/* * Copyright (C) 2017 Amlogic, Inc. All rights reserved. * * 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. * Description: adpcm decoder */ /* * filename: flac_decode.c * * * * * */ #include #include #include #include "codec.h" #ifndef __MW__ #include #else #include "types.h" #endif #include "crc.h" #include "avcodec.h" #include "get_bits.h" #include "golomb.h" #include "flac.h" #include "flacdata.h" #include "adec-armdec-mgt.h" //#include "../../amadec/adec-armdec-mgt.h" #define LOG_TAG "FlacDecoder" #include #define audio_codec_print(...) __android_log_print(ANDROID_LOG_INFO,LOG_TAG,__VA_ARGS__) #define DefaultReadSize 1024*10 //read count from kernel audio buf one time #define DefaultOutBufSize 1024*1024 typedef struct OutData { uint8_t * outb; int byte; int index; } OutData; typedef struct FLACContext { FLACSTREAMINFO AVCodecContext *avctx; ///< parent AVCodecContext GetBitContext gb; ///< GetBitContext initialized to start at the current frame int blocksize; ///< number of samples in the current frame int curr_bps; ///< bps for current subframe, adjusted for channel correlation and wasted bits int sample_shift; ///< shift required to make output samples 16-bit or 32-bit int is32; ///< flag to indicate if output should be 32-bit instead of 16-bit int ch_mode; ///< channel decorrelation type in the current frame int got_streaminfo; ///< indicates if the STREAMINFO has been read int32_t *decoded[FLAC_MAX_CHANNELS]; ///< decoded samples uint8_t *bitstream; unsigned int bitstream_size; unsigned int bitstream_index; unsigned int allocated_bitstream_size; unsigned int error_num; ///< error decoded num(for get error num) } FLACContext; static const int sample_size_table[] = { 0, 8, 12, 0, 16, 20, 24, 0 }; static AVCodecContext acodec; static FLACContext flactext; static OutData outbuffer; #define INPUT_BUF_SIZE 32768 static int64_t get_utf8(GetBitContext *gb) { int64_t val; GET_UTF8(val, get_bits(gb, 8), return -1;) return val; } void *av_malloc(unsigned int size) { void *ptr = NULL; ptr = malloc(size); return ptr; } void *av_realloc(void *ptr, unsigned int size) { return realloc(ptr, size); } void av_freep(void **arg) { if (*arg) { free(*arg); } *arg = NULL; } void av_free(void *arg) { if (arg) { free(arg); arg = NULL; } } void *av_mallocz(unsigned int size) { void *ptr = malloc(size); if (ptr) { memset(ptr, 0, size); } return ptr; } static inline uint32_t bytestream_get_be32(const uint8_t** ptr) { uint32_t tmp; tmp = (*ptr)[3] | ((*ptr)[2] << 8) | ((*ptr)[1] << 16) | ((*ptr)[0] << 24); *ptr += 4; return tmp; } static inline uint32_t bytestream_get_be24(const uint8_t** ptr) { uint32_t tmp; tmp = (*ptr)[2] | ((*ptr)[1] << 8) | ((*ptr)[0] << 16) ; *ptr += 3; return tmp; } static inline uint8_t bytestream_get_byte(const uint8_t** ptr) { uint8_t tmp; tmp = **ptr; *ptr += 1; return tmp; } void *av_fast_realloc(void *ptr, unsigned int *size, unsigned int min_size) { if (min_size < *size) { return ptr; } *size = FFMAX(17 * min_size / 16 + 32, min_size); ptr = av_realloc(ptr, *size); 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 *size = 0; } return ptr; } static void allocate_buffers(FLACContext *s) { int i; if (s->max_framesize == 0 && s->max_blocksize) { s->max_framesize = am_flac_get_max_frame_size(s->max_blocksize, s->channels, s->bps); } for (i = 0; i < s->channels; i++) { s->decoded[i] = av_realloc(s->decoded[i], sizeof(int32_t) * s->max_blocksize); if (s->decoded[i] == NULL) { audio_codec_print("av_realloc decoded[%d] NULL, %s,%d", i, __func__, __LINE__); } } } /** * Parse the STREAMINFO from an inline header. * @param s the flac decoding context * @param buf input buffer, starting with the "fLaC" marker * @param buf_size buffer size * @return non-zero if metadata is invalid */ static int parse_streaminfo(FLACContext *s, const uint8_t *buf, int buf_size) { int metadata_type, metadata_size; if (buf_size < FLAC_STREAMINFO_SIZE + 8) { /* need more data */ return 0; } am_flac_parse_block_header(&buf[4], NULL, &metadata_type, &metadata_size); if (metadata_type != FLAC_METADATA_TYPE_STREAMINFO || metadata_size != FLAC_STREAMINFO_SIZE) { return -1;//AVERROR_INVALIDDATA; } am_flac_parse_streaminfo(s->avctx, (FLACStreaminfo *)s, &buf[8]); allocate_buffers(s); s->got_streaminfo = 1; return 0; } /** * Determine the size of an inline header. * @param buf input buffer, starting with the "fLaC" marker * @param buf_size buffer size * @return