PeerStreamer

/**

 * ALAC audio encoder

 * Copyright (c) 2008  Jaikrishnan Menon <realityman@gmx.net>

 *

 * This file is part of FFmpeg.

 *

 * FFmpeg 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 2.1 of the License, or (at your option) any later version.

 *

 * FFmpeg 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 FFmpeg; 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 "put_bits.h"

#include "dsputil.h"

#include "lpc.h"

#include "mathops.h"

#define DEFAULT_FRAME_SIZE        4096

#define DEFAULT_SAMPLE_SIZE       16

#define MAX_CHANNELS              8

#define ALAC_EXTRADATA_SIZE       36

#define ALAC_FRAME_HEADER_SIZE    55

#define ALAC_FRAME_FOOTER_SIZE    3

#define ALAC_ESCAPE_CODE          0x1FF

#define ALAC_MAX_LPC_ORDER        30

#define DEFAULT_MAX_PRED_ORDER    6

#define DEFAULT_MIN_PRED_ORDER    4

#define ALAC_MAX_LPC_PRECISION    9

#define ALAC_MAX_LPC_SHIFT        9

#define ALAC_CHMODE_LEFT_RIGHT    0

#define ALAC_CHMODE_LEFT_SIDE     1

#define ALAC_CHMODE_RIGHT_SIDE    2

#define ALAC_CHMODE_MID_SIDE      3

typedef struct RiceContext {

    int history_mult;

    int initial_history;

    int k_modifier;

    int rice_modifier;

} RiceContext;

typedef struct LPCContext {

    int lpc_order;

    int lpc_coeff[ALAC_MAX_LPC_ORDER+1];

    int lpc_quant;

} LPCContext;

typedef struct AlacEncodeContext {

    int compression_level;

    int min_prediction_order;

    int max_prediction_order;

    int max_coded_frame_size;

    int write_sample_size;

    int32_t sample_buf[MAX_CHANNELS][DEFAULT_FRAME_SIZE];

    int32_t predictor_buf[DEFAULT_FRAME_SIZE];

    int interlacing_shift;

    int interlacing_leftweight;

    PutBitContext pbctx;

    RiceContext rc;

    LPCContext lpc[MAX_CHANNELS];

    DSPContext dspctx;

    AVCodecContext *avctx;

} AlacEncodeContext;

static void init_sample_buffers(AlacEncodeContext *s, int16_t *input_samples)

{

    int ch, i;

    for(ch=0;ch<s->avctx->channels;ch++) {

        int16_t *sptr = input_samples + ch;

        for(i=0;i<s->avctx->frame_size;i++) {

            s->sample_buf[ch][i] = *sptr;

            sptr += s->avctx->channels;

        }

    }

}

static void encode_scalar(AlacEncodeContext *s, int x, int k, int write_sample_size)

{

    int divisor, q, r;

    k = FFMIN(k, s->rc.k_modifier);

    divisor = (1<<k) - 1;

    q = x / divisor;

    r = x % divisor;

    if(q > 8) {

        // write escape code and sample value directly

        put_bits(&s->pbctx, 9, ALAC_ESCAPE_CODE);

        put_bits(&s->pbctx, write_sample_size, x);

    } else {

        if(q)

            put_bits(&s->pbctx, q, (1<<q) - 1);

        put_bits(&s->pbctx, 1, 0);

        if(k != 1) {

            if(r > 0)

                put_bits(&s->pbctx, k, r+1);

            else

                put_bits(&s->pbctx, k-1, 0);

        }

    }

}

static void write_frame_header(AlacEncodeContext *s, int is_verbatim)

{

    put_bits(&s->pbctx, 3,  s->avctx->channels-1);          // No. of channels -1

    put_bits(&s->pbctx, 16, 0);                             // Seems to be zero

    put_bits(&s->pbctx, 1,  1);                             // Sample count is in the header

    put_bits(&s->pbctx, 2,  0);                             // FIXME: Wasted bytes field

    put_bits(&s->pbctx, 1,  is_verbatim);                   // Audio block is verbatim

    put_bits(&s->pbctx, 32, s->avctx->frame_size);          // No. of samples in the frame

}

static void calc_predictor_params(AlacEncodeContext *s, int ch)

{

    int32_t coefs[MAX_LPC_ORDER][MAX_LPC_ORDER];

    int shift[MAX_LPC_ORDER];

    int opt_order;

    opt_order = ff_lpc_calc_coefs(&s->dspctx, s->sample_buf[ch], s->avctx->frame_size, s->min_prediction_order, s->max_prediction_order,

