diff libmpcodecs/ad_flac.c @ 11004:d48eccbbb984

FLAC decoding support via imported libmpflac. TODO: fix FLAC-in-ogg decoding.

--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/libmpcodecs/ad_flac.c	Sat Oct 04 22:00:25 2003 +0000
@@ -0,0 +1,538 @@
+/*
+ * This is FLAC decoder for MPlayer using stream_decoder from libFLAC
+ * (directly or from libmpflac).
+ * This file is part of MPlayer, see http://mplayerhq.hu/ for info.  
+ * Copyright (C) 2003  Dmitry Baryshkov <mitya at school.ioffe.ru>
+ * 
+ * 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
+ *
+ * parse_double_, grabbag__replaygain_load_from_vorbiscomment, grabbag__replaygain_compute_scale_factor
+ * functions are imported from FLAC project (from grabbag lib sources (replaygain.c)) and are
+ * Copyright (C) 2002,2003  Josh Coalson under the terms of GPL.
+ */
+
+/*
+ * TODO:
+ * in demux_audio use data from seektable block for seeking.
+ * support FLAC-in-Ogg.
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <math.h>
+
+#include "config.h"
+#ifdef HAVE_FLAC
+#include "ad_internal.h"
+#include "mp_msg.h"
+
+static ad_info_t info =  {
+	"FLAC audio decoder",  // name of the driver
+	"flac",    // driver name. should be the same as filename without ad_
+	"Dmitry Baryshkov",     // writer/maintainer of _this_ file
+	"http://flac.sf.net/",          // writer/maintainer/site of the _codec_
+	""           // comments
+};
+
+LIBAD_EXTERN(flac)
+
+#ifdef USE_MPFLAC_DECODER
+#include "FLAC_stream_decoder.h"
+#include "FLAC_assert.h"
+#include "FLAC_metadata.h"
+#else
+#include "FLAC/stream_decoder.h"
+#include "FLAC/assert.h"
+#include "FLAC/metadata.h"
+#endif
+
+/* dithering & replaygain always from libmpflac */
+#include "dither.h"
+#include "replaygain_synthesis.h"
+
+/* Some global constants. Thay have to be configurable, so leaved them as globals. */
+static const FLAC__bool album_mode = true;
+static const int preamp = 0;
+static const FLAC__bool hard_limit = false;
+static const int noise_shaping = 1;
+static const FLAC__bool dither = true;
+typedef struct flac_struct_st
+{
+	FLAC__StreamDecoder *flac_dec; /*decoder handle*/
+	sh_audio_t *sh; /* link back to corresponding sh */
+	
+	/* set this fields before calling FLAC__stream_decoder_process_single */
+	unsigned char *buf; 
+	int minlen;
+	int maxlen;
+	/* Here goes number written at write_callback */
+	int written;
+
+	/* replaygain and dithering via plugin_common */
+	FLAC__bool has_replaygain;
+	double replay_scale;
+	DitherContext dither_context;
+	int bits_per_sample;
+} flac_struct_t;
+
+FLAC__StreamDecoderReadStatus flac_read_callback (const FLAC__StreamDecoder *decoder, FLAC__byte buffer[], unsigned *bytes, void *client_data)
+{
+	int b = demux_read_data(((flac_struct_t*)client_data)->sh->ds, buffer,  *bytes);
+	mp_msg(MSGT_DECAUDIO, MSGL_DBG2, "\nread %d bytes\n", b);
+	*bytes = b;
+	if (b <= 0)
+		return FLAC__STREAM_DECODER_READ_STATUS_END_OF_STREAM;
+	return FLAC__STREAM_DECODER_READ_STATUS_CONTINUE;
+}
+
+/*FIXME: we need to support format conversion:(flac specs allow bits/sample to be from 4 to 32. Not only 8 and 16 !!!)*/
