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-rw-r--r--src/decoder/_flac_common.c310
-rw-r--r--src/decoder/_flac_common.h13
-rw-r--r--src/decoder/audiofile_plugin.c8
-rw-r--r--src/decoder/faad_plugin.c6
-rw-r--r--src/decoder/ffmpeg_plugin.c42
-rw-r--r--src/decoder/flac_metadata.c192
-rw-r--r--src/decoder/flac_metadata.h45
-rw-r--r--src/decoder/flac_pcm.c101
-rw-r--r--src/decoder/flac_pcm.h31
-rw-r--r--src/decoder/flac_plugin.c62
-rw-r--r--src/decoder/mad_plugin.c18
-rw-r--r--src/decoder/mikmod_plugin.c6
-rw-r--r--src/decoder/modplug_plugin.c8
-rw-r--r--src/decoder/mp4ff_plugin.c22
-rw-r--r--src/decoder/mpcdec_plugin.c4
-rw-r--r--src/decoder/mpg123_decoder_plugin.c211
-rw-r--r--src/decoder/oggflac_plugin.c35
-rw-r--r--src/decoder/sidplay_plugin.cxx280
-rw-r--r--src/decoder/sndfile_decoder_plugin.c244
-rwxr-xr-x[-rw-r--r--]src/decoder/vorbis_plugin.c12
-rw-r--r--src/decoder/wavpack_plugin.c28
21 files changed, 1241 insertions, 437 deletions
diff --git a/src/decoder/_flac_common.c b/src/decoder/_flac_common.c
index e096750f3..2ccd2e536 100644
--- a/src/decoder/_flac_common.c
+++ b/src/decoder/_flac_common.c
@@ -22,6 +22,8 @@
*/
#include "_flac_common.h"
+#include "flac_metadata.h"
+#include "flac_pcm.h"
#include <glib.h>
@@ -31,6 +33,8 @@ void
flac_data_init(struct flac_data *data, struct decoder * decoder,
struct input_stream *input_stream)
{
+ pcm_buffer_init(&data->buffer);
+
data->time = 0;
data->position = 0;
data->bit_rate = 0;
@@ -40,158 +44,16 @@ flac_data_init(struct flac_data *data, struct decoder * decoder,
data->tag = NULL;
}
-static void
-flac_find_float_comment(const FLAC__StreamMetadata *block,
- const char *cmnt, float *fl, bool *found_r)
-{
- int offset;
- size_t pos;
- int len;
- unsigned char tmp, *p;
-
- offset = FLAC__metadata_object_vorbiscomment_find_entry_from(block, 0,
- cmnt);
- if (offset < 0)
- return;
-
- pos = strlen(cmnt) + 1; /* 1 is for '=' */
- len = block->data.vorbis_comment.comments[offset].length - pos;
- if (len <= 0)
- return;
-
- p = &block->data.vorbis_comment.comments[offset].entry[pos];
- tmp = p[len];
- p[len] = '\0';
- *fl = (float)atof((char *)p);
- p[len] = tmp;
-
- *found_r = true;
-}
-
-static void
-flac_parse_replay_gain(const FLAC__StreamMetadata *block,
- struct flac_data *data)
+void
+flac_data_deinit(struct flac_data *data)
{
- bool found = false;
-
- if (data->replay_gain_info)
- replay_gain_info_free(data->replay_gain_info);
+ pcm_buffer_deinit(&data->buffer);
- data->replay_gain_info = replay_gain_info_new();
-
- flac_find_float_comment(block, "replaygain_album_gain",
- &data->replay_gain_info->tuples[REPLAY_GAIN_ALBUM].gain,
- &found);
- flac_find_float_comment(block, "replaygain_album_peak",
- &data->replay_gain_info->tuples[REPLAY_GAIN_ALBUM].peak,
- &found);
- flac_find_float_comment(block, "replaygain_track_gain",
- &data->replay_gain_info->tuples[REPLAY_GAIN_TRACK].gain,
- &found);
- flac_find_float_comment(block, "replaygain_track_peak",
- &data->replay_gain_info->tuples[REPLAY_GAIN_TRACK].peak,
- &found);
-
- if (!found) {
+ if (data->replay_gain_info != NULL)
replay_gain_info_free(data->replay_gain_info);
- data->replay_gain_info = NULL;
- }
-}
-
-/**
- * Checks if the specified name matches the entry's name, and if yes,
- * returns the comment value (not null-temrinated).
- */
-static const char *
-flac_comment_value(const FLAC__StreamMetadata_VorbisComment_Entry *entry,
- const char *name, const char *char_tnum, size_t *length_r)
-{
- size_t name_length = strlen(name);
- size_t char_tnum_length = 0;
- const char *comment = (const char*)entry->entry;
-
- if (entry->length <= name_length ||
- g_ascii_strncasecmp(comment, name, name_length) != 0)
- return NULL;
-
- if (char_tnum != NULL) {
- char_tnum_length = strlen(char_tnum);
- if (entry->length > name_length + char_tnum_length + 2 &&
- comment[name_length] == '[' &&
- g_ascii_strncasecmp(comment + name_length + 1,
- char_tnum, char_tnum_length) == 0 &&
- comment[name_length + char_tnum_length + 1] == ']')
- name_length = name_length + char_tnum_length + 2;
- else if (entry->length > name_length + char_tnum_length &&
- g_ascii_strncasecmp(comment + name_length,
- char_tnum, char_tnum_length) == 0)
- name_length = name_length + char_tnum_length;
- }
-
- if (comment[name_length] == '=') {
- *length_r = entry->length - name_length - 1;
- return comment + name_length + 1;
- }
-
- return NULL;
-}
-
-/**
- * Check if the comment's name equals the passed name, and if so, copy
- * the comment value into the tag.
- */
-static bool
-flac_copy_comment(struct tag *tag,
- const FLAC__StreamMetadata_VorbisComment_Entry *entry,
- const char *name, enum tag_type tag_type,
- const char *char_tnum)
-{
- const char *value;
- size_t value_length;
-
- value = flac_comment_value(entry, name, char_tnum, &value_length);
- if (value != NULL) {
- tag_add_item_n(tag, tag_type, value, value_length);
- return true;
- }
-
- return false;
-}
-
-/* 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 void
-flac_parse_comment(struct tag *tag, const char *char_tnum,
- const FLAC__StreamMetadata_VorbisComment_Entry *entry)
-{
- assert(tag != NULL);
-
- if (flac_copy_comment(tag, entry, VORBIS_COMMENT_TRACK_KEY,
- TAG_ITEM_TRACK, char_tnum) ||
- flac_copy_comment(tag, entry, VORBIS_COMMENT_DISC_KEY,
- TAG_ITEM_DISC, char_tnum) ||
- flac_copy_comment(tag, entry, "album artist",
- TAG_ITEM_ALBUM_ARTIST, char_tnum))
- return;
- for (unsigned i = 0; i < TAG_NUM_OF_ITEM_TYPES; ++i)
- if (flac_copy_comment(tag, entry,
- tag_item_names[i], i, char_tnum))
- return;
-}
-
-void
-flac_vorbis_comments_to_tag(struct tag *tag, const char *char_tnum,
- const FLAC__StreamMetadata *block)
-{
- FLAC__StreamMetadata_VorbisComment_Entry *comments =
- block->data.vorbis_comment.comments;
-
- for (unsigned i = block->data.vorbis_comment.num_comments; i > 0; --i)
- flac_parse_comment(tag, char_tnum, comments++);
+ if (data->tag != NULL)
+ tag_free(data->tag);
}
void flac_metadata_common_cb(const FLAC__StreamMetadata * block,
@@ -201,16 +63,18 @@ void flac_metadata_common_cb(const FLAC__StreamMetadata * block,
switch (block->type) {
case FLAC__METADATA_TYPE_STREAMINFO:
- data->audio_format.bits = (int8_t)si->bits_per_sample;
- data->audio_format.sample_rate = si->sample_rate;
- data->audio_format.channels = (int8_t)si->channels;
+ audio_format_init(&data->audio_format, si->sample_rate,
+ si->bits_per_sample, si->channels);
data->total_time = ((float)si->total_samples) / (si->sample_rate);
break;
case FLAC__METADATA_TYPE_VORBIS_COMMENT:
- flac_parse_replay_gain(block, data);
+ if (data->replay_gain_info)
+ replay_gain_info_free(data->replay_gain_info);
+ data->replay_gain_info = flac_parse_replay_gain(block);
if (data->tag != NULL)
- flac_vorbis_comments_to_tag(data->tag, NULL, block);
