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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 "codec_internal.h"
27 #include "encode.h"
28 #include "mathops.h"
29 
30 #define ROQ_FRAME_SIZE           735
31 #define ROQ_HEADER_SIZE   8
32 
33 #define MAX_DPCM (127*127)
34 
35 
36 typedef struct ROQDPCMContext {
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 
roq_dpcm_encode_close(AVCodecContext * avctx)45 static 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 
roq_dpcm_encode_init(AVCodecContext * avctx)54 static av_cold int roq_dpcm_encode_init(AVCodecContext *avctx)
55 {
56     ROQDPCMContext *context = avctx->priv_data;
57     int channels = avctx->ch_layout.nb_channels;
58 
59     if (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 * channels) *
70                         (22050 / ROQ_FRAME_SIZE) * 8;
71 
72     context->frame_buffer = av_malloc(8 * ROQ_FRAME_SIZE * channels *
73                                       sizeof(*context->frame_buffer));
74     if (!context->frame_buffer)
75         return AVERROR(ENOMEM);
76 
77     context->lastSample[0] = context->lastSample[1] = 0;
78 
79     return 0;
80 }
81 
dpcm_predict(short * previous,short current)82 static unsigned char dpcm_predict(short *previous, short current)
83 {
84     int diff;
85     int negative;
86     int result;
87     int predicted;
88 
89     diff = current - *previous;
90 
91     negative = diff<0;
92     diff = FFABS(diff);
93 
94     if (diff >= MAX_DPCM)
95         result = 127;
96     else {
97         result = ff_sqrt(diff);
98         result += diff > result*result+result;
99     }
100 
101     /* See if this overflows */
102  retry:
103     diff = result*result;
104     if (negative)
105         diff = -diff;
106     predicted = *previous + diff;
107 
108     /* If it overflows, back off a step */
109     if (predicted > 32767 || predicted < -32768) {
110         result--;
111         goto retry;
112     }
113 
114     /* Add the sign bit */
115     result |= negative << 7;   //if (negative) result |= 128;
116 
117     *previous = predicted;
118 
119     return result;
120 }
121 
roq_dpcm_encode_frame(AVCodecContext * avctx,AVPacket * avpkt,const AVFrame * frame,int * got_packet_ptr)122 static int roq_dpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
123                                  const AVFrame *frame, int *got_packet_ptr)
124 {
125     int i, stereo, data_size, ret;
126     const int16_t *in = frame ? (const int16_t *)frame->data[0] : NULL;
127     int channels = avctx->ch_layout.nb_channels;
128     uint8_t *out;
129     ROQDPCMContext *context = avctx->priv_data;
130 
131     stereo = (channels == 2);
132 
133     if (!in && context->input_frames >= 8)
134         return 0;
135 
136     if (in && context->input_frames < 8) {
137         memcpy(&context->frame_buffer[context->buffered_samples * channels],
138                in, avctx->frame_size * channels * sizeof(*in));
139         context->buffered_samples += avctx->frame_size;
140         if (context->input_frames == 0)
141             context->first_pts = frame->pts;
142         if (context->input_frames < 7) {
143             context->input_frames++;
144             return 0;
145         }
146     }
147     if (context->input_frames < 8)
148         in = context->frame_buffer;
149 
150     if (stereo) {
151         context->lastSample[0] &= 0xFF00;
152         context->lastSample[1] &= 0xFF00;
153     }
154 
155     if (context->input_frames == 7)
156         data_size = channels * context->buffered_samples;
157     else
158         data_size = channels * avctx->frame_size;
159 
160     ret = ff_get_encode_buffer(avctx, avpkt, ROQ_HEADER_SIZE + data_size, 0);
161     if (ret < 0)
162         return ret;
163     out = avpkt->data;
164 
165     bytestream_put_byte(&out, stereo ? 0x21 : 0x20);
166     bytestream_put_byte(&out, 0x10);
167     bytestream_put_le32(&out, data_size);
168 
169     if (stereo) {
170         bytestream_put_byte(&out, (context->lastSample[1])>>8);
171         bytestream_put_byte(&out, (context->lastSample[0])>>8);
172     } else
173         bytestream_put_le16(&out, context->lastSample[0]);
174 
175     /* Write the actual samples */
176     for (i = 0; i < data_size; i++)
177         *out++ = dpcm_predict(&context->lastSample[i & 1], *in++);
178 
179     avpkt->pts      = context->input_frames <= 7 ? context->first_pts : frame->pts;
180     avpkt->duration = data_size / channels;
181 
182     context->input_frames++;
183     if (!in)
184         context->input_frames = FFMAX(context->input_frames, 8);
185 
186     *got_packet_ptr = 1;
187     return 0;
188 }
189 
190 const FFCodec ff_roq_dpcm_encoder = {
191     .p.name         = "roq_dpcm",
192     .p.long_name    = NULL_IF_CONFIG_SMALL("id RoQ DPCM"),
193     .p.type         = AVMEDIA_TYPE_AUDIO,
194     .p.id           = AV_CODEC_ID_ROQ_DPCM,
195     .p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_DELAY,
196     .priv_data_size = sizeof(ROQDPCMContext),
197     .init           = roq_dpcm_encode_init,
198     FF_CODEC_ENCODE_CB(roq_dpcm_encode_frame),
199     .close          = roq_dpcm_encode_close,
200     .p.sample_fmts  = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
201                                                      AV_SAMPLE_FMT_NONE },
202     .caps_internal  = FF_CODEC_CAP_INIT_THREADSAFE,
203 };
204