+FLAC__StreamDecoderWriteStatus flac_write_callback(const FLAC__StreamDecoder *decoder, const FLAC__Frame *frame, const FLAC__int32 * const buffer[], void *client_data)
+{
+	FLAC__byte *buf = ((flac_struct_t*)(client_data))->buf;
+	int channel, sample;
+	int bps = ((flac_struct_t*)(client_data))->sh->samplesize;
+	mp_msg(MSGT_DECAUDIO, MSGL_DBG2, "\nWrite callback (%d bytes)!!!!\n", bps*frame->header.blocksize*frame->header.channels);
+	if (buf == NULL)
+	{
+		/* This is used in control for skipping 1 audio frame */
+		return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE;
+	}
+#if 0
+	for (sample = 0; sample < frame->header.blocksize; sample ++)
+		for (channel = 0; channel < frame->header.channels; channel ++)
+			switch (bps)
+			{
+				case 3:
+					buf[bps*(sample*frame->header.channels+channel)+2] = (FLAC__byte)(buffer[channel][sample]>>16);
+				case 2:
+					buf[bps*(sample*frame->header.channels+channel)+1] = (FLAC__byte)(buffer[channel][sample]>>8);
+					buf[bps*(sample*frame->header.channels+channel)+0] = (FLAC__byte)(buffer[channel][sample]);
+					break;
+				case 1:
+					buf[bps*(sample*frame->header.channels+channel)] = buffer[channel][sample]^0x80;
+					break;
+			}
+#else
+	FLAC__plugin_common__apply_gain(
+				buf,
+				buffer,
+				frame->header.blocksize,
+				frame->header.channels,
+				((flac_struct_t*)(client_data))->bits_per_sample,
+				((flac_struct_t*)(client_data))->sh->samplesize * 8,
+				((flac_struct_t*)(client_data))->replay_scale,
+				hard_limit,
+				dither,
+				&(((flac_struct_t*)(client_data))->dither_context)
+		);
+#endif
+	((flac_struct_t*)(client_data))->written += bps*frame->header.blocksize*frame->header.channels;
+	return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE;
+}
+
+#ifdef local_min
+#undef local_min
+#endif
+#define local_min(a,b) ((a)<(b)?(a):(b))
+
+static FLAC__bool parse_double_(const FLAC__StreamMetadata_VorbisComment_Entry *entry, double *val)
+{
+	char s[32], *end;
+	const char *p, *q;
+	double v;
+
+	FLAC__ASSERT(0 != entry);
+	FLAC__ASSERT(0 != val);
+
+	p = (const char *)entry->entry;
+	q = strchr(p, '=');
+	if(0 == q)
+		return false;
+	q++;
+	memset(s, 0, sizeof(s)-1);
+	strncpy(s, q, local_min(sizeof(s)-1, entry->length - (q-p)));
+
+	v = strtod(s, &end);
+	if(end == s)
+		return false;
+
+	*val = v;
+	return true;
+}
+
+FLAC__bool grabbag__replaygain_load_from_vorbiscomment(const FLAC__StreamMetadata *block, FLAC__bool album_mode, double *gain, double *peak)
+{
+	int gain_offset, peak_offset;
+static const FLAC__byte *tag_title_gain_ = "REPLAYGAIN_TRACK_GAIN";
+static const FLAC__byte *tag_title_peak_ = "REPLAYGAIN_TRACK_PEAK";
+static const FLAC__byte *tag_album_gain_ = "REPLAYGAIN_ALBUM_GAIN";
+static const FLAC__byte *tag_album_peak_ = "REPLAYGAIN_ALBUM_PEAK";
+
+	FLAC__ASSERT(0 != block);
+	FLAC__ASSERT(block->type == FLAC__METADATA_TYPE_VORBIS_COMMENT);
+
+	if(0 > (gain_offset = FLAC__metadata_object_vorbiscomment_find_entry_from(block, /*offset=*/0, (const char *)(album_mode? tag_album_gain_ : tag_title_gain_))))
+		return false;
+	if(0 > (peak_offset = FLAC__metadata_object_vorbiscomment_find_entry_from(block, /*offset=*/0, (const char *)(album_mode? tag_album_peak_ : tag_title_peak_))))
+		return false;
+
