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ffmpeg / libavcodec / roqaudioenc.c @ 2912e87a

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/*
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 * RoQ audio encoder
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 *
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 * Copyright (c) 2005 Eric Lasota
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 *    Based on RoQ specs (c)2001 Tim Ferguson
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 *
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 * This file is part of Libav.
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 *
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 * Libav is free software; you can redistribute it and/or
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 * modify it under the terms of the GNU Lesser General Public
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 * License as published by the Free Software Foundation; either
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 * version 2.1 of the License, or (at your option) any later version.
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 *
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 * Libav is distributed in the hope that it will be useful,
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 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
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 * Lesser General Public License for more details.
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 *
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 * You should have received a copy of the GNU Lesser General Public
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 * License along with Libav; if not, write to the Free Software
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 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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 */
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#include "libavutil/intmath.h"
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#include "avcodec.h"
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#include "bytestream.h"
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#define ROQ_FIRST_FRAME_SIZE     (735*8)
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#define ROQ_FRAME_SIZE           735
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#define MAX_DPCM (127*127)
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typedef struct
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{
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    short lastSample[2];
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} ROQDPCMContext;
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static av_cold int roq_dpcm_encode_init(AVCodecContext *avctx)
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{
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    ROQDPCMContext *context = avctx->priv_data;
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    if (avctx->channels > 2) {
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        av_log(avctx, AV_LOG_ERROR, "Audio must be mono or stereo\n");
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        return -1;
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    }
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    if (avctx->sample_rate != 22050) {
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        av_log(avctx, AV_LOG_ERROR, "Audio must be 22050 Hz\n");
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        return -1;
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    }
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    if (avctx->sample_fmt != AV_SAMPLE_FMT_S16) {
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        av_log(avctx, AV_LOG_ERROR, "Audio must be signed 16-bit\n");
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        return -1;
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    }
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    avctx->frame_size = ROQ_FIRST_FRAME_SIZE;
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    context->lastSample[0] = context->lastSample[1] = 0;
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    avctx->coded_frame= avcodec_alloc_frame();
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    avctx->coded_frame->key_frame= 1;
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    return 0;
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}
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static unsigned char dpcm_predict(short *previous, short current)
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{
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    int diff;
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    int negative;
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    int result;
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    int predicted;
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    diff = current - *previous;
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    negative = diff<0;
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    diff = FFABS(diff);
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    if (diff >= MAX_DPCM)
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        result = 127;
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    else {
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        result = ff_sqrt(diff);
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        result += diff > result*result+result;
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    }
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    /* See if this overflows */
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 retry:
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    diff = result*result;
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    if (negative)
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        diff = -diff;
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    predicted = *previous + diff;
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    /* If it overflows, back off a step */
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    if (predicted > 32767 || predicted < -32768) {
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        result--;
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        goto retry;
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    }
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    /* Add the sign bit */
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    result |= negative << 7;   //if (negative) result |= 128;
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    *previous = predicted;
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    return result;
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}
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static int roq_dpcm_encode_frame(AVCodecContext *avctx,
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                unsigned char *frame, int buf_size, void *data)
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{
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    int i, samples, stereo, ch;
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    const short *in;
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    unsigned char *out;
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    ROQDPCMContext *context = avctx->priv_data;
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    stereo = (avctx->channels == 2);
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    if (stereo) {
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        context->lastSample[0] &= 0xFF00;
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        context->lastSample[1] &= 0xFF00;
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    }
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    out = frame;
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    in = data;
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    bytestream_put_byte(&out, stereo ? 0x21 : 0x20);
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    bytestream_put_byte(&out, 0x10);
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    bytestream_put_le32(&out, avctx->frame_size*avctx->channels);
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    if (stereo) {
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        bytestream_put_byte(&out, (context->lastSample[1])>>8);
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        bytestream_put_byte(&out, (context->lastSample[0])>>8);
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    } else
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        bytestream_put_le16(&out, context->lastSample[0]);
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    /* Write the actual samples */
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    samples = avctx->frame_size;
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    for (i=0; i<samples; i++)
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        for (ch=0; ch<avctx->channels; ch++)
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            *out++ = dpcm_predict(&context->lastSample[ch], *in++);
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    /* Use smaller frames from now on */
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    avctx->frame_size = ROQ_FRAME_SIZE;
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    /* Return the result size */
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    return out - frame;
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}
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static av_cold int roq_dpcm_encode_close(AVCodecContext *avctx)
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{
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    av_freep(&avctx->coded_frame);
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    return 0;
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}
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AVCodec ff_roq_dpcm_encoder = {
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    "roq_dpcm",
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    AVMEDIA_TYPE_AUDIO,
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    CODEC_ID_ROQ_DPCM,
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    sizeof(ROQDPCMContext),
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    roq_dpcm_encode_init,
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    roq_dpcm_encode_frame,
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    roq_dpcm_encode_close,
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    NULL,
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    .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
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    .long_name = NULL_IF_CONFIG_SMALL("id RoQ DPCM"),
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};