Mercurial > libavcodec.hg
comparison binkaudio.c @ 11067:91b1e4327340 libavcodec
Bink Audio decoder
author | pross |
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date | Sun, 31 Jan 2010 12:51:15 +0000 |
parents | |
children | 28f00789adcd |
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11066:86bf7e0db6ea | 11067:91b1e4327340 |
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1 /* | |
2 * Bink Audio decoder | |
3 * Copyright (c) 2007-2010 Peter Ross (pross@xvid.org) | |
4 * Copyright (c) 2009 Daniel Verkamp (daniel@drv.nu) | |
5 * | |
6 * This file is part of FFmpeg. | |
7 * | |
8 * FFmpeg is free software; you can redistribute it and/or | |
9 * modify it under the terms of the GNU Lesser General Public | |
10 * License as published by the Free Software Foundation; either | |
11 * version 2.1 of the License, or (at your option) any later version. | |
12 * | |
13 * FFmpeg is distributed in the hope that it will be useful, | |
14 * but WITHOUT ANY WARRANTY; without even the implied warranty of | |
15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | |
16 * Lesser General Public License for more details. | |
17 * | |
18 * You should have received a copy of the GNU Lesser General Public | |
19 * License along with FFmpeg; if not, write to the Free Software | |
20 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA | |
21 */ | |
22 | |
23 /** | |
24 * @file libavcodec/binkaudio.c | |
25 * Bink Audio decoder | |
26 * | |
27 * Technical details here: | |
28 * http://wiki.multimedia.cx/index.php?title=Bink_Audio | |
29 */ | |
30 | |
31 #include "avcodec.h" | |
32 #define ALT_BITSTREAM_READER_LE | |
33 #include "get_bits.h" | |
34 #include "dsputil.h" | |
35 extern const uint16_t ff_wma_critical_freqs[25]; | |
36 | |
37 #define MAX_CHANNELS 2 | |
38 #define BINK_BLOCK_MAX_SIZE (MAX_CHANNELS << 11) | |
39 | |
40 typedef struct { | |
41 AVCodecContext *avctx; | |
42 GetBitContext gb; | |
43 DSPContext dsp; | |
44 int first; | |
45 int channels; | |
46 int frame_len; ///< transform size (samples) | |
47 int overlap_len; ///< overlap size (samples) | |
48 int block_size; | |
49 int num_bands; | |
50 unsigned int *bands; | |
51 float root; | |
52 DECLARE_ALIGNED_16(FFTSample, coeffs[BINK_BLOCK_MAX_SIZE]); | |
53 DECLARE_ALIGNED_16(short, previous[BINK_BLOCK_MAX_SIZE / 16]); ///< coeffs from previous audio block | |
54 float *coeffs_ptr[MAX_CHANNELS]; ///< pointers to the coeffs arrays for float_to_int16_interleave | |
55 union { | |
56 RDFTContext rdft; | |
57 DCTContext dct; | |
58 } trans; | |
59 } BinkAudioContext; | |
60 | |
61 | |
62 static av_cold int decode_init(AVCodecContext *avctx) | |
63 { | |
64 BinkAudioContext *s = avctx->priv_data; | |
65 int sample_rate = avctx->sample_rate; | |
66 int sample_rate_half; | |
67 int i; | |
68 int frame_len_bits; | |
69 | |
70 s->avctx = avctx; | |
71 dsputil_init(&s->dsp, avctx); | |
72 | |
73 /* determine frame length */ | |
74 if (avctx->sample_rate < 22050) { | |
75 frame_len_bits = 9; | |
76 } else if (avctx->sample_rate < 44100) { | |
77 frame_len_bits = 10; | |
78 } else { | |
79 frame_len_bits = 11; | |
80 } | |
81 s->frame_len = 1 << frame_len_bits; | |
82 | |
83 if (s->channels > MAX_CHANNELS) { | |
84 av_log(s->avctx, AV_LOG_ERROR, "too many channels: %d\n", s->channels); | |
