Mercurial > libavcodec.hg
view flacenc.c @ 3362:c43fcf831f7c libavcodec
Do not read full byte when less than 8 bits are still to be read.
Does not make a difference with any of my samples, but current
code does not make much sense.
author | reimar |
---|---|
date | Tue, 27 Jun 2006 12:13:01 +0000 |
parents | 4ae69b5b596b |
children | 84f29207af3a |
line wrap: on
line source
/** * FLAC audio encoder * Copyright (c) 2006 Justin Ruggles <jruggle@earthlink.net> * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library 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 * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public * License along with this library; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ #include "avcodec.h" #include "bitstream.h" #include "crc.h" #include "golomb.h" #define FLAC_MAX_CH 8 #define FLAC_MIN_BLOCKSIZE 16 #define FLAC_MAX_BLOCKSIZE 65535 #define FLAC_SUBFRAME_CONSTANT 0 #define FLAC_SUBFRAME_VERBATIM 1 #define FLAC_SUBFRAME_FIXED 8 #define FLAC_SUBFRAME_LPC 32 #define FLAC_CHMODE_NOT_STEREO 0 #define FLAC_CHMODE_LEFT_RIGHT 1 #define FLAC_CHMODE_LEFT_SIDE 8 #define FLAC_CHMODE_RIGHT_SIDE 9 #define FLAC_CHMODE_MID_SIDE 10 #define FLAC_STREAMINFO_SIZE 34 typedef struct FlacSubframe { int type; int type_code; int obits; int order; int32_t samples[FLAC_MAX_BLOCKSIZE]; int32_t residual[FLAC_MAX_BLOCKSIZE]; } FlacSubframe; typedef struct FlacFrame { FlacSubframe subframes[FLAC_MAX_CH]; int blocksize; int bs_code[2]; uint8_t crc8; int ch_mode; } FlacFrame; typedef struct FlacEncodeContext { PutBitContext pb; int channels; int ch_code; int samplerate; int sr_code[2]; int blocksize; int max_framesize; uint32_t frame_count; FlacFrame frame; } FlacEncodeContext; static const int flac_samplerates[16] = { 0, 0, 0, 0, 8000, 16000, 22050, 24000, 32000, 44100, 48000, 96000, 0, 0, 0, 0 }; static const int flac_blocksizes[16] = { 0, 192, 576, 1152, 2304, 4608, 0, 0, 256, 512, 1024, 2048, 4096, 8192, 16384, 32768 }; /** * Writes streaminfo metadata block to byte array */ static void write_streaminfo(FlacEncodeContext *s, uint8_t *header) { PutBitContext pb; memset(header, 0, FLAC_STREAMINFO_SIZE); init_put_bits(&pb, header, FLAC_STREAMINFO_SIZE); /* streaminfo metadata block */ put_bits(&pb, 16, s->blocksize); put_bits(&pb, 16, s->blocksize); put_bits(&pb, 24, 0); put_bits(&pb, 24, s->max_framesize); put_bits(&pb, 20, s->samplerate); put_bits(&pb, 3, s->channels-1); put_bits(&pb, 5, 15); /* bits per sample - 1 */ flush_put_bits(&pb); /* total samples = 0 */ /* MD5 signature = 0 */ } #define BLOCK_TIME_MS 105 /** * Sets blocksize based on samplerate * Chooses the closest predefined blocksize >= BLOCK_TIME_MS milliseconds */ static int select_blocksize(int samplerate) { int i; int target; int blocksize; assert(samplerate > 0); blocksize = flac_blocksizes[1]; target = (samplerate * BLOCK_TIME_MS) / 1000; for(i=0; i<16; i++) { if(target >= flac_blocksizes[i] && flac_blocksizes[i] > blocksize) { blocksize = flac_blocksizes[i]; } } return blocksize; } static int flac_encode_init(AVCodecContext *avctx) { int freq = avctx->sample_rate; int channels = avctx->channels; FlacEncodeContext *s = avctx->priv_data; int i; uint8_t *streaminfo; if(avctx->sample_fmt != SAMPLE_FMT_S16) { return -1; } if(channels < 1 || channels > FLAC_MAX_CH) { return -1; } s->channels = channels; s->ch_code = s->channels-1; /* find samplerate in table */ if(freq < 1) return -1; for(i=4; i<12; i++) { if(freq == flac_samplerates[i]) { s->samplerate = flac_samplerates[i]; s->sr_code[0] = i; s->sr_code[1] = 0; break; } } /* if not in table, samplerate is non-standard */ if(i == 12) { if(freq % 1000 == 0 && freq < 255000) { s->sr_code[0] = 12; s->sr_code[1] = freq / 1000; } else if(freq % 10 == 0 && freq < 655350) { s->sr_code[0] = 14; s->sr_code[1] = freq / 10; } else if(freq < 65535) { s->sr_code[0] = 13; s->sr_code[1] = freq; } else { return -1; } s->samplerate = freq; } s->blocksize = select_blocksize(s->samplerate); avctx->frame_size = s->blocksize; /* set maximum encoded frame size in verbatim mode */ if(s->channels == 2) { s->max_framesize = 14 + ((s->blocksize * 33 + 7) >> 3); } else { s->max_framesize = 14 + (s->blocksize * s->channels * 2); } streaminfo = av_malloc(FLAC_STREAMINFO_SIZE); write_streaminfo(s, streaminfo); avctx->extradata = streaminfo; avctx->extradata_size = FLAC_STREAMINFO_SIZE; s->frame_count = 0; avctx->coded_frame = avcodec_alloc_frame(); avctx->coded_frame->key_frame = 1; return 0; } static void init_frame(FlacEncodeContext *s) { int i, ch; FlacFrame *frame; frame = &s->frame; for(i=0; i<16; i++) { if(s->blocksize == flac_blocksizes[i]) { frame->blocksize = flac_blocksizes[i]; frame->bs_code[0] = i; frame->bs_code[1] = 0; break; } } if(i == 16) { frame->blocksize = s->blocksize; if(frame->blocksize <= 256) { frame->bs_code[0] = 6; frame->bs_code[1] = frame->blocksize-1; } else { frame->bs_code[0] = 7; frame->bs_code[1] = frame->blocksize-1; } } for(ch=0; ch<s->channels; ch++) { frame->subframes[ch].obits = 16; } } /** * Copy channel-interleaved input samples into separate subframes */ static void copy_samples(FlacEncodeContext *s, int16_t *samples) { int i, j, ch; FlacFrame *frame; frame = &s->frame; for(i=0,j=0; i<frame->blocksize; i++) { for(ch=0; ch<s->channels; ch++,j++) { frame->subframes[ch].samples[i] = samples[j]; } } } static int estimate_stereo_mode(int32_t *left_ch, int32_t *right_ch, int n) { int i, best; int32_t lt, rt; uint64_t left, right, mid, side; uint64_t score[4]; /* calculate sum of squares for each channel */ left = right = mid = side = 0; for(i=2; i<n; i++) { lt = left_ch[i] - 2*left_ch[i-1] + left_ch[i-2]; rt = right_ch[i] - 2*right_ch[i-1] + right_ch[i-2]; mid += ABS((lt + rt) >> 1); side += ABS(lt - rt); left += ABS(lt); right += ABS(rt); } /* calculate score for each mode */ score[0] = left + right; score[1] = left + side; score[2] = right + side; score[3] = mid + side; /* return mode with lowest score */ best = 0; for(i=1; i<4; i++) { if(score[i] < score[best]) { best = i; } } if(best == 0) { return FLAC_CHMODE_LEFT_RIGHT; } else if(best == 1) { return FLAC_CHMODE_LEFT_SIDE; } else if(best == 2) { return FLAC_CHMODE_RIGHT_SIDE; } else { return FLAC_CHMODE_MID_SIDE; } } /** * Perform stereo channel decorrelation */ static void channel_decorrelation(FlacEncodeContext *ctx) { FlacFrame *frame; int32_t *left, *right; int i, n; frame = &ctx->frame; n = frame->blocksize; left = frame->subframes[0].samples; right = frame->subframes[1].samples; if(ctx->channels != 2) { frame->ch_mode = FLAC_CHMODE_NOT_STEREO; return; } frame->ch_mode = estimate_stereo_mode(left, right, n); /* perform decorrelation and adjust bits-per-sample */ if(frame->ch_mode == FLAC_CHMODE_LEFT_RIGHT) { return; } if(frame->ch_mode == FLAC_CHMODE_MID_SIDE) { int32_t tmp; for(i=0; i<n; i++) { tmp = left[i]; left[i] = (tmp + right[i]) >> 1; right[i] = tmp - right[i]; } frame->subframes[1].obits++; } else if(frame->ch_mode == FLAC_CHMODE_LEFT_SIDE) { for(i=0; i<n; i++) { right[i] = left[i] - right[i]; } frame->subframes[1].obits++; } else { for(i=0; i<n; i++) { left[i] -= right[i]; } frame->subframes[0].obits++; } } static void encode_residual_verbatim(FlacEncodeContext *s, int ch) { FlacFrame *frame; FlacSubframe *sub; int32_t *res; int32_t *smp; int n; frame = &s->frame; sub = &frame->subframes[ch]; res = sub->residual; smp = sub->samples; n = frame->blocksize; sub->order = 0; sub->type = FLAC_SUBFRAME_VERBATIM; sub->type_code = sub->type; memcpy(res, smp, n * sizeof(int32_t)); } static void encode_residual_fixed(int32_t *res, int32_t *smp, int n, int order) { int i; for(i=0; i<order; i++) { res[i] = smp[i]; } if(order==0){ for(i=order; i<n; i++) res[i]= smp[i]; }else if(order==1){ for(i=order; i<n; i++) res[i]= smp[i] - smp[i-1]; }else if(order==2){ for(i=order; i<n; i++) res[i]= smp[i] - 2*smp[i-1] + smp[i-2]; }else if(order==3){ for(i=order; i<n; i++) res[i]= smp[i] - 3*smp[i-1] + 3*smp[i-2] - smp[i-3]; }else{ for(i=order; i<n; i++) res[i]= smp[i] - 4*smp[i-1] + 6*smp[i-2] - 4*smp[i-3] + smp[i-4]; } } static void encode_residual(FlacEncodeContext *s, int ch) { FlacFrame *frame; FlacSubframe *sub; int32_t *res; int32_t *smp; int n; frame = &s->frame; sub = &frame->subframes[ch]; res = sub->residual; smp = sub->samples; n = frame->blocksize; sub->order = 2; sub->type = FLAC_SUBFRAME_FIXED; sub->type_code = sub->type | sub->order; encode_residual_fixed(res, smp, n, sub->order); } static void put_sbits(PutBitContext *pb, int bits, int32_t val) { assert(bits >= 0 && bits <= 31); put_bits(pb, bits, val & ((1<<bits)-1)); } static void write_utf8(PutBitContext *pb, uint32_t val) { int bytes, shift; if(val < 0x80){ put_bits(pb, 8, val); return; } bytes= (av_log2(val)+4) / 5; shift = (bytes - 1) * 6; put_bits(pb, 8, (256 - (256>>bytes)) | (val >> shift)); while(shift >= 6){ shift -= 6; put_bits(pb, 8, 0x80 | ((val >> shift) & 0x3F)); } } static void output_frame_header(FlacEncodeContext *s) { FlacFrame *frame; int crc; frame = &s->frame; put_bits(&s->pb, 16, 0xFFF8); put_bits(&s->pb, 4, frame->bs_code[0]); put_bits(&s->pb, 4, s->sr_code[0]); if(frame->ch_mode == FLAC_CHMODE_NOT_STEREO) { put_bits(&s->pb, 4, s->ch_code); } else { put_bits(&s->pb, 4, frame->ch_mode); } put_bits(&s->pb, 3, 4); /* bits-per-sample code */ put_bits(&s->pb, 1, 0); write_utf8(&s->pb, s->frame_count); if(frame->bs_code[0] == 6) { put_bits(&s->pb, 8, frame->bs_code[1]); } else if(frame->bs_code[0] == 7) { put_bits(&s->pb, 16, frame->bs_code[1]); } if(s->sr_code[0] == 12) { put_bits(&s->pb, 8, s->sr_code[1]); } else if(s->sr_code[0] > 12) { put_bits(&s->pb, 16, s->sr_code[1]); } flush_put_bits(&s->pb); crc = av_crc(av_crc07, 0, s->pb.buf, put_bits_count(&s->pb)>>3); put_bits(&s->pb, 8, crc); } static void output_subframe_verbatim(FlacEncodeContext *s, int