+ flac_vorbis_comments_to_tag(data->tag, NULL,
+ &block->data.vorbis_comment);
default:
break;
@@ -239,133 +103,35 @@ void flac_error_common_cb(const char *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(struct flac_data *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;
+ size_t buffer_size = frame->header.blocksize *
+ audio_format_frame_size(&data->audio_format);
+ void *buffer;
+
+ buffer = pcm_buffer_get(&data->buffer, buffer_size);
+
+ flac_convert(buffer, data->audio_format.channels,
+ data->audio_format.bits, buf,
+ 0, frame->header.blocksize);
+
+ cmd = decoder_data(data->decoder, data->input_stream,
+ buffer, buffer_size,
+ data->time, data->bit_rate,
+ data->replay_gain_info);
+ switch (cmd) {
+ case DECODE_COMMAND_NONE:
+ case DECODE_COMMAND_START:
+ break;
- if (bytes_per_sample != 1 && bytes_per_sample != 2 &&
- bytes_per_sample != 4)
- /* exotic unsupported bit rate */
+ case DECODE_COMMAND_STOP:
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->input_stream,
- data->chunk,
- num_samples * bytes_per_channel,
- data->time, data->bit_rate,
- data->replay_gain_info);
- 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;
- }
+ case DECODE_COMMAND_SEEK:
+ return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE;
}
return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE;
diff --git a/src/decoder/_flac_common.h b/src/decoder/_flac_common.h
index 68de7e969..e0ddc6027 100644
--- a/src/decoder/_flac_common.h
+++ b/src/decoder/_flac_common.h
@@ -24,7 +24,8 @@
#ifndef MPD_FLAC_COMMON_H
#define MPD_FLAC_COMMON_H
-#include "../decoder_api.h"
+#include "decoder_api.h"
+#include "pcm_buffer.h"
#include "config.h"
#include <glib.h>
@@ -145,7 +146,8 @@ typedef size_t flac_read_status_size_t;
#define FLAC_CHUNK_SIZE 4080
struct flac_data {
- unsigned char chunk[FLAC_CHUNK_SIZE];
+ struct pcm_buffer buffer;
+
float time;
unsigned int bit_rate;
struct audio_format audio_format;
@@ -162,6 +164,9 @@ void
flac_data_init(struct flac_data *data, struct decoder * decoder,
struct input_stream *input_stream);
+void
+flac_data_deinit(struct flac_data *data);
+
void flac_metadata_common_cb(const FLAC__StreamMetadata * block,
struct flac_data *data);
@@ -169,10 +174,6 @@ void flac_error_common_cb(const char *plugin,
FLAC__StreamDecoderErrorStatus status,
struct flac_data *data);
-void
-flac_vorbis_comments_to_tag(struct tag *tag, const char *char_tnum,
- const FLAC__StreamMetadata *block);
-
FLAC__StreamDecoderWriteStatus
flac_common_write(struct flac_data *data, const FLAC__Frame * frame,
const FLAC__int32 *const buf[]);
diff --git a/src/decoder/audiofile_plugin.c b/src/decoder/audiofile_plugin.c
index f66d90dc1..b4959f6c2 100644
--- a/src/decoder/audiofile_plugin.c
+++ b/src/decoder/audiofile_plugin.c
@@ -136,11 +136,9 @@ audiofile_stream_decode(struct decoder *decoder, struct input_stream *is)
afSetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK,
AF_SAMPFMT_TWOSCOMP, bits);
afGetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits);
- audio_format.bits = (uint8_t)bits;
- audio_format.sample_rate =
- (unsigned int)afGetRate(af_fp, AF_DEFAULT_TRACK);
- audio_format.channels =
- (uint8_t)afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK);
+
+ audio_format_init(&audio_format, afGetRate(af_fp, AF_DEFAULT_TRACK),
+ bits, afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK));
if (!audio_format_valid(&audio_format)) {
g_warning("Invalid audio format: %u:%u:%u\n",
diff --git a/src/decoder/faad_plugin.c b/src/decoder/faad_plugin.c
index 7b2806a4c..516f741c7 100644
--- a/src/decoder/faad_plugin.c
+++ b/src/decoder/faad_plugin.c
@@ -266,11 +266,7 @@ faad_decoder_init(faacDecHandle decoder, struct decoder_buffer *buffer,
decoder_buffer_consume(buffer, nbytes);
- *audio_format = (struct audio_format){
- .bits = 16,
- .channels = channels,
- .sample_rate = sample_rate,
- };
+ audio_format_init(audio_format, sample_rate, 16, channels);
return true;
}
diff --git a/src/decoder/ffmpeg_plugin.c b/src/decoder/ffmpeg_plugin.c
index 2f7ed6d1f..63f0f7ece 100644
--- a/src/decoder/ffmpeg_plugin.c
+++ b/src/decoder/ffmpeg_plugin.c
@@ -267,6 +267,7 @@ ffmpeg_decode_internal(struct ffmpeg_context *ctx)
struct audio_format audio_format;
enum decoder_command cmd;
int total_time;
+ uint8_t bits;
total_time = 0;
@@ -275,13 +276,13 @@ ffmpeg_decode_internal(struct ffmpeg_context *ctx)
}
#if LIBAVCODEC_VERSION_INT >= ((51<<16)+(41<<8)+0)
- audio_format.bits = (uint8_t) av_get_bits_per_sample_format(codec_context->sample_fmt);
+ bits = (uint8_t) av_get_bits_per_sample_format(codec_context->sample_fmt);
#else
/* XXX fixme 16-bit for older ffmpeg (13 Aug 2007) */
- audio_format.bits = (uint8_t) 16;
+ bits = (uint8_t) 16;
#endif
- audio_format.sample_rate = (unsigned int)codec_context->sample_rate;
- audio_format.channels = codec_context->channels;
+ audio_format_init(&audio_format, codec_context->sample_rate, bits,
+ codec_context->channels);
if (!audio_format_valid(&audio_format)) {
g_warning("Invalid audio format: %u:%u:%u\n",
@@ -342,8 +343,9 @@ static bool
ffmpeg_copy_metadata(struct tag *tag, AVMetadata *m,
enum tag_type type, const char *name)
{
- AVMetadataTag *mt = av_metadata_get(m, name, NULL, 0);
- if (mt != NULL)
+ AVMetadataTag *mt = NULL;
+
+ while ((mt = av_metadata_get(m, name, mt, 0)) != NULL)
tag_add_item(tag, type, mt->value);
return mt != NULL;
}
@@ -361,35 +363,35 @@ static bool ffmpeg_tag_internal(struct ffmpeg_context *ctx)
#if LIBAVFORMAT_VERSION_INT >= ((52<<16)+(31<<8)+0)
av_metadata_conv(f, NULL, f->iformat->metadata_conv);
- ffmpeg_copy_metadata(tag, f->metadata, TAG_ITEM_TITLE, "title");
- ffmpeg_copy_metadata(tag, f->metadata, TAG_ITEM_ARTIST, "author");
- ffmpeg_copy_metadata(tag, f->metadata, TAG_ITEM_ALBUM, "album");
- ffmpeg_copy_metadata(tag, f->metadata, TAG_ITEM_COMMENT, "comment");
- ffmpeg_copy_metadata(tag, f->metadata, TAG_ITEM_GENRE, "genre");
- ffmpeg_copy_metadata(tag, f->metadata, TAG_ITEM_TRACK, "track");
- ffmpeg_copy_metadata(tag, f->metadata, TAG_ITEM_DATE, "year");
+ ffmpeg_copy_metadata(tag, f->metadata, TAG_TITLE, "title");
+ ffmpeg_copy_metadata(tag, f->metadata, TAG_ARTIST, "author");
+ ffmpeg_copy_metadata(tag, f->metadata, TAG_ALBUM, "album");
+ ffmpeg_copy_metadata(tag, f->metadata, TAG_COMMENT, "comment");
+ ffmpeg_copy_metadata(tag, f->metadata, TAG_GENRE, "genre");
+ ffmpeg_copy_metadata(tag, f->metadata, TAG_TRACK, "track");
+ ffmpeg_copy_metadata(tag, f->metadata, TAG_DATE, "year");
#else
if (f->author[0])
- tag_add_item(tag, TAG_ITEM_ARTIST, f->author);
+ tag_add_item(tag, TAG_ARTIST, f->author);
if (f->title[0])
- tag_add_item(tag, TAG_ITEM_TITLE, f->title);
+ tag_add_item(tag, TAG_TITLE, f->title);
if (f->album[0])
