libavcodec/adxenc.c
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 /*
  * ADX ADPCM codecs
  * Copyright (c) 2001,2003 BERO
  *
  * 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
  */
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 #include "avcodec.h"
 #include "adx.h"
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 #include "bytestream.h"
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 #include "internal.h"
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 #include "put_bits.h"
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 /**
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  * @file
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  * SEGA CRI adx codecs.
  *
  * Reference documents:
  * http://ku-www.ss.titech.ac.jp/~yatsushi/adx.html
  * adx2wav & wav2adx http://www.geocities.co.jp/Playtown/2004/
  */
 
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 static void adx_encode(ADXContext *c, uint8_t *adx, const int16_t *wav,
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                        ADXChannelState *prev, int channels)
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 {
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     PutBitContext pb;
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     int scale;
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     int i, j;
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     int s0, s1, s2, d;
     int max = 0;
     int min = 0;
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     s1 = prev->s1;
     s2 = prev->s2;
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     for (i = 0, j = 0; j < 32; i += channels, j++) {
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         s0 = wav[i];
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         d = ((s0 << COEFF_BITS) - c->coeff[0] * s1 - c->coeff[1] * s2) >> COEFF_BITS;
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         if (max < d)
             max = d;
         if (min > d)
             min = d;
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         s2 = s1;
         s1 = s0;
     }
 
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     if (max == 0 && min == 0) {
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         prev->s1 = s1;
         prev->s2 = s2;
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         memset(adx, 0, BLOCK_SIZE);
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         return;
     }
 
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     if (max / 7 > -min / 8)
         scale = max / 7;
     else
         scale = -min / 8;
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     if (scale == 0)
         scale = 1;
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     AV_WB16(adx, scale);
 
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     init_put_bits(&pb, adx + 2, 16);
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     s1 = prev->s1;
     s2 = prev->s2;
     for (i = 0, j = 0; j < 32; i += channels, j++) {
         d = ((wav[i] << COEFF_BITS) - c->coeff[0] * s1 - c->coeff[1] * s2) >> COEFF_BITS;
 
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         d = av_clip_intp2(ROUNDED_DIV(d, scale), 3);
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         put_sbits(&pb, 4, d);
 
         s0 = ((d << COEFF_BITS) * scale + c->coeff[0] * s1 + c->coeff[1] * s2) >> COEFF_BITS;
         s2 = s1;
         s1 = s0;
     }
     prev->s1 = s1;
     prev->s2 = s2;
 
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     flush_put_bits(&pb);
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 }
 
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 #define HEADER_SIZE 36
 
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 static int adx_encode_header(AVCodecContext *avctx, uint8_t *buf, int bufsize)
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 {
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     ADXContext *c = avctx->priv_data;
 
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     bytestream_put_be16(&buf, 0x8000);              /* header signature */
     bytestream_put_be16(&buf, HEADER_SIZE - 4);     /* copyright offset */
     bytestream_put_byte(&buf, 3);                   /* encoding */
     bytestream_put_byte(&buf, BLOCK_SIZE);          /* block size */
     bytestream_put_byte(&buf, 4);                   /* sample size */
     bytestream_put_byte(&buf, avctx->channels);     /* channels */
     bytestream_put_be32(&buf, avctx->sample_rate);  /* sample rate */
     bytestream_put_be32(&buf, 0);                   /* total sample count */
     bytestream_put_be16(&buf, c->cutoff);           /* cutoff frequency */
     bytestream_put_byte(&buf, 3);                   /* version */
     bytestream_put_byte(&buf, 0);                   /* flags */
     bytestream_put_be32(&buf, 0);                   /* unknown */
     bytestream_put_be32(&buf, 0);                   /* loop enabled */
     bytestream_put_be16(&buf, 0);                   /* padding */
     bytestream_put_buffer(&buf, "(c)CRI", 6);       /* copyright signature */
 
     return HEADER_SIZE;
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 }
 
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 static av_cold int adx_encode_init(AVCodecContext *avctx)
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 {
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     ADXContext *c = avctx->priv_data;
 
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     if (avctx->channels > 2) {
         av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");
         return AVERROR(EINVAL);
     }
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     avctx->frame_size = BLOCK_SAMPLES;
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     /* the cutoff can be adjusted, but this seems to work pretty well */
     c->cutoff = 500;
     ff_adx_calculate_coeffs(c->cutoff, avctx->sample_rate, COEFF_BITS, c->coeff);
 
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     return 0;
 }
 
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 static int adx_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
                             const AVFrame *frame, int *got_packet_ptr)
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 {
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     ADXContext *c          = avctx->priv_data;
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     const int16_t *samples = (const int16_t *)frame->data[0];
     uint8_t *dst;
     int ch, out_size, ret;
 
     out_size = BLOCK_SIZE * avctx->channels + !c->header_parsed * HEADER_SIZE;
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     if ((ret = ff_alloc_packet2(avctx, avpkt, out_size, 0)) < 0)
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         return ret;
     dst = avpkt->data;
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     if (!c->header_parsed) {
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         int hdrsize;
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         if ((hdrsize = adx_encode_header(avctx, dst, avpkt->size)) < 0) {
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             av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n");
             return AVERROR(EINVAL);
         }
         dst      += hdrsize;
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         c->header_parsed = 1;
     }
 
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     for (ch = 0; ch < avctx->channels; ch++) {
         adx_encode(c, dst, samples + ch, &c->prev[ch], avctx->channels);
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         dst += BLOCK_SIZE;
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     }
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     *got_packet_ptr = 1;
     return 0;
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 }
 
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 AVCodec ff_adpcm_adx_encoder = {
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     .name           = "adpcm_adx",
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     .long_name      = NULL_IF_CONFIG_SMALL("SEGA CRI ADX ADPCM"),
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     .type           = AVMEDIA_TYPE_AUDIO,
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     .id             = AV_CODEC_ID_ADPCM_ADX,
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     .priv_data_size = sizeof(ADXContext),
     .init           = adx_encode_init,
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     .encode2        = adx_encode_frame,
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     .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16,
                                                       AV_SAMPLE_FMT_NONE },
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 };