annotate alacenc.c @ 11104:bb877c9cb102 libavcodec

Detect spatial direct MBs partitioned smaller than 16x16 that can be partitioned as 16x16 (except ones changing interlacing relative to the colocated MB). 20 cycles slower during MV generation 175 cycles faster during MC

/**
 * ALAC audio encoder
 * Copyright (c) 2008  Jaikrishnan Menon <[e-mail]>
 *
 * 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 "get_bits.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_bits32(&s->pbctx, 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;

    if (s->compression_level == 1) {
        s->lpc[ch].lpc_order = 6;
        s->lpc[ch].lpc_quant = 6;
        s->lpc[ch].lpc_coeff[0] =  160;
        s->lpc[ch].lpc_coeff[1] = -190;
        s->lpc[ch].lpc_coeff[2] =  170;
        s->lpc[ch].lpc_coeff[3] = -130;
        s->lpc[ch].lpc_coeff[4] =   80;
        s->lpc[ch].lpc_coeff[5] =  -25;
    } else {
        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;

    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 = 2;
    else
        s->compression_level = av_clip(avctx->compression_level, 0, 2);

    // 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 = 8 + (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)"),
};