- tag_add_item(tag, TAG_ITEM_ALBUM, f->album);
+ tag_add_item(tag, TAG_ALBUM, f->album);
if (f->track > 0) {
char buffer[16];
snprintf(buffer, sizeof(buffer), "%d", f->track);
- tag_add_item(tag, TAG_ITEM_TRACK, buffer);
+ tag_add_item(tag, TAG_TRACK, buffer);
}
if (f->comment[0])
- tag_add_item(tag, TAG_ITEM_COMMENT, f->comment);
+ tag_add_item(tag, TAG_COMMENT, f->comment);
if (f->genre[0])
- tag_add_item(tag, TAG_ITEM_GENRE, f->genre);
+ tag_add_item(tag, TAG_GENRE, f->genre);
if (f->year > 0) {
char buffer[16];
snprintf(buffer, sizeof(buffer), "%d", f->year);
- tag_add_item(tag, TAG_ITEM_DATE, buffer);
+ tag_add_item(tag, TAG_DATE, buffer);
}
#endif
diff --git a/src/decoder/flac_metadata.c b/src/decoder/flac_metadata.c
new file mode 100644
index 000000000..912559c8d
--- /dev/null
+++ b/src/decoder/flac_metadata.c
@@ -0,0 +1,192 @@
+/*
+ * Copyright (C) 2003-2009 The Music Player Daemon Project
+ * http://www.musicpd.org
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with this program; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include "flac_metadata.h"
+#include "replay_gain.h"
+#include "tag.h"
+
+#include <glib.h>
+
+#include <assert.h>
+#include <stdbool.h>
+#include <stdlib.h>
+
+static bool
+flac_find_float_comment(const FLAC__StreamMetadata *block,
+ const char *cmnt, float *fl)
+{
+ int offset;
+ size_t pos;
+ int len;
+ unsigned char tmp, *p;
+
+ offset = FLAC__metadata_object_vorbiscomment_find_entry_from(block, 0,
+ cmnt);
+ if (offset < 0)
+ return false;
+
+ pos = strlen(cmnt) + 1; /* 1 is for '=' */
+ len = block->data.vorbis_comment.comments[offset].length - pos;
+ if (len <= 0)
+ return false;
+
+ p = &block->data.vorbis_comment.comments[offset].entry[pos];
+ tmp = p[len];
+ p[len] = '\0';
+ *fl = (float)atof((char *)p);
+ p[len] = tmp;
+
+ return true;
+}
+
+struct replay_gain_info *
+flac_parse_replay_gain(const FLAC__StreamMetadata *block)
+{
+ struct replay_gain_info *rgi;
+ bool found = false;
+
+ rgi = replay_gain_info_new();
+
+ found = flac_find_float_comment(block, "replaygain_album_gain",
+ &rgi->tuples[REPLAY_GAIN_ALBUM].gain) ||
+ flac_find_float_comment(block, "replaygain_album_peak",
+ &rgi->tuples[REPLAY_GAIN_ALBUM].peak) ||
+ flac_find_float_comment(block, "replaygain_track_gain",
+ &rgi->tuples[REPLAY_GAIN_TRACK].gain) ||
+ flac_find_float_comment(block, "replaygain_track_peak",
+ &rgi->tuples[REPLAY_GAIN_TRACK].peak);
+ if (!found) {
+ replay_gain_info_free(rgi);
+ rgi = NULL;
+ }
+
+ return rgi;
+}
+
+/**
+ * Checks if the specified name matches the entry's name, and if yes,
+ * returns the comment value (not null-temrinated).
+ */
+static const char *
+flac_comment_value(const FLAC__StreamMetadata_VorbisComment_Entry *entry,
+ const char *name, const char *char_tnum, size_t *length_r)
+{
+ size_t name_length = strlen(name);
+ size_t char_tnum_length = 0;
+ const char *comment = (const char*)entry->entry;
+
+ if (entry->length <= name_length ||
+ g_ascii_strncasecmp(comment, name, name_length) != 0)
+ return NULL;
+
+ if (char_tnum != NULL) {
+ char_tnum_length = strlen(char_tnum);
+ if (entry->length > name_length + char_tnum_length + 2 &&
+ comment[name_length] == '[' &&
+ g_ascii_strncasecmp(comment + name_length + 1,
+ char_tnum, char_tnum_length) == 0 &&
+ comment[name_length + char_tnum_length + 1] == ']')
+ name_length = name_length + char_tnum_length + 2;
+ else if (entry->length > name_length + char_tnum_length &&
+ g_ascii_strncasecmp(comment + name_length,
+ char_tnum, char_tnum_length) == 0)
+ name_length = name_length + char_tnum_length;
+ }
+
+ if (comment[name_length] == '=') {
+ *length_r = entry->length - name_length - 1;
+ return comment + name_length + 1;
+ }
+
+ return NULL;
+}
+
+/**
+ * Check if the comment's name equals the passed name, and if so, copy
+ * the comment value into the tag.
+ */
+static bool
+flac_copy_comment(struct tag *tag,
+ const FLAC__StreamMetadata_VorbisComment_Entry *entry,
+ const char *name, enum tag_type tag_type,
+ const char *char_tnum)
+{
+ const char *value;
+ size_t value_length;
+
+ value = flac_comment_value(entry, name, char_tnum, &value_length);
+ if (value != NULL) {
+ tag_add_item_n(tag, tag_type, value, value_length);
+ return true;
+ }
+
+ return false;
+}
+
+/* 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 void
+flac_parse_comment(struct tag *tag, const char *char_tnum,
+ const FLAC__StreamMetadata_VorbisComment_Entry *entry)
+{
+ assert(tag != NULL);
+
+ if (flac_copy_comment(tag, entry, VORBIS_COMMENT_TRACK_KEY,
+ TAG_TRACK, char_tnum) ||
+ flac_copy_comment(tag, entry, VORBIS_COMMENT_DISC_KEY,
+ TAG_DISC, char_tnum) ||
+ flac_copy_comment(tag, entry, "album artist",
+ TAG_ALBUM_ARTIST, char_tnum))
+ return;
+
+ for (unsigned i = 0; i < TAG_NUM_OF_ITEM_TYPES; ++i)
+ if (flac_copy_comment(tag, entry,
+ tag_item_names[i], i, char_tnum))
+ return;
+}
+
+void
+flac_vorbis_comments_to_tag(struct tag *tag, const char *char_tnum,
+ const FLAC__StreamMetadata_VorbisComment *comment)
+{
+ for (unsigned i = 0; i < comment->num_comments; ++i)
+ flac_parse_comment(tag, char_tnum, &comment->comments[i]);
+}
+
+void
+flac_tag_apply_metadata(struct tag *tag, const char *track,
+ const FLAC__StreamMetadata *block)
+{
+ switch (block->type) {
+ case FLAC__METADATA_TYPE_VORBIS_COMMENT:
+ flac_vorbis_comments_to_tag(tag, track,
+ &block->data.vorbis_comment);
+ break;
+
+ case FLAC__METADATA_TYPE_STREAMINFO:
+ tag->time = flac_duration(&block->data.stream_info);
+ break;
+
+ default:
+ break;
+ }
+}
diff --git a/src/decoder/flac_metadata.h b/src/decoder/flac_metadata.h
new file mode 100644
index 000000000..ef97288d5
--- /dev/null
+++ b/src/decoder/flac_metadata.h
@@ -0,0 +1,45 @@
+/*
+ * Copyright (C) 2003-2009 The Music Player Daemon Project
+ * http://www.musicpd.org
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with this program; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#ifndef MPD_FLAC_METADATA_H
+#define MPD_FLAC_METADATA_H
+
+#include <FLAC/metadata.h>
+
+struct tag;
+
+static inline unsigned
+flac_duration(const FLAC__StreamMetadata_StreamInfo *stream_info)
+{
+ return (stream_info->total_samples + stream_info->sample_rate - 1) /
+ stream_info->sample_rate;
+}
+
+struct replay_gain_info *
+flac_parse_replay_gain(const FLAC__StreamMetadata *block);
+
+void
+flac_vorbis_comments_to_tag(struct tag *tag, const char *char_tnum,
+ const FLAC__StreamMetadata_VorbisComment *comment);
+
+void
+flac_tag_apply_metadata(struct tag *tag, const char *track,
+ const FLAC__StreamMetadata *block);
+
+#endif
diff --git a/src/decoder/flac_pcm.c b/src/decoder/flac_pcm.c
new file mode 100644
index 000000000..d37ef54e2
--- /dev/null
+++ b/src/decoder/flac_pcm.c
@@ -0,0 +1,101 @@
+/*
+ * Copyright (C) 2003-2009 The Music Player Daemon Project
