/*
- * FLAC (Free Lossless Audio Codec) decoder
- * Copyright (c) 2003 Alex Beregszaszi
+ * FLAC common code
+ * Copyright (c) 2009 Justin Ruggles
*
* This file is part of FFmpeg.
*
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
-/**
- * @file flac.c
- * FLAC (Free Lossless Audio Codec) decoder
- * @author Alex Beregszaszi
- *
- * For more information on the FLAC format, visit:
- * http://flac.sourceforge.net/
- *
- * This decoder can be used in 1 of 2 ways: Either raw FLAC data can be fed
- * through, starting from the initial 'fLaC' signature; or by passing the
- * 34-byte streaminfo structure through avctx->extradata[_size] followed
- * by data starting with the 0xFFF8 marker.
- */
-
-#include <limits.h>
-
-#define ALT_BITSTREAM_READER
-#include "libavutil/crc.h"
-#include "avcodec.h"
-#include "bitstream.h"
-#include "golomb.h"
#include "flac.h"
-#undef NDEBUG
-#include <assert.h>
-
-#define MAX_CHANNELS 8
-#define MAX_BLOCKSIZE 65535
-#define FLAC_STREAMINFO_SIZE 34
-
-enum decorrelation_type {
- INDEPENDENT,
- LEFT_SIDE,
- RIGHT_SIDE,
- MID_SIDE,
-};
-
-typedef struct FLACContext {
- FLACSTREAMINFO
-
- AVCodecContext *avctx;
- GetBitContext gb;
-
- int blocksize/*, last_blocksize*/;
- int curr_bps;
- enum decorrelation_type decorrelation;
-
- int32_t *decoded[MAX_CHANNELS];
- uint8_t *bitstream;
- int bitstream_size;
- int bitstream_index;
- unsigned int allocated_bitstream_size;
-} FLACContext;
-
-#define METADATA_TYPE_STREAMINFO 0
-
-static const int sample_rate_table[] =
-{ 0, 0, 0, 0,
- 8000, 16000, 22050, 24000, 32000, 44100, 48000, 96000,
- 0, 0, 0, 0 };
-
-static const int sample_size_table[] =
-{ 0, 8, 12, 0, 16, 20, 24, 0 };
-
-static const int blocksize_table[] = {
- 0, 192, 576<<0, 576<<1, 576<<2, 576<<3, 0, 0,
-256<<0, 256<<1, 256<<2, 256<<3, 256<<4, 256<<5, 256<<6, 256<<7
-};
-
-static int64_t get_utf8(GetBitContext *gb){
- int64_t val;
- GET_UTF8(val, get_bits(gb, 8), return -1;)
- return val;
-}
-
-static void allocate_buffers(FLACContext *s);
-static int metadata_parse(FLACContext *s);
-
-static av_cold int flac_decode_init(AVCodecContext * avctx)
+int ff_flac_get_max_frame_size(int blocksize, int ch, int bps)
{
- FLACContext *s = avctx->priv_data;
- s->avctx = avctx;
-
- if (avctx->extradata_size > 4) {
- /* initialize based on the demuxer-supplied streamdata header */
- if (avctx->extradata_size == FLAC_STREAMINFO_SIZE) {
- ff_flac_parse_streaminfo(avctx, (FLACStreaminfo *)s, avctx->extradata);
- allocate_buffers(s);
- } else {
- init_get_bits(&s->gb, avctx->extradata, avctx->extradata_size*8);
- metadata_parse(s);
- }
- }
-
- return 0;
-}
-
-static void dump_headers(AVCodecContext *avctx, FLACStreaminfo *s)
-{
- av_log(avctx, AV_LOG_DEBUG, " Blocksize: %d .. %d\n", s->min_blocksize, s->max_blocksize);
- av_log(avctx, AV_LOG_DEBUG, " Max Framesize: %d\n", s->max_framesize);
