3 * Copyright (c) 2007-2010 Peter Ross (pross@xvid.org)
4 * Copyright (c) 2009 Daniel Verkamp (daniel@drv.nu)
6 * This file is part of FFmpeg.
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.
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.
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
27 * Technical details here:
28 * http://wiki.multimedia.cx/index.php?title=Bink_Audio
32 #define ALT_BITSTREAM_READER_LE
36 #include "fmtconvert.h"
38 extern const uint16_t ff_wma_critical_freqs[25];
40 #define MAX_CHANNELS 2
41 #define BINK_BLOCK_MAX_SIZE (MAX_CHANNELS << 11)
46 FmtConvertContext fmt_conv;
49 int frame_len; ///< transform size (samples)
50 int overlap_len; ///< overlap size (samples)
55 DECLARE_ALIGNED(16, FFTSample, coeffs)[BINK_BLOCK_MAX_SIZE];
56 DECLARE_ALIGNED(16, short, previous)[BINK_BLOCK_MAX_SIZE / 16]; ///< coeffs from previous audio block
57 float *coeffs_ptr[MAX_CHANNELS]; ///< pointers to the coeffs arrays for float_to_int16_interleave
65 static av_cold int decode_init(AVCodecContext *avctx)
67 BinkAudioContext *s = avctx->priv_data;
68 int sample_rate = avctx->sample_rate;
73 dsputil_init(&s->dsp, avctx);
74 ff_fmt_convert_init(&s->fmt_conv, avctx);
76 /* determine frame length */
77 if (avctx->sample_rate < 22050) {
79 } else if (avctx->sample_rate < 44100) {
84 s->frame_len = 1 << frame_len_bits;
86 if (avctx->channels > MAX_CHANNELS) {
87 av_log(avctx, AV_LOG_ERROR, "too many channels: %d\n", avctx->channels);
91 if (avctx->codec->id == CODEC_ID_BINKAUDIO_RDFT) {
92 // audio is already interleaved for the RDFT format variant
93 sample_rate *= avctx->channels;
94 s->frame_len *= avctx->channels;
96 if (avctx->channels == 2)
99 s->channels = avctx->channels;
102 s->overlap_len = s->frame_len / 16;
103 s->block_size = (s->frame_len - s->overlap_len) * s->channels;
104 sample_rate_half = (sample_rate + 1) / 2;
105 s->root = 2.0 / sqrt(s->frame_len);
107 /* calculate number of bands */
108 for (s->num_bands = 1; s->num_bands < 25; s->num_bands++)
109 if (sample_rate_half <= ff_wma_critical_freqs[s->num_bands - 1])
112 s->bands = av_malloc((s->num_bands + 1) * sizeof(*s->bands));
114 return AVERROR(ENOMEM);
116 /* populate bands data */
118 for (i = 1; i < s->num_bands; i++)
119 s->bands[i] = (ff_wma_critical_freqs[i - 1] * s->frame_len / sample_rate_half) & ~1;
120 s->bands[s->num_bands] = s->frame_len;
123 avctx->sample_fmt = AV_SAMPLE_FMT_S16;
125 for (i = 0; i < s->channels; i++)
126 s->coeffs_ptr[i] = s->coeffs + i * s->frame_len;
128 if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == CODEC_ID_BINKAUDIO_RDFT)
129 ff_rdft_init(&s->trans.rdft, frame_len_bits, DFT_C2R);
130 else if (CONFIG_BINKAUDIO_DCT_DECODER)
131 ff_dct_init(&s->trans.dct, frame_len_bits, DCT_III);
138 static float get_float(GetBitContext *gb)
140 int power = get_bits(gb, 5);
141 float f = ldexpf(get_bits_long(gb, 23), power - 23);
147 static const uint8_t rle_length_tab[16] = {
148 2, 3, 4, 5, 6, 8, 9, 10, 11, 12, 13, 14, 15, 16, 32, 64
