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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 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 #include "internal.h"
27 #include "mathops.h"
28 
29 #define ROQ_FRAME_SIZE 735
30 #define ROQ_HEADER_SIZE 8
31 
32 #define MAX_DPCM (127*127)
33 
34 
35 typedef struct
36 {
37  short lastSample[2];
40  int16_t *frame_buffer;
41  int64_t first_pts;
43 
44 
46 {
47  ROQDPCMContext *context = avctx->priv_data;
48 
49  av_freep(&context->frame_buffer);
50 
51  return 0;
52 }
53 
55 {
56  ROQDPCMContext *context = avctx->priv_data;
57  int ret;
58 
59  if (avctx->channels > 2) {
60  av_log(avctx, AV_LOG_ERROR, "Audio must be mono or stereo\n");
61  return AVERROR(EINVAL);
62  }
63  if (avctx->sample_rate != 22050) {
64  av_log(avctx, AV_LOG_ERROR, "Audio must be 22050 Hz\n");
65  return AVERROR(EINVAL);
66  }
67 
68  avctx->frame_size = ROQ_FRAME_SIZE;
69  avctx->bit_rate = (ROQ_HEADER_SIZE + ROQ_FRAME_SIZE * avctx->channels) *
70  (22050 / ROQ_FRAME_SIZE) * 8;
71 
72  context->frame_buffer = av_malloc(8 * ROQ_FRAME_SIZE * avctx->channels *
73  sizeof(*context->frame_buffer));
74  if (!context->frame_buffer) {
75  ret = AVERROR(ENOMEM);
76  goto error;
77  }
78 
79  context->lastSample[0] = context->lastSample[1] = 0;
80 
81  return 0;
82 error:
83  roq_dpcm_encode_close(avctx);
84  return ret;
85 }
86 
87 static unsigned char dpcm_predict(short *previous, short current)
88 {
89  int diff;
90  int negative;
91  int result;
92  int predicted;
93 
94  diff = current - *previous;
95 
96  negative = diff<0;
97  diff = FFABS(diff);
98 
99  if (diff >= MAX_DPCM)
100  result = 127;
101  else {
102  result = ff_sqrt(diff);
103  result += diff > result*result+result;
104  }
105 
106  /* See if this overflows */
107  retry:
108  diff = result*result;
109  if (negative)
110  diff = -diff;
111  predicted = *previous + diff;
112 
113  /* If it overflows, back off a step */
114  if (predicted > 32767 || predicted < -32768) {
115  result--;
116  goto retry;
117  }
118 
119  /* Add the sign bit */
120  result |= negative << 7; //if (negative) result |= 128;
121 
122  *previous = predicted;
123 
124  return result;
125 }
126 
128  const AVFrame *frame, int *got_packet_ptr)
129 {
130  int i, stereo, data_size, ret;
131  const int16_t *in = frame ? (const int16_t *)frame->data[0] : NULL;
132  uint8_t *out;
133  ROQDPCMContext *context = avctx->priv_data;
134 
135  stereo = (avctx->channels == 2);
136 
137  if (!in && context->input_frames >= 8)
138  return 0;
139 
140  if (in && context->input_frames < 8) {
141  memcpy(&context->frame_buffer[context->buffered_samples * avctx->channels],
142  in, avctx->frame_size * avctx->channels * sizeof(*in));
143  context->buffered_samples += avctx->frame_size;
144  if (context->input_frames == 0)
145  context->first_pts = frame->pts;
146  if (context->input_frames < 7) {
147  context->input_frames++;
148  return 0;
149  }
150  }
151  if (context->input_frames < 8) {
152  in = context->frame_buffer;
153  }
154 
155  if (stereo) {
156  context->lastSample[0] &= 0xFF00;
157  context->lastSample[1] &= 0xFF00;
158  }
159 
160  if (context->input_frames == 7)
161  data_size = avctx->channels * context->buffered_samples;
162  else
163  data_size = avctx->channels * avctx->frame_size;
164 
165  if ((ret = ff_alloc_packet2(avctx, avpkt, ROQ_HEADER_SIZE + data_size)) < 0)
166  return ret;
167  out = avpkt->data;
168 
169  bytestream_put_byte(&out, stereo ? 0x21 : 0x20);
170  bytestream_put_byte(&out, 0x10);
171  bytestream_put_le32(&out, data_size);
172 
173  if (stereo) {
174  bytestream_put_byte(&out, (context->lastSample[1])>>8);
175  bytestream_put_byte(&out, (context->lastSample[0])>>8);
176  } else
177  bytestream_put_le16(&out, context->lastSample[0]);
178 
179  /* Write the actual samples */
180  for (i = 0; i < data_size; i++)
181  *out++ = dpcm_predict(&context->lastSample[i & 1], *in++);
182 
183  avpkt->pts = context->input_frames <= 7 ? context->first_pts : frame->pts;
184  avpkt->duration = data_size / avctx->channels;
185 
186  context->input_frames++;
187  if (!in)
188  context->input_frames = FFMAX(context->input_frames, 8);
189 
190  *got_packet_ptr = 1;
191  return 0;
192 }
193 
195  .name = "roq_dpcm",
196  .long_name = NULL_IF_CONFIG_SMALL("id RoQ DPCM"),
197  .type = AVMEDIA_TYPE_AUDIO,
198  .id = AV_CODEC_ID_ROQ_DPCM,
199  .priv_data_size = sizeof(ROQDPCMContext),
201  .encode2 = roq_dpcm_encode_frame,
203  .capabilities = CODEC_CAP_DELAY,
204  .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
206 };