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/*

 * 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"

#include "internal.h"

#include "mathops.h"

#define ROQ_FRAME_SIZE           735

#define ROQ_HEADER_SIZE   8

#define MAX_DPCM (127*127)

typedef struct

{

    short lastSample[2];

    int input_frames;

    int buffered_samples;

    int16_t *frame_buffer;

    int64_t first_pts;

} ROQDPCMContext;

static av_cold int roq_dpcm_encode_close(AVCodecContext *avctx)

{

    ROQDPCMContext *context = avctx->priv_data;

    av_freep(&context->frame_buffer);

    return 0;

}

static av_cold int roq_dpcm_encode_init(AVCodecContext *avctx)

{

    ROQDPCMContext *context = avctx->priv_data;

    int ret;

    if (avctx->channels > 2) {

        av_log(avctx, AV_LOG_ERROR, "Audio must be mono or stereo\n");

        return AVERROR(EINVAL);

    }

    if (avctx->sample_rate != 22050) {

        av_log(avctx, AV_LOG_ERROR, "Audio must be 22050 Hz\n");

        return AVERROR(EINVAL);

    }

    avctx->frame_size = ROQ_FRAME_SIZE;

    avctx->bit_rate   = (ROQ_HEADER_SIZE + ROQ_FRAME_SIZE * avctx->channels) *

                        (22050 / ROQ_FRAME_SIZE) * 8;

    context->frame_buffer = av_malloc(8 * ROQ_FRAME_SIZE * avctx->channels *

                                      sizeof(*context->frame_buffer));

    if (!context->frame_buffer) {

        ret = AVERROR(ENOMEM);

        goto error;

    }

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

    return 0;

error:

    roq_dpcm_encode_close(avctx);

    return ret;

}

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, AVPacket *avpkt,

                                 const AVFrame *frame, int *got_packet_ptr)

{

    int i, stereo, data_size, ret;

    const int16_t *in = frame ? (const int16_t *)frame->data[0] : NULL;

    uint8_t *out;

    ROQDPCMContext *context = avctx->priv_data;

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

    if (!in && context->input_frames >= 8)

        return 0;

    if (in && context->input_frames < 8) {

        memcpy(&context->frame_buffer[context->buffered_samples * avctx->channels],

               in, avctx->frame_size * avctx->channels * sizeof(*in));

        context->buffered_samples += avctx->frame_size;

        if (context->input_frames == 0)

            context->first_pts = frame->pts;

        if (context->input_frames < 7) {

            context->input_frames++;

            return 0;

        }

    }

    if (context->input_frames < 8) {

        in = context->frame_buffer;

    }

    if (stereo) {

        context->lastSample[0] &= 0xFF00;

        context->lastSample[1] &= 0xFF00;

    }

    if (context->input_frames == 7)

        data_size = avctx->channels * context->buffered_samples;

    else

        data_size = avctx->channels * avctx->frame_size;

    if ((ret = ff_alloc_packet2(avctx, avpkt, ROQ_HEADER_SIZE + data_size)) < 0)

        return ret;

    out = avpkt->data;

    bytestream_put_byte(&out, stereo ? 0x21 : 0x20);

    bytestream_put_byte(&out, 0x10);

    bytestream_put_le32(&out, data_size);

    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 */

    for (i = 0; i < data_size; i++)

        *out++ = dpcm_predict(&context->lastSample[i & 1], *in++);

    avpkt->pts      = context->input_frames <= 7 ? context->first_pts : frame->pts;

    avpkt->duration = data_size / avctx->channels;

    context->input_frames++;

    if (!in)

        context->input_frames = FFMAX(context->input_frames, 8);

    *got_packet_ptr = 1;

    return 0;

}

AVCodec ff_roq_dpcm_encoder = {

    .name           = "roq_dpcm",

    .long_name      = NULL_IF_CONFIG_SMALL("id RoQ DPCM"),

    .type           = AVMEDIA_TYPE_AUDIO,

    .id             = AV_CODEC_ID_ROQ_DPCM,

    .priv_data_size = sizeof(ROQDPCMContext),

    .init           = roq_dpcm_encode_init,

    .encode2        = roq_dpcm_encode_frame,

    .close          = roq_dpcm_encode_close,

    .capabilities   = CODEC_CAP_DELAY,

    .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,

                                                     AV_SAMPLE_FMT_NONE },

};