FFmpeg: libavcodec/adxenc.c Source File

FFmpeg
adxenc.c
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1 /*
2  * ADX ADPCM codecs
3  * Copyright (c) 2001,2003 BERO
4  *
5  * This file is part of FFmpeg.
6  *
7  * FFmpeg is free software; you can redistribute it and/or
8  * modify it under the terms of the GNU Lesser General Public
9  * License as published by the Free Software Foundation; either
10  * version 2.1 of the License, or (at your option) any later version.
11  *
12  * FFmpeg is distributed in the hope that it will be useful,
13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15  * Lesser General Public License for more details.
16  *
17  * You should have received a copy of the GNU Lesser General Public
18  * License along with FFmpeg; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 
22 #include "avcodec.h"
23 #include "adx.h"
24 #include "bytestream.h"
25 #include "internal.h"
26 #include "put_bits.h"
27 
28 /**
29  * @file
30  * SEGA CRI adx codecs.
31  *
32  * Reference documents:
33  * http://ku-www.ss.titech.ac.jp/~yatsushi/adx.html
34  * adx2wav & wav2adx http://www.geocities.co.jp/Playtown/2004/
35  */
36 
37  static void adx_encode(ADXContext *c, uint8_t *adx, const int16_t *wav,
38  ADXChannelState *prev, int channels)
39 {
40  PutBitContext pb;
41  int scale;
42  int i, j;
43  int s0, s1, s2, d;
44  int max = 0;
45  int min = 0;
46  int data[BLOCK_SAMPLES];
47 
48  s1 = prev->s1;
49  s2 = prev->s2;
50  for (i = 0, j = 0; j < 32; i += channels, j++) {
51  s0 = wav[i];
52  d = ((s0 << COEFF_BITS) - c->coeff[0] * s1 - c->coeff[1] * s2) >> COEFF_BITS;
53  data[j] = d;
54  if (max < d)
55  max = d;
56  if (min > d)
57  min = d;
58  s2 = s1;
59  s1 = s0;
60  }
61  prev->s1 = s1;
62  prev->s2 = s2;
63 
64  if (max == 0 && min == 0) {
65  memset(adx, 0, BLOCK_SIZE);
66  return;
67  }
68 
69  if (max / 7 > -min / 8)
70  scale = max / 7;
71  else
72  scale = -min / 8;
73 
74  if (scale == 0)
75  scale = 1;
76 
77  AV_WB16(adx, scale);
78 
79  init_put_bits(&pb, adx + 2, 16);
80  for (i = 0; i < BLOCK_SAMPLES; i++)
81  put_sbits(&pb, 4, av_clip(data[i] / scale, -8, 7));
82  flush_put_bits(&pb);
83 }
84 
85  #define HEADER_SIZE 36
86 
87  static int adx_encode_header(AVCodecContext *avctx, uint8_t *buf, int bufsize)
88 {
89  ADXContext *c = avctx->priv_data;
90 
91  bytestream_put_be16(&buf, 0x8000); /* header signature */
92  bytestream_put_be16(&buf, HEADER_SIZE - 4); /* copyright offset */
93  bytestream_put_byte(&buf, 3); /* encoding */
94  bytestream_put_byte(&buf, BLOCK_SIZE); /* block size */
95  bytestream_put_byte(&buf, 4); /* sample size */
96  bytestream_put_byte(&buf, avctx->channels); /* channels */
97  bytestream_put_be32(&buf, avctx->sample_rate); /* sample rate */
98  bytestream_put_be32(&buf, 0); /* total sample count */
99  bytestream_put_be16(&buf, c->cutoff); /* cutoff frequency */
100  bytestream_put_byte(&buf, 3); /* version */
101  bytestream_put_byte(&buf, 0); /* flags */
102  bytestream_put_be32(&buf, 0); /* unknown */
103  bytestream_put_be32(&buf, 0); /* loop enabled */
104  bytestream_put_be16(&buf, 0); /* padding */
105  bytestream_put_buffer(&buf, "(c)CRI", 6); /* copyright signature */
106 
107  return HEADER_SIZE;
108 }
109 
110  static av_cold int adx_encode_init(AVCodecContext *avctx)
111 {
112  ADXContext *c = avctx->priv_data;
113 
114  if (avctx->channels > 2) {
115  av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");
116  return AVERROR(EINVAL);
117  }
118  avctx->frame_size = BLOCK_SAMPLES;
119 
120 #if FF_API_OLD_ENCODE_AUDIO
121  avcodec_get_frame_defaults(&c->frame);
122  avctx->coded_frame = &c->frame;
123 #endif
124 
125  /* the cutoff can be adjusted, but this seems to work pretty well */
126  c->cutoff = 500;
127  ff_adx_calculate_coeffs(c->cutoff, avctx->sample_rate, COEFF_BITS, c->coeff);
128 
129  return 0;
130 }
131 
132  static int adx_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
133  const AVFrame *frame, int *got_packet_ptr)
134 {
135  ADXContext *c = avctx->priv_data;
136  const int16_t *samples = (const int16_t *)frame->data[0];
137  uint8_t *dst;
138  int ch, out_size, ret;
139 
140  out_size = BLOCK_SIZE * avctx->channels + !c->header_parsed * HEADER_SIZE;
141  if ((ret = ff_alloc_packet2(avctx, avpkt, out_size)) < 0)
142  return ret;
143  dst = avpkt->data;
144 
145  if (!c->header_parsed) {
146  int hdrsize;
147  if ((hdrsize = adx_encode_header(avctx, dst, avpkt->size)) < 0) {
148  av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n");
149  return AVERROR(EINVAL);
150  }
151  dst += hdrsize;
152  c->header_parsed = 1;
153  }
154 
155  for (ch = 0; ch < avctx->channels; ch++) {
156  adx_encode(c, dst, samples + ch, &c->prev[ch], avctx->channels);
157  dst += BLOCK_SIZE;
158  }
159 
160  *got_packet_ptr = 1;
161  return 0;
162 }
163 
164  AVCodec ff_adpcm_adx_encoder = {
165  .name = "adpcm_adx",
166  .type = AVMEDIA_TYPE_AUDIO,
167  .id = AV_CODEC_ID_ADPCM_ADX,
168  .priv_data_size = sizeof(ADXContext),
169  .init = adx_encode_init,
170  .encode2 = adx_encode_frame,
171  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16,
172  AV_SAMPLE_FMT_NONE },
173  .long_name = NULL_IF_CONFIG_SMALL("SEGA CRI ADX ADPCM"),
174 };
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