984 lines
29 KiB
C
984 lines
29 KiB
C
/*
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* Copyright (C) 2017 Amlogic, Inc. All rights reserved.
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful, but WITHOUT
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* ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or
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* FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for
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* more details.
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*
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* You should have received a copy of the GNU General Public License along
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* with this program; if not, write to the Free Software Foundation, Inc.,
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* 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
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* Description: adpcm decoder
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*/
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/*
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* filename: flac_decode.c
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*
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*
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*
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*
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*
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*/
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#include <stdio.h>
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#include <stdlib.h>
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#include <limits.h>
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#include "codec.h"
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#ifndef __MW__
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#include <inttypes.h>
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#else
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#include "types.h"
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#endif
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#include "crc.h"
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#include "avcodec.h"
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#include "get_bits.h"
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#include "golomb.h"
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#include "flac.h"
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#include "flacdata.h"
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#include "adec-armdec-mgt.h"
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//#include "../../amadec/adec-armdec-mgt.h"
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#define LOG_TAG "FlacDecoder"
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#include <android/log.h>
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#define audio_codec_print(...) __android_log_print(ANDROID_LOG_INFO,LOG_TAG,__VA_ARGS__)
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#define DefaultReadSize 1024*10 //read count from kernel audio buf one time
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#define DefaultOutBufSize 1024*1024
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typedef struct OutData {
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uint8_t * outb;
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int byte;
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int index;
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} OutData;
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typedef struct FLACContext {
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FLACSTREAMINFO
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AVCodecContext *avctx; ///< parent AVCodecContext
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GetBitContext gb; ///< GetBitContext initialized to start at the current frame
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int blocksize; ///< number of samples in the current frame
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int curr_bps; ///< bps for current subframe, adjusted for channel correlation and wasted bits
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int sample_shift; ///< shift required to make output samples 16-bit or 32-bit
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int is32; ///< flag to indicate if output should be 32-bit instead of 16-bit
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int ch_mode; ///< channel decorrelation type in the current frame
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int got_streaminfo; ///< indicates if the STREAMINFO has been read
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int32_t *decoded[FLAC_MAX_CHANNELS]; ///< decoded samples
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uint8_t *bitstream;
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unsigned int bitstream_size;
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unsigned int bitstream_index;
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unsigned int allocated_bitstream_size;
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unsigned int error_num; ///< error decoded num(for get error num)
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} FLACContext;
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static const int sample_size_table[] =
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{ 0, 8, 12, 0, 16, 20, 24, 0 };
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static AVCodecContext acodec;
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static FLACContext flactext;
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static OutData outbuffer;
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#define INPUT_BUF_SIZE 32768
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static int64_t get_utf8(GetBitContext *gb)
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{
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int64_t val;
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GET_UTF8(val, get_bits(gb, 8), return -1;)
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return val;
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}
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void *av_malloc(unsigned int size)
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{
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void *ptr = NULL;
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ptr = malloc(size);
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return ptr;
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}
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void *av_realloc(void *ptr, unsigned int size)
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{
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return realloc(ptr, size);
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}
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void av_freep(void **arg)
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{
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if (*arg) {
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free(*arg);
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}
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*arg = NULL;
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}
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void av_free(void *arg)
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{
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if (arg) {
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free(arg);
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arg = NULL;
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}
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}
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void *av_mallocz(unsigned int size)
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{
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void *ptr = malloc(size);
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if (ptr) {
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memset(ptr, 0, size);
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}
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return ptr;
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}
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static inline uint32_t bytestream_get_be32(const uint8_t** ptr)
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{
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uint32_t tmp;
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tmp = (*ptr)[3] | ((*ptr)[2] << 8) | ((*ptr)[1] << 16) | ((*ptr)[0] << 24);
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*ptr += 4;
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return tmp;
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}
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static inline uint32_t bytestream_get_be24(const uint8_t** ptr)
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{
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uint32_t tmp;
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tmp = (*ptr)[2] | ((*ptr)[1] << 8) | ((*ptr)[0] << 16) ;
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*ptr += 3;
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return tmp;
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}
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static inline uint8_t bytestream_get_byte(const uint8_t** ptr)
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{
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uint8_t tmp;
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tmp = **ptr;
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*ptr += 1;
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return tmp;
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}
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void *av_fast_realloc(void *ptr, unsigned int *size, unsigned int min_size)
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{
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if (min_size < *size) {
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return ptr;
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}
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*size = FFMAX(17 * min_size / 16 + 32, min_size);
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ptr = av_realloc(ptr, *size);
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if (!ptr) { //we could set this to the unmodified min_size but this is safer if the user lost the ptr and uses NULL now
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*size = 0;
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}
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return ptr;
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}
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static void allocate_buffers(FLACContext *s)
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{
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int i;
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if (s->max_framesize == 0 && s->max_blocksize) {
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s->max_framesize = am_flac_get_max_frame_size(s->max_blocksize,
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s->channels, s->bps);
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}
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for (i = 0; i < s->channels; i++) {
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s->decoded[i] = av_realloc(s->decoded[i], sizeof(int32_t) * s->max_blocksize);
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if (s->decoded[i] == NULL) {
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audio_codec_print("av_realloc decoded[%d] NULL, %s,%d", i, __func__, __LINE__);
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}
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}
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}
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/**
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* Parse the STREAMINFO from an inline header.
