545 files changed, 230976 insertions, 0 deletions
diff --git a/audio_codec/libflac/flac_decode.c b/audio_codec/libflac/flac_decode.c new file mode 100644 index 0000000..e1ddaaf --- a/dev/null +++ b/audio_codec/libflac/flac_decode.c @@ -0,0 +1,943 @@ +/* +* filename: flac_decode.c +* +* +* +* +* +*/ + +#include <stdio.h> +#include <stdlib.h> +#include <limits.h> +#include "codec.h" +#ifndef __MW__ +#include <inttypes.h> +#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 "../../amadec/adec-armdec-mgt.h" + +#include <android/log.h> + +#define LOG_TAG "FlacDecoder" +#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; +} 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); + } +} +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 = ff_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); + } +} + + +/** + * 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; + } + ff_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; + } + ff_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 { + ff_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 ff_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 sp=%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 ff_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 ff_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 mid-stream is not supported\n"); + return -1; + } + s->ch_mode = fi.ch_mode; + + if (fi.bps && fi.bps != s->bps) { + audio_codec_print("switching bps mid-stream is not supported\n"); + 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, input_buf_size = 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; + + unsigned 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"); + //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("\n\n[%s]BuildDate--%s BuildTime--%s", __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) { + return 0; + } + if (!ff_flac_is_extradata_valid(avctx, &format, &streaminfo)) { + return -1; + } + + /* initialize based on the demuxer-supplied streamdata header */ + ff_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); + } + + for (i = 0; i < flactext.channels; i++) { + av_freep(&flactext.decoded[i]); + } + // av_free(flactext.bitstream); + + return 0; +} + +int audio_dec_getinfo(audio_decoder_operations_t *adec_ops, void *pAudioInfo) +{ + return 0; +} + |