                                   ALAC_MAX_LPC_PRECISION, coefs, shift, 1, ORDER_METHOD_EST, ALAC_MAX_LPC_SHIFT, 1);

    s->lpc[ch].lpc_order = opt_order;

    s->lpc[ch].lpc_quant = shift[opt_order-1];

    memcpy(s->lpc[ch].lpc_coeff, coefs[opt_order-1], opt_order*sizeof(int));

}

static int estimate_stereo_mode(int32_t *left_ch, int32_t *right_ch, int n)

{

    int i, best;

    int32_t lt, rt;

    uint64_t sum[4];

    uint64_t score[4];

    /* calculate sum of 2nd order residual for each channel */

    sum[0] = sum[1] = sum[2] = sum[3] = 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];

        sum[2] += FFABS((lt + rt) >> 1);

        sum[3] += FFABS(lt - rt);

        sum[0] += FFABS(lt);

        sum[1] += FFABS(rt);

    }

    /* calculate score for each mode */

    score[0] = sum[0] + sum[1];

    score[1] = sum[0] + sum[3];

    score[2] = sum[1] + sum[3];

    score[3] = sum[2] + sum[3];

    /* return mode with lowest score */

    best = 0;

    for(i=1; i<4; i++) {

        if(score[i] < score[best]) {

            best = i;

        }

    }

    return best;

}

static void alac_stereo_decorrelation(AlacEncodeContext *s)

{

    int32_t *left = s->sample_buf[0], *right = s->sample_buf[1];

    int i, mode, n = s->avctx->frame_size;

    int32_t tmp;

    mode = estimate_stereo_mode(left, right, n);

    switch(mode)

    {

        case ALAC_CHMODE_LEFT_RIGHT:

            s->interlacing_leftweight = 0;

            s->interlacing_shift = 0;

            break;

        case ALAC_CHMODE_LEFT_SIDE:

            for(i=0; i<n; i++) {

                right[i] = left[i] - right[i];

            }

            s->interlacing_leftweight = 1;

            s->interlacing_shift = 0;

            break;

        case ALAC_CHMODE_RIGHT_SIDE:

            for(i=0; i<n; i++) {

                tmp = right[i];

                right[i] = left[i] - right[i];

                left[i] = tmp + (right[i] >> 31);

            }

            s->interlacing_leftweight = 1;

            s->interlacing_shift = 31;

            break;

        default:

            for(i=0; i<n; i++) {

                tmp = left[i];

                left[i] = (tmp + right[i]) >> 1;

                right[i] = tmp - right[i];

            }

            s->interlacing_leftweight = 1;

            s->interlacing_shift = 1;

            break;

    }

}

static void alac_linear_predictor(AlacEncodeContext *s, int ch)

{

    int i;

    LPCContext lpc = s->lpc[ch];

    if(lpc.lpc_order == 31) {

        s->predictor_buf[0] = s->sample_buf[ch][0];

        for(i=1; i<s->avctx->frame_size; i++)

            s->predictor_buf[i] = s->sample_buf[ch][i] - s->sample_buf[ch][i-1];

        return;

    }

    // generalised linear predictor

    if(lpc.lpc_order > 0) {

        int32_t *samples  = s->sample_buf[ch];

        int32_t *residual = s->predictor_buf;

        // generate warm-up samples

        residual[0] = samples[0];

        for(i=1;i<=lpc.lpc_order;i++)

            residual[i] = samples[i] - samples[i-1];

        // perform lpc on remaining samples

        for(i = lpc.lpc_order + 1; i < s->avctx->frame_size; i++) {

            int sum = 1 << (lpc.lpc_quant - 1), res_val, j;

            for (j = 0; j < lpc.lpc_order; j++) {

                sum += (samples[lpc.lpc_order-j] - samples[0]) *

                        lpc.lpc_coeff[j];

            }

            sum >>= lpc.lpc_quant;

            sum += samples[0];

            residual[i] = sign_extend(samples[lpc.lpc_order+1] - sum,

                                      s->write_sample_size);

            res_val = residual[i];

            if(res_val) {

                int index = lpc.lpc_order - 1;

                int neg = (res_val < 0);

                while(index >= 0 && (neg ? (res_val < 0):(res_val > 0))) {

                    int val = samples[0] - samples[lpc.lpc_order - index];

                    int sign = (val ? FFSIGN(val) : 0);

                    if(neg)

