view roqaudioenc.c @ 10874:bcfe2acbf190 libavcodec

AAC: Compress codebook tables and optimise sign bit handling The codebooks each consist of small number of values repeated in groups of 2 or 4. Storing the codebooks as a packed list of 2- or 4-bit indexes into a table reduces their size substantially (from 7.5k to 1.5k), resulting in less cache pressure. For the band types with sign bits in the bitstream, storing the number and position of non-zero codebook values using a few bits avoids multiple get_bits() calls and floating-point comparisons which gcc handles miserably. Some float/int type punning also avoids gcc brain damage. Overall speedup 20-35% on Cortex-A8, 20% on Core i7.

/*
 * RoQ audio encoder
 *
 * Copyright (c) 2005 Eric Lasota
 *    Based on RoQ specs (c)2001 Tim Ferguson
 *
 * 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 "bytestream.h"

#define ROQ_FIRST_FRAME_SIZE     (735*8)
#define ROQ_FRAME_SIZE           735


#define MAX_DPCM (127*127)


typedef struct
{
    short lastSample[2];
} ROQDPCMContext;

static av_cold int roq_dpcm_encode_init(AVCodecContext *avctx)
{
    ROQDPCMContext *context = avctx->priv_data;

    if (avctx->channels > 2) {
        av_log(avctx, AV_LOG_ERROR, "Audio must be mono or stereo\n");
        return -1;
    }
    if (avctx->sample_rate != 22050) {
        av_log(avctx, AV_LOG_ERROR, "Audio must be 22050 Hz\n");
        return -1;
    }
    if (avctx->sample_fmt != SAMPLE_FMT_S16) {
        av_log(avctx, AV_LOG_ERROR, "Audio must be signed 16-bit\n");
        return -1;
    }

    avctx->frame_size = ROQ_FIRST_FRAME_SIZE;

    context->lastSample[0] = context->lastSample[1] = 0;

    avctx->coded_frame= avcodec_alloc_frame();
    avctx->coded_frame->key_frame= 1;

    return 0;
}

static unsigned char dpcm_predict(short *previous, short current)
{
    int diff;
    int negative;
    int result;
    int predicted;

    diff = current - *previous;

    negative = diff<0;
    diff = FFABS(diff);

    if (diff >= MAX_DPCM)
        result = 127;
    else {
        result = ff_sqrt(diff);
        result += diff > result*result+result;
    }

    /* See if this overflows */
 retry:
    diff = result*result;
    if (negative)
        diff = -diff;
    predicted = *previous + diff;

    /* If it overflows, back off a step */
    if (predicted > 32767 || predicted < -32768) {
        result--;
        goto retry;
    }

    /* Add the sign bit */
    result |= negative << 7;   //if (negative) result |= 128;

    *previous = predicted;

    return result;
}

static int roq_dpcm_encode_frame(AVCodecContext *avctx,
                unsigned char *frame, int buf_size, void *data)
{
    int i, samples, stereo, ch;
    short *in;
    unsigned char *out;

    ROQDPCMContext *context = avctx->priv_data;

    stereo = (avctx->channels == 2);

    if (stereo) {
        context->lastSample[0] &= 0xFF00;
        context->lastSample[1] &= 0xFF00;
    }

    out = frame;
    in = data;

    bytestream_put_byte(&out, stereo ? 0x21 : 0x20);
    bytestream_put_byte(&out, 0x10);
    bytestream_put_le32(&out, avctx->frame_size*avctx->channels);

    if (stereo) {
        bytestream_put_byte(&out, (context->lastSample[1])>>8);
        bytestream_put_byte(&out, (context->lastSample[0])>>8);
    } else
        bytestream_put_le16(&out, context->lastSample[0]);

    /* Write the actual samples */
    samples = avctx->frame_size;
    for (i=0; i<samples; i++)
        for (ch=0; ch<avctx->channels; ch++)
            *out++ = dpcm_predict(&context->lastSample[ch], *in++);

    /* Use smaller frames from now on */
    avctx->frame_size = ROQ_FRAME_SIZE;

    /* Return the result size */
    return out - frame;
}

static av_cold int roq_dpcm_encode_close(AVCodecContext *avctx)
{
    av_freep(&avctx->coded_frame);

    return 0;
}

AVCodec roq_dpcm_encoder = {
    "roq_dpcm",
    CODEC_TYPE_AUDIO,
    CODEC_ID_ROQ_DPCM,
    sizeof(ROQDPCMContext),
    roq_dpcm_encode_init,
    roq_dpcm_encode_frame,
    roq_dpcm_encode_close,
    NULL,
    .sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
    .long_name = NULL_IF_CONFIG_SMALL("id RoQ DPCM"),
};