+ * http://www.musicpd.org
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with this program; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include "flac_pcm.h"
+
+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];
+}
+
+void
+flac_convert(void *dest,
+ unsigned int num_channels, unsigned sample_format,
+ const FLAC__int32 *const buf[],
+ unsigned int position, unsigned int end)
+{
+ switch (sample_format) {
+ case 16:
+ 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 24:
+ case 32:
+ flac_convert_32((int32_t*)dest, num_channels, buf,
+ position, end);
+ break;
+
+ case 8:
+ flac_convert_8((int8_t*)dest, num_channels, buf,
+ position, end);
+ break;
+ }
+}
diff --git a/src/decoder/flac_pcm.h b/src/decoder/flac_pcm.h
new file mode 100644
index 000000000..dca9d6824
--- /dev/null
+++ b/src/decoder/flac_pcm.h
@@ -0,0 +1,31 @@
+/*
+ * Copyright (C) 2003-2009 The Music Player Daemon Project
+ * http://www.musicpd.org
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with this program; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#ifndef MPD_FLAC_PCM_H
+#define MPD_FLAC_PCM_H
+
+#include <FLAC/ordinals.h>
+
+void
+flac_convert(void *dest,
+ unsigned int num_channels, unsigned sample_format,
+ const FLAC__int32 *const buf[],
+ unsigned int position, unsigned int end);
+
+#endif
diff --git a/src/decoder/flac_plugin.c b/src/decoder/flac_plugin.c
index 0c0d994b7..17d0e8519 100644
--- a/src/decoder/flac_plugin.c
+++ b/src/decoder/flac_plugin.c
@@ -18,6 +18,7 @@
*/
#include "_flac_common.h"
+#include "flac_metadata.h"
#include <glib.h>
@@ -268,12 +269,8 @@ flac_tag_load(const char *file, const char *char_tnum)
block = FLAC__metadata_simple_iterator_get_block(it);
if (!block)
break;
- if (block->type == FLAC__METADATA_TYPE_VORBIS_COMMENT) {
- flac_vorbis_comments_to_tag(tag, char_tnum, block);
- } else if (block->type == FLAC__METADATA_TYPE_STREAMINFO) {
- tag->time = ((float)block->data.stream_info.total_samples) /
- block->data.stream_info.sample_rate + 0.5;
- }
+
+ flac_tag_apply_metadata(tag, char_tnum, block);
FLAC__metadata_object_delete(block);
} while (FLAC__metadata_simple_iterator_next(it));
@@ -300,6 +297,8 @@ flac_cue_tag_load(const char *file)
FLAC__uint64 track_time = 0;
#ifdef HAVE_CUE /* libcue */
FLAC__StreamMetadata* vc;
+ char* cs_filename;
+ FILE* cs_file;
#endif /* libcue */
FLAC__StreamMetadata* si = FLAC__metadata_object_new(FLAC__METADATA_TYPE_STREAMINFO);
FLAC__StreamMetadata* cs;
@@ -329,16 +328,25 @@ flac_cue_tag_load(const char *file)
FLAC__metadata_object_delete(vc);
}
+
+ if (tag == NULL) {
+ cs_filename = g_strconcat(file, ".cue", NULL);
+
+ cs_file = fopen(cs_filename, "rt");
+ g_free(cs_filename);
+
+ if (cs_file != NULL) {
+ tag = cue_tag_file(cs_file, tnum);
+ fclose(cs_file);
+ }
+ }
#endif /* libcue */
if (tag == NULL)
tag = flac_tag_load(file, char_tnum);
- if (char_tnum != NULL)
- {
- tag_add_item( tag,
- TAG_ITEM_TRACK,
- char_tnum);
+ if (char_tnum != NULL) {
+ tag_add_item(tag, TAG_TRACK, char_tnum);
g_free(char_tnum);
}
@@ -474,13 +482,8 @@ flac_decode_internal(struct decoder * decoder,
}
fail:
- if (data.replay_gain_info)
- replay_gain_info_free(data.replay_gain_info);
-
- tag_free(data.tag);
-
- if (flac_dec)
- flac_delete(flac_dec);
+ flac_data_deinit(&data);
+ flac_delete(flac_dec);
if (err)
g_warning("%s\n", err);
@@ -653,11 +656,8 @@ fail:
if (pathname)
g_free(pathname);
- if (data.replay_gain_info)
- replay_gain_info_free(data.replay_gain_info);
-
- if (flac_dec)
- flac_delete(flac_dec);
+ flac_data_deinit(&data);
+ flac_delete(flac_dec);
if (err)
g_warning("%s\n", err);
@@ -776,11 +776,8 @@ flac_filedecode_internal(struct decoder* decoder,
}
fail:
- if (data.replay_gain_info)
- replay_gain_info_free(data.replay_gain_info);
-
- if (flac_dec)
- flac_delete(flac_dec);
+ flac_data_deinit(&data);
+ flac_delete(flac_dec);
if (err)
g_warning("%s\n", err);
@@ -836,13 +833,8 @@ oggflac_tag_dup(const char *file)
do {
if (!(block = FLAC__metadata_iterator_get_block(it)))
break;
- if (block->type == FLAC__METADATA_TYPE_VORBIS_COMMENT) {
- flac_vorbis_comments_to_tag(ret, NULL, block);
- } else if (block->type == FLAC__METADATA_TYPE_STREAMINFO) {
- ret->time = ((float)block->data.stream_info.
- total_samples) /
- block->data.stream_info.sample_rate + 0.5;
- }
+
+ flac_tag_apply_metadata(ret, NULL, block);
} while (FLAC__metadata_iterator_next(it));
FLAC__metadata_iterator_delete(it);
diff --git a/src/decoder/mad_plugin.c b/src/decoder/mad_plugin.c
index 1ef7183fa..27ddf655d 100644
--- a/src/decoder/mad_plugin.c
+++ b/src/decoder/mad_plugin.c
@@ -779,10 +779,10 @@ mp3_frame_duration(const struct mad_frame *frame)
MAD_UNITS_MILLISECONDS) / 1000.0;
}
-static off_t
+static goffset
mp3_this_frame_offset(const struct mp3_data *data)
{
- off_t offset = data->input_stream->offset;
+ goffset offset = data->input_stream->offset;
if (data->stream.this_frame != NULL)
offset -= data->stream.bufend - data->stream.this_frame;
@@ -792,7 +792,7 @@ mp3_this_frame_offset(const struct mp3_data *data)
return offset;
}
-static off_t
+static goffset
mp3_rest_including_this_frame(const struct mp3_data *data)
{
return data->input_stream->size - mp3_this_frame_offset(data);
@@ -804,7 +804,7 @@ mp3_rest_including_this_frame(const struct mp3_data *data)
static void
mp3_filesize_to_song_length(struct mp3_data *data)
{
- off_t rest = mp3_rest_including_this_frame(data);
+ goffset rest = mp3_rest_including_this_frame(data);
if (rest > 0) {
float frame_duration = mp3_frame_duration(&data->frame);
@@ -1170,13 +1170,6 @@ mp3_read(struct mp3_data *data, struct replay_gain_info **replay_gain_info_r)
return ret != DECODE_BREAK;
}
-static void mp3_audio_format(struct mp3_data *data, struct audio_format *af)
-{
- af->bits = 24;
- af->sample_rate = (data->frame).header.samplerate;
- af->channels = MAD_NCHANNELS(&(data->frame).header);
-}
-
static void
mp3_decode(struct decoder *decoder, struct input_stream *input_stream)
{
@@ -1192,7 +1185,8 @@ mp3_decode(struct decoder *decoder, struct input_stream *input_stream)
return;
}
- mp3_audio_format(&data, &audio_format);
+ audio_format_init(&audio_format, data.frame.header.samplerate, 24,
+ MAD_NCHANNELS(&data.frame.header));
decoder_initialized(decoder, &audio_format,
data.input_stream->seekable, data.total_time);
diff --git a/src/decoder/mikmod_plugin.c b/src/decoder/mikmod_plugin.c
index 065c34319..850232f34 100644
--- a/src/decoder/mikmod_plugin.c
+++ b/src/decoder/mikmod_plugin.c
@@ -175,9 +175,7 @@ mod_decode(struct decoder *decoder, const char *path)
return;
}
- audio_format.bits = 16;
- audio_format.sample_rate = 44100;
- audio_format.channels = 2;
+ audio_format_init(&audio_format, 44100, 16, 2);
secPerByte =
1.0 / ((audio_format.bits * audio_format.channels / 8.0) *