+	if(!parse_double_(block->data.vorbis_comment.comments + gain_offset, gain))
+		return false;
+	if(!parse_double_(block->data.vorbis_comment.comments + peak_offset, peak))
+		return false;
+
+	return true;
+}
+
+double grabbag__replaygain_compute_scale_factor(double peak, double gain, double preamp, FLAC__bool prevent_clipping)
+{
+	double scale;
+	FLAC__ASSERT(peak >= 0.0);
+ 	gain += preamp;
+	scale = (float) pow(10.0, gain * 0.05);
+	if(prevent_clipping && peak > 0.0) {
+		const double max_scale = (float)(1.0 / peak);
+		if(scale > max_scale)
+			scale = max_scale;
+	}
+	return scale;
+}
+
+void flac_metadata_callback (const FLAC__StreamDecoder *decoder, const FLAC__StreamMetadata *metadata, void *client_data)
+{
+	int i, j;
+	sh_audio_t *sh = ((flac_struct_t*)client_data)->sh;
+	mp_msg(MSGT_DECAUDIO, MSGL_DBG2, "Metadata received\n");
+	switch (metadata->type)
+	{
+		case FLAC__METADATA_TYPE_STREAMINFO:
+			mp_msg(MSGT_DECAUDIO, MSGL_V, "STREAMINFO block (%u bytes):\n", metadata->length);
+			mp_msg(MSGT_DECAUDIO, MSGL_V, "min_blocksize: %u samples\n", metadata->data.stream_info.min_blocksize);
+			mp_msg(MSGT_DECAUDIO, MSGL_V, "max_blocksize: %u samples\n", metadata->data.stream_info.max_blocksize);
+			mp_msg(MSGT_DECAUDIO, MSGL_V, "min_framesize: %u bytes\n", metadata->data.stream_info.min_framesize);
+			mp_msg(MSGT_DECAUDIO, MSGL_V, "max_framesize: %u bytes\n", metadata->data.stream_info.max_framesize);
+			mp_msg(MSGT_DECAUDIO, MSGL_V, "sample_rate: %u Hz\n", metadata->data.stream_info.sample_rate);
+			sh->samplerate = metadata->data.stream_info.sample_rate;
+			mp_msg(MSGT_DECAUDIO, MSGL_V, "channels: %u\n", metadata->data.stream_info.channels);
+			sh->channels = metadata->data.stream_info.channels;
+			mp_msg(MSGT_DECAUDIO, MSGL_V, "bits_per_sample: %u\n", metadata->data.stream_info.bits_per_sample);
+			((flac_struct_t*)client_data)->bits_per_sample = metadata->data.stream_info.bits_per_sample;
+			sh->samplesize = (metadata->data.stream_info.bits_per_sample<=8)?1:2;
+			/* FIXME: need to support dithering to samplesize 4 */
+			sh->sample_format=(sh->samplesize==1)?AFMT_U8:AFMT_S16_LE; // sample format, see libao2/afmt.h
+			sh->o_bps = sh->samplesize * metadata->data.stream_info.channels * metadata->data.stream_info.sample_rate;
+			sh->i_bps = metadata->data.stream_info.bits_per_sample * metadata->data.stream_info.channels * metadata->data.stream_info.sample_rate / 8 / 2;
+			// input data rate (compressed bytes per second)
+			// Compression rate is near 0.5 
+			mp_msg(MSGT_DECAUDIO, MSGL_V, "total_samples: %llu\n", metadata->data.stream_info.total_samples);
+			mp_msg(MSGT_DECAUDIO, MSGL_V, "md5sum: ");
+			for (i = 0; i < 16; i++)
+				mp_msg(MSGT_DECAUDIO, MSGL_V, "%02hhx", metadata->data.stream_info.md5sum[i]);
+			mp_msg(MSGT_DECAUDIO, MSGL_V, "\n");
+			
+			break;
+		case FLAC__METADATA_TYPE_PADDING:
+			mp_msg(MSGT_DECAUDIO, MSGL_V, "PADDING block (%u bytes)\n", metadata->length);
+			break;
+		case FLAC__METADATA_TYPE_APPLICATION:
+			mp_msg(MSGT_DECAUDIO, MSGL_V, "APPLICATION block (%u bytes):\n", metadata->length);
+			mp_msg(MSGT_DECAUDIO, MSGL_V, "Application id: 0x");
+			for (i = 0; i < 4; i++)