ch) { int i; FlacFrame *frame; FlacSubframe *sub; int32_t res; frame = &s->frame; sub = &frame->subframes[ch]; for(i=0; i<frame->blocksize; i++) { res = sub->residual[i]; put_sbits(&s->pb, sub->obits, res); } } static void output_residual(FlacEncodeContext *ctx, int ch) { int i, j, p; int k, porder, psize, res_cnt; FlacFrame *frame; FlacSubframe *sub; frame = &ctx->frame; sub = &frame->subframes[ch]; /* rice-encoded block */ put_bits(&ctx->pb, 2, 0); /* partition order */ porder = 0; psize = frame->blocksize; //porder = sub->rc.porder; //psize = frame->blocksize >> porder; put_bits(&ctx->pb, 4, porder); res_cnt = psize - sub->order; /* residual */ j = sub->order; for(p=0; p<(1 << porder); p++) { //k = sub->rc.params[p]; k = 9; put_bits(&ctx->pb, 4, k); if(p == 1) res_cnt = psize; for(i=0; i<res_cnt && j<frame->blocksize; i++, j++) { set_sr_golomb_flac(&ctx->pb, sub->residual[j], k, INT32_MAX, 0); } } } static void output_subframe_fixed(FlacEncodeContext *ctx, int ch) { int i; FlacFrame *frame; FlacSubframe *sub; frame = &ctx->frame; sub = &frame->subframes[ch]; /* warm-up samples */ for(i=0; i<sub->order; i++) { put_sbits(&ctx->pb, sub->obits, sub->residual[i]); } /* residual */ output_residual(ctx, ch); } static void output_subframes(FlacEncodeContext *s) { FlacFrame *frame; FlacSubframe *sub; int ch; frame = &s->frame; for(ch=0; ch<s->channels; ch++) { sub = &frame->subframes[ch]; /* subframe header */ put_bits(&s->pb, 1, 0); put_bits(&s->pb, 6, sub->type_code); put_bits(&s->pb, 1, 0); /* no wasted bits */ /* subframe */ if(sub->type == FLAC_SUBFRAME_VERBATIM) { output_subframe_verbatim(s, ch); } else { output_subframe_fixed(s, ch); } } } static void output_frame_footer(FlacEncodeContext *s) { int crc; flush_put_bits(&s->pb); crc = bswap_16(av_crc(av_crc8005, 0, s->pb.buf, put_bits_count(&s->pb)>>3)); put_bits(&s->pb, 16, crc); flush_put_bits(&s->pb); } static int flac_encode_frame(AVCodecContext *avctx, uint8_t *frame, int buf_size, void *data) { int ch; FlacEncodeContext *s; int16_t *samples = data; int out_bytes; s = avctx->priv_data; s->blocksize = avctx->frame_size; init_frame(s); copy_samples(s, samples); channel_decorrelation(s); for(ch=0; ch<s->channels; ch++) { encode_residual(s, ch); } init_put_bits(&s->pb, frame, buf_size); output_frame_header(s); output_subframes(s); output_frame_footer(s); out_bytes = put_bits_count(&s->pb) >> 3; if(out_bytes > s->max_framesize || out_bytes >= buf_size) { /* frame too large. use verbatim mode */ for(ch=0; ch<s->channels; ch++) { encode_residual_verbatim(s, ch); } init_put_bits(&s->pb, frame, buf_size); output_frame_header(s); output_subframes(s); output_frame_footer(s); out_bytes = put_bits_count(&s->pb) >> 3; if(out_bytes > s->max_framesize || out_bytes >= buf_size) { /* still too large. must be an error. */ av_log(avctx, AV_LOG_ERROR, "error encoding frame\n"); return -1; } } s->frame_count++; return out_bytes; } static int flac_encode_close(AVCodecContext *avctx) { av_freep(&avctx->extradata); avctx->extradata_size = 0; av_freep(&avctx->coded_frame); return 0; } AVCodec flac_encoder = { "flac", CODEC_TYPE_AUDIO, CODEC_ID_FLAC, sizeof(FlacEncodeContext), flac_encode_init, flac_encode_frame, flac_encode_close, NULL, .capabilities = CODEC_CAP_SMALL_LAST_FRAME, };