@@ -222,7 +220,7 @@ static struct tag *modTagDup(const char *file)
title = g_strdup(Player_LoadTitle(path2));
g_free(path2);
if (title)
- tag_add_item(ret, TAG_ITEM_TITLE, title);
+ tag_add_item(ret, TAG_TITLE, title);
return ret;
}
diff --git a/src/decoder/modplug_plugin.c b/src/decoder/modplug_plugin.c
index f636f2fa6..301c80674 100644
--- a/src/decoder/modplug_plugin.c
+++ b/src/decoder/modplug_plugin.c
@@ -121,9 +121,7 @@ mod_decode(struct decoder *decoder, struct input_stream *is)
return;
}
- audio_format.bits = 16;
- audio_format.sample_rate = 44100;
- audio_format.channels = 2;
+ audio_format_init(&audio_format, 44100, 16, 2);
sec_perbyte =
1.0 / ((audio_format.bits * audio_format.channels / 8.0) *
@@ -186,11 +184,11 @@ static struct tag *mod_tagdup(const char *file)
return NULL;
}
ret = tag_new();
- ret->time = 0;
+ ret->time = ModPlug_GetLength(f) / 1000;
title = g_strdup(ModPlug_GetName(f));
if (title)
- tag_add_item(ret, TAG_ITEM_TITLE, title);
+ tag_add_item(ret, TAG_TITLE, title);
g_free(title);
ModPlug_Unload(f);
diff --git a/src/decoder/mp4ff_plugin.c b/src/decoder/mp4ff_plugin.c
index cf9382904..745d1e619 100644
--- a/src/decoder/mp4ff_plugin.c
+++ b/src/decoder/mp4ff_plugin.c
@@ -131,11 +131,7 @@ mp4_faad_new(mp4ff_t *mp4fh, int *track_r, struct audio_format *audio_format)
}
*track_r = track;
- *audio_format = (struct audio_format){
- .bits = 16,
- .channels = channels,
- .sample_rate = sample_rate,
- };
+ audio_format_init(audio_format, sample_rate, 16, channels);
if (!audio_format_valid(audio_format)) {
g_warning("Invalid audio format: %u:%u:%u\n",
@@ -395,22 +391,22 @@ mp4_tag_dup(const char *file)
mp4ff_meta_get_by_index(mp4fh, i, &item, &value);
if (0 == g_ascii_strcasecmp("artist", item)) {
- tag_add_item(ret, TAG_ITEM_ARTIST, value);
+ tag_add_item(ret, TAG_ARTIST, value);
} else if (0 == g_ascii_strcasecmp("title", item)) {
- tag_add_item(ret, TAG_ITEM_TITLE, value);
+ tag_add_item(ret, TAG_TITLE, value);
} else if (0 == g_ascii_strcasecmp("album", item)) {
- tag_add_item(ret, TAG_ITEM_ALBUM, value);
+ tag_add_item(ret, TAG_ALBUM, value);
} else if (0 == g_ascii_strcasecmp("track", item)) {
- tag_add_item(ret, TAG_ITEM_TRACK, value);
+ tag_add_item(ret, TAG_TRACK, value);
} else if (0 == g_ascii_strcasecmp("disc", item)) {
/* Is that the correct id? */
- tag_add_item(ret, TAG_ITEM_DISC, value);
+ tag_add_item(ret, TAG_DISC, value);
} else if (0 == g_ascii_strcasecmp("genre", item)) {
- tag_add_item(ret, TAG_ITEM_GENRE, value);
+ tag_add_item(ret, TAG_GENRE, value);
} else if (0 == g_ascii_strcasecmp("date", item)) {
- tag_add_item(ret, TAG_ITEM_DATE, value);
+ tag_add_item(ret, TAG_DATE, value);
} else if (0 == g_ascii_strcasecmp("writer", item)) {
- tag_add_item(ret, TAG_ITEM_COMPOSER, value);
+ tag_add_item(ret, TAG_COMPOSER, value);
}
free(item);
diff --git a/src/decoder/mpcdec_plugin.c b/src/decoder/mpcdec_plugin.c
index 26349f93a..a684da104 100644
--- a/src/decoder/mpcdec_plugin.c
+++ b/src/decoder/mpcdec_plugin.c
@@ -193,9 +193,7 @@ mpcdec_decode(struct decoder *mpd_decoder, struct input_stream *is)
mpc_demux_get_info(demux, &info);
#endif
- audio_format.bits = 24;
- audio_format.channels = info.channels;
- audio_format.sample_rate = info.sample_freq;
+ audio_format_init(&audio_format, info.sample_freq, 24, info.channels);
if (!audio_format_valid(&audio_format)) {
#ifndef MPC_IS_OLD_API
diff --git a/src/decoder/mpg123_decoder_plugin.c b/src/decoder/mpg123_decoder_plugin.c
new file mode 100644
index 000000000..20d9c4a54
--- /dev/null
+++ b/src/decoder/mpg123_decoder_plugin.c
@@ -0,0 +1,211 @@
+/*
+ * Copyright (C) 2003-2009 The Music Player Daemon Project
+ * http://www.musicpd.org
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with this program; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include "config.h" /* must be first for large file support */
+#include "decoder_api.h"
+
+#include <glib.h>
+
+#include <mpg123.h>
+
+#undef G_LOG_DOMAIN
+#define G_LOG_DOMAIN "mpg123"
+
+static bool
+mpd_mpg123_init(G_GNUC_UNUSED const struct config_param *param)
+{
+ mpg123_init();
+
+ return true;
+}
+
+static void
+mpd_mpg123_finish(void)
+{
+ mpg123_exit();
+}
+
+/**
+ * Opens a file with an existing #mpg123_handle.
+ *
+ * @param handle a handle which was created before; on error, this
+ * function will not free it
+ * @param audio_format this parameter is filled after successful
+ * return
+ * @return true on success
+ */
+static bool
+mpd_mpg123_open(mpg123_handle *handle, const char *path_fs,
+ struct audio_format *audio_format)
+{
+ char *path_dup;
+ int error;
+ int channels, encoding;
+ long rate;
+
+ /* mpg123_open() wants a writable string :-( */
+ path_dup = g_strdup(path_fs);
+
+ error = mpg123_open(handle, path_dup);
+ g_free(path_dup);
+ if (error != MPG123_OK) {
+ g_warning("libmpg123 failed to open %s: %s",
+ path_fs, mpg123_plain_strerror(error));
+ return false;
+ }
+
+ /* obtain the audio format */
+
+ error = mpg123_getformat(handle, &rate, &channels, &encoding);
+ if (error != MPG123_OK) {
+ g_warning("mpg123_getformat() failed: %s",
+ mpg123_plain_strerror(error));
+ return false;
+ }
+
+ if (encoding != MPG123_ENC_SIGNED_16) {
+ /* other formats not yet implemented */
+ g_warning("expected MPG123_ENC_SIGNED_16, got %d", encoding);
+ return false;
+ }
+
+ audio_format_init(audio_format, rate, 16, channels);
+ if (!audio_format_valid(audio_format)) {
+ g_warning("invalid audio format");
+ return false;
+ }
+
+ return true;
+}
+
+static void
+mpd_mpg123_file_decode(struct decoder *decoder, const char *path_fs)
+{
+ struct audio_format audio_format;
+ mpg123_handle *handle;
+ int error;
+ off_t num_samples, position;
+ enum decoder_command cmd;
+
+ /* open the file */
+
+ handle = mpg123_new(NULL, &error);
+ if (handle == NULL) {
+ g_warning("mpg123_new() failed: %s",
+ mpg123_plain_strerror(error));
+ return;
+ }
+
+ if (!mpd_mpg123_open(handle, path_fs, &audio_format)) {
+ mpg123_delete(handle);
+ return;
+ }
+
+ num_samples = mpg123_length(handle);
+
+ /* tell MPD core we're ready */
+
+ decoder_initialized(decoder, &audio_format, false,
+ (float)num_samples /
+ (float)audio_format.sample_rate);
+
+ /* the decoder main loop */
+
+ do {
+ unsigned char buffer[8192];
+ size_t nbytes;
+
+ position = mpg123_tell(handle);
+
+ /* decode */
+
+ error = mpg123_read(handle, buffer, sizeof(buffer), &nbytes);
+ if (error != MPG123_OK) {
+ if (error != MPG123_DONE)
+ g_warning("mpg123_read() failed: %s",
+ mpg123_plain_strerror(error));
+ break;
+ }
+
+ /* send to MPD */
+
+ cmd = decoder_data(decoder, NULL, buffer, nbytes,
+ (float)position /
+ (float)audio_format.sample_rate,
+ 0, NULL);
+
+ /* seeking not yet implemented */
+ } while (cmd == DECODE_COMMAND_NONE);
+
