
+ scale = frameInfo.samplerate;
+#endif
+ audio_format.sample_rate = scale;
+ audio_format.channels = frameInfo.channels;
+ decoder_initialized(mpd_decoder, &audio_format,
+ total_time);
+ initialized = true;
+ }
+
+ if (channels * (unsigned long)(dur + offset) > frameInfo.samples) {
+ dur = frameInfo.samples / channels;
+ offset = 0;
+ }
+
+ sampleCount = (unsigned long)(dur * channels);
+
+ if (sampleCount > 0) {
+ initial = 0;
+ bitRate = frameInfo.bytesconsumed * 8.0 *
+ frameInfo.channels * scale /
+ frameInfo.samples / 1000 + 0.5;
+ }
+
+ sampleBufferLen = sampleCount * 2;
+
+ sampleBuffer += offset * channels * 2;
+
+ decoder_data(mpd_decoder, inStream, 1, sampleBuffer,
+ sampleBufferLen, file_time,
+ bitRate, NULL);
+ if (decoder_get_command(mpd_decoder) == DECODE_COMMAND_STOP)
+ break;
+ }
+
+ free(seekTable);
+ faacDecClose(decoder);
+ mp4ff_close(mp4fh);
+ free(mp4cb);
+
+ if (!initialized)
+ return -1;
+
+ if (decoder_get_command(mpd_decoder) == DECODE_COMMAND_SEEK && seeking) {
+ decoder_clear(mpd_decoder);
+ decoder_command_finished(mpd_decoder);
+ }
+ decoder_flush(mpd_decoder);
+
+ return 0;
+}
+
+static struct tag *mp4DataDup(char *file, int *mp4MetadataFound)
+{
+ struct tag *ret = NULL;
+ InputStream inStream;
+ mp4ff_t *mp4fh;
+ mp4ff_callback_t *callback;
+ int32_t track;
+ int32_t file_time;
+ int32_t scale;
+ int i;
+
+ *mp4MetadataFound = 0;
+
+ if (openInputStream(&inStream, file) < 0) {
+ DEBUG("mp4DataDup: Failed to open file: %s\n", file);
+ return NULL;
+ }
+
+ callback = xmalloc(sizeof(mp4ff_callback_t));
+ callback->read = mp4_inputStreamReadCallback;
+ callback->seek = mp4_inputStreamSeekCallback;
+ callback->user_data = &inStream;
+
+ mp4fh = mp4ff_open_read(callback);
+ if (!mp4fh) {
+ free(callback);
+ closeInputStream(&inStream);
+ return NULL;
+ }
+
+ track = mp4_getAACTrack(mp4fh);
+ if (track < 0) {
+ mp4ff_close(mp4fh);
+ closeInputStream(&inStream);
+ free(callback);
+ return NULL;
+ }
+
+ ret = tag_new();
+ file_time = mp4ff_get_track_duration_use_offsets(mp4fh, track);
+ scale = mp4ff_time_scale(mp4fh, track);
+ if (scale < 0) {
+ mp4ff_close(mp4fh);
+ closeInputStream(&inStream);
+ free(callback);
+ tag_free(ret);
+ return NULL;
+ }
+ ret->time = ((float)file_time) / scale + 0.5;
+
+ for (i = 0; i < mp4ff_meta_get_num_items(mp4fh); i++) {
+ char *item;
+ char *value;
+
+ mp4ff_meta_get_by_index(mp4fh, i, &item, &value);
+
+ if (0 == strcasecmp("artist", item)) {
+ tag_add_item(ret, TAG_ITEM_ARTIST, value);
+ *mp4MetadataFound = 1;
+ } else if (0 == strcasecmp("title", item)) {
+ tag_add_item(ret, TAG_ITEM_TITLE, value);
+ *mp4MetadataFound = 1;
+ } else if (0 == strcasecmp("album", item)) {
+ tag_add_item(ret, TAG_ITEM_ALBUM, value);
+ *mp4MetadataFound = 1;
+ } else if (0 == strcasecmp("track", item)) {
+ tag_add_item(ret, TAG_ITEM_TRACK, value);
+ *mp4MetadataFound = 1;
+ } else if (0 == strcasecmp("disc", item)) { /* Is that the correct id? */
+ tag_add_item(ret, TAG_ITEM_DISC, value);
+ *mp4MetadataFound = 1;
+ } else if (0 == strcasecmp("genre", item)) {
+ tag_add_item(ret, TAG_ITEM_GENRE, value);
+ *mp4MetadataFound = 1;
+ } else if (0 == strcasecmp("date", item)) {
+ tag_add_item(ret, TAG_ITEM_DATE, value);
+ *mp4MetadataFound = 1;
+ }
+
+ free(item);
+ free(value);
+ }
+
+ mp4ff_close(mp4fh);
+ closeInputStream(&inStream);
+
+ return ret;
+}
+
+static struct tag *mp4TagDup(char *file)
+{
+ struct tag *ret = NULL;
+ int mp4MetadataFound = 0;
+
+ ret = mp4DataDup(file, &mp4MetadataFound);
+ if (!ret)
+ return NULL;
+ if (!mp4MetadataFound) {
+ struct tag *temp = tag_id3_load(file);
+ if (temp) {
+ temp->time = ret->time;
+ tag_free(ret);
+ ret = temp;
+ }
+ }
+
+ return ret;
+}
+
+static const char *mp4_suffixes[] = { "m4a", "mp4", NULL };
+static const char *mp4_mimeTypes[] = { "audio/mp4", "audio/m4a", NULL };
+
+struct decoder_plugin mp4Plugin = {
+ .name = "mp4",
+ .stream_decode = mp4_decode,
+ .tag_dup = mp4TagDup,
+ .stream_types = INPUT_PLUGIN_STREAM_FILE | INPUT_PLUGIN_STREAM_URL,
+ .suffixes = mp4_suffixes,
+ .mime_types = mp4_mimeTypes,
+};
diff --git a/src/decoder/mpc_plugin.c b/src/decoder/mpc_plugin.c
new file mode 100644
index 000000000..fb1b0b56c
--- /dev/null
+++ b/src/decoder/mpc_plugin.c
@@ -0,0 +1,308 @@
+/* the Music Player Daemon (MPD)
+ * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
+ * This project's homepage is: 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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include "../decoder_api.h"
+#include "../utils.h"
+#include "../log.h"
+
+#include <mpcdec/mpcdec.h>
+
+typedef struct _MpcCallbackData {
+ InputStream *inStream;
+ struct decoder *decoder;
+} MpcCallbackData;
+
+static mpc_int32_t mpc_read_cb(void *vdata, void *ptr, mpc_int32_t size)
+{
+ MpcCallbackData *data = (MpcCallbackData *) vdata;
+
+ return decoder_read(data->decoder, data->inStream, ptr, size);
+}
+
+static mpc_bool_t mpc_seek_cb(void *vdata, mpc_int32_t offset)
+{
+ MpcCallbackData *data = (MpcCallbackData *) vdata;
+
+ return seekInputStream(data->inStream, offset, SEEK_SET) < 0 ? 0 : 1;
+}
+
+static mpc_int32_t mpc_tell_cb(void *vdata)
+{
+ MpcCallbackData *data = (MpcCallbackData *) vdata;
+
+ return (long)(data->inStream->offset);
+}
+
+static mpc_bool_t mpc_canseek_cb(void *vdata)
+{
+ MpcCallbackData *data = (MpcCallbackData *) vdata;