- av_log(avctx, AV_LOG_DEBUG, " Samplerate: %d\n", s->samplerate);
- av_log(avctx, AV_LOG_DEBUG, " Channels: %d\n", s->channels);
- av_log(avctx, AV_LOG_DEBUG, " Bits: %d\n", s->bps);
-}
-
-static void allocate_buffers(FLACContext *s){
- int i;
-
- assert(s->max_blocksize);
-
- if(s->max_framesize == 0 && s->max_blocksize){
- s->max_framesize= (s->channels * s->bps * s->max_blocksize + 7)/ 8; //FIXME header overhead
- }
-
- for (i = 0; i < s->channels; i++)
- {
- s->decoded[i] = av_realloc(s->decoded[i], sizeof(int32_t)*s->max_blocksize);
- }
-
- if(s->allocated_bitstream_size < s->max_framesize)
- s->bitstream= av_fast_realloc(s->bitstream, &s->allocated_bitstream_size, s->max_framesize);
-}
-
-void ff_flac_parse_streaminfo(AVCodecContext *avctx, struct FLACStreaminfo *s,
- const uint8_t *buffer)
-{
- GetBitContext gb;
- init_get_bits(&gb, buffer, FLAC_STREAMINFO_SIZE*8);
-
- /* mandatory streaminfo */
- s->min_blocksize = get_bits(&gb, 16);
- s->max_blocksize = get_bits(&gb, 16);
-
- skip_bits(&gb, 24); /* skip min frame size */
- s->max_framesize = get_bits_long(&gb, 24);
-
- s->samplerate = get_bits_long(&gb, 20);
- s->channels = get_bits(&gb, 3) + 1;
- s->bps = get_bits(&gb, 5) + 1;
-
- avctx->channels = s->channels;
- avctx->sample_rate = s->samplerate;
-
- skip_bits(&gb, 36); /* total num of samples */
-
- skip_bits(&gb, 64); /* md5 sum */
- skip_bits(&gb, 64); /* md5 sum */
-
- dump_headers(avctx, s);
-}
-
-/**
- * Parse a list of metadata blocks. This list of blocks must begin with
- * the fLaC marker.
- * @param s the flac decoding context containing the gb bit reader used to
- * parse metadata
- * @return 1 if some metadata was read, 0 if no fLaC marker was found
- */
-static int metadata_parse(FLACContext *s)
-{
- int i, metadata_last, metadata_type, metadata_size, streaminfo_updated=0;
-
- if (show_bits_long(&s->gb, 32) == MKBETAG('f','L','a','C')) {
- skip_bits(&s->gb, 32);
-
- av_log(s->avctx, AV_LOG_DEBUG, "STREAM HEADER\n");
- do {
- metadata_last = get_bits1(&s->gb);
- metadata_type = get_bits(&s->gb, 7);
- metadata_size = get_bits_long(&s->gb, 24);
-
- av_log(s->avctx, AV_LOG_DEBUG,
- " metadata block: flag = %d, type = %d, size = %d\n",
- metadata_last, metadata_type, metadata_size);
- if (metadata_size) {
- switch (metadata_type) {
- case METADATA_TYPE_STREAMINFO:
- ff_flac_parse_streaminfo(s->avctx, (FLACStreaminfo *)s, s->gb.buffer+get_bits_count(&s->gb)/8);
- streaminfo_updated = 1;
-
- default:
- for (i=0; i<metadata_size; i++)
- skip_bits(&s->gb, 8);
- }
- }
- } while (!metadata_last);
-
- if (streaminfo_updated)
- allocate_buffers(s);
- return 1;
- }
- return 0;
-}
-
-static int decode_residuals(FLACContext *s, int channel, int pred_order)
-{
- int i, tmp, partition, method_type, rice_order;
- int sample = 0, samples;
-
- method_type = get_bits(&s->gb, 2);
- if (method_type > 1){
