Imported Debian version 2.5.3~trusty1
[deb_ffmpeg.git] / ffmpeg / libavcodec / binkaudio.c
1 /*
2 * Bink Audio decoder
3 * Copyright (c) 2007-2011 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
25 * Bink Audio decoder
26 *
27 * Technical details here:
28 * http://wiki.multimedia.cx/index.php?title=Bink_Audio
29 */
30
31 #include "libavutil/channel_layout.h"
32 #include "avcodec.h"
33 #define BITSTREAM_READER_LE
34 #include "get_bits.h"
35 #include "dct.h"
36 #include "rdft.h"
37 #include "fmtconvert.h"
38 #include "internal.h"
39 #include "wma_freqs.h"
40 #include "libavutil/intfloat.h"
41
42 static float quant_table[96];
43
44 #define MAX_CHANNELS 2
45 #define BINK_BLOCK_MAX_SIZE (MAX_CHANNELS << 11)
46
47 typedef struct {
48 GetBitContext gb;
49 int version_b; ///< Bink version 'b'
50 int first;
51 int channels;
52 int frame_len; ///< transform size (samples)
53 int overlap_len; ///< overlap size (samples)
54 int block_size;
55 int num_bands;
56 unsigned int *bands;
57 float root;
58 DECLARE_ALIGNED(32, FFTSample, coeffs)[BINK_BLOCK_MAX_SIZE];
59 float previous[MAX_CHANNELS][BINK_BLOCK_MAX_SIZE / 16]; ///< coeffs from previous audio block
60 uint8_t *packet_buffer;
61 union {
62 RDFTContext rdft;
63 DCTContext dct;
64 } trans;
65 } BinkAudioContext;
66
67
68 static av_cold int decode_init(AVCodecContext *avctx)
69 {
70 BinkAudioContext *s = avctx->priv_data;
71 int sample_rate = avctx->sample_rate;
72 int sample_rate_half;
73 int i;
74 int frame_len_bits;
75
76 /* determine frame length */
77 if (avctx->sample_rate < 22050) {
78 frame_len_bits = 9;
79 } else if (avctx->sample_rate < 44100) {
80 frame_len_bits = 10;
81 } else {
82 frame_len_bits = 11;
83 }
84
85 if (avctx->channels < 1 || avctx->channels > MAX_CHANNELS) {
86 av_log(avctx, AV_LOG_ERROR, "invalid number of channels: %d\n", avctx->channels);
87 return AVERROR_INVALIDDATA;
88 }
89 avctx->channel_layout = avctx->channels == 1 ? AV_CH_LAYOUT_MONO :
90 AV_CH_LAYOUT_STEREO;
91
92 s->version_b = avctx->extradata_size >= 4 && avctx->extradata[3] == 'b';
93
94 if (avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT) {
95 // audio is already interleaved for the RDFT format variant
96 avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
97 sample_rate *= avctx->channels;
98 s->channels = 1;
99 if (!s->version_b)
100 frame_len_bits += av_log2(avctx->channels);
101 } else {
102 s->channels = avctx->channels;
103 avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
104 }
105
106 s->frame_len = 1 << frame_len_bits;
107 s->overlap_len = s->frame_len / 16;
108 s->block_size = (s->frame_len - s->overlap_len) * s->channels;
109 sample_rate_half = (sample_rate + 1) / 2;
110 if (avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT)
111 s->root = 2.0 / (sqrt(s->frame_len) * 32768.0);
112 else
113 s->root = s->frame_len / (sqrt(s->frame_len) * 32768.0);
114 for (i = 0; i < 96; i++) {
115 /* constant is result of 0.066399999/log10(M_E) */
116 quant_table[i] = expf(i * 0.15289164787221953823f) * s->root;
117 }
118
119 /* calculate number of bands */
120 for (s->num_bands = 1; s->num_bands < 25; s->num_bands++)