+				mp_msg(MSGT_DECAUDIO, MSGL_V, "%02hhx", metadata->data.application.id[i]);
+			mp_msg(MSGT_DECAUDIO, MSGL_V, "\nData: \n");
+			for (i = 0; i < (metadata->length-4)/8; i++)
+			{
+				for(j = 0; j < 8; j++)
+					mp_msg(MSGT_DECAUDIO, MSGL_V, "%c", (unsigned char)metadata->data.application.data[i*8+j]<0x20?'.':metadata->data.application.data[i*8+j]);
+				mp_msg(MSGT_DECAUDIO, MSGL_V, "  |  ");
+				for(j = 0; j < 8; j++)
+					mp_msg(MSGT_DECAUDIO, MSGL_V, "%#02hhx ", metadata->data.application.data[i*8+j]);
+				mp_msg(MSGT_DECAUDIO, MSGL_V, "\n");
+			}
+			if (metadata->length-4-i*8 != 0)
+			{
+				for(j = 0; j < metadata->length-4-i*8; j++)
+					mp_msg(MSGT_DECAUDIO, MSGL_V, "%c", (unsigned char)metadata->data.application.data[i*8+j]<0x20?'.':metadata->data.application.data[i*8+j]);
+				for(; j <8; j++)
+					mp_msg(MSGT_DECAUDIO, MSGL_V, " ");
+				mp_msg(MSGT_DECAUDIO, MSGL_V, "  |  ");
+				for(j = 0; j < metadata->length-4-i*8; j++)
+					mp_msg(MSGT_DECAUDIO, MSGL_V, "%#02hhx ", metadata->data.application.data[i*8+j]);
+				mp_msg(MSGT_DECAUDIO, MSGL_V, "\n");
+			}
+			break;
+		case FLAC__METADATA_TYPE_SEEKTABLE:
+			mp_msg(MSGT_DECAUDIO, MSGL_V, "SEEKTABLE block (%u bytes):\n", metadata->length);
+			mp_msg(MSGT_DECAUDIO, MSGL_V, "%d seekpoints:\n", metadata->data.seek_table.num_points);
+			for (i = 0; i < metadata->data.seek_table.num_points; i++)
+				if (metadata->data.seek_table.points[i].sample_number != FLAC__STREAM_METADATA_SEEKPOINT_PLACEHOLDER)
+					mp_msg(MSGT_DECAUDIO, MSGL_V, "  %3d) sample_number=%llu stream_offset=%llu frame_samples=%u\n", i,
+						metadata->data.seek_table.points[i].sample_number,
+						metadata->data.seek_table.points[i].stream_offset,
+						metadata->data.seek_table.points[i].frame_samples);
+				else
+					mp_msg(MSGT_DECAUDIO, MSGL_V, "  %3d) PLACEHOLDER\n", i);
+			break;
+		case FLAC__METADATA_TYPE_VORBIS_COMMENT:
+			mp_msg(MSGT_DECAUDIO, MSGL_V, "VORBISCOMMENT block (%u bytes):\n", metadata->length);
+			{
+				char entry[metadata->data.vorbis_comment.vendor_string.length+1];
+				memcpy(&entry, metadata->data.vorbis_comment.vendor_string.entry, metadata->data.vorbis_comment.vendor_string.length);
+				entry[metadata->data.vorbis_comment.vendor_string.length] = '\0';
+				mp_msg(MSGT_DECAUDIO, MSGL_V, "vendor_string: %s\n", entry);
+			}
+			mp_msg(MSGT_DECAUDIO, MSGL_V, "%d comment(s):\n",  metadata->data.vorbis_comment.num_comments);
+			for (i = 0; i < metadata->data.vorbis_comment.num_comments; i++)
+			{
+				char entry[metadata->data.vorbis_comment.comments[i].length];
+				memcpy(&entry, metadata->data.vorbis_comment.comments[i].entry, metadata->data.vorbis_comment.comments[i].length);
+				entry[metadata->data.vorbis_comment.comments[i].length] = '\0';
+				mp_msg(MSGT_DECAUDIO, MSGL_V, "%s\n", entry);
+			}
+			{
+				double gain, peak;
+				if(grabbag__replaygain_load_from_vorbiscomment(metadata, album_mode, &gain, &peak))
+				{
+					((flac_struct_t*)client_data)->has_replaygain = true;
+					((flac_struct_t*)client_data)->replay_scale = grabbag__replaygain_compute_scale_factor(peak, gain, (double)preamp, /*prevent_clipping=*/!hard_limit);