- av_log(s->avctx, AV_LOG_DEBUG, "illegal residual coding method %d\n", method_type);
- return -1;
- }
-
- rice_order = get_bits(&s->gb, 4);
-
- samples= s->blocksize >> rice_order;
- if (pred_order > samples) {
- av_log(s->avctx, AV_LOG_ERROR, "invalid predictor order: %i > %i\n", pred_order, samples);
- return -1;
- }
-
- sample=
- i= pred_order;
- for (partition = 0; partition < (1 << rice_order); partition++)
- {
- tmp = get_bits(&s->gb, method_type == 0 ? 4 : 5);
- if (tmp == (method_type == 0 ? 15 : 31))
- {
- av_log(s->avctx, AV_LOG_DEBUG, "fixed len partition\n");
- tmp = get_bits(&s->gb, 5);
- for (; i < samples; i++, sample++)
- s->decoded[channel][sample] = get_sbits(&s->gb, tmp);
- }
- else
- {
-// av_log(s->avctx, AV_LOG_DEBUG, "rice coded partition k=%d\n", tmp);
- for (; i < samples; i++, sample++){
- s->decoded[channel][sample] = get_sr_golomb_flac(&s->gb, tmp, INT_MAX, 0);
- }
- }
- i= 0;
- }
-
-// av_log(s->avctx, AV_LOG_DEBUG, "partitions: %d, samples: %d\n", 1 << rice_order, sample);
-
- return 0;
-}
-
-static int decode_subframe_fixed(FLACContext *s, int channel, int pred_order)
-{
- const int blocksize = s->blocksize;
- int32_t *decoded = s->decoded[channel];
- int a, b, c, d, i;
-
-// av_log(s->avctx, AV_LOG_DEBUG, " SUBFRAME FIXED\n");
-
- /* warm up samples */
-// av_log(s->avctx, AV_LOG_DEBUG, " warm up samples: %d\n", pred_order);
-
- for (i = 0; i < pred_order; i++)
- {
- decoded[i] = get_sbits(&s->gb, s->curr_bps);
-// av_log(s->avctx, AV_LOG_DEBUG, " %d: %d\n", i, s->decoded[channel][i]);
- }
-
- if (decode_residuals(s, channel, pred_order) < 0)
- return -1;
-
- if(pred_order > 0)
- a = decoded[pred_order-1];
- if(pred_order > 1)
- b = a - decoded[pred_order-2];
- if(pred_order > 2)
- c = b - decoded[pred_order-2] + decoded[pred_order-3];
- if(pred_order > 3)
- d = c - decoded[pred_order-2] + 2*decoded[pred_order-3] - decoded[pred_order-4];
-
- switch(pred_order)
- {
- case 0:
- break;
- case 1:
- for (i = pred_order; i < blocksize; i++)
- decoded[i] = a += decoded[i];
- break;
- case 2:
- for (i = pred_order; i < blocksize; i++)
- decoded[i] = a += b += decoded[i];
- break;
- case 3:
- for (i = pred_order; i < blocksize; i++)
- decoded[i] = a += b += c += decoded[i];
- break;
- case 4:
- for (i = pred_order; i < blocksize; i++)
- decoded[i] = a += b += c += d += decoded[i];
- break;
- default:
- av_log(s->avctx, AV_LOG_ERROR, "illegal pred order %d\n", pred_order);
- return -1;
- }
-
- return 0;
-}
-
-static int decode_subframe_lpc(FLACContext *s, int channel, int pred_order)
-{
- int i, j;
- int coeff_prec, qlevel;
- int coeffs[pred_order];
- int32_t *decoded = s->decoded[channel];
-
-// av_log(s->avctx, AV_LOG_DEBUG, " SUBFRAME LPC\n");
-
- /* warm up samples */
-// av_log(s->avctx, AV_LOG_DEBUG, " warm up samples: %d\n", pred_order);
-
- for (i = 0; i < pred_order; i++)