+
+ return data->inStream->seekable;
+}
+
+static mpc_int32_t mpc_getsize_cb(void *vdata)
+{
+ MpcCallbackData *data = (MpcCallbackData *) vdata;
+
+ return data->inStream->size;
+}
+
+/* this _looks_ performance-critical, don't de-inline -- eric */
+static inline int16_t convertSample(MPC_SAMPLE_FORMAT sample)
+{
+ /* only doing 16-bit audio for now */
+ int32_t val;
+
+ const int clip_min = -1 << (16 - 1);
+ const int clip_max = (1 << (16 - 1)) - 1;
+
+#ifdef MPC_FIXED_POINT
+ const int shift = 16 - MPC_FIXED_POINT_SCALE_SHIFT;
+
+ if (sample > 0) {
+ sample <<= shift;
+ } else if (shift < 0) {
+ sample >>= -shift;
+ }
+ val = sample;
+#else
+ const int float_scale = 1 << (16 - 1);
+
+ val = sample * float_scale;
+#endif
+
+ if (val < clip_min)
+ val = clip_min;
+ else if (val > clip_max)
+ val = clip_max;
+
+ return val;
+}
+
+static int mpc_decode(struct decoder * mpd_decoder, InputStream * inStream)
+{
+ mpc_decoder decoder;
+ mpc_reader reader;
+ mpc_streaminfo info;
+ struct audio_format audio_format;
+
+ MpcCallbackData data;
+
+ MPC_SAMPLE_FORMAT sample_buffer[MPC_DECODER_BUFFER_LENGTH];
+
+ int eof = 0;
+ long ret;
+#define MPC_CHUNK_SIZE 4096
+ char chunk[MPC_CHUNK_SIZE];
+ int chunkpos = 0;
+ long bitRate = 0;
+ int16_t *s16 = (int16_t *) chunk;
+ unsigned long samplePos = 0;
+ mpc_uint32_t vbrUpdateAcc;
+ mpc_uint32_t vbrUpdateBits;
+ float total_time;
+ int i;
+ ReplayGainInfo *replayGainInfo = NULL;
+
+ data.inStream = inStream;
+ data.decoder = mpd_decoder;
+
+ reader.read = mpc_read_cb;
+ reader.seek = mpc_seek_cb;
+ reader.tell = mpc_tell_cb;
+ reader.get_size = mpc_getsize_cb;
+ reader.canseek = mpc_canseek_cb;
+ reader.data = &data;
+
+ mpc_streaminfo_init(&info);
+
+ if ((ret = mpc_streaminfo_read(&info, &reader)) != ERROR_CODE_OK) {
+ if (decoder_get_command(mpd_decoder) != DECODE_COMMAND_STOP) {
+ ERROR("Not a valid musepack stream\n");
+ return -1;
+ }
+ return 0;
+ }
+
+ mpc_decoder_setup(&decoder, &reader);
+
+ if (!mpc_decoder_initialize(&decoder, &info)) {
+ if (decoder_get_command(mpd_decoder) != DECODE_COMMAND_STOP) {
+ ERROR("Not a valid musepack stream\n");
+ return -1;
+ }
+ return 0;
+ }
+
+ audio_format.bits = 16;
+ audio_format.channels = info.channels;
+ audio_format.sample_rate = info.sample_freq;
+
+ replayGainInfo = newReplayGainInfo();
+ replayGainInfo->albumGain = info.gain_album * 0.01;
+ replayGainInfo->albumPeak = info.peak_album / 32767.0;
+ replayGainInfo->trackGain = info.gain_title * 0.01;
+ replayGainInfo->trackPeak = info.peak_title / 32767.0;
+
+ decoder_initialized(mpd_decoder, &audio_format,
+ mpc_streaminfo_get_length(&info));
+
+ while (!eof) {
+ if (decoder_get_command(mpd_decoder) == DECODE_COMMAND_SEEK) {
+ samplePos = decoder_seek_where(mpd_decoder) *
+ audio_format.sample_rate;
+ if (mpc_decoder_seek_sample(&decoder, samplePos)) {
+ decoder_clear(mpd_decoder);
+ s16 = (int16_t *) chunk;
+ chunkpos = 0;
+ decoder_command_finished(mpd_decoder);
+ } else
+ decoder_seek_error(mpd_decoder);
+ }
+
+ vbrUpdateAcc = 0;
+ vbrUpdateBits = 0;
+ ret = mpc_decoder_decode(&decoder, sample_buffer,
+ &vbrUpdateAcc, &vbrUpdateBits);
+
+ if (ret <= 0 || decoder_get_command(mpd_decoder) == DECODE_COMMAND_STOP) {
+ eof = 1;
+ break;
+ }
+
+ samplePos += ret;
+
+ /* ret is in samples, and we have stereo */
+ ret *= 2;
+
+ for (i = 0; i < ret; i++) {
+ /* 16 bit audio again */
+ *s16 = convertSample(sample_buffer[i]);
+ chunkpos += 2;
+ s16++;
+
+ if (chunkpos >= MPC_CHUNK_SIZE) {
+ total_time = ((float)samplePos) /
+ audio_format.sample_rate;
+
+ bitRate = vbrUpdateBits *
+ audio_format.sample_rate / 1152 / 1000;
+
+ decoder_data(mpd_decoder, inStream,
+ inStream->seekable,
+ chunk, chunkpos,
+ total_time,
+ bitRate, replayGainInfo);
+
+ chunkpos = 0;
+ s16 = (int16_t *) chunk;
+ if (decoder_get_command(mpd_decoder) == DECODE_COMMAND_STOP) {
+ eof = 1;
+ break;
+ }
+ }
+ }
+ }
+
+ if (decoder_get_command(mpd_decoder) != DECODE_COMMAND_STOP &&
+ chunkpos > 0) {
+ total_time = ((float)samplePos) / audio_format.sample_rate;
+
+ bitRate =
+ vbrUpdateBits * audio_format.sample_rate / 1152 / 1000;
+
+ decoder_data(mpd_decoder, NULL, inStream->seekable,
+ chunk, chunkpos, total_time, bitRate,
+ replayGainInfo);
+ }
+
+ decoder_flush(mpd_decoder);
+
+ freeReplayGainInfo(replayGainInfo);
+
+ return 0;
+}
+
+static float mpcGetTime(char *file)
+{
+ InputStream inStream;
+ float total_time = -1;
+
+ mpc_reader reader;
+ mpc_streaminfo info;
+ MpcCallbackData data;
+
+ data.inStream = &inStream;
+ data.decoder = NULL;
+
+ reader.read = mpc_read_cb;
+ reader.seek = mpc_seek_cb;
+ reader.tell = mpc_tell_cb;
+ reader.get_size = mpc_getsize_cb;
+ reader.canseek = mpc_canseek_cb;
+ reader.data = &data;
+
+ mpc_streaminfo_init(&info);
+
+ if (openInputStream(&inStream, file) < 0) {
+ DEBUG("mpcGetTime: Failed to open file: %s\n", file);
+ return -1;
+ }
+
+ if (mpc_streaminfo_read(&info, &reader) != ERROR_CODE_OK) {
+ closeInputStream(&inStream);
+ return -1;
+ }
+
+ total_time = mpc_streaminfo_get_length(&info);
+
+ closeInputStream(&inStream);
+
+ return total_time;
+}
+
+static struct tag *mpcTagDup(char *file)
+{
+ struct tag *ret = NULL;
+ float total_time = mpcGetTime(file);
+
+ if (total_time < 0) {
+ DEBUG("mpcTagDup: Failed to get Songlength of file: %s\n",
+ file);
+ return NULL;