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* @param s the flac decoding context
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* @param buf input buffer, starting with the "fLaC" marker
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* @param buf_size buffer size
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* @return non-zero if metadata is invalid
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*/
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static int parse_streaminfo(FLACContext *s, const uint8_t *buf, int buf_size)
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{
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int metadata_type, metadata_size;
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if (buf_size < FLAC_STREAMINFO_SIZE + 8) {
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/* need more data */
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return 0;
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}
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am_flac_parse_block_header(&buf[4], NULL, &metadata_type, &metadata_size);
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if (metadata_type != FLAC_METADATA_TYPE_STREAMINFO ||
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metadata_size != FLAC_STREAMINFO_SIZE) {
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return -1;//AVERROR_INVALIDDATA;
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}
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am_flac_parse_streaminfo(s->avctx, (FLACStreaminfo *)s, &buf[8]);
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allocate_buffers(s);
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s->got_streaminfo = 1;
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return 0;
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}
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/**
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* Determine the size of an inline header.
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* @param buf input buffer, starting with the "fLaC" marker
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* @param buf_size buffer size
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* @return number of bytes in the header, or 0 if more data is needed
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*/
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static int get_metadata_size(const uint8_t *buf, int buf_size)
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{
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int metadata_last, metadata_size;
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const uint8_t *buf_end = buf + buf_size;
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buf += 4;
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do {
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am_flac_parse_block_header(buf, &metadata_last, NULL, &metadata_size);
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buf += 4;
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if (buf + metadata_size > buf_end) {
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/* need more data in order to read the complete header */
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return 0;
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}
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buf += metadata_size;
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} while (!metadata_last);
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return buf_size - (buf_end - buf);
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}
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void am_flac_parse_streaminfo(AVCodecContext *avctx, struct FLACStreaminfo *s,
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const uint8_t *buffer)
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{
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GetBitContext gb;
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init_get_bits(&gb, buffer, FLAC_STREAMINFO_SIZE * 8);
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skip_bits(&gb, 16); /* skip min blocksize */
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s->max_blocksize = get_bits(&gb, 16);
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if (s->max_blocksize < FLAC_MIN_BLOCKSIZE) {
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audio_codec_print("invalid max blocksize: %d\n", s->max_blocksize);
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s->max_blocksize = 16;
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}
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skip_bits(&gb, 24); /* skip min frame size */
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s->max_framesize = get_bits_long(&gb, 24);
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s->samplerate = get_bits_long(&gb, 20);
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s->channels = get_bits(&gb, 3) + 1;
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s->bps = get_bits(&gb, 5) + 1;
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audio_codec_print("## METADATA sr=%d, ch=%d, bps=%d,-------------\n",