+					mp_msg(MSGT_DECAUDIO, MSGL_V, "calculated replay_scale: %lf\n", ((flac_struct_t*)client_data)->replay_scale);
+				}
+			}
+			break;
+		case FLAC__METADATA_TYPE_CUESHEET:
+			mp_msg(MSGT_DECAUDIO, MSGL_V, "CUESHEET block (%u bytes):\n", metadata->length);
+			mp_msg(MSGT_DECAUDIO, MSGL_V, "mcn: '%s'\n", metadata->data.cue_sheet.media_catalog_number);
+			mp_msg(MSGT_DECAUDIO, MSGL_V, "lead_in: %llu\n", metadata->data.cue_sheet.lead_in);
+			mp_msg(MSGT_DECAUDIO, MSGL_V, "is_cd: %s\n", metadata->data.cue_sheet.is_cd?"true":"false");
+			mp_msg(MSGT_DECAUDIO, MSGL_V, "num_tracks: %u\n", metadata->data.cue_sheet.num_tracks);
+			for (i = 0; i < metadata->data.cue_sheet.num_tracks; i++)
+			{
+				mp_msg(MSGT_DECAUDIO, MSGL_V, "track[%d]:\n", i);
+				mp_msg(MSGT_DECAUDIO, MSGL_V, "offset: %llu\n", metadata->data.cue_sheet.tracks[i].offset);
+				mp_msg(MSGT_DECAUDIO, MSGL_V, "number: %hhu%s\n", metadata->data.cue_sheet.tracks[i].number, metadata->data.cue_sheet.tracks[i].number==170?"(LEAD-OUT)":"");
+				mp_msg(MSGT_DECAUDIO, MSGL_V, "isrc: '%s'\n", metadata->data.cue_sheet.tracks[i].isrc);
+				mp_msg(MSGT_DECAUDIO, MSGL_V, "type: %s\n", metadata->data.cue_sheet.tracks[i].type?"non-audio":"audio");
+				mp_msg(MSGT_DECAUDIO, MSGL_V, "pre_emphasis: %s\n", metadata->data.cue_sheet.tracks[i].pre_emphasis?"true":"false");
+				mp_msg(MSGT_DECAUDIO, MSGL_V, "num_indices: %hhu\n", metadata->data.cue_sheet.tracks[i].num_indices);
+				for (j = 0; j < metadata->data.cue_sheet.tracks[i].num_indices; j++)
+				{
+					mp_msg(MSGT_DECAUDIO, MSGL_V, "index[%d]:\n", j);
+					mp_msg(MSGT_DECAUDIO, MSGL_V, "offset:%llu\n", metadata->data.cue_sheet.tracks[i].indices[j].offset);
+					mp_msg(MSGT_DECAUDIO, MSGL_V, "number:%hhu\n", metadata->data.cue_sheet.tracks[i].indices[j].number);
+				}
+			}
+			break;
+		default: if (metadata->type >= FLAC__METADATA_TYPE_UNDEFINED)
+			mp_msg(MSGT_DECAUDIO, MSGL_V, "UNKNOWN block (%u bytes):\n", metadata->length);
+			else
+			mp_msg(MSGT_DECAUDIO, MSGL_V, "Strange block: UNKNOWN #%d < FLAC__METADATA_TYPE_UNDEFINED (%u bytes):\n", metadata->type, metadata->length);
+			for (i = 0; i < (metadata->length)/8; i++)
+			{
+				for(j = 0; j < 8; j++)
+					mp_msg(MSGT_DECAUDIO, MSGL_V, "%c", (unsigned char)metadata->data.unknown.data[i*8+j]<0x20?'.':metadata->data.unknown.data[i*8+j]);
+				mp_msg(MSGT_DECAUDIO, MSGL_V, "  |  ");
+				for(j = 0; j < 8; j++)
+					mp_msg(MSGT_DECAUDIO, MSGL_V, "%#02hhx ", metadata->data.unknown.data[i*8+j]);
+				mp_msg(MSGT_DECAUDIO, MSGL_V, "\n");
+			}
+			if (metadata->length-i*8 != 0)
+			{
+				for(j = 0; j < metadata->length-i*8; j++)
+					mp_msg(MSGT_DECAUDIO, MSGL_V, "%c", (unsigned char)metadata->data.unknown.data[i*8+j]<0x20?'.':metadata->data.unknown.data[i*8+j]);
+				for(; j <8; j++)
+					mp_msg(MSGT_DECAUDIO, MSGL_V, " ");
+				mp_msg(MSGT_DECAUDIO, MSGL_V, "  |  ");
+				for(j = 0; j < metadata->length-i*8; j++)
+					mp_msg(MSGT_DECAUDIO, MSGL_V, "%#02hhx ", metadata->data.unknown.data[i*8+j]);
+				mp_msg(MSGT_DECAUDIO, MSGL_V, "\n");
+			}
+			break;
+	}
+}
+
+void flac_error_callback(const FLAC__StreamDecoder *decoder, FLAC__StreamDecoderErrorStatus status, void *client_data)