+ }
+
+ ret = tag_ape_load(file);
+ if (!ret)
+ ret = tag_id3_load(file);
+ if (!ret)
+ ret = tag_new();
+ ret->time = total_time;
+
+ return ret;
+}
+
+static const char *mpcSuffixes[] = { "mpc", NULL };
+
+struct decoder_plugin mpcPlugin = {
+ .name = "mpc",
+ .stream_decode = mpc_decode,
+ .tag_dup = mpcTagDup,
+ .stream_types = INPUT_PLUGIN_STREAM_URL | INPUT_PLUGIN_STREAM_FILE,
+ .suffixes = mpcSuffixes,
+};
diff --git a/src/decoder/oggflac_plugin.c b/src/decoder/oggflac_plugin.c
new file mode 100644
index 000000000..091b00988
--- /dev/null
+++ b/src/decoder/oggflac_plugin.c
@@ -0,0 +1,355 @@
+/* the Music Player Daemon (MPD)
+ * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
+ * This project's homepage is: http://www.musicpd.org
+ *
+ * OggFLAC support (half-stolen from flac_plugin.c :))
+ * (c) 2005 by Eric Wong <normalperson@yhbt.net>
+ *
+ * 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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include "_flac_common.h"
+#include "_ogg_common.h"
+
+#include "../utils.h"
+#include "../log.h"
+
+#include <OggFLAC/seekable_stream_decoder.h>
+
+static void oggflac_cleanup(FlacData * data,
+ OggFLAC__SeekableStreamDecoder * decoder)
+{
+ if (data->replayGainInfo)
+ freeReplayGainInfo(data->replayGainInfo);
+ if (decoder)
+ OggFLAC__seekable_stream_decoder_delete(decoder);
+}
+
+static OggFLAC__SeekableStreamDecoderReadStatus of_read_cb(mpd_unused const
+ OggFLAC__SeekableStreamDecoder
+ * decoder,
+ FLAC__byte buf[],
+ unsigned *bytes,
+ void *fdata)
+{
+ FlacData *data = (FlacData *) fdata;
+ size_t r;
+
+ r = decoder_read(data->decoder, data->inStream, (void *)buf, *bytes);
+ *bytes = r;
+
+ if (r == 0 && !inputStreamAtEOF(data->inStream) &&
+ decoder_get_command(data->decoder) == DECODE_COMMAND_NONE)
+ return OggFLAC__SEEKABLE_STREAM_DECODER_READ_STATUS_ERROR;
+
+ return OggFLAC__SEEKABLE_STREAM_DECODER_READ_STATUS_OK;
+}
+
+static OggFLAC__SeekableStreamDecoderSeekStatus of_seek_cb(mpd_unused const
+ OggFLAC__SeekableStreamDecoder
+ * decoder,
+ FLAC__uint64 offset,
+ void *fdata)
+{
+ FlacData *data = (FlacData *) fdata;
+
+ if (seekInputStream(data->inStream, offset, SEEK_SET) < 0) {
+ return OggFLAC__SEEKABLE_STREAM_DECODER_SEEK_STATUS_ERROR;
+ }
+
+ return OggFLAC__SEEKABLE_STREAM_DECODER_SEEK_STATUS_OK;
+}
+
+static OggFLAC__SeekableStreamDecoderTellStatus of_tell_cb(mpd_unused const
+ OggFLAC__SeekableStreamDecoder
+ * decoder,
+ FLAC__uint64 *
+ offset, void *fdata)
+{
+ FlacData *data = (FlacData *) fdata;
+
+ *offset = (long)(data->inStream->offset);
+
+ return OggFLAC__SEEKABLE_STREAM_DECODER_TELL_STATUS_OK;
+}
+
+static OggFLAC__SeekableStreamDecoderLengthStatus of_length_cb(mpd_unused const
+ OggFLAC__SeekableStreamDecoder
+ * decoder,
+ FLAC__uint64 *
+ length,
+ void *fdata)
+{
+ FlacData *data = (FlacData *) fdata;
+
+ *length = (size_t) (data->inStream->size);
+
+ return OggFLAC__SEEKABLE_STREAM_DECODER_LENGTH_STATUS_OK;
+}
+
+static FLAC__bool of_EOF_cb(mpd_unused const OggFLAC__SeekableStreamDecoder * decoder,
+ void *fdata)
+{
+ FlacData *data = (FlacData *) fdata;
+
+ return (decoder_get_command(data->decoder) != DECODE_COMMAND_NONE &&
+ decoder_get_command(data->decoder) != DECODE_COMMAND_SEEK) ||
+ inputStreamAtEOF(data->inStream);
+}
+
+static void of_error_cb(mpd_unused const OggFLAC__SeekableStreamDecoder * decoder,
+ FLAC__StreamDecoderErrorStatus status, void *fdata)
+{
+ flac_error_common_cb("oggflac", status, (FlacData *) fdata);
+}
+
+static void oggflacPrintErroredState(OggFLAC__SeekableStreamDecoderState state)
+{
+ switch (state) {
+ case OggFLAC__SEEKABLE_STREAM_DECODER_MEMORY_ALLOCATION_ERROR:
+ ERROR("oggflac allocation error\n");
+ break;
+ case OggFLAC__SEEKABLE_STREAM_DECODER_READ_ERROR:
+ ERROR("oggflac read error\n");
+ break;
+ case OggFLAC__SEEKABLE_STREAM_DECODER_SEEK_ERROR:
+ ERROR("oggflac seek error\n");
+ break;
+ case OggFLAC__SEEKABLE_STREAM_DECODER_STREAM_DECODER_ERROR:
+ ERROR("oggflac seekable stream error\n");
+ break;
+ case OggFLAC__SEEKABLE_STREAM_DECODER_ALREADY_INITIALIZED:
+ ERROR("oggflac decoder already initialized\n");
+ break;
+ case OggFLAC__SEEKABLE_STREAM_DECODER_INVALID_CALLBACK:
+ ERROR("invalid oggflac callback\n");
+ break;
+ case OggFLAC__SEEKABLE_STREAM_DECODER_UNINITIALIZED:
+ ERROR("oggflac decoder uninitialized\n");
+ break;
+ case OggFLAC__SEEKABLE_STREAM_DECODER_OK:
+ case OggFLAC__SEEKABLE_STREAM_DECODER_SEEKING:
+ case OggFLAC__SEEKABLE_STREAM_DECODER_END_OF_STREAM:
+ break;
+ }
+}
+
+static FLAC__StreamDecoderWriteStatus oggflacWrite(mpd_unused const
+ OggFLAC__SeekableStreamDecoder
+ * decoder,
+ const FLAC__Frame * frame,
+ const FLAC__int32 *
+ const buf[], void *vdata)
+{
+ FlacData *data = (FlacData *) vdata;
+ FLAC__uint32 samples = frame->header.blocksize;
+ float timeChange;
+
+ timeChange = ((float)samples) / frame->header.sample_rate;
+ data->time += timeChange;
+
+ return flac_common_write(data, frame, buf);
+}
+
+/* used by TagDup */
+static void of_metadata_dup_cb(mpd_unused const OggFLAC__SeekableStreamDecoder * decoder,
+ const FLAC__StreamMetadata * block, void *vdata)
+{
+ FlacData *data = (FlacData *) vdata;
+
+ switch (block->type) {
+ case FLAC__METADATA_TYPE_STREAMINFO:
+ if (!data->tag)
+ data->tag = tag_new();
+ data->tag->time = ((float)block->data.stream_info.