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s->samplerate, s->channels, s->bps);
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avctx->channels = s->channels;
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avctx->sample_rate = s->samplerate;
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avctx->bits_per_raw_sample = s->bps;
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s->samples = get_bits_long(&gb, 32) << 4;
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s->samples |= get_bits(&gb, 4);
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skip_bits_long(&gb, 64); /* md5 sum */
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skip_bits_long(&gb, 64); /* md5 sum */
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}
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static void allocate_buffers(FLACContext *s);
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int am_flac_is_extradata_valid(AVCodecContext *avctx,
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enum FLACExtradataFormat *format,
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uint8_t **streaminfo_start)
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{
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if (!avctx->extradata || avctx->extradata_size < FLAC_STREAMINFO_SIZE) {
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audio_codec_print("extradata NULL or too small.\n");
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return 0;
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}
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if (AV_RL32(avctx->extradata) != MKTAG('f', 'L', 'a', 'C')) {
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/* extradata contains STREAMINFO only */
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if (avctx->extradata_size != FLAC_STREAMINFO_SIZE) {
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audio_codec_print("extradata contains %d bytes too many.\n",
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FLAC_STREAMINFO_SIZE - avctx->extradata_size);
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}
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*format = FLAC_EXTRADATA_FORMAT_STREAMINFO;
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*streaminfo_start = avctx->extradata;
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} else {
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if (avctx->extradata_size < 8 + FLAC_STREAMINFO_SIZE) {
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audio_codec_print("extradata too small.\n");
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return 0;
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}
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*format = FLAC_EXTRADATA_FORMAT_FULL_HEADER;
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*streaminfo_start = &avctx->extradata[8];
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}
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return 1;
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}
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static int decode_residuals(FLACContext *s, int channel, int pred_order)
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{
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int i, tmp, partition, method_type, rice_order;
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int sample = 0, samples;
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method_type = get_bits(&s->gb, 2);
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if (method_type > 1) {
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audio_codec_print("illegal residual coding method %d\n", method_type);
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return -1;
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}
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rice_order = get_bits(&s->gb, 4);
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samples = s->blocksize >> rice_order;
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if (pred_order > samples) {
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audio_codec_print("invalid predictor order: %i > %i\n", pred_order, samples);
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return -1;
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}
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sample =
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i = pred_order;
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for (partition = 0; partition < (1 << rice_order); partition++) {
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tmp = get_bits(&s->gb, method_type == 0 ? 4 : 5);
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if (tmp == (method_type == 0 ? 15 : 31)) {
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tmp = get_bits(&s->gb, 5);
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for (; i < samples; i++, sample++) {
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s->decoded[channel][sample] = get_sbits_long(&s->gb, tmp);
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}
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} else {