85 return -1; | |
86 } | |
87 | |
88 if (avctx->codec->id == CODEC_ID_BINKAUDIO_RDFT) { | |
89 // audio is already interleaved for the RDFT format variant | |
90 sample_rate *= avctx->channels; | |
91 s->frame_len *= avctx->channels; | |
92 s->channels = 1; | |
93 if (avctx->channels == 2) | |
94 frame_len_bits++; | |
95 } else { | |
96 s->channels = avctx->channels; | |
97 } | |
98 | |
99 s->overlap_len = s->frame_len / 16; | |
100 s->block_size = (s->frame_len - s->overlap_len) * s->channels; | |
101 sample_rate_half = (sample_rate + 1) / 2; | |
102 s->root = 2.0 / sqrt(s->frame_len); | |
103 | |
104 /* calculate number of bands */ | |
105 for (s->num_bands = 1; s->num_bands < 25; s->num_bands++) | |
106 if (sample_rate_half <= ff_wma_critical_freqs[s->num_bands - 1]) | |
107 break; | |
108 | |
109 s->bands = av_malloc((s->num_bands + 1) * sizeof(*s->bands)); | |
110 if (!s->bands) | |
111 return AVERROR(ENOMEM); | |
112 | |
113 /* populate bands data */ | |
114 s->bands[0] = 1; | |
115 for (i = 1; i < s->num_bands; i++) | |
116 s->bands[i] = ff_wma_critical_freqs[i - 1] * (s->frame_len / 2) / sample_rate_half; | |
117 s->bands[s->num_bands] = s->frame_len / 2; | |
118 | |
119 s->first = 1; | |
120 avctx->sample_fmt = SAMPLE_FMT_S16; | |
121 | |
122 for (i = 0; i < s->channels; i++) | |
123 s->coeffs_ptr[i] = s->coeffs + i * s->frame_len; | |
124 | |
125 if (avctx->codec->id == CODEC_ID_BINKAUDIO_RDFT) | |
126 ff_rdft_init(&s->trans.rdft, frame_len_bits, IRIDFT); | |
127 else | |
128 ff_dct_init(&s->trans.dct, frame_len_bits, 0); | |
129 | |
130 return 0; | |
131 } | |
132 | |
133 static float get_float(GetBitContext *gb) | |
134 { | |
135 int power = get_bits(gb, 5); | |
136 float f = ldexpf(get_bits_long(gb, 23), power - 23); | |
137 if (get_bits1(gb)) | |
138 f = -f; | |
139 return f; | |
140 } | |
141 | |
142 static const uint8_t rle_length_tab[16] = { | |
143 2, 3, 4, 5, 6, 8, 9, 10, 11, 12, 13, 14, 15, 16, 32, 64 | |
144 }; | |
145 | |
146 /** | |
147 * Decode Bink Audio block | |
148 * @param[out] out Output buffer (must contain s->block_size elements) | |
149 */ | |
150 static void decode_block(BinkAudioContext *s, short *out, int use_dct) | |
151 { | |
152 int ch, i, j, k; | |
153 float q, quant[25]; | |
154 int width, coeff; | |
155 GetBitContext *gb = &s->gb; | |
156 | |
157 if (use_dct) | |
158 skip_bits(gb, 2); | |
159 | |
160 for (ch = 0; ch < s->channels; ch++) { | |
161 FFTSample *coeffs = s->coeffs_ptr[ch]; | |
162 q = 0.0f; | |
163 coeffs[0] = get_float(gb) * s->root; | |
164 coeffs[1] = get_float(gb) * s->root; | |
165 | |
166 for (i = 0; i < s->num_bands; i++) { | |
167 /* constant is result of 0.066399999/log10(M_E) */ | |
168 int value = get_bits(gb, 8); | |
169 quant[i] = expf(FFMIN(value, 95) * 0.15289164787221953823f) * s->root; | |
170 } | |
171 | |
172 // find band (k) | |
173 for (k = 0; s->bands[k] < 1; k++) { | |
174 q = quant[k]; | |
175 } | |
176 | |
177 // parse coefficients | |
178 i = 2; | |
179 while (i < s->frame_len) { | |
180 if (get_bits1(gb)) { | |
181 j = i + rle_length_tab[get_bits(gb, 4)] * 8; | |
182 } else { | |
183 j = i + 8; | |
184 } | |
185 | |
186 j = FFMIN(j, s->frame_len); | |
187 | |
188 width = get_bits(gb, 4); | |
189 if (width == 0) { | |
190 memset(coeffs + i, 0, (j - i) * sizeof(*coeffs)); | |