+ /* cleanup */
+
+ mpg123_delete(handle);
+}
+
+static struct tag *
+mpd_mpg123_tag_dup(const char *path_fs)
+{
+ struct audio_format audio_format;
+ mpg123_handle *handle;
+ int error;
+ off_t num_samples;
+ struct tag *tag;
+
+ handle = mpg123_new(NULL, &error);
+ if (handle == NULL) {
+ g_warning("mpg123_new() failed: %s",
+ mpg123_plain_strerror(error));
+ return NULL;
+ }
+
+ if (!mpd_mpg123_open(handle, path_fs, &audio_format)) {
+ mpg123_delete(handle);
+ return NULL;
+ }
+
+ num_samples = mpg123_length(handle);
+ if (num_samples <= 0) {
+ mpg123_delete(handle);
+ return NULL;
+ }
+
+ tag = tag_new();
+
+ tag->time = num_samples / audio_format.sample_rate;
+
+ /* ID3 tag support not yet implemented */
+
+ mpg123_delete(handle);
+ return tag;
+}
+
+static const char *const mpg123_suffixes[] = {
+ "mp3",
+ NULL
+};
+
+const struct decoder_plugin mpg123_decoder_plugin = {
+ .name = "mpg123",
+ .init = mpd_mpg123_init,
+ .finish = mpd_mpg123_finish,
+ .file_decode = mpd_mpg123_file_decode,
+ /* streaming not yet implemented */
+ .tag_dup = mpd_mpg123_tag_dup,
+ .suffixes = mpg123_suffixes,
+};
diff --git a/src/decoder/oggflac_plugin.c b/src/decoder/oggflac_plugin.c
index c0e7e35e1..d4ba7b213 100644
--- a/src/decoder/oggflac_plugin.c
+++ b/src/decoder/oggflac_plugin.c
@@ -23,17 +23,15 @@
#include "_flac_common.h"
#include "_ogg_common.h"
+#include "flac_metadata.h"
#include <glib.h>
#include <OggFLAC/seekable_stream_decoder.h>
#include <assert.h>
#include <unistd.h>
-static void oggflac_cleanup(struct flac_data *data,
- OggFLAC__SeekableStreamDecoder * decoder)
+static void oggflac_cleanup(OggFLAC__SeekableStreamDecoder * decoder)
{
- if (data->replay_gain_info)
- replay_gain_info_free(data->replay_gain_info);
if (decoder)
OggFLAC__seekable_stream_decoder_delete(decoder);
}
@@ -173,17 +171,7 @@ static void of_metadata_dup_cb(G_GNUC_UNUSED const OggFLAC__SeekableStreamDecode
assert(data->tag != NULL);
- switch (block->type) {
- case FLAC__METADATA_TYPE_STREAMINFO:
- 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:
- flac_vorbis_comments_to_tag(data->tag, NULL, block);
- default:
- break;
- }
+ flac_tag_apply_metadata(data->tag, NULL, block);
}
/* used by decode */
@@ -264,6 +252,7 @@ oggflac_tag_dup(const char *file)
struct input_stream input_stream;
OggFLAC__SeekableStreamDecoder *decoder;
struct flac_data data;
+ struct tag *tag;
if (!input_stream_open(&input_stream, file))
return NULL;
@@ -284,15 +273,18 @@ oggflac_tag_dup(const char *file)
* 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);
+ oggflac_cleanup(decoder);
input_stream_close(&input_stream);
- if (!tag_is_defined(data.tag)) {
- tag_free(data.tag);
+ if (tag_is_defined(data.tag)) {
+ tag = data.tag;
data.tag = NULL;
- }
+ } else
+ tag = NULL;
+
+ flac_data_deinit(&data);
- return data.tag;
+ return tag;
}
static void
@@ -352,7 +344,8 @@ oggflac_decode(struct decoder * mpd_decoder, struct input_stream *input_stream)
}
fail:
- oggflac_cleanup(&data, decoder);
+ oggflac_cleanup(decoder);
+ flac_data_deinit(&data);
}
static const char *const oggflac_suffixes[] = { "ogg", "oga", NULL };
diff --git a/src/decoder/sidplay_plugin.cxx b/src/decoder/sidplay_plugin.cxx
index c62e6b4b6..079b178fb 100644
--- a/src/decoder/sidplay_plugin.cxx
+++ b/src/decoder/sidplay_plugin.cxx
@@ -21,14 +21,180 @@ extern "C" {
#include "../decoder_api.h"
}
+#include <errno.h>
+#include <stdlib.h>
#include <glib.h>
#include <sidplay/sidplay2.h>
#include <sidplay/builders/resid.h>
+#include <sidplay/utils/SidTuneMod.h>
#undef G_LOG_DOMAIN
#define G_LOG_DOMAIN "sidplay"
+#define SUBTUNE_PREFIX "tune_"
+
+static GPatternSpec *path_with_subtune;
+static const char *songlength_file;
+static GKeyFile *songlength_database;
+
+static bool all_files_are_containers;
+static unsigned default_songlength;
+
+static bool filter_setting;
+
+static GKeyFile *
+sidplay_load_songlength_db(const char *path)
+{
+ GError *error = NULL;
+ gchar *data;
+ gsize size;
+
+ if (!g_file_get_contents(path, &data, &size, &error)) {
+ g_warning("unable to read songlengths file %s: %s",
+ path, error->message);
+ g_error_free(error);
+ return NULL;
+ }
+
+ /* replace any ; comment characters with # */
+ for (gsize i = 0; i < size; i++)
+ if (data[i] == ';')
+ data[i] = '#';
+
+ GKeyFile *db = g_key_file_new();
+ bool success = g_key_file_load_from_data(db, data, size,
+ G_KEY_FILE_NONE, &error);
+ g_free(data);
+ if (!success) {
+ g_warning("unable to parse songlengths file %s: %s",
+ path, error->message);
+ g_error_free(error);
+ g_key_file_free(db);
+ return NULL;
+ }
+
+ g_key_file_set_list_separator(db, ' ');
+ return db;
+}
+
+static bool
+sidplay_init(const struct config_param *param)
+{
+ /* read the songlengths database file */
+ songlength_file=config_get_block_string(param,
+ "songlength_database", NULL);
+ if (songlength_file != NULL)
+ songlength_database = sidplay_load_songlength_db(songlength_file);
+
+ default_songlength=config_get_block_unsigned(param,
+ "default_songlength", 0);
+
+ all_files_are_containers=config_get_block_bool(param,
+ "all_files_are_containers", true);
+
+ path_with_subtune=g_pattern_spec_new(
+ "*/" SUBTUNE_PREFIX "???.sid");
+
+ filter_setting=config_get_block_bool(param, "filter", true);
+
+ return true;
+}
+
+void
+sidplay_finish()
+{
+ g_pattern_spec_free(path_with_subtune);
+
+ if(songlength_database)
+ g_key_file_free(songlength_database);
+}
+
+/**
+ * returns the file path stripped of any /tune_xxx.sid subtune
+ * suffix
+ */
+static char *
+get_container_name(const char *path_fs)
+{
+ char *path_container=g_strdup(path_fs);
+
+ if(!g_pattern_match(path_with_subtune,
+ strlen(path_container), path_container, NULL))
+ return path_container;
+
+ char *ptr=g_strrstr(path_container, "/" SUBTUNE_PREFIX);
+ if(ptr) *ptr='\0';
+
+ return path_container;
+}
+
+/**
+ * returns tune number from file.sid/tune_xxx.sid style path or 1 if
+ * no subtune is appended
+ */
+static int
+get_song_num(const char *path_fs)
+{
+ if(g_pattern_match(path_with_subtune,
+ strlen(path_fs), path_fs, NULL)) {
+ char *sub=g_strrstr(path_fs, "/" SUBTUNE_PREFIX);
+ if(!sub) return 1;
+
+ sub+=strlen("/" SUBTUNE_PREFIX);
+ int song_num=strtol(sub, NULL, 10);
+
+ if (errno == EINVAL)
+ return 1;
+ else
+ return song_num;
+ } else
+ return 1;
+}
+
+/* get the song length in seconds */
+static int
+get_song_length(const char *path_fs)
+{
+ if (songlength_database == NULL)
+ return -1;
+
+ gchar *sid_file=get_container_name(path_fs);
+ SidTuneMod tune(sid_file);
+ g_free(sid_file);
+ if(!tune) {
+ g_warning("failed to load file for calculating md5 sum");
+ return -1;
+ }
+ char md5sum[SIDTUNE_MD5_LENGTH+1];
+ tune.createMD5(md5sum);
+
+ int song_num=get_song_num(path_fs);
+
+ gsize num_items;
