libavcodec/roqaudioenc.c

   1 /*
   2  * RoQ audio encoder
   3  *
   4  * Copyright (c) 2005 Eric Lasota
   5  *    Based on RoQ specs (c)2001 Tim Ferguson
   6  *
   7  * This file is part of FFmpeg.
   8  *
   9  * FFmpeg is free software; you can redistribute it and/or
  10  * modify it under the terms of the GNU Lesser General Public
  11  * License as published by the Free Software Foundation; either
  12  * version 2.1 of the License, or (at your option) any later version.
  13  *
  14  * FFmpeg is distributed in the hope that it will be useful,
  15  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  16  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  17  * Lesser General Public License for more details.
  18  *
  19  * You should have received a copy of the GNU Lesser General Public
  20  * License along with FFmpeg; if not, write to the Free Software
  21  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  22  */
  23
  24 #include "avcodec.h"
  25 #include "bytestream.h"
  26
  27 #define ROQ_FIRST_FRAME_SIZE     (735*8)
  28 #define ROQ_FRAME_SIZE           735
  29
  30
  31 #define MAX_DPCM (127*127)
  32
  33
  34 typedef struct
  35 {
  36     short lastSample[2];
  37 } ROQDPCMContext;
  38
  39 static av_cold int roq_dpcm_encode_init(AVCodecContext *avctx)
  40 {
  41     ROQDPCMContext *context = avctx->priv_data;
  42
  43     if (avctx->channels > 2) {
  44         av_log(avctx, AV_LOG_ERROR, "Audio must be mono or stereo\n");
  45         return -1;
  46     }
  47     if (avctx->sample_rate != 22050) {
  48         av_log(avctx, AV_LOG_ERROR, "Audio must be 22050 Hz\n");
  49         return -1;
  50     }
  51     if (avctx->sample_fmt != SAMPLE_FMT_S16) {
  52         av_log(avctx, AV_LOG_ERROR, "Audio must be signed 16-bit\n");
  53         return -1;
  54     }
  55
  56     avctx->frame_size = ROQ_FIRST_FRAME_SIZE;
  57
  58     context->lastSample[0] = context->lastSample[1] = 0;
  59
  60     avctx->coded_frame= avcodec_alloc_frame();
  61     avctx->coded_frame->key_frame= 1;
  62
  63     return 0;
  64 }
  65
  66 static unsigned char dpcm_predict(short *previous, short current)
  67 {
  68     int diff;
  69     int negative;
  70     int result;
  71     int predicted;
  72
  73     diff = current - *previous;
  74
  75     negative = diff<0;
  76     diff = FFABS(diff);
  77
  78     if (diff >= MAX_DPCM)
  79         result = 127;
  80     else {
  81         result = ff_sqrt(diff);
  82         result += diff > result*result+result;
  83     }
  84
  85     /* See if this overflows */
  86  retry:
  87     diff = result*result;
  88     if (negative)
  89         diff = -diff;
  90     predicted = *previous + diff;
  91
  92     /* If it overflows, back off a step */
  93     if (predicted > 32767 || predicted < -32768) {
  94         result--;
  95         goto retry;
  96     }
  97
  98     /* Add the sign bit */
  99     result |= negative << 7;   //if (negative) result |= 128;
 100
 101     *previous = predicted;
 102
 103     return result;
 104 }
 105
 106 static int roq_dpcm_encode_frame(AVCodecContext *avctx,
 107                 unsigned char *frame, int buf_size, void *data)
 108 {
 109     int i, samples, stereo, ch;
 110     short *in;
 111     unsigned char *out;
 112
 113     ROQDPCMContext *context = avctx->priv_data;
 114
 115     stereo = (avctx->channels == 2);
 116
 117     if (stereo) {
 118         context->lastSample[0] &= 0xFF00;
 119         context->lastSample[1] &= 0xFF00;
 120     }
 121
 122     out = frame;
 123     in = data;
 124
 125     bytestream_put_byte(&out, stereo ? 0x21 : 0x20);
 126     bytestream_put_byte(&out, 0x10);
 127     bytestream_put_le32(&out, avctx->frame_size*avctx->channels);
 128
 129     if (stereo) {
 130         bytestream_put_byte(&out, (context->lastSample[1])>>8);
 131         bytestream_put_byte(&out, (context->lastSample[0])>>8);
 132     } else
 133         bytestream_put_le16(&out, context->lastSample[0]);
 134
 135     /* Write the actual samples */
 136     samples = avctx->frame_size;
 137     for (i=0; i<samples; i++)
 138         for (ch=0; ch<avctx->channels; ch++)
 139             *out++ = dpcm_predict(&context->lastSample[ch], *in++);
 140
 141     /* Use smaller frames from now on */
 142     avctx->frame_size = ROQ_FRAME_SIZE;
 143
 144     /* Return the result size */
 145     return out - frame;
 146 }
 147
 148 static av_cold int roq_dpcm_encode_close(AVCodecContext *avctx)
 149 {
 150     av_freep(&avctx->coded_frame);
 151
 152     return 0;
 153 }
 154
 155 AVCodec roq_dpcm_encoder = {
 156     "roq_dpcm",
 157     CODEC_TYPE_AUDIO,
 158     CODEC_ID_ROQ_DPCM,
 159     sizeof(ROQDPCMContext),
 160     roq_dpcm_encode_init,
 161     roq_dpcm_encode_frame,
 162     roq_dpcm_encode_close,
 163     NULL,
 164     .sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
 165     .long_name = NULL_IF_CONFIG_SMALL("id RoQ DPCM"),
 166 };