aboutsummaryrefslogtreecommitdiffstats
path: root/src/inputPlugins
diff options
context:
space:
mode:
Diffstat (limited to 'src/inputPlugins')
-rw-r--r--src/inputPlugins/_flac_common.c323
-rw-r--r--src/inputPlugins/_flac_common.h168
-rw-r--r--src/inputPlugins/_ogg_common.c49
-rw-r--r--src/inputPlugins/_ogg_common.h31
-rw-r--r--src/inputPlugins/aac_plugin.c602
-rw-r--r--src/inputPlugins/audiofile_plugin.c147
-rw-r--r--src/inputPlugins/ffmpeg_plugin.c419
-rw-r--r--src/inputPlugins/flac_plugin.c459
-rw-r--r--src/inputPlugins/mod_plugin.c278
-rw-r--r--src/inputPlugins/mp3_plugin.c1086
-rw-r--r--src/inputPlugins/mp4_plugin.c423
-rw-r--r--src/inputPlugins/mpc_plugin.c308
-rw-r--r--src/inputPlugins/oggflac_plugin.c355
-rw-r--r--src/inputPlugins/oggvorbis_plugin.c387
-rw-r--r--src/inputPlugins/wavpack_plugin.c574
15 files changed, 0 insertions, 5609 deletions
diff --git a/src/inputPlugins/_flac_common.c b/src/inputPlugins/_flac_common.c
deleted file mode 100644
index db43e0003..000000000
--- a/src/inputPlugins/_flac_common.c
+++ /dev/null
@@ -1,323 +0,0 @@
-/* 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/inputPlugins/_flac_common.h b/src/inputPlugins/_flac_common.h
deleted file mode 100644
index 45714b4bd..000000000
--- a/src/inputPlugins/_flac_common.h
+++ /dev/null
@@ -1,168 +0,0 @@
-/* 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/inputPlugins/_ogg_common.c b/src/inputPlugins/_ogg_common.c
deleted file mode 100644
index 841b2ad3f..000000000
--- a/src/inputPlugins/_ogg_common.c
+++ /dev/null
@@ -1,49 +0,0 @@
-/* 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/inputPlugins/_ogg_common.h b/src/inputPlugins/_ogg_common.h
deleted file mode 100644
index 7c9e7b630..000000000
--- a/src/inputPlugins/_ogg_common.h
+++ /dev/null
@@ -1,31 +0,0 @@
-/* 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/inputPlugins/aac_plugin.c b/src/inputPlugins/aac_plugin.c
deleted file mode 100644
index 7842bcc22..000000000
--- a/src/inputPlugins/aac_plugin.c
+++ /dev/null
@@ -1,602 +0,0 @@
-/* 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/inputPlugins/audiofile_plugin.c b/src/inputPlugins/audiofile_plugin.c
deleted file mode 100644
index 99846e853..000000000
--- a/src/inputPlugins/audiofile_plugin.c
+++ /dev/null
@@ -1,147 +0,0 @@
-/* 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/inputPlugins/ffmpeg_plugin.c b/src/inputPlugins/ffmpeg_plugin.c
deleted file mode 100644
index 6455cd1ce..000000000
--- a/src/inputPlugins/ffmpeg_plugin.c
+++ /dev/null
@@ -1,419 +0,0 @@
-/* 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/inputPlugins/flac_plugin.c b/src/inputPlugins/flac_plugin.c
deleted file mode 100644
index 7b9fce27d..000000000
--- a/src/inputPlugins/flac_plugin.c
+++ /dev/null
@@ -1,459 +0,0 @@
-/* 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/inputPlugins/mod_plugin.c b/src/inputPlugins/mod_plugin.c
deleted file mode 100644
index 5916a24ab..000000000
--- a/src/inputPlugins/mod_plugin.c
+++ /dev/null
@@ -1,278 +0,0 @@
-/* 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/inputPlugins/mp3_plugin.c b/src/inputPlugins/mp3_plugin.c
deleted file mode 100644
index a0de30ba7..000000000
--- a/src/inputPlugins/mp3_plugin.c
+++ /dev/null
@@ -1,1086 +0,0 @@
-/* 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/inputPlugins/mp4_plugin.c b/src/inputPlugins/mp4_plugin.c
deleted file mode 100644
index 4a613744e..000000000
--- a/src/inputPlugins/mp4_plugin.c
+++ /dev/null
@@ -1,423 +0,0 @@
-/* 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/inputPlugins/mpc_plugin.c b/src/inputPlugins/mpc_plugin.c
deleted file mode 100644
index fb1b0b56c..000000000
--- a/src/inputPlugins/mpc_plugin.c
+++ /dev/null
@@ -1,308 +0,0 @@
-/* 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/inputPlugins/oggflac_plugin.c b/src/inputPlugins/oggflac_plugin.c
deleted file mode 100644
index 091b00988..000000000
--- a/src/inputPlugins/oggflac_plugin.c
+++ /dev/null
@@ -1,355 +0,0 @@
-/* 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/inputPlugins/oggvorbis_plugin.c b/src/inputPlugins/oggvorbis_plugin.c
deleted file mode 100644
index 0eecb783f..000000000
--- a/src/inputPlugins/oggvorbis_plugin.c
+++ /dev/null
@@ -1,387 +0,0 @@
-/* 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/inputPlugins/wavpack_plugin.c b/src/inputPlugins/wavpack_plugin.c
deleted file mode 100644
index 14b7e5f69..000000000
--- a/src/inputPlugins/wavpack_plugin.c
+++ /dev/null
@@ -1,574 +0,0 @@
-/* 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 2024-07-07 22:54:16 +0000