+ total_samples) /
+ block->data.stream_info.sample_rate + 0.5;
+ return;
+ case FLAC__METADATA_TYPE_VORBIS_COMMENT:
+ copyVorbisCommentBlockToMpdTag(block, data->tag);
+ default:
+ break;
+ }
+}
+
+/* used by decode */
+static void of_metadata_decode_cb(mpd_unused const OggFLAC__SeekableStreamDecoder * dec,
+ const FLAC__StreamMetadata * block,
+ void *vdata)
+{
+ flac_metadata_common_cb(block, (FlacData *) vdata);
+}
+
+static OggFLAC__SeekableStreamDecoder
+ * full_decoder_init_and_read_metadata(FlacData * data,
+ unsigned int metadata_only)
+{
+ OggFLAC__SeekableStreamDecoder *decoder = NULL;
+ unsigned int s = 1;
+
+ if (!(decoder = OggFLAC__seekable_stream_decoder_new()))
+ return NULL;
+
+ if (metadata_only) {
+ s &= OggFLAC__seekable_stream_decoder_set_metadata_callback
+ (decoder, of_metadata_dup_cb);
+ s &= OggFLAC__seekable_stream_decoder_set_metadata_respond
+ (decoder, FLAC__METADATA_TYPE_STREAMINFO);
+ } else {
+ s &= OggFLAC__seekable_stream_decoder_set_metadata_callback
+ (decoder, of_metadata_decode_cb);
+ }
+
+ s &= OggFLAC__seekable_stream_decoder_set_read_callback(decoder,
+ of_read_cb);
+ s &= OggFLAC__seekable_stream_decoder_set_seek_callback(decoder,
+ of_seek_cb);
+ s &= OggFLAC__seekable_stream_decoder_set_tell_callback(decoder,
+ of_tell_cb);
+ s &= OggFLAC__seekable_stream_decoder_set_length_callback(decoder,
+ of_length_cb);
+ s &= OggFLAC__seekable_stream_decoder_set_eof_callback(decoder,
+ of_EOF_cb);
+ s &= OggFLAC__seekable_stream_decoder_set_write_callback(decoder,
+ oggflacWrite);
+ s &= OggFLAC__seekable_stream_decoder_set_metadata_respond(decoder,
+ FLAC__METADATA_TYPE_VORBIS_COMMENT);
+ s &= OggFLAC__seekable_stream_decoder_set_error_callback(decoder,
+ of_error_cb);
+ s &= OggFLAC__seekable_stream_decoder_set_client_data(decoder,
+ (void *)data);
+
+ if (!s) {
+ ERROR("oggflac problem before init()\n");
+ goto fail;
+ }
+ if (OggFLAC__seekable_stream_decoder_init(decoder) !=
+ OggFLAC__SEEKABLE_STREAM_DECODER_OK) {
+ ERROR("oggflac problem doing init()\n");
+ goto fail;
+ }
+ if (!OggFLAC__seekable_stream_decoder_process_until_end_of_metadata
+ (decoder)) {
+ ERROR("oggflac problem reading metadata\n");
+ goto fail;
+ }
+
+ return decoder;
+
+fail:
+ oggflacPrintErroredState(OggFLAC__seekable_stream_decoder_get_state
+ (decoder));
+ OggFLAC__seekable_stream_decoder_delete(decoder);
+ return NULL;
+}
+
+/* public functions: */
+static struct tag *oggflac_TagDup(char *file)
+{
+ InputStream inStream;
+ OggFLAC__SeekableStreamDecoder *decoder;
+ FlacData data;
+
+ if (openInputStream(&inStream, file) < 0)
+ return NULL;
+ if (ogg_stream_type_detect(&inStream) != FLAC) {
+ closeInputStream(&inStream);
+ return NULL;
+ }
+
+ init_FlacData(&data, NULL, &inStream);
+
+ /* errors here won't matter,
+ * data.tag will be set or unset, that's all we care about */
+ decoder = full_decoder_init_and_read_metadata(&data, 1);
+
+ oggflac_cleanup(&data, decoder);
+ closeInputStream(&inStream);
+
+ return data.tag;
+}
+
+static bool oggflac_try_decode(InputStream * inStream)
+{
+ if (!inStream->seekable)
+ /* we cannot seek after the detection, so don't bother
+ checking */
+ return true;
+
+ return ogg_stream_type_detect(inStream) == FLAC;
+}
+
+static int oggflac_decode(struct decoder * mpd_decoder, InputStream * inStream)
+{
+ OggFLAC__SeekableStreamDecoder *decoder = NULL;
+ FlacData data;
+ int ret = 0;
+
+ init_FlacData(&data, mpd_decoder, inStream);
+
+ if (!(decoder = full_decoder_init_and_read_metadata(&data, 0))) {
+ ret = -1;
+ goto fail;
+ }
+
+ decoder_initialized(mpd_decoder, &data.audio_format, data.total_time);
+
+ while (1) {
+ OggFLAC__seekable_stream_decoder_process_single(decoder);
+ if (OggFLAC__seekable_stream_decoder_get_state(decoder) !=
+ OggFLAC__SEEKABLE_STREAM_DECODER_OK) {
+ break;
+ }
+ if (decoder_get_command(mpd_decoder) == DECODE_COMMAND_SEEK) {
+ FLAC__uint64 sampleToSeek = decoder_seek_where(mpd_decoder) *
+ data.audio_format.sample_rate + 0.5;
+ if (OggFLAC__seekable_stream_decoder_seek_absolute
+ (decoder, sampleToSeek)) {
+ decoder_clear(mpd_decoder);
+ data.time = ((float)sampleToSeek) /
+ data.audio_format.sample_rate;
+ data.position = 0;
+ decoder_command_finished(mpd_decoder);
+ } else
+ decoder_seek_error(mpd_decoder);
+ }
+ }
+
+ if (decoder_get_command(mpd_decoder) == DECODE_COMMAND_NONE) {
+ oggflacPrintErroredState
+ (OggFLAC__seekable_stream_decoder_get_state(decoder));
+ OggFLAC__seekable_stream_decoder_finish(decoder);
+ }
+
+fail:
+ oggflac_cleanup(&data, decoder);
+
+ return ret;
+}
+
+static const char *oggflac_Suffixes[] = { "ogg", "oga",NULL };
+static const char *oggflac_mime_types[] = { "audio/x-flac+ogg",
+ "application/ogg",
+ "application/x-ogg",
+ NULL };
+
+struct decoder_plugin oggflacPlugin = {
+ .name = "oggflac",
+ .try_decode = oggflac_try_decode,
+ .stream_decode = oggflac_decode,
+ .tag_dup = oggflac_TagDup,
+ .stream_types = INPUT_PLUGIN_STREAM_URL | INPUT_PLUGIN_STREAM_FILE,
+ .suffixes = oggflac_Suffixes,
+ .mime_types = oggflac_mime_types
+};
diff --git a/src/decoder/oggvorbis_plugin.c b/src/decoder/oggvorbis_plugin.c
new file mode 100644
index 000000000..0eecb783f
--- /dev/null
+++ b/src/decoder/oggvorbis_plugin.c
@@ -0,0 +1,387 @@
+/* the Music Player Daemon (MPD)
+ * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
+ * This project's homepage is: 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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+/* TODO 'ogg' should probably be replaced with 'oggvorbis' in all instances */
+
+#include "_ogg_common.h"
+#include "../utils.h"
+#include "../log.h"
+
+#ifndef HAVE_TREMOR
+#include <vorbis/vorbisfile.h>
+#else
+#include <tremor/ivorbisfile.h>
+/* Macros to make Tremor's API look like libogg. Tremor always
+ returns host-byte-order 16-bit signed data, and uses integer
+ milliseconds where libogg uses double seconds.