- {
- decoded[i] = get_sbits(&s->gb, s->curr_bps);
-// av_log(s->avctx, AV_LOG_DEBUG, " %d: %d\n", i, decoded[i]);
- }
-
- coeff_prec = get_bits(&s->gb, 4) + 1;
- if (coeff_prec == 16)
- {
- av_log(s->avctx, AV_LOG_DEBUG, "invalid coeff precision\n");
- return -1;
- }
-// av_log(s->avctx, AV_LOG_DEBUG, " qlp coeff prec: %d\n", coeff_prec);
- qlevel = get_sbits(&s->gb, 5);
-// av_log(s->avctx, AV_LOG_DEBUG, " quant level: %d\n", qlevel);
- if(qlevel < 0){
- av_log(s->avctx, AV_LOG_DEBUG, "qlevel %d not supported, maybe buggy stream\n", qlevel);
- return -1;
- }
-
- for (i = 0; i < pred_order; i++)
- {
- coeffs[i] = get_sbits(&s->gb, coeff_prec);
-// av_log(s->avctx, AV_LOG_DEBUG, " %d: %d\n", i, coeffs[i]);
- }
+ /* Technically, there is no limit to FLAC frame size, but an encoder
+ should not write a frame that is larger than if verbatim encoding mode
+ were to be used. */
- if (decode_residuals(s, channel, pred_order) < 0)
- return -1;
+ int count;
- if (s->bps > 16) {
- int64_t sum;
- for (i = pred_order; i < s->blocksize; i++)
- {
- sum = 0;
- for (j = 0; j < pred_order; j++)
- sum += (int64_t)coeffs[j] * decoded[i-j-1];
- decoded[i] += sum >> qlevel;
- }
+ count = 16; /* frame header */
+ count += ch * ((7+bps+7)/8); /* subframe headers */
+ if (ch == 2) {
+ /* for stereo, need to account for using decorrelation */
+ count += (( 2*bps+1) * blocksize + 7) / 8;
} else {
- for (i = pred_order; i < s->blocksize-1; i += 2)
- {
- int c;
- int d = decoded[i-pred_order];
- int s0 = 0, s1 = 0;
- for (j = pred_order-1; j > 0; j--)
- {
- c = coeffs[j];
- s0 += c*d;
- d = decoded[i-j];
- s1 += c*d;
- }
- c = coeffs[0];
- s0 += c*d;
- d = decoded[i] += s0 >> qlevel;
- s1 += c*d;
- decoded[i+1] += s1 >> qlevel;
- }
- if (i < s->blocksize)
- {
- int sum = 0;
- for (j = 0; j < pred_order; j++)
- sum += coeffs[j] * decoded[i-j-1];
- decoded[i] += sum >> qlevel;
- }
- }
-
- return 0;
-}
-
-static inline int decode_subframe(FLACContext *s, int channel)
-{
- int type, wasted = 0;
- int i, tmp;
-
- s->curr_bps = s->bps;
- if(channel == 0){
- if(s->decorrelation == RIGHT_SIDE)
- s->curr_bps++;
- }else{
- if(s->decorrelation == LEFT_SIDE || s->decorrelation == MID_SIDE)
- s->curr_bps++;
- }
-
- if (get_bits1(&s->gb))
- {
- av_log(s->avctx, AV_LOG_ERROR, "invalid subframe padding\n");
- return -1;
+ count += ( ch*bps * blocksize + 7) / 8;
}
- type = get_bits(&s->gb, 6);
-// wasted = get_bits1(&s->gb);
+ count += 2; /* frame footer */
-// if (wasted)
-// {
-// while (!get_bits1(&s->gb))
-// wasted++;
-// if (wasted)
-// wasted++;
-// s->curr_bps -= wasted;
-// }
-#if 0
- wasted= 16 - av_log2(show_bits(&s->gb, 17));
- skip_bits(&s->gb, wasted+1);
- s->curr_bps -= wasted;
-#else
- if (get_bits1(&s->gb))
- {
- wasted = 1;
- while (!get_bits1(&s->gb))
- wasted++;
- s->curr_bps -= wasted;