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for (; i < samples; i++, sample++) {
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s->decoded[channel][sample] = get_sr_golomb_flac(&s->gb, tmp, INT_MAX, 0);
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}
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}
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i = 0;
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}
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return 0;
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}
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static int decode_subframe_fixed(FLACContext *s, int channel, int pred_order)
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{
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const int blocksize = s->blocksize;
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int32_t *decoded = s->decoded[channel];
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int av_uninit(a), av_uninit(b), av_uninit(c), av_uninit(d), i;
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/* warm up samples */
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for (i = 0; i < pred_order; i++) {
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decoded[i] = get_sbits_long(&s->gb, s->curr_bps);
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}
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if (decode_residuals(s, channel, pred_order) < 0) {
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return -1;
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}
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if (pred_order > 0) {
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a = decoded[pred_order - 1];
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}
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if (pred_order > 1) {
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b = a - decoded[pred_order - 2];
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}
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if (pred_order > 2) {
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c = b - decoded[pred_order - 2] + decoded[pred_order - 3];
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}
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if (pred_order > 3) {
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d = c - decoded[pred_order - 2] + 2 * decoded[pred_order - 3] - decoded[pred_order - 4];
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}
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switch (pred_order) {
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case 0:
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break;
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case 1:
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for (i = pred_order; i < blocksize; i++) {
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decoded[i] = a += decoded[i];
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}
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break;
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case 2:
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for (i = pred_order; i < blocksize; i++) {
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decoded[i] = a += b += decoded[i];
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}
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break;
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case 3:
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for (i = pred_order; i < blocksize; i++) {
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decoded[i] = a += b += c += decoded[i];
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}
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break;
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case 4:
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for (i = pred_order; i < blocksize; i++) {
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decoded[i] = a += b += c += d += decoded[i];
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}
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break;
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default:
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audio_codec_print("illegal pred order %d\n", pred_order);
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return -1;
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}
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return 0;
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}
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static int decode_subframe_lpc(FLACContext *s, int channel, int pred_order)
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{
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int i, j;
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int coeff_prec, qlevel;
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int coeffs[32];
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int32_t *decoded = s->decoded[channel];
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/* warm up samples */