191 i = j; | |
192 while (s->bands[k] * 2 < i) | |
193 q = quant[k++]; | |
194 } else { | |
195 while (i < j) { | |
196 if (s->bands[k] * 2 == i) | |
197 q = quant[k++]; | |
198 coeff = get_bits(gb, width); | |
199 if (coeff) { | |
200 if (get_bits1(gb)) | |
201 coeffs[i] = -q * coeff; | |
202 else | |
203 coeffs[i] = q * coeff; | |
204 } else { | |
205 coeffs[i] = 0.0f; | |
206 } | |
207 i++; | |
208 } | |
209 } | |
210 } | |
211 | |
212 if (use_dct) | |
213 ff_dct_calc (&s->trans.dct, coeffs); | |
214 else | |
215 ff_rdft_calc(&s->trans.rdft, coeffs); | |
216 } | |
217 | |
218 s->dsp.float_to_int16_interleave(out, (const float **)s->coeffs_ptr, s->frame_len, s->channels); | |
219 | |
220 if (!s->first) { | |
221 int count = s->overlap_len * s->channels; | |
222 int shift = av_log2(count); | |
223 for (i = 0; i < count; i++) { | |
224 out[i] = (s->previous[i] * (count - i) + out[i] * i) >> shift; | |
225 } | |
226 } | |
227 | |
228 memcpy(s->previous, out + s->block_size, | |
229 s->overlap_len * s->channels * sizeof(*out)); | |
230 | |
231 s->first = 0; | |
232 } | |
233 | |
234 static av_cold int decode_end(AVCodecContext *avctx) | |
235 { | |
236 BinkAudioContext * s = avctx->priv_data; | |
237 av_freep(&s->bands); | |
238 if (avctx->codec->id == CODEC_ID_BINKAUDIO_RDFT) | |
239 ff_rdft_end(&s->trans.rdft); | |
240 else | |
241 ff_dct_end(&s->trans.dct); | |
242 return 0; | |
243 } | |
244 | |
245 static void get_bits_align32(GetBitContext *s) | |
246 { | |
247 int n = (-get_bits_count(s)) & 31; | |
248 if (n) skip_bits(s, n); | |
249 } | |
250 | |
251 static int decode_frame(AVCodecContext *avctx, | |
252 void *data, int *data_size, | |
253 AVPacket *avpkt) | |
254 { | |
255 BinkAudioContext *s = avctx->priv_data; | |
256 const uint8_t *buf = avpkt->data; | |
257 int buf_size = avpkt->size; | |
258 short *samples = data; | |
259 short *samples_end = (short*)((uint8_t*)data + *data_size); | |
260 int reported_size; | |
261 GetBitContext *gb = &s->gb; | |
262 | |
263 init_get_bits(gb, buf, buf_size * 8); | |
264 | |
265 reported_size = get_bits_long(gb, 32); | |
266 while (get_bits_count(gb) / 8 < buf_size && | |
267 samples + s->block_size <= samples_end) { | |
268 decode_block(s, samples, avctx->codec->id == CODEC_ID_BINKAUDIO_DCT); | |
269 samples += s->block_size; | |
270 get_bits_align32(gb); | |
271 } | |
272 | |
273 *data_size = (uint8_t*)samples - (uint8_t*)data; | |
274 if (reported_size != *data_size) { | |
275 av_log(avctx, AV_LOG_WARNING, "reported data size (%d) does not match output data size (%d)\n", | |
276 reported_size, *data_size); | |
277 } | |
278 return buf_size; | |
279 } | |
280 | |
281 AVCodec binkaudio_rdft_decoder = { | |
282 "binkaudio_rdft", | |
283 CODEC_TYPE_AUDIO, | |
284 CODEC_ID_BINKAUDIO_RDFT, | |
285 sizeof(BinkAudioContext), | |
286 decode_init, | |
287 NULL, | |
288 decode_end, | |
289 decode_frame, | |
290 .long_name = NULL_IF_CONFIG_SMALL("Bink Audio (RDFT)") | |
291 }; | |
292 | |
293 AVCodec binkaudio_dct_decoder = { | |
294 "binkaudio_dct", | |
295 CODEC_TYPE_AUDIO, | |
296 CODEC_ID_BINKAUDIO_DCT, | |
297 sizeof(BinkAudioContext), | |
298 decode_init, | |
299 NULL, | |
300 decode_end, | |
301 decode_frame, | |
302 .long_name = NULL_IF_CONFIG_SMALL("Bink Audio (DCT)") | |
303 }; |