FFmpeg coverage

Directory: ../../../ffmpeg/
File: src/libavcodec/roqaudioenc.c
Date: 2024-11-20 23:03:26
Exec Total Coverage
Lines: 74 84 88.1%
Functions: 4 4 100.0%
Branches: 34 46 73.9%
Line Branch Exec Source
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 "libavutil/mem.h"
25 #include "avcodec.h"
26 #include "bytestream.h"
27 #include "codec_internal.h"
28 #include "encode.h"
29 #include "mathops.h"
30
31 #define ROQ_FRAME_SIZE 735
32 #define ROQ_HEADER_SIZE 8
33
34 #define MAX_DPCM (127*127)
35
36
37 typedef struct ROQDPCMContext {
38 short lastSample[2];
39 int input_frames;
40 int buffered_samples;
41 int16_t *frame_buffer;
42 int64_t first_pts;
43 } ROQDPCMContext;
44
45
46 1 static av_cold int roq_dpcm_encode_close(AVCodecContext *avctx)
47 {
48 1 ROQDPCMContext *context = avctx->priv_data;
49
50 1 av_freep(&context->frame_buffer);
51
52 1 return 0;
53 }
54
55 1 static av_cold int roq_dpcm_encode_init(AVCodecContext *avctx)
56 {
57 1 ROQDPCMContext *context = avctx->priv_data;
58 1 int channels = avctx->ch_layout.nb_channels;
59
60
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 1 if (channels > 2) {
61 ✗ av_log(avctx, AV_LOG_ERROR, "Audio must be mono or stereo\n");
62 ✗ return AVERROR(EINVAL);
63 }
64
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 1 if (avctx->sample_rate != 22050) {
65 ✗ av_log(avctx, AV_LOG_ERROR, "Audio must be 22050 Hz\n");
66 ✗ return AVERROR(EINVAL);
67 }
68
69 1 avctx->frame_size = ROQ_FRAME_SIZE;
70 1 avctx->bit_rate = (ROQ_HEADER_SIZE + ROQ_FRAME_SIZE * channels) *
71 1 (22050 / ROQ_FRAME_SIZE) * 8;
72
73 1 context->frame_buffer = av_malloc(8 * ROQ_FRAME_SIZE * channels *
74 sizeof(*context->frame_buffer));
75
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 1 if (!context->frame_buffer)
76 ✗ return AVERROR(ENOMEM);
77
78 1 context->lastSample[0] = context->lastSample[1] = 0;
79
80 1 return 0;
81 }
82
83 264600 static unsigned char dpcm_predict(short *previous, short current)
84 {
85 int diff;
86 int negative;
87 int result;
88 int predicted;
89
90 264600 diff = current - *previous;
91
92 264600 negative = diff<0;
93 264600 diff = FFABS(diff);
94
95
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 264600 if (diff >= MAX_DPCM)
96 10896 result = 127;
97 else {
98 253704 result = ff_sqrt(diff);
99 253704 result += diff > result*result+result;
100 }
101
102 /* See if this overflows */
103 264600 retry:
104 264600 diff = result*result;
105
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 264600 if (negative)
106 131603 diff = -diff;
107 264600 predicted = *previous + diff;
108
109 /* If it overflows, back off a step */
110
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 264600 if (predicted > 32767 || predicted < -32768) {
111 ✗ result--;
112 ✗ goto retry;
113 }
114
115 /* Add the sign bit */
116 264600 result |= negative << 7; //if (negative) result |= 128;
117
118 264600 *previous = predicted;
119
120 264600 return result;
121 }
122
123 181 static int roq_dpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
124 const AVFrame *frame, int *got_packet_ptr)
125 {
126 int i, stereo, data_size, ret;
127
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 181 const int16_t *in = frame ? (const int16_t *)frame->data[0] : NULL;
128 181 int channels = avctx->ch_layout.nb_channels;
129 uint8_t *out;
130 181 ROQDPCMContext *context = avctx->priv_data;
131
132 181 stereo = (channels == 2);
133
134
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 181 if (!in && context->input_frames >= 8)
135 1 return 0;
136
137
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 180 if (in && context->input_frames < 8) {
138 8 memcpy(&context->frame_buffer[context->buffered_samples * channels],
139 8 in, avctx->frame_size * channels * sizeof(*in));
140 8 context->buffered_samples += avctx->frame_size;
141
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 8 if (context->input_frames == 0)
142 1 context->first_pts = frame->pts;
143
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 8 if (context->input_frames < 7) {
144 7 context->input_frames++;
145 7 return 0;
146 }
147 }
148
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 173 if (context->input_frames < 8)
149 1 in = context->frame_buffer;
150
151
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 173 if (stereo) {
152 173 context->lastSample[0] &= 0xFF00;
153 173 context->lastSample[1] &= 0xFF00;
154 }
155
156
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 173 if (context->input_frames == 7)
157 1 data_size = channels * context->buffered_samples;
158 else
159 172 data_size = channels * avctx->frame_size;
160
161 173 ret = ff_get_encode_buffer(avctx, avpkt, ROQ_HEADER_SIZE + data_size, 0);
162
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 173 if (ret < 0)
163 ✗ return ret;
164 173 out = avpkt->data;
165
166
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 173 bytestream_put_byte(&out, stereo ? 0x21 : 0x20);
167 173 bytestream_put_byte(&out, 0x10);
168 173 bytestream_put_le32(&out, data_size);
169
170
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 173 if (stereo) {
171 173 bytestream_put_byte(&out, (context->lastSample[1])>>8);
172 173 bytestream_put_byte(&out, (context->lastSample[0])>>8);
173 } else
174 ✗ bytestream_put_le16(&out, context->lastSample[0]);
175
176 /* Write the actual samples */
177
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 264773 for (i = 0; i < data_size; i++)
178 264600 *out++ = dpcm_predict(&context->lastSample[(i & 1) & stereo], *in++);
179
180
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 173 avpkt->pts = context->input_frames <= 7 ? context->first_pts : frame->pts;
181 173 avpkt->duration = data_size / channels;
182
183 173 context->input_frames++;
184
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 173 if (!in)
185 ✗ context->input_frames = FFMAX(context->input_frames, 8);
186
187 173 *got_packet_ptr = 1;
188 173 return 0;
189 }
190
191 const FFCodec ff_roq_dpcm_encoder = {
192 .p.name = "roq_dpcm",
193 CODEC_LONG_NAME("id RoQ DPCM"),
194 .p.type = AVMEDIA_TYPE_AUDIO,
195 .p.id = AV_CODEC_ID_ROQ_DPCM,
196 .p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_DELAY,
197 .priv_data_size = sizeof(ROQDPCMContext),
198 .init = roq_dpcm_encode_init,
199 FF_CODEC_ENCODE_CB(roq_dpcm_encode_frame),
200 .close = roq_dpcm_encode_close,
201 .p.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
202 AV_SAMPLE_FMT_NONE },
203 };
204