+*/
+#define ov_read(VF, BUFFER, LENGTH, BIGENDIANP, WORD, SGNED, BITSTREAM) \
+ ov_read(VF, BUFFER, LENGTH, BITSTREAM)
+#define ov_time_total(VF, I) ((double)ov_time_total(VF, I)/1000)
+#define ov_time_tell(VF) ((double)ov_time_tell(VF)/1000)
+#define ov_time_seek_page(VF, S) (ov_time_seek_page(VF, (S)*1000))
+#endif /* HAVE_TREMOR */
+
+#ifdef WORDS_BIGENDIAN
+#define OGG_DECODE_USE_BIGENDIAN 1
+#else
+#define OGG_DECODE_USE_BIGENDIAN 0
+#endif
+
+typedef struct _OggCallbackData {
+ InputStream *inStream;
+ struct decoder *decoder;
+} OggCallbackData;
+
+static size_t ogg_read_cb(void *ptr, size_t size, size_t nmemb, void *vdata)
+{
+ size_t ret;
+ OggCallbackData *data = (OggCallbackData *) vdata;
+
+ ret = decoder_read(data->decoder, data->inStream, ptr, size * nmemb);
+
+ errno = 0;
+ /*if(ret<0) errno = ((InputStream *)inStream)->error; */
+
+ return ret / size;
+}
+
+static int ogg_seek_cb(void *vdata, ogg_int64_t offset, int whence)
+{
+ const OggCallbackData *data = (const OggCallbackData *) vdata;
+ if(decoder_get_command(data->decoder) == DECODE_COMMAND_STOP)
+ return -1;
+ return seekInputStream(data->inStream, offset, whence);
+}
+
+/* TODO: check Ogg libraries API and see if we can just not have this func */
+static int ogg_close_cb(mpd_unused void *vdata)
+{
+ return 0;
+}
+
+static long ogg_tell_cb(void *vdata)
+{
+ const OggCallbackData *data = (const OggCallbackData *) vdata;
+
+ return (long)(data->inStream->offset);
+}
+
+static const char *ogg_parseComment(const char *comment, const char *needle)
+{
+ int len = strlen(needle);
+
+ if (strncasecmp(comment, needle, len) == 0 && *(comment + len) == '=') {
+ return comment + len + 1;
+ }
+
+ return NULL;
+}
+
+static void ogg_getReplayGainInfo(char **comments, ReplayGainInfo ** infoPtr)
+{
+ const char *temp;
+ int found = 0;
+
+ if (*infoPtr)
+ freeReplayGainInfo(*infoPtr);
+ *infoPtr = newReplayGainInfo();
+
+ while (*comments) {
+ if ((temp =
+ ogg_parseComment(*comments, "replaygain_track_gain"))) {
+ (*infoPtr)->trackGain = atof(temp);
+ found = 1;
+ } else if ((temp = ogg_parseComment(*comments,
+ "replaygain_album_gain"))) {
+ (*infoPtr)->albumGain = atof(temp);
+ found = 1;
+ } else if ((temp = ogg_parseComment(*comments,
+ "replaygain_track_peak"))) {
+ (*infoPtr)->trackPeak = atof(temp);
+ found = 1;
+ } else if ((temp = ogg_parseComment(*comments,
+ "replaygain_album_peak"))) {
+ (*infoPtr)->albumPeak = atof(temp);
+ found = 1;
+ }
+
+ comments++;
+ }
+
+ if (!found) {
+ freeReplayGainInfo(*infoPtr);
+ *infoPtr = NULL;
+ }
+}
+
+static const char *VORBIS_COMMENT_TRACK_KEY = "tracknumber";
+static const char *VORBIS_COMMENT_DISC_KEY = "discnumber";
+
+static unsigned int ogg_parseCommentAddToTag(char *comment,
+ unsigned int itemType,
+ struct tag ** tag)
+{
+ const char *needle;
+ unsigned int len;
+ switch (itemType) {
+ case TAG_ITEM_TRACK:
+ needle = VORBIS_COMMENT_TRACK_KEY;
+ break;
+ case TAG_ITEM_DISC:
+ needle = VORBIS_COMMENT_DISC_KEY;
+ break;
+ default:
+ needle = mpdTagItemKeys[itemType];
+ }
+ len = strlen(needle);
+
+ if (strncasecmp(comment, needle, len) == 0 && *(comment + len) == '=') {
+ if (!*tag)
+ *tag = tag_new();
+
+ tag_add_item(*tag, itemType, comment + len + 1);
+
+ return 1;
+ }
+
+ return 0;
+}
+
+static struct tag *oggCommentsParse(char **comments)
+{
+ struct tag *tag = NULL;
+
+ while (*comments) {
+ int j;
+ for (j = TAG_NUM_OF_ITEM_TYPES; --j >= 0;) {
+ if (ogg_parseCommentAddToTag(*comments, j, &tag))
+ break;
+ }
+ comments++;
+ }
+
+ return tag;
+}
+
+static void putOggCommentsIntoOutputBuffer(char *streamName,
+ char **comments)
+{
+ struct tag *tag;
+
+ tag = oggCommentsParse(comments);
+ if (!tag && streamName) {
+ tag = tag_new();
+ }
+ if (!tag)
+ return;
+
+ if (streamName) {
+ tag_clear_items_by_type(tag, TAG_ITEM_NAME);
+ tag_add_item(tag, TAG_ITEM_NAME, streamName);
+ }
+
+ tag_free(tag);
+}
+
+/* public */
+static int oggvorbis_decode(struct decoder * decoder, InputStream * inStream)
+{
+ OggVorbis_File vf;
+ ov_callbacks callbacks;
+ OggCallbackData data;
+ struct audio_format audio_format;
+ int current_section;
+ int prev_section = -1;
+ long ret;
+#define OGG_CHUNK_SIZE 4096
+ char chunk[OGG_CHUNK_SIZE];
+ int chunkpos = 0;
+ long bitRate = 0;
+ long test;
+ ReplayGainInfo *replayGainInfo = NULL;
+ char **comments;
+ const char *errorStr;
+ int initialized = 0;
+
+ data.inStream = inStream;
+ data.decoder = decoder;
+
+ callbacks.read_func = ogg_read_cb;
+ callbacks.seek_func = ogg_seek_cb;
+ callbacks.close_func = ogg_close_cb;
+ callbacks.tell_func = ogg_tell_cb;
+ if ((ret = ov_open_callbacks(&data, &vf, NULL, 0, callbacks)) < 0) {
+ if (decoder_get_command(decoder) != DECODE_COMMAND_NONE)
+ return 0;
+
+ switch (ret) {
+ case OV_EREAD:
+ errorStr = "read error";
+ break;
+ case OV_ENOTVORBIS:
+ errorStr = "not vorbis stream";
+ break;
+ case OV_EVERSION:
+ errorStr = "vorbis version mismatch";
+ break;
+ case OV_EBADHEADER:
+ errorStr = "invalid vorbis header";
+ break;
+ case OV_EFAULT:
+ errorStr = "internal logic error";
+ break;
+ default:
+ errorStr = "unknown error";
+ break;
+ }
+ ERROR("Error decoding Ogg Vorbis stream: %s\n",
+ errorStr);
+ return -1;
+ }
+ audio_format.bits = 16;
+
+ while (1) {
+ if (decoder_get_command(decoder) == DECODE_COMMAND_SEEK) {
+ double seek_where = decoder_seek_where(decoder);
