--- /dev/null
+/**
+ * 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
+};
+
+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; ch<s->channels; 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; i<frame->blocksize; i++) {
+ for(ch=0; ch<s->channels; 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; i<order; i++) {
+ res[i] = smp[i];
+ }
+ for(i=order; i<n; i++) {
+ pred = 0;
+ switch(order) {
+ case 0: pred = 0;
+ break;
+ case 1: pred = smp[i-1];
+ break;
+ case 2: pred = 2*smp[i-1] - smp[i-2];
+ break;
+ case 3: pred = 3*smp[i-1] - 3*smp[i-2] + smp[i-3];
+ break;
+ case 4: pred = 4*smp[i-1] - 6*smp[i-2] + 4*smp[i-3] - smp[i-4];
+ break;
+ }
+ res[i] = smp[i] - pred;
+ }
+}
+
+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)
+{
+ 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<bytes-1; i++) {
+ 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[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; 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;
+ if(init_frame(s)) {
+ return 0;
+ }
+
+ copy_samples(s, samples);
+
+ 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->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,
+};
projects / platform / upstream / libav.git / commitdiff