- av_log(s->avctx, AV_LOG_DEBUG, "%d wasted bits\n", wasted);
- }
-#endif
-//FIXME use av_log2 for types
- if (type == 0)
- {
- av_log(s->avctx, AV_LOG_DEBUG, "coding type: constant\n");
- tmp = get_sbits(&s->gb, s->curr_bps);
- for (i = 0; i < s->blocksize; i++)
- s->decoded[channel][i] = tmp;
- }
- else if (type == 1)
- {
- av_log(s->avctx, AV_LOG_DEBUG, "coding type: verbatim\n");
- for (i = 0; i < s->blocksize; i++)
- s->decoded[channel][i] = get_sbits(&s->gb, s->curr_bps);
- }
- else if ((type >= 8) && (type <= 12))
- {
-// av_log(s->avctx, AV_LOG_DEBUG, "coding type: fixed\n");
- if (decode_subframe_fixed(s, channel, type & ~0x8) < 0)
- return -1;
- }
- else if (type >= 32)
- {
-// av_log(s->avctx, AV_LOG_DEBUG, "coding type: lpc\n");
- if (decode_subframe_lpc(s, channel, (type & ~0x20)+1) < 0)
- return -1;
- }
- else
- {
- av_log(s->avctx, AV_LOG_ERROR, "invalid coding type\n");
- return -1;
- }
-
- if (wasted)
- {
- int i;
- for (i = 0; i < s->blocksize; i++)
- s->decoded[channel][i] <<= wasted;
- }
-
- return 0;
+ return count;
}
-
-static int decode_frame(FLACContext *s, int alloc_data_size)
-{
- int blocksize_code, sample_rate_code, sample_size_code, assignment, i, crc8;
- int decorrelation, bps, blocksize, samplerate;
-
- blocksize_code = get_bits(&s->gb, 4);
-
- sample_rate_code = get_bits(&s->gb, 4);
-
- assignment = get_bits(&s->gb, 4); /* channel assignment */
- if (assignment < 8 && s->channels == assignment+1)
- decorrelation = INDEPENDENT;
- else if (assignment >=8 && assignment < 11 && s->channels == 2)
- decorrelation = LEFT_SIDE + assignment - 8;
- else
- {
- av_log(s->avctx, AV_LOG_ERROR, "unsupported channel assignment %d (channels=%d)\n", assignment, s->channels);
- return -1;
- }
-
- sample_size_code = get_bits(&s->gb, 3);
- if(sample_size_code == 0)
- bps= s->bps;
- else if((sample_size_code != 3) && (sample_size_code != 7))
- bps = sample_size_table[sample_size_code];
- else
- {
- av_log(s->avctx, AV_LOG_ERROR, "invalid sample size code (%d)\n", sample_size_code);
- return -1;
- }
-
- if (get_bits1(&s->gb))
- {
- av_log(s->avctx, AV_LOG_ERROR, "broken stream, invalid padding\n");
- return -1;
- }
-
- if(get_utf8(&s->gb) < 0){
- av_log(s->avctx, AV_LOG_ERROR, "utf8 fscked\n");
- return -1;
- }
-#if 0
- if (/*((blocksize_code == 6) || (blocksize_code == 7)) &&*/
- (s->min_blocksize != s->max_blocksize)){
- }else{
- }
-#endif
-
- if (blocksize_code == 0)
- blocksize = s->min_blocksize;
- else if (blocksize_code == 6)
- blocksize = get_bits(&s->gb, 8)+1;
- else if (blocksize_code == 7)
- blocksize = get_bits(&s->gb, 16)+1;
- else
- blocksize = blocksize_table[blocksize_code];
-
- if(blocksize > s->max_blocksize){
- av_log(s->avctx, AV_LOG_ERROR, "blocksize %d > %d\n", blocksize, s->max_blocksize);
- return -1;
- }
-
- if(blocksize * s->channels * sizeof(int16_t) > alloc_data_size)
- return -1;
-