+{
+	if (status != FLAC__STREAM_DECODER_ERROR_STATUS_LOST_SYNC)
+		mp_msg(MSGT_DECAUDIO, MSGL_ERR, "\nError callback called (%s)!!!\n", FLAC__StreamDecoderErrorStatusString[status]);
+}
+
+static int preinit(sh_audio_t *sh){
+  // there are default values set for buffering, but you can override them:
+  
+  sh->audio_out_minsize=8*4*65535; // due to specs: we assume max 8 channels,
+                                  // 4 bytes/sample and 65535 samples/frame
+				  // So allocating 2Mbytes buffer :)
+  
+  // minimum input buffer size (set only if you need input buffering)
+  // (should be the max compressed frame size)
+  sh->audio_in_minsize=2048; // Default: 0 (no input buffer)
+  
+  // if you set audio_in_minsize non-zero, the buffer will be allocated
+  // before the init() call by the core, and you can access it via
+  // pointer: sh->audio_in_buffer
+  // it will free'd after uninit(), so you don't have to use malloc/free here!
+
+  return 1; // return values: 1=OK 0=ERROR
+}
+
+static int init(sh_audio_t *sh_audio){
+	flac_struct_t *context = (flac_struct_t*)calloc(sizeof(flac_struct_t), 1);
+  
+	sh_audio->context = context;
+	context->sh = sh_audio;
+	if (context == NULL)
+	{
+		mp_msg(MSGT_DECAUDIO, MSGL_FATAL, "flac_init: error allocating context.\n");
+		return 0;
+	}
+
+	context->flac_dec = FLAC__stream_decoder_new();
+	if (context->flac_dec == NULL)
+	{
+		mp_msg(MSGT_DECAUDIO, MSGL_ERR, "flac_init: error allocaing FLAC decoder.\n");
+		return 0;
+	}
+  
+	if (!FLAC__stream_decoder_set_client_data(context->flac_dec, context))
+	{
+		mp_msg(MSGT_DECAUDIO, MSGL_ERR, "error setting private data for callbacks.\n");
+		return 0;
+	}
+
+	if (!FLAC__stream_decoder_set_read_callback(context->flac_dec, &flac_read_callback))
+	{
+		mp_msg(MSGT_DECAUDIO, MSGL_ERR, "error setting read callback.\n");
+		return 0;
+	}
+
+	if (!FLAC__stream_decoder_set_write_callback(context->flac_dec, &flac_write_callback))
+	{
+		mp_msg(MSGT_DECAUDIO, MSGL_ERR, "error setting write callback.\n");
+		return 0;
+	}
+
+	if (!FLAC__stream_decoder_set_metadata_callback(context->flac_dec, &flac_metadata_callback))
+	{
+		mp_msg(MSGT_DECAUDIO, MSGL_ERR, "error setting metadata callback.\n");
+		return 0;
+	}
+
+	if (!FLAC__stream_decoder_set_error_callback(context->flac_dec, &flac_error_callback))
+	{
+		mp_msg(MSGT_DECAUDIO, MSGL_ERR, "error setting error callback.\n");
+		return 0;
+	}
+
+	if (!FLAC__stream_decoder_set_metadata_respond_all(context->flac_dec))
+	{
+		mp_msg(MSGT_DECAUDIO, MSGL_ERR, "error during setting metadata_respond_all.\n");
+		return 0;
+	}
+
+	if (FLAC__stream_decoder_init(context->flac_dec) != FLAC__STREAM_DECODER_SEARCH_FOR_METADATA)
+	{
+		mp_msg(MSGT_DECAUDIO, MSGL_ERR, "Error initializing decoder!\n");
+		return 0;
+	}
+
+	context->buf = NULL;
+	context->minlen = context->maxlen = 0;
+	context->replay_scale = 1.0;
+
+	FLAC__stream_decoder_process_until_end_of_metadata(context->flac_dec);
+
+	FLAC__plugin_common__init_dither_context(&(context->dither_context), sh_audio->samplesize * 8, noise_shaping);
+	
+	return 1; // return values: 1=OK 0=ERROR
+}
+
+static void uninit(sh_audio_t *sh){
+  // uninit the decoder etc...