+ if (0 == ov_time_seek_page(&vf, seek_where)) {
+ decoder_clear(decoder);
+ chunkpos = 0;
+ decoder_command_finished(decoder);
+ } else
+ decoder_seek_error(decoder);
+ }
+ ret = ov_read(&vf, chunk + chunkpos,
+ OGG_CHUNK_SIZE - chunkpos,
+ OGG_DECODE_USE_BIGENDIAN, 2, 1, &current_section);
+ if (current_section != prev_section) {
+ /*printf("new song!\n"); */
+ vorbis_info *vi = ov_info(&vf, -1);
+ audio_format.channels = vi->channels;
+ audio_format.sample_rate = vi->rate;
+ if (!initialized) {
+ float total_time = ov_time_total(&vf, -1);
+ if (total_time < 0)
+ total_time = 0;
+ decoder_initialized(decoder, &audio_format,
+ total_time);
+ initialized = 1;
+ }
+ comments = ov_comment(&vf, -1)->user_comments;
+ putOggCommentsIntoOutputBuffer(inStream->metaName,
+ comments);
+ ogg_getReplayGainInfo(comments, &replayGainInfo);
+ }
+
+ prev_section = current_section;
+
+ if (ret <= 0) {
+ if (ret == OV_HOLE) /* bad packet */
+ ret = 0;
+ else /* break on EOF or other error */
+ break;
+ }
+
+ chunkpos += ret;
+
+ if (chunkpos >= OGG_CHUNK_SIZE) {
+ if ((test = ov_bitrate_instant(&vf)) > 0) {
+ bitRate = test / 1000;
+ }
+ decoder_data(decoder, inStream,
+ inStream->seekable,
+ chunk, chunkpos,
+ ov_pcm_tell(&vf) / audio_format.sample_rate,
+ bitRate, replayGainInfo);
+ chunkpos = 0;
+ if (decoder_get_command(decoder) == DECODE_COMMAND_STOP)
+ break;
+ }
+ }
+
+ if (decoder_get_command(decoder) == DECODE_COMMAND_NONE &&
+ chunkpos > 0) {
+ decoder_data(decoder, NULL, inStream->seekable,
+ chunk, chunkpos,
+ ov_time_tell(&vf), bitRate,
+ replayGainInfo);
+ }
+
+ if (replayGainInfo)
+ freeReplayGainInfo(replayGainInfo);
+
+ ov_clear(&vf);
+
+ decoder_flush(decoder);
+
+ return 0;
+}
+
+static struct tag *oggvorbis_TagDup(char *file)
+{
+ struct tag *ret;
+ FILE *fp;
+ OggVorbis_File vf;
+
+ fp = fopen(file, "r");
+ if (!fp) {
+ DEBUG("oggvorbis_TagDup: Failed to open file: '%s', %s\n",
+ file, strerror(errno));
+ return NULL;
+ }
+ if (ov_open(fp, &vf, NULL, 0) < 0) {
+ fclose(fp);
+ return NULL;
+ }
+
+ ret = oggCommentsParse(ov_comment(&vf, -1)->user_comments);
+
+ if (!ret)
+ ret = tag_new();
+ ret->time = (int)(ov_time_total(&vf, -1) + 0.5);
+
+ ov_clear(&vf);
+
+ return ret;
+}
+
+static bool oggvorbis_try_decode(InputStream * inStream)
+{
+ if (!inStream->seekable)
+ /* we cannot seek after the detection, so don't bother
+ checking */
+ return true;
+
+ return ogg_stream_type_detect(inStream) == VORBIS;
+}
+
+static const char *oggvorbis_Suffixes[] = { "ogg","oga", NULL };
+static const char *oggvorbis_MimeTypes[] = { "application/ogg",
+ "audio/x-vorbis+ogg",
+ "application/x-ogg",
+ NULL };
+
+struct decoder_plugin oggvorbisPlugin = {
+ .name = "oggvorbis",
+ .try_decode = oggvorbis_try_decode,
+ .stream_decode = oggvorbis_decode,
+ .tag_dup = oggvorbis_TagDup,
+ .stream_types = INPUT_PLUGIN_STREAM_URL | INPUT_PLUGIN_STREAM_FILE,
+ .suffixes = oggvorbis_Suffixes,
+ .mime_types = oggvorbis_MimeTypes
+};
diff --git a/src/decoder/wavpack_plugin.c b/src/decoder/wavpack_plugin.c
new file mode 100644
index 000000000..14b7e5f69
--- /dev/null
+++ b/src/decoder/wavpack_plugin.c
@@ -0,0 +1,574 @@
+/* the Music Player Daemon (MPD)
+ * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
+ * This project's homepage is: http://www.musicpd.org
+ *
+ * WavPack support added by Laszlo Ashin <kodest@gmail.com>
+ *
+ * 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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include "../decoder_api.h"
+#include "../utils.h"
+#include "../log.h"
+#include "../path.h"
+
+#include <wavpack/wavpack.h>
+
+/* pick 1020 since its devisible for 8,16,24, and 32-bit audio */
+#define CHUNK_SIZE 1020
+
+#define ERRORLEN 80
+
+static struct {
+ const char *name;
+ int type;
+} tagtypes[] = {
+ { "artist", TAG_ITEM_ARTIST },
+ { "album", TAG_ITEM_ALBUM },
+ { "title", TAG_ITEM_TITLE },
+ { "track", TAG_ITEM_TRACK },
+ { "name", TAG_ITEM_NAME },
+ { "genre", TAG_ITEM_GENRE },
+ { "date", TAG_ITEM_DATE },
+ { "composer", TAG_ITEM_COMPOSER },
+ { "performer", TAG_ITEM_PERFORMER },
+ { "comment", TAG_ITEM_COMMENT },
+ { "disc", TAG_ITEM_DISC },
+ { NULL, 0 }
+};
+
+/*
+ * This function has been borrowed from the tiny player found on
+ * wavpack.com. Modifications were required because mpd only handles
+ * max 16 bit samples.
+ */
+static void format_samples_int(int Bps, void *buffer, uint32_t samcnt)
+{
+ int32_t temp;
+ uchar *dst = (uchar *)buffer;
+ int32_t *src = (int32_t *)buffer;
+
+ switch (Bps) {
+ case 1:
+ while (samcnt--)
+ *dst++ = *src++;
+ break;
+ case 2:
+ while (samcnt--) {
+ temp = *src++;
+#ifdef WORDS_BIGENDIAN
+ *dst++ = (uchar)(temp >> 8);
+ *dst++ = (uchar)(temp);
+#else
+ *dst++ = (uchar)(temp);
+ *dst++ = (uchar)(temp >> 8);
+#endif
+ }
+ break;
+ case 3:
+ /* downscale to 16 bits */
+ while (samcnt--) {
+ temp = *src++;
+#ifdef WORDS_BIGENDIAN
+ *dst++ = (uchar)(temp >> 16);
+ *dst++ = (uchar)(temp >> 8);
+#else
+ *dst++ = (uchar)(temp >> 8);
+ *dst++ = (uchar)(temp >> 16);
+#endif
+ }
+ break;
+ case 4:
+ /* downscale to 16 bits */
+ while (samcnt--) {
+ temp = *src++;
+#ifdef WORDS_BIGENDIAN
+ *dst++ = (uchar)(temp >> 24);
+ *dst++ = (uchar)(temp >> 16);
+
+#else
+ *dst++ = (uchar)(temp >> 16);
+ *dst++ = (uchar)(temp >> 24);
+#endif
+ }
+ break;
+ }
+}
+
+/*
+ * This function converts floating point sample data to 16 bit integer.