                        sign*=-1;

                    lpc.lpc_coeff[index] -= sign;

                    val *= sign;

                    res_val -= ((val >> lpc.lpc_quant) *

                            (lpc.lpc_order - index));

                    index--;

                }

            }

            samples++;

        }

    }

}

static void alac_entropy_coder(AlacEncodeContext *s)

{

    unsigned int history = s->rc.initial_history;

    int sign_modifier = 0, i, k;

    int32_t *samples = s->predictor_buf;

    for(i=0;i < s->avctx->frame_size;) {

        int x;

        k = av_log2((history >> 9) + 3);

        x = -2*(*samples)-1;

        x ^= (x>>31);

        samples++;

        i++;

        encode_scalar(s, x - sign_modifier, k, s->write_sample_size);

        history += x * s->rc.history_mult

                   - ((history * s->rc.history_mult) >> 9);

        sign_modifier = 0;

        if(x > 0xFFFF)

            history = 0xFFFF;

        if((history < 128) && (i < s->avctx->frame_size)) {

            unsigned int block_size = 0;

            k = 7 - av_log2(history) + ((history + 16) >> 6);

            while((*samples == 0) && (i < s->avctx->frame_size)) {

                samples++;

                i++;

                block_size++;

            }

            encode_scalar(s, block_size, k, 16);

            sign_modifier = (block_size <= 0xFFFF);

            history = 0;

        }

    }

}

static void write_compressed_frame(AlacEncodeContext *s)

{

    int i, j;

    /* only simple mid/side decorrelation supported as of now */

    if(s->avctx->channels == 2)

        alac_stereo_decorrelation(s);

    put_bits(&s->pbctx, 8, s->interlacing_shift);

    put_bits(&s->pbctx, 8, s->interlacing_leftweight);

    for(i=0;i<s->avctx->channels;i++) {

        calc_predictor_params(s, i);

        put_bits(&s->pbctx, 4, 0);  // prediction type : currently only type 0 has been RE'd

        put_bits(&s->pbctx, 4, s->lpc[i].lpc_quant);

        put_bits(&s->pbctx, 3, s->rc.rice_modifier);

        put_bits(&s->pbctx, 5, s->lpc[i].lpc_order);

        // predictor coeff. table

        for(j=0;j<s->lpc[i].lpc_order;j++) {

            put_sbits(&s->pbctx, 16, s->lpc[i].lpc_coeff[j]);

        }

    }

    // apply lpc and entropy coding to audio samples

    for(i=0;i<s->avctx->channels;i++) {

        alac_linear_predictor(s, i);

        alac_entropy_coder(s);

    }

}

static av_cold int alac_encode_init(AVCodecContext *avctx)

{

    AlacEncodeContext *s    = avctx->priv_data;

    uint8_t *alac_extradata = av_mallocz(ALAC_EXTRADATA_SIZE+1);

    avctx->frame_size      = DEFAULT_FRAME_SIZE;

    avctx->bits_per_coded_sample = DEFAULT_SAMPLE_SIZE;

    if(avctx->sample_fmt != SAMPLE_FMT_S16) {

        av_log(avctx, AV_LOG_ERROR, "only pcm_s16 input samples are supported\n");

        return -1;

    }

    // Set default compression level

    if(avctx->compression_level == FF_COMPRESSION_DEFAULT)

        s->compression_level = 1;

    else

        s->compression_level = av_clip(avctx->compression_level, 0, 1);

    // Initialize default Rice parameters

    s->rc.history_mult    = 40;

    s->rc.initial_history = 10;

    s->rc.k_modifier      = 14;

    s->rc.rice_modifier   = 4;

    s->max_coded_frame_size = (ALAC_FRAME_HEADER_SIZE + ALAC_FRAME_FOOTER_SIZE +

                               avctx->frame_size*avctx->channels*avctx->bits_per_coded_sample)>>3;

    s->write_sample_size  = avctx->bits_per_coded_sample + avctx->channels - 1; // FIXME: consider wasted_bytes

    AV_WB32(alac_extradata,    ALAC_EXTRADATA_SIZE);

    AV_WB32(alac_extradata+4,  MKBETAG('a','l','a','c'));

    AV_WB32(alac_extradata+12, avctx->frame_size);

    AV_WB8 (alac_extradata+17, avctx->bits_per_coded_sample);

