first rudimentary version of (Justin Ruggles jruggle earthlink net) flac encoder

Originally committed as revision 5514 to svn://svn.ffmpeg.org/ffmpeg/trunk

1 files changed, 570 insertions, 0 deletions

diff --git a/libavcodec/flacenc.c b/libavcodec/flacenc.c
new file mode 100644
index 0000000000..76855ffabe
--- /dev/null
+++ b/libavcodec/flacenc.c
@@ -0,0 +1,570 @@
+/**
+ * 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,
+};