+ */
+static void format_samples_float(mpd_unused int Bps, void *buffer,
+ uint32_t samcnt)
+{
+ int16_t *dst = (int16_t *)buffer;
+ float *src = (float *)buffer;
+
+ while (samcnt--) {
+ *dst++ = (int16_t)(*src++);
+ }
+}
+
+/*
+ * This does the main decoding thing.
+ * Requires an already opened WavpackContext.
+ */
+static void wavpack_decode(struct decoder * decoder,
+ WavpackContext *wpc, int canseek,
+ ReplayGainInfo *replayGainInfo)
+{
+ struct audio_format audio_format;
+ void (*format_samples)(int Bps, void *buffer, uint32_t samcnt);
+ char chunk[CHUNK_SIZE];
+ float file_time;
+ int samplesreq, samplesgot;
+ int allsamples;
+ int position, outsamplesize;
+ int Bps;
+
+ audio_format.sample_rate = WavpackGetSampleRate(wpc);
+ audio_format.channels = WavpackGetReducedChannels(wpc);
+ audio_format.bits = WavpackGetBitsPerSample(wpc);
+
+ if (audio_format.bits > 16)
+ audio_format.bits = 16;
+
+ if ((WavpackGetMode(wpc) & MODE_FLOAT) == MODE_FLOAT)
+ format_samples = format_samples_float;
+ else
+ format_samples = format_samples_int;
+/*
+ if ((WavpackGetMode(wpc) & MODE_WVC) == MODE_WVC)
+ ERROR("decoding WITH wvc!!!\n");
+ else
+ ERROR("decoding without wvc\n");
+*/
+ allsamples = WavpackGetNumSamples(wpc);
+ Bps = WavpackGetBytesPerSample(wpc);
+
+ outsamplesize = Bps;
+ if (outsamplesize > 2)
+ outsamplesize = 2;
+ outsamplesize *= audio_format.channels;
+
+ samplesreq = sizeof(chunk) / (4 * audio_format.channels);
+
+ decoder_initialized(decoder, &audio_format,
+ (float)allsamples / audio_format.sample_rate);
+
+ position = 0;
+
+ do {
+ if (decoder_get_command(decoder) == DECODE_COMMAND_SEEK) {
+ if (canseek) {
+ int where;
+
+ decoder_clear(decoder);
+
+ where = decoder_seek_where(decoder) *
+ audio_format.sample_rate;
+ if (WavpackSeekSample(wpc, where)) {
+ position = where;
+ decoder_command_finished(decoder);
+ } else
+ decoder_seek_error(decoder);
+ } else {
+ decoder_seek_error(decoder);
+ }
+ }
+
+ if (decoder_get_command(decoder) == DECODE_COMMAND_STOP)
+ break;
+
+ samplesgot = WavpackUnpackSamples(wpc,
+ (int32_t *)chunk, samplesreq);
+ if (samplesgot > 0) {
+ int bitrate = (int)(WavpackGetInstantBitrate(wpc) /
+ 1000 + 0.5);
+ position += samplesgot;
+ file_time = (float)position / audio_format.sample_rate;
+
+ format_samples(Bps, chunk,
+ samplesgot * audio_format.channels);
+
+ decoder_data(decoder, NULL, 0, chunk,
+ samplesgot * outsamplesize,
+ file_time, bitrate,
+ replayGainInfo);
+ }
+ } while (samplesgot == samplesreq);
+
+ decoder_flush(decoder);
+}
+
+static char *wavpack_tag(WavpackContext *wpc, char *key)
+{
+ char *value = NULL;
+ int size;
+
+ size = WavpackGetTagItem(wpc, key, NULL, 0);
+ if (size > 0) {
+ size++;
+ value = xmalloc(size);
+ if (!value)
+ return NULL;
+ WavpackGetTagItem(wpc, key, value, size);
+ }
+
+ return value;
+}
+
+static ReplayGainInfo *wavpack_replaygain(WavpackContext *wpc)
+{
+ static char replaygain_track_gain[] = "replaygain_track_gain";
+ static char replaygain_album_gain[] = "replaygain_album_gain";
+ static char replaygain_track_peak[] = "replaygain_track_peak";
+ static char replaygain_album_peak[] = "replaygain_album_peak";
+ ReplayGainInfo *replayGainInfo;
+ int found = 0;
+ char *value;
+
+ replayGainInfo = newReplayGainInfo();
+
+ value = wavpack_tag(wpc, replaygain_track_gain);
+ if (value) {
+ replayGainInfo->trackGain = atof(value);
+ free(value);
+ found = 1;
+ }
+
+ value = wavpack_tag(wpc, replaygain_album_gain);
+ if (value) {
+ replayGainInfo->albumGain = atof(value);
+ free(value);
+ found = 1;
+ }
+
+ value = wavpack_tag(wpc, replaygain_track_peak);
+ if (value) {
+ replayGainInfo->trackPeak = atof(value);
+ free(value);
+ found = 1;
+ }
+
+ value = wavpack_tag(wpc, replaygain_album_peak);
+ if (value) {
+ replayGainInfo->albumPeak = atof(value);
+ free(value);
+ found = 1;
+ }
+
+
+ if (found)
+ return replayGainInfo;
+
+ freeReplayGainInfo(replayGainInfo);
+
+ return NULL;
+}
+
+/*
+ * Reads metainfo from the specified file.