+ gchar **values=g_key_file_get_string_list(songlength_database,
+ "Database", md5sum, &num_items, NULL);
+ if(!values || song_num>num_items) {
+ g_strfreev(values);
+ return -1;
+ }
+
+ int minutes=strtol(values[song_num-1], NULL, 10);
+ if(errno==EINVAL) minutes=0;
+
+ int seconds;
+ char *ptr=strchr(values[song_num-1], ':');
+ if(ptr) {
+ seconds=strtol(ptr+1, NULL, 10);
+ if(errno==EINVAL) seconds=0;
+ } else
+ seconds=0;
+
+ g_strfreev(values);
+
+ return (minutes*60)+seconds;
+}
+
static void
sidplay_file_decode(struct decoder *decoder, const char *path_fs)
{
@@ -36,13 +202,19 @@ sidplay_file_decode(struct decoder *decoder, const char *path_fs)
/* load the tune */
- SidTune tune(path_fs, NULL, true);
+ char *path_container=get_container_name(path_fs);
+ SidTune tune(path_container, NULL, true);
+ g_free(path_container);
if (!tune) {
g_warning("failed to load file");
return;
}
- tune.selectSong(1);
+ int song_num=get_song_num(path_fs);
+ tune.selectSong(song_num);
+
+ int song_len=get_song_length(path_fs);
+ if(song_len==-1) song_len=default_songlength;
/* initialize the player */
@@ -67,7 +239,7 @@ sidplay_file_decode(struct decoder *decoder, const char *path_fs)
return;
}
- builder.filter(false);
+ builder.filter(filter_setting);
if (!builder) {
g_warning("ReSIDBuilder.filter() failed");
return;
@@ -103,14 +275,14 @@ sidplay_file_decode(struct decoder *decoder, const char *path_fs)
/* initialize the MPD decoder */
struct audio_format audio_format;
- audio_format.sample_rate = 48000;
- audio_format.bits = 16;
- audio_format.channels = 2;
+ audio_format_init(&audio_format, 48000, 16, 2);
- decoder_initialized(decoder, &audio_format, false, -1);
+ decoder_initialized(decoder, &audio_format, true, (float)song_len);
/* .. and play */
+ float data_time=0;
+ int timebase=player.timebase();
enum decoder_command cmd;
do {
char buffer[4096];
@@ -121,29 +293,105 @@ sidplay_file_decode(struct decoder *decoder, const char *path_fs)
break;
cmd = decoder_data(decoder, NULL, buffer, nbytes,
- 0, 0, NULL);
- } while (cmd == DECODE_COMMAND_NONE);
+ data_time, 0, NULL);
+
+ data_time=player.time()/timebase;
+
+ if(cmd==DECODE_COMMAND_SEEK) {
+ int target_time=decoder_seek_where(decoder);
+
+ /* can't rewind so return to zero and seek forward */
+ if(target_time<data_time) {
+ player.stop();
+ data_time=0;
+ }
+
+ /* ignore data until target time is reached */
+ while(data_time<target_time) {
+ nbytes=player.play(buffer, sizeof(buffer));
+ if(nbytes==0)
+ break;
+ data_time=player.time()/timebase;
+ }
+
+ decoder_command_finished(decoder);
+ }
+
+ if(song_len && data_time>=(float)song_len)
+ break;
+
+ } while (cmd != DECODE_COMMAND_STOP);
}
static struct tag *
sidplay_tag_dup(const char *path_fs)
{
- SidTune tune(path_fs, NULL, true);
+ int song_num=get_song_num(path_fs);
+ char *path_container=get_container_name(path_fs);
+
+ SidTune tune(path_container, NULL, true);
+ g_free(path_container);
if (!tune)
return NULL;
const SidTuneInfo &info = tune.getInfo();
struct tag *tag = tag_new();
+ /* title */
+ const char *title;
if (info.numberOfInfoStrings > 0 && info.infoString[0] != NULL)
- tag_add_item(tag, TAG_ITEM_TITLE, info.infoString[0]);
-
+ title=info.infoString[0];
+ else
+ title="";
+
+ if(info.songs>1) {
+ char *tag_title=g_strdup_printf("%s (%d/%d)",
+ title, song_num, info.songs);
+ tag_add_item(tag, TAG_TITLE, tag_title);
+ g_free(tag_title);
+ } else
+ tag_add_item(tag, TAG_TITLE, title);
+
+ /* artist */
if (info.numberOfInfoStrings > 1 && info.infoString[1] != NULL)
- tag_add_item(tag, TAG_ITEM_ARTIST, info.infoString[1]);
+ tag_add_item(tag, TAG_ARTIST, info.infoString[1]);
+
+ /* track */
+ char *track=g_strdup_printf("%d", song_num);
+ tag_add_item(tag, TAG_TRACK, track);
+ g_free(track);
+
+ /* time */
+ int song_len=get_song_length(path_fs);
+ if(song_len!=-1) tag->time=song_len;
return tag;
}
+static char *
+sidplay_container_scan(const char *path_fs, const unsigned int tnum)
+{
+ SidTune tune(path_fs, NULL, true);
+ if (!tune)
+ return NULL;
+
+ const SidTuneInfo &info=tune.getInfo();
+
+ /* Don't treat sids containing a single tune
+ as containers */
+ if(!all_files_are_containers && info.songs<2)
+ return NULL;
+
+ /* Construct container/tune path names, eg.
+ Delta.sid/tune_001.sid */
+ if(tnum<=info.songs) {
+ char *subtune= g_strdup_printf(
+ SUBTUNE_PREFIX "%03u.sid", tnum);
+ return subtune;
+ } else
+ return NULL;
+}
+
static const char *const sidplay_suffixes[] = {
"sid",
NULL
@@ -152,12 +400,12 @@ static const char *const sidplay_suffixes[] = {
extern const struct decoder_plugin sidplay_decoder_plugin;
const struct decoder_plugin sidplay_decoder_plugin = {
"sidplay",
- NULL, /* init() */
- NULL, /* finish() */
+ sidplay_init,
+ sidplay_finish,
NULL, /* stream_decode() */
sidplay_file_decode,
sidplay_tag_dup,
- NULL, /* container_scan */
+ sidplay_container_scan,
sidplay_suffixes,
NULL, /* mime_types */
};
diff --git a/src/decoder/sndfile_decoder_plugin.c b/src/decoder/sndfile_decoder_plugin.c
new file mode 100644
index 000000000..59993ef89
--- /dev/null
+++ b/src/decoder/sndfile_decoder_plugin.c
@@ -0,0 +1,244 @@
+/*
+ * Copyright (C) 2003-2009 The Music Player Daemon Project
+ * http://www.musicpd.org
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with this program; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include "decoder_api.h"
+
+#include <sndfile.h>
+
+#undef G_LOG_DOMAIN
+#define G_LOG_DOMAIN "sndfile"
+
+static sf_count_t
+sndfile_vio_get_filelen(void *user_data)
+{
+ const struct input_stream *is = user_data;
+
+ return is->size;
+}
+
+static sf_count_t
+sndfile_vio_seek(sf_count_t offset, int whence, void *user_data)
+{
+ struct input_stream *is = user_data;
+ bool success;
+
+ success = input_stream_seek(is, offset, whence);
+ if (!success)
+ return -1;
+
+ return is->offset;
+}
+
+static sf_count_t
+sndfile_vio_read(void *ptr, sf_count_t count, void *user_data)
+{
+ struct input_stream *is = user_data;
+ size_t nbytes;
+
+ nbytes = input_stream_read(is, ptr, count);
+ if (nbytes == 0 && is->error != 0)
+ return -1;
+
+ return nbytes;
+}
+
+static sf_count_t
+sndfile_vio_write(G_GNUC_UNUSED const void *ptr,
+ G_GNUC_UNUSED sf_count_t count,
+ G_GNUC_UNUSED void *user_data)
+{
+ /* no writing! */
+ return -1;
+}
+
+static sf_count_t
+sndfile_vio_tell(void *user_data)
+{
+ const struct input_stream *is = user_data;
+
+ return is->offset;
+}
+
+/**
+ * This SF_VIRTUAL_IO implementation wraps MPD's #input_stream to a
+ * libsndfile stream.
+ */
+static SF_VIRTUAL_IO vio = {
+ .get_filelen = sndfile_vio_get_filelen,
+ .seek = sndfile_vio_seek,
+ .read = sndfile_vio_read,
+ .write = sndfile_vio_write,
+ .tell = sndfile_vio_tell,
+};
+
+/**
+ * Converts a frame number to a timestamp (in seconds).
