@@ -68,6 +68,9 @@ endif

 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

@@ -72,6 +72,9 @@ void avcodec_register_all(void)

     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);

@@ -2066,6 +2066,7 @@ extern AVCodec mp3lame_encoder;

 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;

new file mode 100644

@@ -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,

+};

@@ -435,6 +435,10 @@ static inline void set_ur_golomb_jpegls(PutBitContext *pb, int i, int k, int lim

     e= (i>>k) + 1;

     if(e<limit){

+        while(e > 31) {

+            put_bits(pb, 31, 0);

+            e -= 31;

+        }

         put_bits(pb, e, 1);

         if(k)

             put_bits(pb, k, i&((1<<k)-1));