121 if (sample_rate_half <= ff_wma_critical_freqs[s->num_bands - 1])
122 break;
123
124 s->bands = av_malloc((s->num_bands + 1) * sizeof(*s->bands));
125 if (!s->bands)
126 return AVERROR(ENOMEM);
127
128 /* populate bands data */
129 s->bands[0] = 2;
130 for (i = 1; i < s->num_bands; i++)
131 s->bands[i] = (ff_wma_critical_freqs[i - 1] * s->frame_len / sample_rate_half) & ~1;
132 s->bands[s->num_bands] = s->frame_len;
133
134 s->first = 1;
135
136 if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT)
137 ff_rdft_init(&s->trans.rdft, frame_len_bits, DFT_C2R);
138 else if (CONFIG_BINKAUDIO_DCT_DECODER)
139 ff_dct_init(&s->trans.dct, frame_len_bits, DCT_III);
140 else
141 return -1;
142
143 return 0;
144 }
145
146 static float get_float(GetBitContext *gb)
147 {
148 int power = get_bits(gb, 5);
149 float f = ldexpf(get_bits_long(gb, 23), power - 23);
150 if (get_bits1(gb))
151 f = -f;
152 return f;
153 }
154
155 static const uint8_t rle_length_tab[16] = {
156 2, 3, 4, 5, 6, 8, 9, 10, 11, 12, 13, 14, 15, 16, 32, 64
157 };
158
159 /**
160 * Decode Bink Audio block
161 * @param[out] out Output buffer (must contain s->block_size elements)
162 * @return 0 on success, negative error code on failure
163 */
164 static int decode_block(BinkAudioContext *s, float **out, int use_dct)
165 {
166 int ch, i, j, k;
167 float q, quant[25];
168 int width, coeff;
169 GetBitContext *gb = &s->gb;
170
171 if (use_dct)
172 skip_bits(gb, 2);
173
174 for (ch = 0; ch < s->channels; ch++) {
175 FFTSample *coeffs = out[ch];
176
177 if (s->version_b) {
178 if (get_bits_left(gb) < 64)
179 return AVERROR_INVALIDDATA;
180 coeffs[0] = av_int2float(get_bits_long(gb, 32)) * s->root;
181 coeffs[1] = av_int2float(get_bits_long(gb, 32)) * s->root;
182 } else {
183 if (get_bits_left(gb) < 58)
184 return AVERROR_INVALIDDATA;
185 coeffs[0] = get_float(gb) * s->root;
186 coeffs[1] = get_float(gb) * s->root;
187 }
188
189 if (get_bits_left(gb) < s->num_bands * 8)
190 return AVERROR_INVALIDDATA;
191 for (i = 0; i < s->num_bands; i++) {
192 int value = get_bits(gb, 8);
193 quant[i] = quant_table[FFMIN(value, 95)];
194 }
195
196 k = 0;
197 q = quant[0];
198
199 // parse coefficients
200 i = 2;
201 while (i < s->frame_len) {
202 if (s->version_b) {
203 j = i + 16;
204 } else {
205 int v = get_bits1(gb);
206 if (v) {
207 v = get_bits(gb, 4);
208 j = i + rle_length_tab[v] * 8;
209 } else {
210 j = i + 8;
211 }
212 }
213
214 j = FFMIN(j, s->frame_len);
215
216 width = get_bits(gb, 4);
217 if (width == 0) {
218 memset(coeffs + i, 0, (j - i) * sizeof(*coeffs));
219 i = j;
220 while (s->bands[k] < i)
221 q = quant[k++];
222 } else {
223 while (i < j) {
224 if (s->bands[k] == i)
225 q = quant[k++];
226 coeff = get_bits(gb, width);
227 if (coeff) {
228 int v;
229 v = get_bits1(gb);
230 if (v)
231 coeffs[i] = -q * coeff;
232 else
233 coeffs[i] = q * coeff;
234 } else {
235 coeffs[i] = 0.0f;
236 }
237 i++;
238 }
239 }
240 }
241
242 if (CONFIG_BINKAUDIO_DCT_DECODER && use_dct) {
243 coeffs[0] /= 0.5;
244 s->trans.dct.dct_calc(&s->trans.dct, coeffs);
245 }
246 else if (CONFIG_BINKAUDIO_RDFT_DECODER)
247 s->trans.rdft.rdft_calc(&s->trans.rdft, coeffs);
248 }
249
250 for (ch = 0; ch < s->channels; ch++) {
