roqaudioenc.c
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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 Libav.
8  *
9  * Libav 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  * Libav 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 Libav; if not, write to the Free Software
21  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
22  */
23 
24 #include "libavutil/intmath.h"
25 #include "avcodec.h"
26 #include "bytestream.h"
27 
28 #define ROQ_FIRST_FRAME_SIZE (735*8)
29 #define ROQ_FRAME_SIZE 735
30 
31 
32 #define MAX_DPCM (127*127)
33 
34 
35 typedef struct
36 {
37  short lastSample[2];
39 
41 {
42  ROQDPCMContext *context = avctx->priv_data;
43 
44  if (avctx->channels > 2) {
45  av_log(avctx, AV_LOG_ERROR, "Audio must be mono or stereo\n");
46  return -1;
47  }
48  if (avctx->sample_rate != 22050) {
49  av_log(avctx, AV_LOG_ERROR, "Audio must be 22050 Hz\n");
50  return -1;
51  }
52  if (avctx->sample_fmt != AV_SAMPLE_FMT_S16) {
53  av_log(avctx, AV_LOG_ERROR, "Audio must be signed 16-bit\n");
54  return -1;
55  }
56 
58 
59  context->lastSample[0] = context->lastSample[1] = 0;
60 
62  avctx->coded_frame->key_frame= 1;
63 
64  return 0;
65 }
66 
67 static unsigned char dpcm_predict(short *previous, short current)
68 {
69  int diff;
70  int negative;
71  int result;
72  int predicted;
73 
74  diff = current - *previous;
75 
76  negative = diff<0;
77  diff = FFABS(diff);
78 
79  if (diff >= MAX_DPCM)
80  result = 127;
81  else {
82  result = ff_sqrt(diff);
83  result += diff > result*result+result;
84  }
85 
86  /* See if this overflows */
87  retry:
88  diff = result*result;
89  if (negative)
90  diff = -diff;
91  predicted = *previous + diff;
92 
93  /* If it overflows, back off a step */
94  if (predicted > 32767 || predicted < -32768) {
95  result--;
96  goto retry;
97  }
98 
99  /* Add the sign bit */
100  result |= negative << 7; //if (negative) result |= 128;
101 
102  *previous = predicted;
103 
104  return result;
105 }
106 
108  unsigned char *frame, int buf_size, void *data)
109 {
110  int i, samples, stereo, ch;
111  const short *in;
112  unsigned char *out;
113 
114  ROQDPCMContext *context = avctx->priv_data;
115 
116  stereo = (avctx->channels == 2);
117 
118  if (stereo) {
119  context->lastSample[0] &= 0xFF00;
120  context->lastSample[1] &= 0xFF00;
121  }
122 
123  out = frame;
124  in = data;
125 
126  bytestream_put_byte(&out, stereo ? 0x21 : 0x20);
127  bytestream_put_byte(&out, 0x10);
128  bytestream_put_le32(&out, avctx->frame_size*avctx->channels);
129 
130  if (stereo) {
131  bytestream_put_byte(&out, (context->lastSample[1])>>8);
132  bytestream_put_byte(&out, (context->lastSample[0])>>8);
133  } else
134  bytestream_put_le16(&out, context->lastSample[0]);
135 
136  /* Write the actual samples */
137  samples = avctx->frame_size;
138  for (i=0; i<samples; i++)
139  for (ch=0; ch<avctx->channels; ch++)
140  *out++ = dpcm_predict(&context->lastSample[ch], *in++);
141 
142  /* Use smaller frames from now on */
143  avctx->frame_size = ROQ_FRAME_SIZE;
144 
145  /* Return the result size */
146  return out - frame;
147 }
148 
150 {
151  av_freep(&avctx->coded_frame);
152 
153  return 0;
154 }
155 
157  .name = "roq_dpcm",
158  .type = AVMEDIA_TYPE_AUDIO,
159  .id = CODEC_ID_ROQ_DPCM,
160  .priv_data_size = sizeof(ROQDPCMContext),
162  .encode = roq_dpcm_encode_frame,
164  .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
165  .long_name = NULL_IF_CONFIG_SMALL("id RoQ DPCM"),
166 };