- if (sample_rate_code == 0){
- samplerate= s->samplerate;
- }else if ((sample_rate_code > 3) && (sample_rate_code < 12))
- samplerate = sample_rate_table[sample_rate_code];
- else if (sample_rate_code == 12)
- samplerate = get_bits(&s->gb, 8) * 1000;
- else if (sample_rate_code == 13)
- samplerate = get_bits(&s->gb, 16);
- else if (sample_rate_code == 14)
- samplerate = get_bits(&s->gb, 16) * 10;
- else{
- av_log(s->avctx, AV_LOG_ERROR, "illegal sample rate code %d\n", sample_rate_code);
- return -1;
- }
-
- skip_bits(&s->gb, 8);
- crc8 = av_crc(av_crc_get_table(AV_CRC_8_ATM), 0,
- s->gb.buffer, get_bits_count(&s->gb)/8);
- if(crc8){
- av_log(s->avctx, AV_LOG_ERROR, "header crc mismatch crc=%2X\n", crc8);
- return -1;
- }
-
- s->blocksize = blocksize;
- s->samplerate = samplerate;
- s->bps = bps;
- s->decorrelation= decorrelation;
-
-// dump_headers(s->avctx, (FLACStreaminfo *)s);
-
- /* subframes */
- for (i = 0; i < s->channels; i++)
- {
-// av_log(s->avctx, AV_LOG_DEBUG, "decoded: %x residual: %x\n", s->decoded[i], s->residual[i]);
- if (decode_subframe(s, i) < 0)
- return -1;
- }
-
- align_get_bits(&s->gb);
-
- /* frame footer */
- skip_bits(&s->gb, 16); /* data crc */
-
- return 0;
-}
-
-static int flac_decode_frame(AVCodecContext *avctx,
- void *data, int *data_size,
- const uint8_t *buf, int buf_size)
-{
- FLACContext *s = avctx->priv_data;
- int tmp = 0, i, j = 0, input_buf_size = 0;
- int16_t *samples = data;
- int alloc_data_size= *data_size;
-
- *data_size=0;
-
- if(s->max_framesize == 0){
- s->max_framesize= 65536; // should hopefully be enough for the first header
- s->bitstream= av_fast_realloc(s->bitstream, &s->allocated_bitstream_size, s->max_framesize);
- }
-
- if(1 && s->max_framesize){//FIXME truncated
- buf_size= FFMAX(FFMIN(buf_size, s->max_framesize - s->bitstream_size), 0);
- input_buf_size= buf_size;
-
- if(s->bitstream_index + s->bitstream_size + buf_size > s->allocated_bitstream_size){
-// printf("memmove\n");
- memmove(s->bitstream, &s->bitstream[s->bitstream_index], s->bitstream_size);
- s->bitstream_index=0;
- }
- memcpy(&s->bitstream[s->bitstream_index + s->bitstream_size], buf, buf_size);
- buf= &s->bitstream[s->bitstream_index];
- buf_size += s->bitstream_size;
- s->bitstream_size= buf_size;
-
- if(buf_size < s->max_framesize && input_buf_size){
-// printf("wanna more data ...\n");
- return input_buf_size;
- }
- }
-
- init_get_bits(&s->gb, buf, buf_size*8);
-
- if(metadata_parse(s))
- goto end;
-
- tmp = show_bits(&s->gb, 16);
- if((tmp & 0xFFFE) != 0xFFF8){
- av_log(s->avctx, AV_LOG_ERROR, "FRAME HEADER not here\n");
- while(get_bits_count(&s->gb)/8+2 < buf_size && (show_bits(&s->gb, 16) & 0xFFFE) != 0xFFF8)
- skip_bits(&s->gb, 8);
- goto end; // we may not have enough bits left to decode a frame, so try next time
- }
- skip_bits(&s->gb, 16);
- if (decode_frame(s, alloc_data_size) < 0){