251 int j;
252 int count = s->overlap_len * s->channels;
253 if (!s->first) {
254 j = ch;
255 for (i = 0; i < s->overlap_len; i++, j += s->channels)
256 out[ch][i] = (s->previous[ch][i] * (count - j) +
257 out[ch][i] * j) / count;
258 }
259 memcpy(s->previous[ch], &out[ch][s->frame_len - s->overlap_len],
260 s->overlap_len * sizeof(*s->previous[ch]));
261 }
262
263 s->first = 0;
264
265 return 0;
266 }
267
268 static av_cold int decode_end(AVCodecContext *avctx)
269 {
270 BinkAudioContext * s = avctx->priv_data;
271 av_freep(&s->bands);
272 av_freep(&s->packet_buffer);
273 if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT)
274 ff_rdft_end(&s->trans.rdft);
275 else if (CONFIG_BINKAUDIO_DCT_DECODER)
276 ff_dct_end(&s->trans.dct);
277
278 return 0;
279 }
280
281 static void get_bits_align32(GetBitContext *s)
282 {
283 int n = (-get_bits_count(s)) & 31;
284 if (n) skip_bits(s, n);
285 }
286
287 static int decode_frame(AVCodecContext *avctx, void *data,
288 int *got_frame_ptr, AVPacket *avpkt)
289 {
290 BinkAudioContext *s = avctx->priv_data;
291 AVFrame *frame = data;
292 GetBitContext *gb = &s->gb;
293 int ret, consumed = 0;
294
295 if (!get_bits_left(gb)) {
296 uint8_t *buf;
297 /* handle end-of-stream */
298 if (!avpkt->size) {
299 *got_frame_ptr = 0;
300 return 0;
301 }
302 if (avpkt->size < 4) {
303 av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
304 return AVERROR_INVALIDDATA;
305 }
306 buf = av_realloc(s->packet_buffer, avpkt->size + FF_INPUT_BUFFER_PADDING_SIZE);
307 if (!buf)
308 return AVERROR(ENOMEM);
309 memset(buf + avpkt->size, 0, FF_INPUT_BUFFER_PADDING_SIZE);
310 s->packet_buffer = buf;
311 memcpy(s->packet_buffer, avpkt->data, avpkt->size);
312 if ((ret = init_get_bits8(gb, s->packet_buffer, avpkt->size)) < 0)
313 return ret;
314 consumed = avpkt->size;
315
316 /* skip reported size */
317 skip_bits_long(gb, 32);
318 }
319
320 /* get output buffer */
321 frame->nb_samples = s->frame_len;
322 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
323 return ret;
324
325 if (decode_block(s, (float **)frame->extended_data,
326 avctx->codec->id == AV_CODEC_ID_BINKAUDIO_DCT)) {
327 av_log(avctx, AV_LOG_ERROR, "Incomplete packet\n");
328 return AVERROR_INVALIDDATA;
329 }
330 get_bits_align32(gb);
331
332 frame->nb_samples = s->block_size / avctx->channels;
333 *got_frame_ptr = 1;
334
335 return consumed;
336 }
337
338 AVCodec ff_binkaudio_rdft_decoder = {
339 .name = "binkaudio_rdft",
340 .long_name = NULL_IF_CONFIG_SMALL("Bink Audio (RDFT)"),
341 .type = AVMEDIA_TYPE_AUDIO,
342 .id = AV_CODEC_ID_BINKAUDIO_RDFT,
343 .priv_data_size = sizeof(BinkAudioContext),
344 .init = decode_init,
345 .close = decode_end,
346 .decode = decode_frame,
347 .capabilities = CODEC_CAP_DELAY | CODEC_CAP_DR1,
348 };
349
350 AVCodec ff_binkaudio_dct_decoder = {
351 .name = "binkaudio_dct",
352 .long_name = NULL_IF_CONFIG_SMALL("Bink Audio (DCT)"),
353 .type = AVMEDIA_TYPE_AUDIO,
354 .id = AV_CODEC_ID_BINKAUDIO_DCT,
355 .priv_data_size = sizeof(BinkAudioContext),
356 .init = decode_init,
357 .close = decode_end,
358 .decode = decode_frame,
359 .capabilities = CODEC_CAP_DELAY | CODEC_CAP_DR1,
360 };