# HG changeset patch # User michael # Date 1151144415 0 # Node ID 5b901881d6ed7209b8112f99e1b0cc3b28edc445 # Parent 608659b58523dd21c49e8fb8dad9431f9f072ff0 first rudimentary version of (Justin Ruggles jruggle earthlink net) flac encoder diff -r 608659b58523 -r 5b901881d6ed Makefile --- a/Makefile Sat Jun 24 09:25:21 2006 +0000 +++ b/Makefile Sat Jun 24 10:20:15 2006 +0000 @@ -68,6 +68,9 @@ ifeq ($(CONFIG_FLAC_DECODER),yes) OBJS+= flac.o endif +ifeq ($(CONFIG_FLAC_ENCODER),yes) + OBJS+= flacenc.o +endif ifeq ($(CONFIG_FLIC_DECODER),yes) OBJS+= flicvideo.o endif diff -r 608659b58523 -r 5b901881d6ed allcodecs.c --- a/allcodecs.c Sat Jun 24 09:25:21 2006 +0000 +++ b/allcodecs.c Sat Jun 24 10:20:15 2006 +0000 @@ -72,6 +72,9 @@ register_avcodec(&faac_encoder); #endif //CONFIG_FAAC_ENCODER #endif +#ifdef CONFIG_FLAC_ENCODER + register_avcodec(&flac_encoder); +#endif #ifdef CONFIG_XVID #ifdef CONFIG_XVID_ENCODER register_avcodec(&xvid_encoder); diff -r 608659b58523 -r 5b901881d6ed avcodec.h --- a/avcodec.h Sat Jun 24 09:25:21 2006 +0000 +++ b/avcodec.h Sat Jun 24 10:20:15 2006 +0000 @@ -2066,6 +2066,7 @@ extern AVCodec oggvorbis_encoder; extern AVCodec oggtheora_encoder; extern AVCodec faac_encoder; +extern AVCodec flac_encoder; extern AVCodec xvid_encoder; extern AVCodec mpeg1video_encoder; extern AVCodec mpeg2video_encoder; diff -r 608659b58523 -r 5b901881d6ed flacenc.c --- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/flacenc.c Sat Jun 24 10:20:15 2006 +0000 @@ -0,0 +1,570 @@ +/** + * FLAC audio encoder + * Copyright (c) 2006 Justin Ruggles + * + * 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 +}; + +static const int flac_blocksizes_ordered[14] = { + 0, 192, 256, 512, 576, 1024, 1152, 2048, 2304, 4096, 4608, 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 = 0; + target = (samplerate * BLOCK_TIME_MS) / 1000; + for(i=13; i>=0; i--) { + if(target >= flac_blocksizes_ordered[i]) { + blocksize = flac_blocksizes_ordered[i]; + break; + } + } + if(blocksize == 0) { + blocksize = flac_blocksizes_ordered[1]; + } + 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(s == NULL) { + return -1; + } + + 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; + + 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 int 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; chchannels; ch++) { + frame->subframes[ch].obits = 16; + } + if(s->channels == 2) { + frame->ch_mode = FLAC_CHMODE_LEFT_RIGHT; + } else { + frame->ch_mode = FLAC_CHMODE_NOT_STEREO; + } + + return 0; +} + +/** + * 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; iblocksize; i++) { + for(ch=0; chchannels; ch++,j++) { + frame->subframes[ch].samples[i] = samples[j]; + } + } +} + +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; + int32_t pred; + + for(i=0; iframe; + 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) +{ + uint32_t uval; + + assert(bits >= 0 && bits <= 31); + uval = (val < 0) ? (1UL << bits) + val : val; + put_bits(pb, bits, uval); +} + +static void +write_utf8(PutBitContext *pb, uint32_t val) +{ + int i, bytes, mask, shift; + + bytes = 1; + if(val >= 0x80) bytes++; + if(val >= 0x800) bytes++; + if(val >= 0x10000) bytes++; + if(val >= 0x200000) bytes++; + if(val >= 0x4000000) bytes++; + + if(bytes == 1) { + put_bits(pb, 8, val); + return; + } + + shift = (bytes - 1) * 6; + mask = 0x80 + ((1 << 7) - (1 << (8 - bytes))); + put_bits(pb, 8, mask | (val >> shift)); + for(i=0; i> 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[1] > 0) { + if(frame->bs_code[1] < 256) { + put_bits(&s->pb, 8, frame->bs_code[1]); + } else { + put_bits(&s->pb, 16, frame->bs_code[1]); + } + } + if(s->sr_code[1] > 0) { + if(s->sr_code[1] < 256) { + put_bits(&s->pb, 8, s->sr_code[1]); + } else { + 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; iblocksize; 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; iblocksize; 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; iorder; 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; chchannels; 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; + if(init_frame(s)) { + return 0; + } + + copy_samples(s, samples); + + for(ch=0; chchannels; 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; chchannels; 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->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, +}; diff -r 608659b58523 -r 5b901881d6ed golomb.h --- a/golomb.h Sat Jun 24 09:25:21 2006 +0000 +++ b/golomb.h Sat Jun 24 10:20:15 2006 +0000 @@ -435,6 +435,10 @@ e= (i>>k) + 1; if(e 31) { + put_bits(pb, 31, 0); + e -= 31; + } put_bits(pb, e, 1); if(k) put_bits(pb, k, i&((1<