3 * Copyright (c) 2007-2011 Peter Ross (pross@xvid.org)
4 * Copyright (c) 2009 Daniel Verkamp (daniel@drv.nu)
6 * This file is part of Libav.
8 * Libav 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.
13 * Libav 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.
18 * You should have received a copy of the GNU Lesser General Public
19 * License along with Libav; if not, write to the Free Software
20 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
27 * Technical details here:
28 * http://wiki.multimedia.cx/index.php?title=Bink_Audio
32 #define ALT_BITSTREAM_READER_LE
37 #include "fmtconvert.h"
38 #include "libavutil/intfloat_readwrite.h"
40 extern const uint16_t ff_wma_critical_freqs[25];
42 #define MAX_CHANNELS 2
43 #define BINK_BLOCK_MAX_SIZE (MAX_CHANNELS << 11)
48 FmtConvertContext fmt_conv;
49 int version_b; ///< Bink version 'b'
52 int frame_len; ///< transform size (samples)
53 int overlap_len; ///< overlap size (samples)
58 DECLARE_ALIGNED(32, FFTSample, coeffs)[BINK_BLOCK_MAX_SIZE];
59 DECLARE_ALIGNED(16, short, previous)[BINK_BLOCK_MAX_SIZE / 16]; ///< coeffs from previous audio block
60 float *coeffs_ptr[MAX_CHANNELS]; ///< pointers to the coeffs arrays for float_to_int16_interleave
68 static av_cold int decode_init(AVCodecContext *avctx)
70 BinkAudioContext *s = avctx->priv_data;
71 int sample_rate = avctx->sample_rate;
76 dsputil_init(&s->dsp, avctx);
77 ff_fmt_convert_init(&s->fmt_conv, avctx);
79 /* determine frame length */
80 if (avctx->sample_rate < 22050) {
82 } else if (avctx->sample_rate < 44100) {
88 if (avctx->channels > MAX_CHANNELS) {
89 av_log(avctx, AV_LOG_ERROR, "too many channels: %d\n", avctx->channels);
93 s->version_b = avctx->extradata && avctx->extradata[3] == 'b';
95 if (avctx->codec->id == CODEC_ID_BINKAUDIO_RDFT) {
96 // audio is already interleaved for the RDFT format variant
97 sample_rate *= avctx->channels;
100 frame_len_bits += av_log2(avctx->channels);
102 s->channels = avctx->channels;
105 s->frame_len = 1 << frame_len_bits;
106 s->overlap_len = s->frame_len / 16;
107 s->block_size = (s->frame_len - s->overlap_len) * s->channels;
108 sample_rate_half = (sample_rate + 1) / 2;
109 s->root = 2.0 / sqrt(s->frame_len);
111 /* calculate number of bands */
112 for (s->num_bands = 1; s->num_bands < 25; s->num_bands++)
113 if (sample_rate_half <= ff_wma_critical_freqs[s->num_bands - 1])
116 s->bands = av_malloc((s->num_bands + 1) * sizeof(*s->bands));
118 return AVERROR(ENOMEM);
120 /* populate bands data */
122 for (i = 1; i < s->num_bands; i++)
123 s->bands[i] = (ff_wma_critical_freqs[i - 1] * s->frame_len / sample_rate_half) & ~1;
124 s->bands[s->num_bands] = s->frame_len;
127 avctx->sample_fmt = AV_SAMPLE_FMT_S16;
129 for (i = 0; i < s->channels; i++)
130 s->coeffs_ptr[i] = s->coeffs + i * s->frame_len;
132 if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == CODEC_ID_BINKAUDIO_RDFT)
133 ff_rdft_init(&s->trans.rdft, frame_len_bits, DFT_C2R);
134 else if (CONFIG_BINKAUDIO_DCT_DECODER)
135 ff_dct_init(&s->trans.dct, frame_len_bits, DCT_III);
142 static float get_float(GetBitContext *gb)
144 int power = get_bits(gb, 5);
145 float f = ldexpf(get_bits_long(gb, 23), power - 23);
151 static const uint8_t rle_length_tab[16] = {
152 2, 3, 4, 5, 6, 8, 9, 10, 11, 12, 13, 14, 15, 16, 32, 64
155 #define GET_BITS_SAFE(out, nbits) do { \
156 if (get_bits_left(gb) < nbits) \
157 return AVERROR_INVALIDDATA; \
158 out = get_bits(gb, nbits); \
162 * Decode Bink Audio block
163 * @param[out] out Output buffer (must contain s->block_size elements)
