/* 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 "../decoder_api.h"
#if defined(HAVE_FLAC) || defined(HAVE_OGGFLAC)
#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->chunk_length = 0;
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 = (mpd_sint8)si->bits_per_sample;
data->audio_format.sampleRate = si->sample_rate;
data->audio_format.channels = (mpd_sint8)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);
}
}
/* keep this inlined, this is just macro but prettier :) */
static inline int flacSendChunk(FlacData * data)
{
if (decoder_data(data->decoder, data->inStream,
1, data->chunk,
data->chunk_length, data->time,
data->bitRate,
data->replayGainInfo) == DECODE_COMMAND_STOP)
return -1;
return 0;
}
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)
{
unsigned int c_chan, i;
FLAC__uint16 u16;
unsigned char *uc;
for (; position < end; ++position) {
for (c_chan = 0; c_chan < num_channels; c_chan++) {
u16 = buf[c_chan][position];
uc = (unsigned char *)&u16;
for (i = 0; i < bytes_per_sample; i++) {
*dest++ = *uc++;
}
}
}
}
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;
assert(data->audio_format.bits > 0);
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;
if (num_channels == 2 && bytes_per_sample == 2)
flac_convert_stereo16((int16_t*)data->chunk,
buf, c_samp,
c_samp + num_samples);
else if (bytes_per_sample == 2)
flac_convert_16((int16_t*)data->chunk,
num_channels, buf, c_samp,
c_samp + num_samples);
else if (bytes_per_sample == 4)
flac_convert_32((int32_t*)data->chunk,
num_channels, buf, c_samp,
c_samp + num_samples);
else if (bytes_per_sample == 1)
flac_convert_8((int8_t*)data->chunk,
num_channels, buf, c_samp,
c_samp + num_samples);
else
flac_convert(data->chunk,
num_channels, bytes_per_sample, buf,
c_samp, c_samp + num_samples);
data->chunk_length = num_samples * bytes_per_channel;
if (flacSendChunk(data) < 0) {
return
FLAC__STREAM_DECODER_WRITE_STATUS_ABORT;
}
data->chunk_length = 0;
if (decoder_get_command(data->decoder) == DECODE_COMMAND_SEEK) {
return
FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE;
}
}
return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE;
}
#endif /* HAVE_FLAC || HAVE_OGGFLAC */