+ */
+static struct tag *wavpack_tagdup(char *fname)
+{
+ WavpackContext *wpc;
+ struct tag *tag;
+ char error[ERRORLEN];
+ char *s;
+ int ssize;
+ int i, j;
+
+ wpc = WavpackOpenFileInput(fname, error, OPEN_TAGS, 0);
+ if (wpc == NULL) {
+ ERROR("failed to open WavPack file \"%s\": %s\n", fname, error);
+ return NULL;
+ }
+
+ tag = tag_new();
+ tag->time =
+ (float)WavpackGetNumSamples(wpc) / WavpackGetSampleRate(wpc);
+
+ ssize = 0;
+ s = NULL;
+
+ for (i = 0; tagtypes[i].name != NULL; ++i) {
+ j = WavpackGetTagItem(wpc, tagtypes[i].name, NULL, 0);
+ if (j > 0) {
+ ++j;
+
+ if (s == NULL) {
+ s = xmalloc(j);
+ if (s == NULL) break;
+ ssize = j;
+ } else if (j > ssize) {
+ char *t = (char *)xrealloc(s, j);
+ if (t == NULL) break;
+ ssize = j;
+ s = t;
+ }
+
+ WavpackGetTagItem(wpc, tagtypes[i].name, s, j);
+ tag_add_item(tag, tagtypes[i].type, s);
+ }
+ }
+
+ if (s != NULL)
+ free(s);
+
+ WavpackCloseFile(wpc);
+
+ return tag;
+}
+
+/*
+ * mpd InputStream <=> WavpackStreamReader wrapper callbacks
+ */
+
+/* This struct is needed for per-stream last_byte storage. */
+typedef struct {
+ struct decoder *decoder;
+ InputStream *is;
+ /* Needed for push_back_byte() */
+ int last_byte;
+} InputStreamPlus;
+
+static int32_t read_bytes(void *id, void *data, int32_t bcount)
+{
+ InputStreamPlus *isp = (InputStreamPlus *)id;
+ uint8_t *buf = (uint8_t *)data;
+ int32_t i = 0;
+
+ if (isp->last_byte != EOF) {
+ *buf++ = isp->last_byte;
+ isp->last_byte = EOF;
+ --bcount;
+ ++i;
+ }
+ return i + decoder_read(isp->decoder, isp->is, buf, bcount);
+}
+
+static uint32_t get_pos(void *id)
+{
+ return ((InputStreamPlus *)id)->is->offset;
+}
+
+static int set_pos_abs(void *id, uint32_t pos)
+{
+ return seekInputStream(((InputStreamPlus *)id)->is, pos, SEEK_SET);
+}
+
+static int set_pos_rel(void *id, int32_t delta, int mode)
+{
+ return seekInputStream(((InputStreamPlus *)id)->is, delta, mode);
+}
+
+static int push_back_byte(void *id, int c)
+{
+ ((InputStreamPlus *)id)->last_byte = c;
+ return 1;
+}
+
+static uint32_t get_length(void *id)
+{
+ return ((InputStreamPlus *)id)->is->size;
+}
+
+static int can_seek(void *id)
+{
+ return ((InputStreamPlus *)id)->is->seekable;
+}
+
+static WavpackStreamReader mpd_is_reader = {
+ .read_bytes = read_bytes,
+ .get_pos = get_pos,
+ .set_pos_abs = set_pos_abs,
+ .set_pos_rel = set_pos_rel,
+ .push_back_byte = push_back_byte,
+ .get_length = get_length,
+ .can_seek = can_seek,
+ .write_bytes = NULL /* no need to write edited tags */
+};
+
+static void
+initInputStreamPlus(InputStreamPlus *isp, struct decoder *decoder,
+ InputStream *is)
+{
+ isp->decoder = decoder;
+ isp->is = is;
+ isp->last_byte = EOF;
+}
+
+/*
+ * Tries to decode the specified stream, and gives true if managed to do it.
+ */
+static bool wavpack_trydecode(InputStream *is)
+{
+ char error[ERRORLEN];
+ WavpackContext *wpc;
+ InputStreamPlus isp;
+
+ initInputStreamPlus(&isp, NULL, is);
+ wpc = WavpackOpenFileInputEx(&mpd_is_reader, &isp, NULL, error,
+ OPEN_STREAMING, 0);
+ if (wpc == NULL)
+ return false;
+
+ WavpackCloseFile(wpc);
+ /* Seek it back in order to play from the first byte. */
+ seekInputStream(is, 0, SEEK_SET);
+
+ return true;
+}
+
+static int wavpack_open_wvc(struct decoder *decoder,
+ InputStream *is_wvc)
+{
+ char tmp[MPD_PATH_MAX];
+ const char *utf8url;
+ size_t len;
+ char *wvc_url = NULL;
+ int ret;
+
+ /*
+ * As we use dc->utf8url, this function will be bad for
+ * single files. utf8url is not absolute file path :/
+ */
+ utf8url = decoder_get_url(decoder, tmp);
+ if (utf8url == NULL)
+ return 0;
+
+ len = strlen(utf8url);
+ if (!len)
+ return 0;
+
+ wvc_url = (char *)xmalloc(len + 2); /* +2: 'c' and EOS */
+ if (wvc_url == NULL)
+ return 0;
+
+ memcpy(wvc_url, utf8url, len);
+ wvc_url[len] = 'c';
+ wvc_url[len + 1] = '\0';
+
+ ret = openInputStream(is_wvc, wvc_url);
+ free(wvc_url);
+
+ if (ret)
+ return 0;
+
+ /*
+ * And we try to buffer in order to get know
+ * about a possible 404 error.
+ */
+ for (;;) {
+ if (inputStreamAtEOF(is_wvc)) {
+ /*
+ * EOF is reached even without
+ * a single byte is read...
+ * So, this is not good :/
+ */
+ closeInputStream(is_wvc);
+ return 0;
+ }
+
+ if (bufferInputStream(is_wvc) >= 0)
+ return 1;
+
+ if (decoder_get_command(decoder) != DECODE_COMMAND_NONE) {
+ closeInputStream(is_wvc);
+ return 0;
+ }
+
+ /* Save some CPU */
+ my_usleep(1000);
+ }
+}
+
+/*
+ * Decodes a stream.
+ */
+static int wavpack_streamdecode(struct decoder * decoder, InputStream *is)
+{
+ char error[ERRORLEN];
+ WavpackContext *wpc;
+ InputStream is_wvc;
+ int open_flags = OPEN_2CH_MAX | OPEN_NORMALIZE /*| OPEN_STREAMING*/;
+ InputStreamPlus isp, isp_wvc;
+
+ if (wavpack_open_wvc(decoder, &is_wvc)) {
+ initInputStreamPlus(&isp_wvc, decoder, &is_wvc);
+ open_flags |= OPEN_WVC;
+ }
+
+ initInputStreamPlus(&isp, decoder, is);
+ wpc = WavpackOpenFileInputEx(&mpd_is_reader, &isp, &isp_wvc, error,
+ open_flags, 15);
+
+ if (wpc == NULL) {
+ ERROR("failed to open WavPack stream: %s\n", error);
+ return -1;
+ }
+
+ wavpack_decode(decoder, wpc, can_seek(&isp), NULL);
+
+ WavpackCloseFile(wpc);
+ if (open_flags & OPEN_WVC)
+ closeInputStream(&is_wvc);
+ closeInputStream(is);
+
+ return 0;
+}
+
+/*
+ * Decodes a file.
+ */
+static int wavpack_filedecode(struct decoder * decoder, char *fname)
+{
+ char error[ERRORLEN];
+ WavpackContext *wpc;
+ ReplayGainInfo *replayGainInfo;
+
+ wpc = WavpackOpenFileInput(fname, error,
+ OPEN_TAGS | OPEN_WVC |
+ OPEN_2CH_MAX | OPEN_NORMALIZE, 15);
+ if (wpc == NULL) {
+ ERROR("failed to open WavPack file \"%s\": %s\n", fname, error);
+ return -1;
+ }
+
+ replayGainInfo = wavpack_replaygain(wpc);
+
+ wavpack_decode(decoder, wpc, 1, replayGainInfo);
+
+ if (replayGainInfo)
+ freeReplayGainInfo(replayGainInfo);
+
+ WavpackCloseFile(wpc);
+
+ return 0;
+}
+
+static char const *wavpackSuffixes[] = { "wv", NULL };
+static char const *wavpackMimeTypes[] = { "audio/x-wavpack", NULL };
+
+struct decoder_plugin wavpackPlugin = {
+ .name = "wavpack",
+ .try_decode = wavpack_trydecode,
+ .stream_decode = wavpack_streamdecode,
+ .file_decode = wavpack_filedecode,
+ .tag_dup = wavpack_tagdup,
+ .stream_types = INPUT_PLUGIN_STREAM_FILE | INPUT_PLUGIN_STREAM_URL,
+ .suffixes = wavpackSuffixes,
+ .mime_types = wavpackMimeTypes
+};
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-08 03:56:44 +0000