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for (i = 0; i < pred_order; i++) {
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decoded[i] = get_sbits_long(&s->gb, s->curr_bps);
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}
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coeff_prec = get_bits(&s->gb, 4) + 1;
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if (coeff_prec == 16) {
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audio_codec_print("invalid coeff precision\n");
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return -1;
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}
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qlevel = get_sbits(&s->gb, 5);
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if (qlevel < 0) {
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audio_codec_print("qlevel %d not supported, maybe buggy stream\n", qlevel);
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return -1;
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}
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for (i = 0; i < pred_order; i++) {
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coeffs[i] = get_sbits(&s->gb, coeff_prec);
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}
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if (decode_residuals(s, channel, pred_order) < 0) {
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return -1;
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}
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if (s->bps > 16) {
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int64_t sum;
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for (i = pred_order; i < s->blocksize; i++) {
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sum = 0;
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for (j = 0; j < pred_order; j++) {
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sum += (int64_t)coeffs[j] * decoded[i - j - 1];
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}
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decoded[i] += sum >> qlevel;
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}
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} else {
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for (i = pred_order; i < s->blocksize - 1; i += 2) {
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int c;
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int d = decoded[i - pred_order];
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int s0 = 0, s1 = 0;
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for (j = pred_order - 1; j > 0; j--) {
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c = coeffs[j];
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s0 += c * d;
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d = decoded[i - j];
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s1 += c * d;
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}
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c = coeffs[0];
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s0 += c * d;
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d = decoded[i] += s0 >> qlevel;
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s1 += c * d;
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decoded[i + 1] += s1 >> qlevel;
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}
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if (i < s->blocksize) {
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int sum = 0;
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for (j = 0; j < pred_order; j++) {
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sum += coeffs[j] * decoded[i - j - 1];
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}
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decoded[i] += sum >> qlevel;
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}
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}
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return 0;
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}
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void am_flac_parse_block_header(const uint8_t *block_header,
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int *last, int *type, int *size)
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{
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int tmp = bytestream_get_byte(&block_header);
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if (last) {
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*last = tmp & 0x80;
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}
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if (type) {
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*type = tmp & 0x7F;
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}
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if (size) {
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*size = bytestream_get_be24(&block_header);
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}
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}
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static inline int decode_subframe(FLACContext *s, int channel)