number of bytes in the header, or 0 if more data is needed */ static int get_metadata_size(const uint8_t *buf, int buf_size) { int metadata_last, metadata_size; const uint8_t *buf_end = buf + buf_size; buf += 4; do { am_flac_parse_block_header(buf, &metadata_last, NULL, &metadata_size); buf += 4; if (buf + metadata_size > buf_end) { /* need more data in order to read the complete header */ return 0; } buf += metadata_size; } while (!metadata_last); return buf_size - (buf_end - buf); } void am_flac_parse_streaminfo(AVCodecContext *avctx, struct FLACStreaminfo *s, const uint8_t *buffer) { GetBitContext gb; init_get_bits(&gb, buffer, FLAC_STREAMINFO_SIZE * 8); skip_bits(&gb, 16); /* skip min blocksize */ s->max_blocksize = get_bits(&gb, 16); if (s->max_blocksize < FLAC_MIN_BLOCKSIZE) { audio_codec_print("invalid max blocksize: %d\n", s->max_blocksize); s->max_blocksize = 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; audio_codec_print("## METADATA sr=%d, ch=%d, bps=%d,-------------\n", s->samplerate, s->channels, s->bps); avctx->channels = s->channels; avctx->sample_rate = s->samplerate; avctx->bits_per_raw_sample = s->bps; s->samples = get_bits_long(&gb, 32) << 4; s->samples |= get_bits(&gb, 4); skip_bits_long(&gb, 64); /* md5 sum */ skip_bits_long(&gb, 64); /* md5 sum */ } static void allocate_buffers(FLACContext *s); int am_flac_is_extradata_valid(AVCodecContext *avctx, enum FLACExtradataFormat *format, uint8_t **streaminfo_start) { if (!avctx->extradata || avctx->extradata_size < FLAC_STREAMINFO_SIZE) { audio_codec_print("extradata NULL or too small.\n"); return 0; } if (AV_RL32(avctx->extradata) != MKTAG('f', 'L', 'a', 'C')) { /* extradata contains STREAMINFO only */ if (avctx->extradata_size != FLAC_STREAMINFO_SIZE) { audio_codec_print("extradata contains %d bytes too many.\n", FLAC_STREAMINFO_SIZE - avctx->extradata_size); } *format = FLAC_EXTRADATA_FORMAT_STREAMINFO; *streaminfo_start = avctx->extradata; } else { if (avctx->extradata_size < 8 + FLAC_STREAMINFO_SIZE) { audio_codec_print("extradata too small.\n"); return 0; } *format = FLAC_EXTRADATA_FORMAT_FULL_HEADER; *streaminfo_start = &avctx->extradata[8]; } return 1; } 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) { audio_codec_print("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) { audio_codec_print("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)) { tmp = get_bits(&s->gb, 5); for (; i < samples; i++, sample++) { s->decoded[channel][sample] = get_sbits_long(&s->gb, tmp); } } else { for (; i < samples; i++, sample++) { s->decoded[channel][sample] = get_sr_golomb_flac(&s->gb, tmp, INT_MAX, 0); } } i = 0; } 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 av_uninit(a), av_uninit(b), av_uninit(c), av_uninit(d), i; /* warm up samples */ for (i = 0; i < pred_order; i++) { decoded[i] = get_sbits_long(&s->gb, s->curr_bps); } 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: audio_codec_print("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[32]; int32_t *decoded = s->decoded[channel]; /* warm up samples */ for (i = 0; i < pred_order; i++) { decoded[i] = get_sbits_long(&s->gb, s->curr_bps); } coeff_prec = get_bits(&s->gb, 4) + 1; if (coeff_prec == 16) { audio_codec_print("invalid coeff precision\n"); return -1; } qlevel = get_sbits(&s->gb, 5); if (qlevel < 0) { audio_codec_print("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); } if (decode_residuals(s, channel, pred_order) < 0) { return -1; } 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; } } 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; } void am_flac_parse_block_header(const uint8_t *block_header, int *last, int *type, int *size) { int tmp = bytestream_get_byte(&block_header); if (last) { *last = tmp & 0x80; } if (type) { *type = tmp & 0x7F; } if (size) { *size = bytestream_get_be24(&block_header); } } 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->ch_mode == FLAC_CHMODE_RIGHT_SIDE) { s->curr_bps++; } } else { if (s->ch_mode == FLAC_CHMODE_LEFT_SIDE || s->ch_mode == FLAC_CHMODE_MID_SIDE) { s->curr_bps++; } } if (get_bits1(&s->gb)) { audio_codec_print("invalid subframe padding\n"); return -1; } type = get_bits(&s->gb, 6); if (get_bits1(&s->gb)) { wasted = 1; while (!get_bits1(&s->gb)) { wasted++; } s->curr_bps -= wasted; } if (s->curr_bps > 32) { audio_codec_print("decorrelated bit depth > 32"); return -1; } //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; }