- av_log(s->avctx, AV_LOG_ERROR, "decode_frame() failed\n");
- s->bitstream_size=0;
- s->bitstream_index=0;
- return -1;
- }
-
-
-#if 0
- /* fix the channel order here */
- if (s->order == MID_SIDE)
- {
- short *left = samples;
- short *right = samples + s->blocksize;
- for (i = 0; i < s->blocksize; i += 2)
- {
- uint32_t x = s->decoded[0][i];
- uint32_t y = s->decoded[0][i+1];
-
- right[i] = x - (y / 2);
- left[i] = right[i] + y;
- }
- *data_size = 2 * s->blocksize;
- }
- else
- {
- for (i = 0; i < s->channels; i++)
- {
- switch(s->order)
- {
- case INDEPENDENT:
- for (j = 0; j < s->blocksize; j++)
- samples[(s->blocksize*i)+j] = s->decoded[i][j];
- break;
- case LEFT_SIDE:
- case RIGHT_SIDE:
- if (i == 0)
- for (j = 0; j < s->blocksize; j++)
- samples[(s->blocksize*i)+j] = s->decoded[0][j];
- else
- for (j = 0; j < s->blocksize; j++)
- samples[(s->blocksize*i)+j] = s->decoded[0][j] - s->decoded[i][j];
- break;
-// case MID_SIDE:
-// av_log(s->avctx, AV_LOG_DEBUG, "mid-side unsupported\n");
- }
- *data_size += s->blocksize;
- }
- }
-#else
-#define DECORRELATE(left, right)\
- assert(s->channels == 2);\
- for (i = 0; i < s->blocksize; i++)\
- {\
- int a= s->decoded[0][i];\
- int b= s->decoded[1][i];\
- *samples++ = ((left) << (24 - s->bps)) >> 8;\
- *samples++ = ((right) << (24 - s->bps)) >> 8;\
- }\
- break;
-
- switch(s->decorrelation)
- {
- case INDEPENDENT:
- for (j = 0; j < s->blocksize; j++)
- {
- for (i = 0; i < s->channels; i++)
- *samples++ = (s->decoded[i][j] << (24 - s->bps)) >> 8;
- }
- break;
- case LEFT_SIDE:
- DECORRELATE(a,a-b)
- case RIGHT_SIDE:
- DECORRELATE(a+b,b)
- case MID_SIDE:
- DECORRELATE( (a-=b>>1) + b, a)
- }
-#endif
-
- *data_size = (int8_t *)samples - (int8_t *)data;
-// av_log(s->avctx, AV_LOG_DEBUG, "data size: %d\n", *data_size);
-
-// s->last_blocksize = s->blocksize;
-end:
- i= (get_bits_count(&s->gb)+7)/8;
- if(i > buf_size){
- av_log(s->avctx, AV_LOG_ERROR, "overread: %d\n", i - buf_size);
- s->bitstream_size=0;
- s->bitstream_index=0;
- return -1;
- }
-
- if(s->bitstream_size){
- s->bitstream_index += i;
- s->bitstream_size -= i;
- return input_buf_size;
- }else
- return i;
-}
-
-static av_cold int flac_decode_close(AVCodecContext *avctx)
-{
- FLACContext *s = avctx->priv_data;
- int i;
-
- for (i = 0; i < s->channels; i++)
- {
- av_freep(&s->decoded[i]);
- }
- av_freep(&s->bitstream);
-
- return 0;
-}
-
-static void flac_flush(AVCodecContext *avctx){
- FLACContext *s = avctx->priv_data;
-
- s->bitstream_size=
- s->bitstream_index= 0;
-}
-
-AVCodec flac_decoder = {
- "flac",
- CODEC_TYPE_AUDIO,
- CODEC_ID_FLAC,
- sizeof(FLACContext),
- flac_decode_init,
- NULL,
- flac_decode_close,
- flac_decode_frame,
- CODEC_CAP_DELAY,
- .flush= flac_flush,
- .long_name= NULL_IF_CONFIG_SMALL("FLAC (Free Lossless Audio Codec)"),
-};
projects / frescor / ffmpeg.git / blobdiff