+  FLAC__stream_decoder_finish(((flac_struct_t*)(sh->context))->flac_dec);
+  FLAC__stream_decoder_delete(((flac_struct_t*)(sh->context))->flac_dec);
+  // again: you don't have to free() a_in_buffer here! it's done by the core.
+}
+
+static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen){
+	FLAC__StreamDecoderState decstate;
+	FLAC__bool status;
+
+  // audio decoding. the most important thing :)
+  // parameters you get:
+  //  buf = pointer to the output buffer, you have to store uncompressed 
+  //        samples there
+  //  minlen = requested minimum size (in bytes!) of output. it's just a
+  //        _recommendation_, you can decode more or less, it just tell you that
+  //        the caller process needs 'minlen' bytes. if it gets less, it will
+  //        call decode_audio() again.
+  //  maxlen = maximum size (bytes) of output. you MUST NOT write more to the
+  //        buffer, it's the upper-most limit!
+  //        note: maxlen will be always greater or equal to sh->audio_out_minsize
+
+// Store params in private context for callback:
+	((flac_struct_t*)(sh_audio->context))->buf = buf;
+	((flac_struct_t*)(sh_audio->context))->minlen = minlen;
+	((flac_struct_t*)(sh_audio->context))->maxlen = maxlen;
+	((flac_struct_t*)(sh_audio->context))->written = 0;
+
+	status = FLAC__stream_decoder_process_single(((flac_struct_t*)(sh_audio->context))->flac_dec);
+	decstate = FLAC__stream_decoder_get_state(((flac_struct_t*)(sh_audio->context))->flac_dec);
+	if (!status || (
+		decstate != FLAC__STREAM_DECODER_SEARCH_FOR_METADATA &&
+		decstate != FLAC__STREAM_DECODER_READ_METADATA &&
+		decstate != FLAC__STREAM_DECODER_SEARCH_FOR_FRAME_SYNC &&
+		decstate != FLAC__STREAM_DECODER_READ_FRAME
+		))
+	{
+		if (decstate == FLAC__STREAM_DECODER_END_OF_STREAM)
+		{
+			/* return what we have decoded */
+			if (((flac_struct_t*)(sh_audio->context))->written != 0)
+				return ((flac_struct_t*)(sh_audio->context))->written;
+			mp_msg(MSGT_DECAUDIO, MSGL_V, "End of stream.\n");
+			return -1;
+		}
+		mp_msg(MSGT_DECAUDIO, MSGL_WARN, "process_single problem: returned %s, state is %s!\n", status?"true":"false", FLAC__StreamDecoderStateString[decstate]);
+		FLAC__stream_decoder_flush(((flac_struct_t*)(sh_audio->context))->flac_dec);
+		return -1;
+	}
+
+
+  return ((flac_struct_t*)(sh_audio->context))->written; // return value: number of _bytes_ written to output buffer,
+              // or -1 for EOF (or uncorrectable error)
+}
+
+static int control(sh_audio_t *sh,int cmd,void* arg, ...){
+    switch(cmd){
+      case ADCTRL_RESYNC_STREAM:
+        // it is called once after seeking, to resync.
+	// Note: sh_audio->a_in_buffer_len=0; is done _before_ this call!
+	FLAC__stream_decoder_flush (((flac_struct_t*)(sh->context))->flac_dec);
+	return CONTROL_TRUE;
+      case ADCTRL_SKIP_FRAME:
+        // it is called to skip (jump over) small amount (1/10 sec or 1 frame)
+	// of audio data - used to sync audio to video after seeking
+	// if you don't return CONTROL_TRUE, it will defaults to:
+	//      ds_fill_buffer(sh_audio->ds);  // skip 1 demux packet
+	((flac_struct_t*)(sh->context))->buf = NULL;
+	((flac_struct_t*)(sh->context))->minlen =
+	((flac_struct_t*)(sh->context))->maxlen = 0;
+	FLAC__stream_decoder_process_single(((flac_struct_t*)(sh->context))->flac_dec);
+	return CONTROL_TRUE;
+    }
+  return CONTROL_UNKNOWN;
+}
+#endif