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{
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int type, wasted = 0;
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int i, tmp;
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s->curr_bps = s->bps;
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if (channel == 0) {
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if (s->ch_mode == FLAC_CHMODE_RIGHT_SIDE) {
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s->curr_bps++;
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}
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} else {
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if (s->ch_mode == FLAC_CHMODE_LEFT_SIDE || s->ch_mode == FLAC_CHMODE_MID_SIDE) {
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s->curr_bps++;
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}
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}
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if (get_bits1(&s->gb)) {
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audio_codec_print("invalid subframe padding\n");
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return -1;
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}
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type = get_bits(&s->gb, 6);
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if (get_bits1(&s->gb)) {
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wasted = 1;
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while (!get_bits1(&s->gb)) {
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wasted++;
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}
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s->curr_bps -= wasted;
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}
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if (s->curr_bps > 32) {
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audio_codec_print("decorrelated bit depth > 32");
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return -1;
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}
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|
|
//FIXME use av_log2 for types
|
|
if (type == 0) {
|
|
tmp = get_sbits_long(&s->gb, s->curr_bps);
|
|
for (i = 0; i < s->blocksize; i++) {
|
|
s->decoded[channel][i] = tmp;
|
|
}
|
|
} else if (type == 1) {
|
|
for (i = 0; i < s->blocksize; i++) {
|
|
s->decoded[channel][i] = get_sbits_long(&s->gb, s->curr_bps);
|
|
}
|
|
} else if ((type >= 8) && (type <= 12)) {
|
|
if (decode_subframe_fixed(s, channel, type & ~0x8) < 0) {
|
|
return -1;
|
|
}
|
|
} else if (type >= 32) {
|
|
if (decode_subframe_lpc(s, channel, (type & ~0x20) + 1) < 0) {
|
|
return -1;
|
|
}
|
|
} else {
|
|
audio_codec_print("invalid coding type\n");
|
|
return -1;
|
|
}
|
|
|
|
if (wasted) {
|
|
int i;
|
|
for (i = 0; i < s->blocksize; i++) {
|
|
s->decoded[channel][i] <<= wasted;
|
|
}
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int decode_frame_header(GetBitContext *gb, FLACFrameInfo *fi)
|
|
{
|
|
int bs_code, sr_code, bps_code;
|
|
|
|
/* frame sync code */
|
|
//skip_bits(gb, 16);
|
|
if ((get_bits(gb, 15) & 0x7FFF) != 0x7FFC) {
|
|
audio_codec_print("invalid sync code-----------------------\n");
|
|
return -1;
|
|
}
|
|
skip_bits(gb, 1);
|
|
|
|
/* block size and sample rate codes */
|
|
bs_code = get_bits(gb, 4);
|
|
sr_code = get_bits(gb, 4);
|
|
|
|
/* channels and decorrelation */
|
|
fi->ch_mode = get_bits(gb, 4);
|
|
if (fi->ch_mode < FLAC_MAX_CHANNELS) {
|
|
fi->channels = fi->ch_mode + 1;
|
|
fi->ch_mode = FLAC_CHMODE_INDEPENDENT;
|
|
} else if (fi->ch_mode <= FLAC_CHMODE_MID_SIDE) {
|
|
fi->channels = 2;
|
|
} else {
|
|
audio_codec_print("invalid channel mode: %d\n", fi->ch_mode);
|
|
return -1;
|
|
}
|
|
|
|
/* bits per sample */
|
|
bps_code = get_bits(gb, 3);
|
|
if (bps_code == 3 || bps_code == 7) {
|
|
audio_codec_print("invalid sample size code (%d)\n", bps_code);
|
|
return -1;
|
|
}
|
|
fi->bps = sample_size_table[bps_code];
|
|
|
|
/* reserved bit */
|
|
if (get_bits1(gb)) {
|
|
audio_codec_print("broken stream, invalid padding\n");
|
|
return -1;
|
|
}
|
|
|
|
/* sample or frame count */
|
|
if (get_utf8(gb) < 0) {
|
|
audio_codec_print("utf8 fscked\n");
|
|
return -1;
|
|
}
|
|
|
|
/* blocksize */
|
|
if (bs_code == 0) {
|
|
audio_codec_print("reserved blocksize code: 0\n");
|
|
return -1;
|
|
} else if (bs_code == 6) {
|
|
fi->blocksize = get_bits(gb, 8) + 1;
|
|
} else if (bs_code == 7) {
|
|
fi->blocksize = get_bits(gb, 16) + 1;
|
|
} else {
|
|
fi->blocksize = ff_flac_blocksize_table[bs_code];
|
|
}
|
|
|
|
/* sample rate */
|
|
if (sr_code < 12) {
|
|
fi->samplerate = ff_flac_sample_rate_table[sr_code];
|
|
} else if (sr_code == 12) {
|
|
fi->samplerate = get_bits(gb, 8) * 1000;
|
|
} else if (sr_code == 13) {
|
|
fi->samplerate = get_bits(gb, 16);
|
|
} else if (sr_code == 14) {
|
|
fi->samplerate = get_bits(gb, 16) * 10;
|
|
} else {
|
|
audio_codec_print("illegal sample rate code %d\n", sr_code);
|
|
return -1;
|
|
}
|
|
|
|
/* header CRC-8 check */
|
|
skip_bits(gb, 8);
|
|
if (av_crc(av_crc_get_table(AV_CRC_8_ATM), 0, gb->buffer,
|
|
get_bits_count(gb) / 8)) {
|