164 * @return 0 on success, negative error code on failure
166 static int decode_block(BinkAudioContext *s, short *out, int use_dct)
171 GetBitContext *gb = &s->gb;
176 for (ch = 0; ch < s->channels; ch++) {
177 FFTSample *coeffs = s->coeffs_ptr[ch];
179 if (get_bits_left(gb) < 64)
180 return AVERROR_INVALIDDATA;
181 coeffs[0] = av_int2flt(get_bits(gb, 32)) * s->root;
182 coeffs[1] = av_int2flt(get_bits(gb, 32)) * s->root;
184 if (get_bits_left(gb) < 58)
185 return AVERROR_INVALIDDATA;
186 coeffs[0] = get_float(gb) * s->root;
187 coeffs[1] = get_float(gb) * s->root;
190 if (get_bits_left(gb) < s->num_bands * 8)
191 return AVERROR_INVALIDDATA;
192 for (i = 0; i < s->num_bands; i++) {
193 /* constant is result of 0.066399999/log10(M_E) */
194 int value = get_bits(gb, 8);
195 quant[i] = expf(FFMIN(value, 95) * 0.15289164787221953823f) * s->root;
201 // parse coefficients
203 while (i < s->frame_len) {
211 j = i + rle_length_tab[v] * 8;
217 j = FFMIN(j, s->frame_len);
219 GET_BITS_SAFE(width, 4);
221 memset(coeffs + i, 0, (j - i) * sizeof(*coeffs));
223 while (s->bands[k] < i)
227 if (s->bands[k] == i)
229 GET_BITS_SAFE(coeff, width);
234 coeffs[i] = -q * coeff;
236 coeffs[i] = q * coeff;
245 if (CONFIG_BINKAUDIO_DCT_DECODER && use_dct) {
247 s->trans.dct.dct_calc(&s->trans.dct, coeffs);
248 s->dsp.vector_fmul_scalar(coeffs, coeffs, s->frame_len / 2, s->frame_len);
250 else if (CONFIG_BINKAUDIO_RDFT_DECODER)
251 s->trans.rdft.rdft_calc(&s->trans.rdft, coeffs);
254 s->fmt_conv.float_to_int16_interleave(out, (const float **)s->coeffs_ptr,
255 s->frame_len, s->channels);
258 int count = s->overlap_len * s->channels;
259 int shift = av_log2(count);
260 for (i = 0; i < count; i++) {
261 out[i] = (s->previous[i] * (count - i) + out[i] * i) >> shift;
265 memcpy(s->previous, out + s->block_size,
266 s->overlap_len * s->channels * sizeof(*out));
273 static av_cold int decode_end(AVCodecContext *avctx)
275 BinkAudioContext * s = avctx->priv_data;
277 if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == CODEC_ID_BINKAUDIO_RDFT)
278 ff_rdft_end(&s->trans.rdft);
279 else if (CONFIG_BINKAUDIO_DCT_DECODER)
280 ff_dct_end(&s->trans.dct);
284 static void get_bits_align32(GetBitContext *s)
286 int n = (-get_bits_count(s)) & 31;
287 if (n) skip_bits(s, n);
290 static int decode_frame(AVCodecContext *avctx,
291 void *data, int *data_size,
294 BinkAudioContext *s = avctx->priv_data;
295 const uint8_t *buf = avpkt->data;
296 int buf_size = avpkt->size;
297 short *samples = data;
298 short *samples_end = (short*)((uint8_t*)data + *data_size);
300 GetBitContext *gb = &s->gb;
303 av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
304 return AVERROR_INVALIDDATA;
307 init_get_bits(gb, buf, buf_size * 8);
309 reported_size = get_bits_long(gb, 32);
310 while (samples + s->block_size <= samples_end) {
311 if (decode_block(s, samples, avctx->codec->id == CODEC_ID_BINKAUDIO_DCT))
313 samples += s->block_size;
314 get_bits_align32(gb);
317 *data_size = FFMIN(reported_size, (uint8_t*)samples - (uint8_t*)data);
321 AVCodec ff_binkaudio_rdft_decoder = {
322 .name = "binkaudio_rdft",
323 .type = AVMEDIA_TYPE_AUDIO,
324 .id = CODEC_ID_BINKAUDIO_RDFT,
325 .priv_data_size = sizeof(BinkAudioContext),
328 .decode = decode_frame,
329 .long_name = NULL_IF_CONFIG_SMALL("Bink Audio (RDFT)")
332 AVCodec ff_binkaudio_dct_decoder = {
333 .name = "binkaudio_dct",
334 .type = AVMEDIA_TYPE_AUDIO,
335 .id = CODEC_ID_BINKAUDIO_DCT,
336 .priv_data_size = sizeof(BinkAudioContext),
339 .decode = decode_frame,
340 .long_name = NULL_IF_CONFIG_SMALL("Bink Audio (DCT)")