152 * Decode Bink Audio block
153 * @param[out] out Output buffer (must contain s->block_size elements)
155 static void decode_block(BinkAudioContext *s, short *out, int use_dct)
160 GetBitContext *gb = &s->gb;
165 for (ch = 0; ch < s->channels; ch++) {
166 FFTSample *coeffs = s->coeffs_ptr[ch];
167 coeffs[0] = get_float(gb) * s->root;
168 coeffs[1] = get_float(gb) * s->root;
170 for (i = 0; i < s->num_bands; i++) {
171 /* constant is result of 0.066399999/log10(M_E) */
172 int value = get_bits(gb, 8);
173 quant[i] = expf(FFMIN(value, 95) * 0.15289164787221953823f) * s->root;
179 // parse coefficients
181 while (i < s->frame_len) {
183 j = i + rle_length_tab[get_bits(gb, 4)] * 8;
188 j = FFMIN(j, s->frame_len);
190 width = get_bits(gb, 4);
192 memset(coeffs + i, 0, (j - i) * sizeof(*coeffs));
194 while (s->bands[k] < i)
198 if (s->bands[k] == i)
200 coeff = get_bits(gb, width);
203 coeffs[i] = -q * coeff;
205 coeffs[i] = q * coeff;
214 if (CONFIG_BINKAUDIO_DCT_DECODER && use_dct) {
216 ff_dct_calc (&s->trans.dct, coeffs);
217 s->dsp.vector_fmul_scalar(coeffs, coeffs, s->frame_len / 2, s->frame_len);
219 else if (CONFIG_BINKAUDIO_RDFT_DECODER)
220 ff_rdft_calc(&s->trans.rdft, coeffs);
223 s->fmt_conv.float_to_int16_interleave(out, (const float **)s->coeffs_ptr,
224 s->frame_len, s->channels);
227 int count = s->overlap_len * s->channels;
228 int shift = av_log2(count);
229 for (i = 0; i < count; i++) {
230 out[i] = (s->previous[i] * (count - i) + out[i] * i) >> shift;
234 memcpy(s->previous, out + s->block_size,
235 s->overlap_len * s->channels * sizeof(*out));
240 static av_cold int decode_end(AVCodecContext *avctx)
242 BinkAudioContext * s = avctx->priv_data;
244 if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == CODEC_ID_BINKAUDIO_RDFT)
245 ff_rdft_end(&s->trans.rdft);
246 else if (CONFIG_BINKAUDIO_DCT_DECODER)
247 ff_dct_end(&s->trans.dct);
251 static void get_bits_align32(GetBitContext *s)
253 int n = (-get_bits_count(s)) & 31;
254 if (n) skip_bits(s, n);
257 static int decode_frame(AVCodecContext *avctx,
258 void *data, int *data_size,
261 BinkAudioContext *s = avctx->priv_data;
262 const uint8_t *buf = avpkt->data;
263 int buf_size = avpkt->size;
264 short *samples = data;
265 short *samples_end = (short*)((uint8_t*)data + *data_size);
267 GetBitContext *gb = &s->gb;
269 init_get_bits(gb, buf, buf_size * 8);
271 reported_size = get_bits_long(gb, 32);
272 while (get_bits_count(gb) / 8 < buf_size &&
273 samples + s->block_size <= samples_end) {
274 decode_block(s, samples, avctx->codec->id == CODEC_ID_BINKAUDIO_DCT);
275 samples += s->block_size;
276 get_bits_align32(gb);
279 *data_size = FFMIN(reported_size, (uint8_t*)samples - (uint8_t*)data);
283 AVCodec ff_binkaudio_rdft_decoder = {
286 CODEC_ID_BINKAUDIO_RDFT,
287 sizeof(BinkAudioContext),
292 .long_name = NULL_IF_CONFIG_SMALL("Bink Audio (RDFT)")
295 AVCodec ff_binkaudio_dct_decoder = {
298 CODEC_ID_BINKAUDIO_DCT,
299 sizeof(BinkAudioContext),
304 .long_name = NULL_IF_CONFIG_SMALL("Bink Audio (DCT)")