|
audio_codec_print("header crc mismatch\n");
|
|
return -1;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int decode_frame(FLACContext *s)
|
|
{
|
|
int i;
|
|
GetBitContext *gb = &s->gb;
|
|
FLACFrameInfo fi;
|
|
|
|
if (decode_frame_header(gb, &fi)) {
|
|
//audio_codec_print("invalid frame header\n");
|
|
return -1;
|
|
}
|
|
|
|
if (fi.channels != s->channels) {
|
|
audio_codec_print("switching channel layout[%d->%d] mid-stream is not supported\n", s->channels, fi.channels);
|
|
return -1;
|
|
}
|
|
s->ch_mode = fi.ch_mode;
|
|
|
|
if (fi.bps && fi.bps != s->bps) {
|
|
audio_codec_print("switching bps[%d->%d] mid-stream is not supported\n", s->bps, fi.bps);
|
|
return -1;
|
|
}
|
|
if (0/*s->bps > 16*/) {
|
|
s->avctx->sample_fmt = SAMPLE_FMT_S32;
|
|
s->sample_shift = 32 - s->bps;
|
|
s->is32 = 1;
|
|
} else {
|
|
// s->avctx->sample_fmt = SAMPLE_FMT_S16;
|
|
s->sample_shift = 16 - s->bps;
|
|
s->is32 = 0;
|
|
}
|
|
|
|
if (fi.blocksize > s->max_blocksize) {
|
|
audio_codec_print("blocksize %d > %d\n", fi.blocksize, s->max_blocksize);
|
|
return -1;
|
|
}
|
|
s->blocksize = fi.blocksize;
|
|
|
|
if (fi.samplerate == 0) {
|
|
fi.samplerate = s->samplerate;
|
|
} else if (fi.samplerate != s->samplerate) {
|
|
audio_codec_print("sample rate changed from %d to %d\n", s->samplerate, fi.samplerate);
|
|
}
|
|
s->samplerate = s->avctx->sample_rate = fi.samplerate;
|
|
|
|
|
|
/* subframes */
|
|
for (i = 0; i < s->channels; i++) {
|
|
if (decode_subframe(s, i) < 0) {
|
|
return -1;
|
|
}
|
|
}
|
|
|
|
align_get_bits(gb);
|
|
|
|
/* frame footer */
|
|
skip_bits(gb, 16); /* data crc */
|
|
|
|
return 0;
|
|
}
|
|
|
|
static inline int av_clipf1(float a, int amin, int amax)
|
|
{
|
|
if (a < amin) {
|
|
return (int)amin;
|
|
} else if (a > amax) {
|
|
return (int)amax;
|
|
} else {
|
|
return (int)a;
|
|
}
|
|
}
|
|
int audio_dec_decode(audio_decoder_operations_t *adec_ops, char *outbuf, int *outlen, char *inbuf, int inlen)
|
|
{
|
|
AVCodecContext *avctx = &acodec;
|
|
FLACContext *s = &flactext;
|
|
int i, j = 0, bytes_read = 0;
|
|
int16_t *samples_16 = (int16_t*)outbuf;
|
|
int32_t *samples_32 = (int32_t*)outbuf;
|
|
int output_size = 0;
|
|
unsigned char *buf;
|
|
unsigned buf_size;
|
|
int last_sync_pos = 0;
|
|
|
|
int max_framesize = FFMAX(17 * s->max_framesize / 16 + 32, s->max_framesize);
|
|
if (inlen < max_framesize) {
|
|
int inbufindex = 0;
|
|
char *pinbufptr = inbuf;
|
|
char para = *(pinbufptr + 2);
|
|
int frame_num = 0;
|
|
|
|
inbufindex += 11;
|
|
pinbufptr += 11;
|
|
while (inbufindex < inlen - FLAC_MIN_FRAME_SIZE + 2) {
|
|
if ((AV_RB16(pinbufptr) & 0xFFFF) != 0xFFF8 || AV_RB8(pinbufptr + 2) != para) {
|
|
pinbufptr++;
|
|
inbufindex++;
|
|
} else {
|
|
inlen = inbufindex;
|
|
frame_num++;
|
|
if (frame_num > 1) {
|
|
audio_codec_print("## next frame found, inbufindex=%d, frame_num=%d,-------\n", inbufindex, frame_num);
|
|
goto decodecontinue;
|
|
}
|
|
}
|
|
}
|
|
//audio_codec_print("##inlen=%d, diff=%d,-------------------\n", inlen, pinbufptr-inbuf);
|
|
return -1;
|
|
}
|
|
|
|
decodecontinue:
|
|
s->bitstream = inbuf;
|
|
s->bitstream_size = inlen;
|
|
s->bitstream_index = 0;
|
|
|
|
buf = s->bitstream;
|
|
buf_size = s->bitstream_size;
|
|
|
|
/* check for inline header */
|
|
if (AV_RB32(buf) == MKBETAG('f', 'L', 'a', 'C')) {
|
|
if (!s->got_streaminfo && parse_streaminfo(s, buf, buf_size)) {
|
|
audio_codec_print("invalid header\n");
|
|
return -1;
|
|
}
|
|
bytes_read = get_metadata_size(buf, buf_size);
|
|
goto end;
|
|
}
|
|
FIND_SYNC_WORD:
|
|
/* check for frame sync code and resync stream if necessary */
|
|
if ((AV_RB16(buf) & 0xFFFE) != 0xFFF8) {
|
|
const uint8_t *buf_end = buf + buf_size;
|
|
// av_log(s->avctx, AV_LOG_ERROR, "FRAME HEADER not here\n");
|
|
while (buf + 2 < buf_end && (AV_RB16(buf) & 0xFFFE) != 0xFFF8) {
|
|
buf++;
|
|
last_sync_pos++;
|
|
}
|
|
bytes_read = buf_size - (buf_end - buf);
|
|
goto end; // we may not have enough bits left to decode a frame, so try next time
|
|
}
|
|
|
|
/* decode frame */
|
|
init_get_bits(&s->gb, buf, buf_size * 8);
|
|
if (decode_frame(s) < 0) {
|
|
audio_codec_print("decode_frame() failed\n");
|
|
flactext.error_num++;
|
|
//some times , flac may seek to fake sync word pos, caused decode failed
|
|
//call resync to fix this issue
|
|
buf = s->bitstream + last_sync_pos + 1;
|
|
buf_size = s->bitstream_size - (last_sync_pos + 1);
|
|
last_sync_pos++;
|
|
goto FIND_SYNC_WORD;
|
|
}
|
|
bytes_read = (get_bits_count(&s->gb) + 7) / 8;
|
|
|
|
/* check if allocated data size is large enough for output */
|
|
output_size = s->blocksize * (s->channels > 2 ? 2 : s->channels) * (s->is32 ? 4 : 2);
|
|
if (output_size > DefaultOutBufSize) {
|
|
audio_codec_print("output data size is larger than allocated data size\n");
|
|
goto end;
|
|
}
|
|
|
|
#define DECORRELATE(left, right)\
|
|
for (i = 0; i < s->blocksize; i++) {\
|
|
int a= s->decoded[0][i];\
|
|
int b= s->decoded[1][i];\
|
|
if (s->is32) {\
|
|
*samples_32++ = (left) << s->sample_shift;\
|
|
*samples_32++ = (right) << s->sample_shift;\
|
|
} else {\
|
|
if (s->sample_shift >= 0) \
|
|
{\
|
|
*samples_16++ = (left) << s->sample_shift;\
|
|
