Mercurial > libavcodec.hg
annotate roqaudioenc.c @ 12488:351a81a23343 libavcodec
Set a constant frame size for encoding G.726 audio.
author | jbr |
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date | Sat, 11 Sep 2010 19:52:09 +0000 |
parents | dde20597f15e |
children |
rev | line source |
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4932 | 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 | |
11417 | 24 #include "libavutil/intmath.h" |
4932 | 25 #include "avcodec.h" |
26 #include "bytestream.h" | |
27 | |
28 #define ROQ_FIRST_FRAME_SIZE (735*8) | |
29 #define ROQ_FRAME_SIZE 735 | |
30 | |
31 | |
32 #define MAX_DPCM (127*127) | |
33 | |
34 | |
35 typedef struct | |
36 { | |
37 short lastSample[2]; | |
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Rename ROQDPCMContext_t to ROQDPCMContext to avoid _t reserved prefix.
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38 } ROQDPCMContext; |
4932 | 39 |
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Add missing av_cold in static init/close functions.
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40 static av_cold int roq_dpcm_encode_init(AVCodecContext *avctx) |
4932 | 41 { |
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42 ROQDPCMContext *context = avctx->priv_data; |
4932 | 43 |
44 if (avctx->channels > 2) { | |
45 av_log(avctx, AV_LOG_ERROR, "Audio must be mono or stereo\n"); | |
46 return -1; | |
47 } | |
48 if (avctx->sample_rate != 22050) { | |
49 av_log(avctx, AV_LOG_ERROR, "Audio must be 22050 Hz\n"); | |
50 return -1; | |
51 } | |
52 if (avctx->sample_fmt != SAMPLE_FMT_S16) { | |
53 av_log(avctx, AV_LOG_ERROR, "Audio must be signed 16-bit\n"); | |
54 return -1; | |
55 } | |
56 | |
57 avctx->frame_size = ROQ_FIRST_FRAME_SIZE; | |
58 | |
59 context->lastSample[0] = context->lastSample[1] = 0; | |
60 | |
61 avctx->coded_frame= avcodec_alloc_frame(); | |
62 avctx->coded_frame->key_frame= 1; | |
63 | |
64 return 0; | |
65 } | |
66 | |
67 static unsigned char dpcm_predict(short *previous, short current) | |
68 { | |
69 int diff; | |
70 int negative; | |
71 int result; | |
72 int predicted; | |
73 | |
74 diff = current - *previous; | |
75 | |
76 negative = diff<0; | |
77 diff = FFABS(diff); | |
78 | |
79 if (diff >= MAX_DPCM) | |
80 result = 127; | |
10427
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Replace big square-root table by a call to ff_sqrt(). Based on a patch
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81 else { |
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Replace big square-root table by a call to ff_sqrt(). Based on a patch
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82 result = ff_sqrt(diff); |
1a3908177e87
Replace big square-root table by a call to ff_sqrt(). Based on a patch
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83 result += diff > result*result+result; |
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84 } |
4932 | 85 |
86 /* See if this overflows */ | |
87 retry: | |
88 diff = result*result; | |
89 if (negative) | |
90 diff = -diff; | |
91 predicted = *previous + diff; | |
92 | |
93 /* If it overflows, back off a step */ | |
94 if (predicted > 32767 || predicted < -32768) { | |
95 result--; | |
96 goto retry; | |
97 } | |
98 | |
99 /* Add the sign bit */ | |
100 result |= negative << 7; //if (negative) result |= 128; | |
101 | |
102 *previous = predicted; | |
103 | |
104 return result; | |
105 } | |
106 | |
107 static int roq_dpcm_encode_frame(AVCodecContext *avctx, | |
108 unsigned char *frame, int buf_size, void *data) | |
109 { | |
110 int i, samples, stereo, ch; | |
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111 const short *in; |
4932 | 112 unsigned char *out; |
113 | |
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114 ROQDPCMContext *context = avctx->priv_data; |
4932 | 115 |
116 stereo = (avctx->channels == 2); | |
117 | |
118 if (stereo) { | |
119 context->lastSample[0] &= 0xFF00; | |
120 context->lastSample[1] &= 0xFF00; | |
121 } | |
122 | |
123 out = frame; | |
124 in = data; | |
125 | |
126 bytestream_put_byte(&out, stereo ? 0x21 : 0x20); | |
127 bytestream_put_byte(&out, 0x10); | |
128 bytestream_put_le32(&out, avctx->frame_size*avctx->channels); | |
129 | |
130 if (stereo) { | |
131 bytestream_put_byte(&out, (context->lastSample[1])>>8); | |
132 bytestream_put_byte(&out, (context->lastSample[0])>>8); | |
133 } else | |
134 bytestream_put_le16(&out, context->lastSample[0]); | |
135 | |
136 /* Write the actual samples */ | |
137 samples = avctx->frame_size; | |
138 for (i=0; i<samples; i++) | |
139 for (ch=0; ch<avctx->channels; ch++) | |
140 *out++ = dpcm_predict(&context->lastSample[ch], *in++); | |
141 | |
142 /* Use smaller frames from now on */ | |
143 avctx->frame_size = ROQ_FRAME_SIZE; | |
144 | |
145 /* Return the result size */ | |
146 return out - frame; | |
147 } | |
148 | |
6517
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Apply 'cold' attribute to init/uninit functions in libavcodec
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149 static av_cold int roq_dpcm_encode_close(AVCodecContext *avctx) |
4932 | 150 { |
151 av_freep(&avctx->coded_frame); | |
152 | |
153 return 0; | |
154 } | |
155 | |
156 AVCodec roq_dpcm_encoder = { | |
157 "roq_dpcm", | |
11560
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Define AVMediaType enum, and use it instead of enum CodecType, which
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158 AVMEDIA_TYPE_AUDIO, |
4932 | 159 CODEC_ID_ROQ_DPCM, |
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Rename ROQDPCMContext_t to ROQDPCMContext to avoid _t reserved prefix.
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diff
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160 sizeof(ROQDPCMContext), |
4932 | 161 roq_dpcm_encode_init, |
162 roq_dpcm_encode_frame, | |
163 roq_dpcm_encode_close, | |
164 NULL, | |
10145
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Make sample_fmts and channel_layouts compound literals const to reduce size of
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165 .sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE}, |
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Make AVCodec long_names definition conditional depending on CONFIG_SMALL.
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166 .long_name = NULL_IF_CONFIG_SMALL("id RoQ DPCM"), |
4932 | 167 }; |