    AV_WB8 (alac_extradata+21, avctx->channels);

    AV_WB32(alac_extradata+24, s->max_coded_frame_size);

    AV_WB32(alac_extradata+28, avctx->sample_rate*avctx->channels*avctx->bits_per_coded_sample); // average bitrate

    AV_WB32(alac_extradata+32, avctx->sample_rate);

    // Set relevant extradata fields

    if(s->compression_level > 0) {

        AV_WB8(alac_extradata+18, s->rc.history_mult);

        AV_WB8(alac_extradata+19, s->rc.initial_history);

        AV_WB8(alac_extradata+20, s->rc.k_modifier);

    }

    s->min_prediction_order = DEFAULT_MIN_PRED_ORDER;

    if(avctx->min_prediction_order >= 0) {

        if(avctx->min_prediction_order < MIN_LPC_ORDER ||

           avctx->min_prediction_order > ALAC_MAX_LPC_ORDER) {

            av_log(avctx, AV_LOG_ERROR, "invalid min prediction order: %d\n", avctx->min_prediction_order);

                return -1;

        }

        s->min_prediction_order = avctx->min_prediction_order;

    }

    s->max_prediction_order = DEFAULT_MAX_PRED_ORDER;

    if(avctx->max_prediction_order >= 0) {

        if(avctx->max_prediction_order < MIN_LPC_ORDER ||

           avctx->max_prediction_order > ALAC_MAX_LPC_ORDER) {

            av_log(avctx, AV_LOG_ERROR, "invalid max prediction order: %d\n", avctx->max_prediction_order);

                return -1;

        }

        s->max_prediction_order = avctx->max_prediction_order;

    }

    if(s->max_prediction_order < s->min_prediction_order) {

        av_log(avctx, AV_LOG_ERROR, "invalid prediction orders: min=%d max=%d\n",

               s->min_prediction_order, s->max_prediction_order);

        return -1;

    }

    avctx->extradata = alac_extradata;

    avctx->extradata_size = ALAC_EXTRADATA_SIZE;

    avctx->coded_frame = avcodec_alloc_frame();

    avctx->coded_frame->key_frame = 1;

    s->avctx = avctx;

    dsputil_init(&s->dspctx, avctx);

    return 0;

}

static int alac_encode_frame(AVCodecContext *avctx, uint8_t *frame,

                             int buf_size, void *data)

{

    AlacEncodeContext *s = avctx->priv_data;

    PutBitContext *pb = &s->pbctx;

    int i, out_bytes, verbatim_flag = 0;

    if(avctx->frame_size > DEFAULT_FRAME_SIZE) {

        av_log(avctx, AV_LOG_ERROR, "input frame size exceeded\n");

        return -1;

    }

    if(buf_size < 2*s->max_coded_frame_size) {

        av_log(avctx, AV_LOG_ERROR, "buffer size is too small\n");

        return -1;

    }

verbatim:

    init_put_bits(pb, frame, buf_size);

    if((s->compression_level == 0) || verbatim_flag) {

        // Verbatim mode

        int16_t *samples = data;

        write_frame_header(s, 1);

        for(i=0; i<avctx->frame_size*avctx->channels; i++) {

            put_sbits(pb, 16, *samples++);

        }

    } else {

        init_sample_buffers(s, data);

        write_frame_header(s, 0);

        write_compressed_frame(s);

    }

    put_bits(pb, 3, 7);

    flush_put_bits(pb);

    out_bytes = put_bits_count(pb) >> 3;

    if(out_bytes > s->max_coded_frame_size) {

        /* frame too large. use verbatim mode */

        if(verbatim_flag || (s->compression_level == 0)) {

            /* still too large. must be an error. */

            av_log(avctx, AV_LOG_ERROR, "error encoding frame\n");

            return -1;

        }

        verbatim_flag = 1;

        goto verbatim;

    }

    return out_bytes;

}

static av_cold int alac_encode_close(AVCodecContext *avctx)

{

    av_freep(&avctx->extradata);

    avctx->extradata_size = 0;

    av_freep(&avctx->coded_frame);

    return 0;

}

AVCodec alac_encoder = {

    "alac",

    CODEC_TYPE_AUDIO,

    CODEC_ID_ALAC,

    sizeof(AlacEncodeContext),

    alac_encode_init,

    alac_encode_frame,

    alac_encode_close,

    .capabilities = CODEC_CAP_SMALL_LAST_FRAME,

    .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),

};