*samples_16++ = (right) << s->sample_shift;\
|
|
}\
|
|
else\
|
|
{\
|
|
*samples_16++ = (left) >> (-1*s->sample_shift);\
|
|
*samples_16++ = (right) >> (-1*s->sample_shift);\
|
|
}\
|
|
}\
|
|
}\
|
|
break;
|
|
|
|
switch (s->ch_mode) {
|
|
case FLAC_CHMODE_INDEPENDENT:
|
|
if (s->channels <= 2) {
|
|
for (j = 0; j < s->blocksize; j++) {
|
|
for (i = 0; i < s->channels; i++) {
|
|
if (s->is32) {
|
|
*samples_32++ = s->decoded[i][j] << s->sample_shift;
|
|
} else {
|
|
if (s->sample_shift >= 0) {
|
|
*samples_16++ = s->decoded[i][j] << s->sample_shift;
|
|
} else {
|
|
*samples_16++ = s->decoded[i][j] >> (-1 * s->sample_shift);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
} else {
|
|
float sum0 = 0, sum1 = 0;
|
|
//downmix here
|
|
//FL + (sum of other surround channels >> 1)
|
|
//FR + (sum of other surround channels >> 1)
|
|
for (j = 0; j < s->blocksize; j++) {
|
|
if (s->is32) {
|
|
sum0 = s->decoded[0][j] << s->sample_shift;
|
|
sum1 = s->decoded[1][j] << s->sample_shift;
|
|
} else {
|
|
if (s->sample_shift >= 0) {
|
|
sum0 = s->decoded[0][j] << s->sample_shift;
|
|
sum1 = s->decoded[1][j] << s->sample_shift;
|
|
} else {
|
|
sum0 = s->decoded[0][j] >> (-1 * s->sample_shift);
|
|
sum1 = s->decoded[1][j] >> (-1 * s->sample_shift);
|
|
}
|
|
}
|
|
for (i = 2; i < s->channels; i++) {
|
|
if (s->is32) {
|
|
sum0 += (s->decoded[i][j] << s->sample_shift) >> 1;
|
|
sum1 += (s->decoded[i][j] << s->sample_shift) >> 1;
|
|
} else {
|
|
if (s->sample_shift >= 0) {
|
|
sum0 += (s->decoded[i][j] << s->sample_shift) >> 1;
|
|
sum1 += (s->decoded[i][j] << s->sample_shift) >> 1;
|
|
} else {
|
|
sum0 += (s->decoded[i][j] >> (-1 * s->sample_shift)) >> 1;
|
|
sum1 += (s->decoded[i][j] >> (-1 * s->sample_shift)) >> 1;
|
|
}
|
|
}
|
|
}
|
|
|
|
if (s->is32) {
|
|
*samples_32++ = av_clipf1(sum0, -0x80000000, 0x7fffffff);
|
|
*samples_32++ = av_clipf1(sum1, -0x80000000, 0x7fffffff);
|
|
} else {
|
|
*samples_16++ = av_clipf1(sum0, -32768, 32767);
|
|
*samples_16++ = av_clipf1(sum1, -32768, 32767);
|
|
}
|
|
}
|
|
}
|
|
break;
|
|
case FLAC_CHMODE_LEFT_SIDE:
|
|
DECORRELATE(a, a - b)
|
|
case FLAC_CHMODE_RIGHT_SIDE:
|
|
DECORRELATE(a + b, b)
|
|
case FLAC_CHMODE_MID_SIDE:
|
|
DECORRELATE((a -= b >> 1) + b, a)
|
|
}
|
|
|
|
end:
|
|
if (bytes_read > inlen) {
|
|
// av_log(s->avctx, AV_LOG_ERROR, "overread: %d\n", bytes_read - buf_size);
|
|
s->bitstream_size = 0;
|
|
s->bitstream_index = 0;
|
|
return -1;
|
|
}
|
|
|
|
if (s->bitstream_size) {
|
|
s->bitstream_index += bytes_read;
|
|
s->bitstream_size -= bytes_read;
|
|
}
|
|
|
|
*outlen = output_size;
|
|
|
|
return bytes_read;
|
|
}
|
|
|
|
int audio_dec_init(audio_decoder_operations_t *adec_ops)
|
|
{
|
|
memset(&flactext, 0, sizeof(flactext));
|
|
memset(&acodec, 0, sizeof(acodec));
|
|
enum FLACExtradataFormat format;
|
|
uint8_t *streaminfo;
|
|
AVCodecContext *avctx = &acodec;
|
|
FLACContext *s = &flactext;
|
|
s->avctx = &acodec;
|
|
audio_codec_print("[%s]BuildDate--%s BuildTime--%s\n", __FUNCTION__, __DATE__, __TIME__);
|
|
avctx->sample_fmt = SAMPLE_FMT_S16;
|
|
avctx->extradata = adec_ops->extradata;
|
|
avctx->extradata_size = adec_ops->extradata_size;
|
|
|
|
if (!avctx->extradata_size) {
|
|
audio_codec_print("%s[%d]extradata_size is %d\n", __func__, __LINE__, avctx->extradata_size);
|
|
return 0;
|
|
}
|
|
if (!am_flac_is_extradata_valid(avctx, &format, &streaminfo)) {
|
|
return -1;
|
|
}
|
|
|
|
/* initialize based on the demuxer-supplied streamdata header */
|
|
am_flac_parse_streaminfo(avctx, (FLACStreaminfo *)s, streaminfo);
|
|
if (s->bps > 16) {
|
|
avctx->sample_fmt = SAMPLE_FMT_S32;
|
|
} else {
|
|
avctx->sample_fmt = SAMPLE_FMT_S16;
|
|
}
|
|
allocate_buffers(s);
|
|
s->got_streaminfo = 1;
|
|
avctx->channels = (avctx->channels > 2 ? 2 : avctx->channels);
|
|
audio_codec_print("applied flac sr %d,ch num %d\n", avctx->sample_rate, avctx->channels);
|
|
|
|
adec_ops->nInBufSize = DefaultReadSize;
|
|
adec_ops->nOutBufSize = DefaultOutBufSize;
|
|
audio_codec_print("ape_Init.--------------------------------\n");
|
|
|
|
return 0;
|
|
}
|
|
|
|
int audio_dec_release(audio_decoder_operations_t *adec_ops)
|
|
{
|
|
int i;
|
|
audio_codec_print("audio_dec_release.--------------------------------\n");
|
|
|
|
if (outbuffer.outb != NULL) {
|
|
//av_freep(&outbuffer.outb);
|
|
free(outbuffer.outb);
|
|
outbuffer.outb = NULL;
|
|
}
|
|
|
|
for (i = 0; i < flactext.channels; i++) {
|
|
if (flactext.decoded[i] != NULL) {
|
|
//av_freep(&flactext.decoded[i]);
|
|
free(flactext.decoded[i]);
|
|
flactext.decoded[i] = NULL;
|
|
}
|
|
}
|
|
//av_free(flactext.bitstream);
|
|
|
|
return 0;
|
|
}
|
|
|
|
int audio_dec_getinfo(audio_decoder_operations_t *adec_ops, void *pAudioInfo)
|
|
{
|
|
if (pAudioInfo != NULL) {
|
|
((AudioInfo *)pAudioInfo)->channels = flactext.channels;
|
|
((AudioInfo *)pAudioInfo)->samplerate = flactext.samplerate;
|
|
((AudioInfo *)pAudioInfo)->error_num = flactext.error_num;
|
|
((AudioInfo *)pAudioInfo)->drop_num = flactext.error_num;
|
|
#if 0
|
|
audio_codec_print("%s:%d ch(%d) sr(%d) stream_error_num(%u) ",
|
|
__FUNCTION__, __LINE__, flactext.channels , flactext.samplerate, flactext.error_num );
|
|
#endif
|
|
}
|
|
return 0;
|
|
}
|
|
|