History

Bink Audio decoder

Originally committed as revision 21570 to svn://svn.ffmpeg.org/ffmpeg/trunk

Peter Ross committed 16 years ago

Commit c0d3f511 parent 2e375df7 files · +313 −2Browse files

Unified · Split

7 files
  1. Changelog +1 −1
  2. configure +2 −0
  3. doc/general.texi +2 −0
  4. libavcodec/Makefile +2 −0
  5. libavcodec/allcodecs.c +2 −0
  6. libavcodec/avcodec.h +1 −1
  7. libavcodec/binkaudio.c +303 −0
Changelog+1−1
@@ -51,7 +51,7 @@ version <next>: · 50 unmodified lines
51- SIPR decoder51- SIPR decoder
52- Adobe Filmstrip muxer and demuxer52- Adobe Filmstrip muxer and demuxer
53- RTP depacketization of H.26353- RTP depacketization of H.263
54- Bink demuxer54- Bink demuxer and Bink Audio decoder
55 55
56 56
57 57
configure+2−0
@@ -1165,6 +1165,8 @@ aac_encoder_select="fft mdct" · 1164 unmodified lines
1165ac3_decoder_select="fft mdct ac3_parser"1165ac3_decoder_select="fft mdct ac3_parser"
1166alac_encoder_select="lpc"1166alac_encoder_select="lpc"
1167atrac3_decoder_select="fft mdct"1167atrac3_decoder_select="fft mdct"
1168binkaudio_dct_decoder_select="fft mdct rdft dct"
1169binkaudio_rdft_decoder_select="fft mdct rdft"
1168cavs_decoder_select="golomb"1170cavs_decoder_select="golomb"
1169cook_decoder_select="fft mdct"1171cook_decoder_select="fft mdct"
1170cscd_decoder_suggest="zlib"1172cscd_decoder_suggest="zlib"
doc/general.texi+2−0
@@ -550,6 +550,8 @@ following image formats are supported: · 549 unmodified lines
550 @tab QuickTime fourcc 'alac'550 @tab QuickTime fourcc 'alac'
551@item Atrac 1 @tab @tab X551@item Atrac 1 @tab @tab X
552@item Atrac 3 @tab @tab X552@item Atrac 3 @tab @tab X
553@item Bink Audio @tab @tab X
554 @tab Used in Bink and Smacker files in many games.
553@item Delphine Software International CIN audio @tab @tab X555@item Delphine Software International CIN audio @tab @tab X
554 @tab Codec used in Delphine Software International games.556 @tab Codec used in Delphine Software International games.
555@item COOK @tab @tab X557@item COOK @tab @tab X
libavcodec/Makefile+2−0
@@ -65,6 +65,8 @@ OBJS-$(CONFIG_AURA2_DECODER) += aura.o · 64 unmodified lines
65OBJS-$(CONFIG_AVS_DECODER) += avs.o65OBJS-$(CONFIG_AVS_DECODER) += avs.o
66OBJS-$(CONFIG_BETHSOFTVID_DECODER) += bethsoftvideo.o66OBJS-$(CONFIG_BETHSOFTVID_DECODER) += bethsoftvideo.o
67OBJS-$(CONFIG_BFI_DECODER) += bfi.o67OBJS-$(CONFIG_BFI_DECODER) += bfi.o
68OBJS-$(CONFIG_BINKAUDIO_DCT_DECODER) += binkaudio.o wma.o
69OBJS-$(CONFIG_BINKAUDIO_RDFT_DECODER) += binkaudio.o wma.o
68OBJS-$(CONFIG_BMP_DECODER) += bmp.o msrledec.o70OBJS-$(CONFIG_BMP_DECODER) += bmp.o msrledec.o
69OBJS-$(CONFIG_BMP_ENCODER) += bmpenc.o71OBJS-$(CONFIG_BMP_ENCODER) += bmpenc.o
70OBJS-$(CONFIG_C93_DECODER) += c93.o72OBJS-$(CONFIG_C93_DECODER) += c93.o
libavcodec/allcodecs.c+2−0
@@ -213,6 +213,8 @@ void avcodec_register_all(void) · 212 unmodified lines
213 REGISTER_DECODER (APE, ape);213 REGISTER_DECODER (APE, ape);
214 REGISTER_DECODER (ATRAC1, atrac1);214 REGISTER_DECODER (ATRAC1, atrac1);
215 REGISTER_DECODER (ATRAC3, atrac3);215 REGISTER_DECODER (ATRAC3, atrac3);
216 REGISTER_DECODER (BINKAUDIO_DCT, binkaudio_dct);
217 REGISTER_DECODER (BINKAUDIO_RDFT, binkaudio_rdft);
216 REGISTER_DECODER (COOK, cook);218 REGISTER_DECODER (COOK, cook);
217 REGISTER_DECODER (DCA, dca);219 REGISTER_DECODER (DCA, dca);
218 REGISTER_DECODER (DSICINAUDIO, dsicinaudio);220 REGISTER_DECODER (DSICINAUDIO, dsicinaudio);
libavcodec/avcodec.h+1−1
@@ -30,7 +30,7 @@ · 29 unmodified lines
30#include "libavutil/avutil.h"30#include "libavutil/avutil.h"
31 31
32#define LIBAVCODEC_VERSION_MAJOR 5232#define LIBAVCODEC_VERSION_MAJOR 52
33#define LIBAVCODEC_VERSION_MINOR 4933#define LIBAVCODEC_VERSION_MINOR 50
34#define LIBAVCODEC_VERSION_MICRO 034#define LIBAVCODEC_VERSION_MICRO 0
35 35
36#define LIBAVCODEC_VERSION_INT AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \36#define LIBAVCODEC_VERSION_INT AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \
libavcodec/binkaudio.c+303−0
@@ -0,0 +1,303 @@
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"
35extern const uint16_t ff_wma_critical_freqs[25];
36
37#define MAX_CHANNELS 2
38#define BINK_BLOCK_MAX_SIZE (MAX_CHANNELS << 11)
39
40typedef 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
62static 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
133static 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
142static 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 */
150static 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
234static 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
245static void get_bits_align32(GetBitContext *s)
246{
247 int n = (-get_bits_count(s)) & 31;
248 if (n) skip_bits(s, n);
249}
250
251static 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
281AVCodec 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
293AVCodec 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};

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