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
35typedef 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
45static av_cold int roq_dpcm_encode_close(AVCodecContext *avctx)
46{
47    ROQDPCMContext *context = avctx->priv_data;
48
49    av_freep(&context->frame_buffer);
50
51    return 0;
52}
53
54static av_cold int roq_dpcm_encode_init(AVCodecContext *avctx)
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;
82error:
83    roq_dpcm_encode_close(avctx);
84    return ret;
85}
86
87static 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
127static int roq_dpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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
194AVCodec ff_roq_dpcm_encoder = {
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),
200    .init           = roq_dpcm_encode_init,
201    .encode2        = roq_dpcm_encode_frame,
202    .close          = roq_dpcm_encode_close,
203    .capabilities   = CODEC_CAP_DELAY,
204    .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
205                                                     AV_SAMPLE_FMT_NONE },
206};
207