+ */
+static float
+frame_to_time(sf_count_t frame, const struct audio_format *audio_format)
+{
+ return (float)frame / (float)audio_format->sample_rate;
+}
+
+/**
+ * Converts a timestamp (in seconds) to a frame number.
+ */
+static sf_count_t
+time_to_frame(float t, const struct audio_format *audio_format)
+{
+ return (sf_count_t)(t * audio_format->sample_rate);
+}
+
+static void
+sndfile_stream_decode(struct decoder *decoder, struct input_stream *is)
+{
+ SNDFILE *sf;
+ SF_INFO info;
+ struct audio_format audio_format;
+ size_t frame_size;
+ sf_count_t read_frames, num_frames, position = 0;
+ int buffer[4096];
+ enum decoder_command cmd;
+
+ info.format = 0;
+
+ sf = sf_open_virtual(&vio, SFM_READ, &info, is);
+ if (sf == NULL) {
+ g_warning("sf_open_virtual() failed");
+ return;
+ }
+
+ /* for now, always read 32 bit samples. Later, we could lower
+ MPD's CPU usage by reading 16 bit samples with
+ sf_readf_short() on low-quality source files. */
+ audio_format_init(&audio_format, info.samplerate, 32, info.channels);
+
+ if (!audio_format_valid(&audio_format)) {
+ g_warning("invalid audio format");
+ return;
+ }
+
+ decoder_initialized(decoder, &audio_format, info.seekable,
+ frame_to_time(info.frames, &audio_format));
+
+ frame_size = audio_format_frame_size(&audio_format);
+ read_frames = sizeof(buffer) / frame_size;
+
+ do {
+ num_frames = sf_readf_int(sf, buffer, read_frames);
+ if (num_frames <= 0)
+ break;
+
+ cmd = decoder_data(decoder, is,
+ buffer, num_frames * frame_size,
+ frame_to_time(position, &audio_format),
+ 0, NULL);
+ if (cmd == DECODE_COMMAND_SEEK) {
+ sf_count_t c =
+ time_to_frame(decoder_seek_where(decoder),
+ &audio_format);
+ c = sf_seek(sf, c, SEEK_SET);
+ if (c < 0)
+ decoder_seek_error(decoder);
+ else
+ decoder_command_finished(decoder);
+ cmd = DECODE_COMMAND_NONE;
+ }
+ } while (cmd == DECODE_COMMAND_NONE);
+
+ sf_close(sf);
+}
+
+static struct tag *
+sndfile_tag_dup(const char *path_fs)
+{
+ SNDFILE *sf;
+ SF_INFO info;
+ struct tag *tag;
+ const char *p;
+
+ info.format = 0;
+
+ sf = sf_open(path_fs, SFM_READ, &info);
+ if (sf == NULL)
+ return NULL;
+
+ if (!audio_valid_sample_rate(info.samplerate)) {
+ sf_close(sf);
+ g_warning("Invalid sample rate in %s\n", path_fs);
+ return NULL;
+ }
+
+ tag = tag_new();
+ tag->time = info.frames / info.samplerate;
+
+ p = sf_get_string(sf, SF_STR_TITLE);
+ if (p != NULL)
+ tag_add_item(tag, TAG_TITLE, p);
+
+ p = sf_get_string(sf, SF_STR_ARTIST);
+ if (p != NULL)
+ tag_add_item(tag, TAG_ARTIST, p);
+
+ p = sf_get_string(sf, SF_STR_DATE);
+ if (p != NULL)
+ tag_add_item(tag, TAG_DATE, p);
+
+ sf_close(sf);
+
+ return tag;
+}
+
+static const char *const sndfile_suffixes[] = {
+ "wav", "aiff", "aif", /* Microsoft / SGI / Apple */
+ "au", "snd", /* Sun / DEC / NeXT */
+ "paf", /* Paris Audio File */
+ "iff", "svx", /* Commodore Amiga IFF / SVX */
+ "sf", /* IRCAM */
+ "voc", /* Creative */
+ "w64", /* Soundforge */
+ "pvf", /* Portable Voice Format */
+ "xi", /* Fasttracker */
+ "htk", /* HMM Tool Kit */
+ "caf", /* Apple */
+ "sd2", /* Sound Designer II */
+
+ /* libsndfile also supports FLAC and Ogg Vorbis, but only by
+ linking with libFLAC and libvorbis - we can do better, we
+ have native plugins for these libraries */
+
+ NULL
+};
+
+static const char *const sndfile_mime_types[] = {
+ "audio/x-wav",
+ "audio/x-aiff",
+
+ /* what are the MIME types of the other supported formats? */
+
+ NULL
+};
+
+const struct decoder_plugin sndfile_decoder_plugin = {
+ .name = "sndfile",
+ .stream_decode = sndfile_stream_decode,
+ .tag_dup = sndfile_tag_dup,
+ .suffixes = sndfile_suffixes,
+ .mime_types = sndfile_mime_types,
+};
diff --git a/src/decoder/vorbis_plugin.c b/src/decoder/vorbis_plugin.c
index d4f81e91f..2ac9c3bcb 100644..100755
--- a/src/decoder/vorbis_plugin.c
+++ b/src/decoder/vorbis_plugin.c
@@ -24,6 +24,7 @@
#include "uri.h"
#ifndef HAVE_TREMOR
+#define OV_EXCLUDE_STATIC_CALLBACKS
#include <vorbis/vorbisfile.h>
#else
#include <tremor/ivorbisfile.h>
@@ -176,11 +177,11 @@ vorbis_parse_comment(struct tag *tag, const char *comment)
assert(tag != NULL);
if (vorbis_copy_comment(tag, comment, VORBIS_COMMENT_TRACK_KEY,
- TAG_ITEM_TRACK) ||
+ TAG_TRACK) ||
vorbis_copy_comment(tag, comment, VORBIS_COMMENT_DISC_KEY,
- TAG_ITEM_DISC) ||
+ TAG_DISC) ||
vorbis_copy_comment(tag, comment, "album artist",
- TAG_ITEM_ALBUM_ARTIST))
+ TAG_ALBUM_ARTIST))
return;
for (unsigned i = 0; i < TAG_NUM_OF_ITEM_TYPES; ++i)
@@ -324,8 +325,7 @@ vorbis_stream_decode(struct decoder *decoder,
vorbis_info *vi = ov_info(&vf, -1);
struct replay_gain_info *new_rgi;
- audio_format.channels = vi->channels;
- audio_format.sample_rate = vi->rate;
+ audio_format_init(&audio_format, vi->rate, 16, vi->channels);
if (!audio_format_valid(&audio_format)) {
g_warning("Invalid audio format: %u:%u:%u\n",
@@ -378,7 +378,7 @@ vorbis_tag_dup(const char *file)
FILE *fp;
OggVorbis_File vf;
- fp = fopen(file, "r");
+ fp = fopen(file, "rb");
if (!fp) {
return NULL;
}
diff --git a/src/decoder/wavpack_plugin.c b/src/decoder/wavpack_plugin.c
index 821536fb5..3e7c0cef7 100644
--- a/src/decoder/wavpack_plugin.c
+++ b/src/decoder/wavpack_plugin.c
@@ -41,17 +41,17 @@ static struct {
const char *name;
enum tag_type 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 },
+ { "artist", TAG_ARTIST },
+ { "album", TAG_ALBUM },
+ { "title", TAG_TITLE },
+ { "track", TAG_TRACK },
+ { "name", TAG_NAME },
+ { "genre", TAG_GENRE },
+ { "date", TAG_DATE },
+ { "composer", TAG_COMPOSER },
+ { "performer", TAG_PERFORMER },
+ { "comment", TAG_COMMENT },
+ { "disc", TAG_DISC },
};
/** A pointer type for format converter function. */
@@ -145,9 +145,9 @@ wavpack_decode(struct decoder *decoder, WavpackContext *wpc, bool can_seek,
int bytes_per_sample, output_sample_size;
int position;
- audio_format.sample_rate = WavpackGetSampleRate(wpc);
- audio_format.channels = WavpackGetReducedChannels(wpc);
- audio_format.bits = WavpackGetBitsPerSample(wpc);
+ audio_format_init(&audio_format, WavpackGetSampleRate(wpc),
+ WavpackGetBitsPerSample(wpc),
+ WavpackGetReducedChannels(wpc));
/* round bitwidth to 8-bit units */
audio_format.bits = (audio_format.bits + 7) & (~7);
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-15 19:20:29 +0000