609beffb4b4bf754286a22cd12661624464638d2
[libav.git] / 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 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 #include "internal.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];
38 int input_frames;
39 int buffered_samples;
40 int16_t *frame_buffer;
41 int64_t first_pts;
42 } ROQDPCMContext;
43
44
45 static av_cold int roq_dpcm_encode_close(AVCodecContext *avctx)
46 {
47 ROQDPCMContext *context = avctx->priv_data;
48
49 #if FF_API_OLD_ENCODE_AUDIO
50 av_freep(&avctx->coded_frame);
51 #endif
52 av_freep(&context->frame_buffer);
53
54 return 0;
55 }
56
57 static av_cold int roq_dpcm_encode_init(AVCodecContext *avctx)
58 {
59 ROQDPCMContext *context = avctx->priv_data;
60 int ret;
61
62 if (avctx->channels > 2) {
63 av_log(avctx, AV_LOG_ERROR, "Audio must be mono or stereo\n");
64 return AVERROR(EINVAL);
65 }
66 if (avctx->sample_rate != 22050) {
67 av_log(avctx, AV_LOG_ERROR, "Audio must be 22050 Hz\n");
68 return AVERROR(EINVAL);
69 }
70
71 avctx->frame_size = ROQ_FRAME_SIZE;
72 avctx->bit_rate = (ROQ_HEADER_SIZE + ROQ_FRAME_SIZE * avctx->channels) *
73 (22050 / ROQ_FRAME_SIZE) * 8;
74
75 context->frame_buffer = av_malloc(8 * ROQ_FRAME_SIZE * avctx->channels *
76 sizeof(*context->frame_buffer));
77 if (!context->frame_buffer) {
78 ret = AVERROR(ENOMEM);
79 goto error;
80 }
81
82 context->lastSample[0] = context->lastSample[1] = 0;
83
84 #if FF_API_OLD_ENCODE_AUDIO
85 avctx->coded_frame= avcodec_alloc_frame();
86 if (!avctx->coded_frame) {
87 ret = AVERROR(ENOMEM);
88 goto error;
89 }
90 #endif
91
92 return 0;
93 error:
94 roq_dpcm_encode_close(avctx);
95 return ret;
96 }
97
98 static unsigned char dpcm_predict(short *previous, short current)
99 {
100 int diff;
101 int negative;
102 int result;
103 int predicted;
104
105 diff = current - *previous;
106
107 negative = diff<0;
108 diff = FFABS(diff);
109
110 if (diff >= MAX_DPCM)
111 result = 127;
112 else {
113 result = ff_sqrt(diff);
114 result += diff > result*result+result;
115 }
116
117 /* See if this overflows */
118 retry:
119 diff = result*result;
120 if (negative)
121 diff = -diff;
122 predicted = *previous + diff;
123
124 /* If it overflows, back off a step */
125 if (predicted > 32767 || predicted < -32768) {
126 result--;
127 goto retry;
128 }
129
130 /* Add the sign bit */
131 result |= negative << 7; //if (negative) result |= 128;
132
133 *previous = predicted;
134
135 return result;
136 }
137
138 static int roq_dpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
139 const AVFrame *frame, int *got_packet_ptr)
140 {
141 int i, stereo, data_size, ret;
142 const int16_t *in = frame ? (const int16_t *)frame->data[0] : NULL;
143 uint8_t *out;
144 ROQDPCMContext *context = avctx->priv_data;
145
146 stereo = (avctx->channels == 2);
147
148 if (!in && context->input_frames >= 8)
149 return 0;
150
151 if (in && context->input_frames < 8) {
152 memcpy(&context->frame_buffer[context->buffered_samples * avctx->channels],
153 in, avctx->frame_size * avctx->channels * sizeof(*in));
154 context->buffered_samples += avctx->frame_size;
155 if (context->input_frames == 0)
156 context->first_pts = frame->pts;
157 if (context->input_frames < 7) {
158 context->input_frames++;
159 return 0;
160 }
161 in = context->frame_buffer;
162 }
163
164 if (stereo) {
165 context->lastSample[0] &= 0xFF00;
166 context->lastSample[1] &= 0xFF00;
167 }
168
169 if (context->input_frames == 7 || !in)
170 data_size = avctx->channels * context->buffered_samples;
171 else
172 data_size = avctx->channels * avctx->frame_size;
173
174 if ((ret = ff_alloc_packet(avpkt, ROQ_HEADER_SIZE + data_size))) {
175 av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
176 return ret;
177 }
178 out = avpkt->data;
179
180 bytestream_put_byte(&out, stereo ? 0x21 : 0x20);
181 bytestream_put_byte(&out, 0x10);
182 bytestream_put_le32(&out, data_size);
183
184 if (stereo) {
185 bytestream_put_byte(&out, (context->lastSample[1])>>8);
186 bytestream_put_byte(&out, (context->lastSample[0])>>8);
187 } else
188 bytestream_put_le16(&out, context->lastSample[0]);
189
190 /* Write the actual samples */
191 for (i = 0; i < data_size; i++)
192 *out++ = dpcm_predict(&context->lastSample[i & 1], *in++);
193
194 avpkt->pts = context->input_frames <= 7 ? context->first_pts : frame->pts;
195 avpkt->duration = data_size / avctx->channels;
196
197 context->input_frames++;
198 if (!in)
199 context->input_frames = FFMAX(context->input_frames, 8);
200
201 *got_packet_ptr = 1;
202 return 0;
203 }
204
205 AVCodec ff_roq_dpcm_encoder = {
206 .name = "roq_dpcm",
207 .type = AVMEDIA_TYPE_AUDIO,
208 .id = CODEC_ID_ROQ_DPCM,
209 .priv_data_size = sizeof(ROQDPCMContext),
210 .init = roq_dpcm_encode_init,
211 .encode2 = roq_dpcm_encode_frame,
212 .close = roq_dpcm_encode_close,
213 .capabilities = CODEC_CAP_DELAY,
214 .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
215 .long_name = NULL_IF_CONFIG_SMALL("id RoQ DPCM"),
216 };