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
31 #include "libavutil/channel_layout.h"
32 #include "libavutil/intfloat.h"
34 #define BITSTREAM_READER_LE
36 #include "bitstream.h"
41 #include "wma_freqs.h"
43 static float quant_table[96];
45 #define MAX_CHANNELS 2
46 #define BINK_BLOCK_MAX_SIZE (MAX_CHANNELS << 11)
48 typedef struct BinkAudioContext {
50 int version_b; ///< Bink version 'b'
53 int frame_len; ///< transform size (samples)
54 int overlap_len; ///< overlap size (samples)
59 DECLARE_ALIGNED(32, FFTSample, coeffs)[BINK_BLOCK_MAX_SIZE];
60 float previous[MAX_CHANNELS][BINK_BLOCK_MAX_SIZE / 16]; ///< coeffs from previous audio block
69 static av_cold int decode_init(AVCodecContext *avctx)
71 BinkAudioContext *s = avctx->priv_data;
72 int sample_rate = avctx->sample_rate;
77 /* determine frame length */
78 if (avctx->sample_rate < 22050) {
80 } else if (avctx->sample_rate < 44100) {
86 if (avctx->channels > MAX_CHANNELS) {
87 av_log(avctx, AV_LOG_ERROR, "too many channels: %d\n", avctx->channels);
90 avctx->channel_layout = avctx->channels == 1 ? AV_CH_LAYOUT_MONO :
93 s->version_b = avctx->extradata && avctx->extradata[3] == 'b';
95 if (avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT) {
96 // audio is already interleaved for the RDFT format variant
97 avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
98 sample_rate *= avctx->channels;
101 frame_len_bits += av_log2(avctx->channels);
103 s->channels = avctx->channels;
104 avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
107 s->frame_len = 1 << frame_len_bits;
108 s->overlap_len = s->frame_len / 16;
109 s->block_size = (s->frame_len - s->overlap_len) * s->channels;
110 sample_rate_half = (sample_rate + 1) / 2;
111 if (avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT)
112 s->root = 2.0 / (sqrt(s->frame_len) * 32768.0);
114 s->root = s->frame_len / (sqrt(s->frame_len) * 32768.0);
115 for (i = 0; i < 96; i++) {
116 /* constant is result of 0.066399999/log10(M_E) */
117 quant_table[i] = expf(i * 0.15289164787221953823f) * s->root;
120 /* calculate number of bands */
121 for (s->num_bands = 1; s->num_bands < 25; s->num_bands++)
122 if (sample_rate_half <= ff_wma_critical_freqs[s->num_bands - 1])
125 s->bands = av_malloc((s->num_bands + 1) * sizeof(*s->bands));
127 return AVERROR(ENOMEM);
129 /* populate bands data */
131 for (i = 1; i < s->num_bands; i++)
132 s->bands[i] = (ff_wma_critical_freqs[i - 1] * s->frame_len / sample_rate_half) & ~1;
133 s->bands[s->num_bands] = s->frame_len;
137 if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT)
138 ff_rdft_init(&s->trans.rdft, frame_len_bits, DFT_C2R);
139 else if (CONFIG_BINKAUDIO_DCT_DECODER)
140 ff_dct_init(&s->trans.dct, frame_len_bits, DCT_III);
144 s->pkt = av_packet_alloc();
146 return AVERROR(ENOMEM);
151 static float get_float(BitstreamContext *bc)
153 int power = bitstream_read(bc, 5);
154 float f = ldexpf(bitstream_read(bc, 23), power - 23);
155 if (bitstream_read_bit(bc))
160 static const uint8_t rle_length_tab[16] = {
161 2, 3, 4, 5, 6, 8, 9, 10, 11, 12, 13, 14, 15, 16, 32, 64
165 * Decode Bink Audio block
166 * @param[out] out Output buffer (must contain s->block_size elements)
167 * @return 0 on success, negative error code on failure
169 static int decode_block(BinkAudioContext *s, float **out, int use_dct)
174 BitstreamContext *bc = &s->bc;
177 bitstream_skip(bc, 2);
179 for (ch = 0; ch < s->channels; ch++) {
180 FFTSample *coeffs = out[ch];
183 if (bitstream_bits_left(bc) < 64)
184 return AVERROR_INVALIDDATA;
185 coeffs[0] = av_int2float(bitstream_read(bc, 32)) * s->root;
186 coeffs[1] = av_int2float(bitstream_read(bc, 32)) * s->root;
188 if (bitstream_bits_left(bc) < 58)
189 return AVERROR_INVALIDDATA;
190 coeffs[0] = get_float(bc) * s->root;
191 coeffs[1] = get_float(bc) * s->root;
194 if (bitstream_bits_left(bc) < s->num_bands * 8)
195 return AVERROR_INVALIDDATA;
196 for (i = 0; i < s->num_bands; i++) {
197 int value = bitstream_read(bc, 8);
198 quant[i] = quant_table[FFMIN(value, 95)];
204 // parse coefficients
206 while (i < s->frame_len) {
210 int v = bitstream_read_bit(bc);
212 v = bitstream_read(bc, 4);
213 j = i + rle_length_tab[v] * 8;
219 j = FFMIN(j, s->frame_len);
221 width = bitstream_read(bc, 4);
223 memset(coeffs + i, 0, (j - i) * sizeof(*coeffs));
225 while (s->bands[k] < i)
229 if (s->bands[k] == i)
231 coeff = bitstream_read(bc, width);
234 v = bitstream_read_bit(bc);
236 coeffs[i] = -q * coeff;
238 coeffs[i] = q * coeff;
247 if (CONFIG_BINKAUDIO_DCT_DECODER && use_dct) {
249 s->trans.dct.dct_calc(&s->trans.dct, coeffs);
251 else if (CONFIG_BINKAUDIO_RDFT_DECODER)
252 s->trans.rdft.rdft_calc(&s->trans.rdft, coeffs);
255 for (ch = 0; ch < s->channels; ch++) {
257 int count = s->overlap_len * s->channels;
260 for (i = 0; i < s->overlap_len; i++, j += s->channels)
261 out[ch][i] = (s->previous[ch][i] * (count - j) +
262 out[ch][i] * j) / count;
264 memcpy(s->previous[ch], &out[ch][s->frame_len - s->overlap_len],
265 s->overlap_len * sizeof(*s->previous[ch]));
273 static av_cold int decode_end(AVCodecContext *avctx)
275 BinkAudioContext * s = avctx->priv_data;
277 if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == AV_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);
282 av_packet_free(&s->pkt);
287 static void get_bits_align32(BitstreamContext *s)
289 int n = (-bitstream_tell(s)) & 31;
291 bitstream_skip(s, n);
294 static int binkaudio_receive_frame(AVCodecContext *avctx, AVFrame *frame)
296 BinkAudioContext *s = avctx->priv_data;
297 BitstreamContext *bc = &s->bc;
301 ret = ff_decode_get_packet(avctx, s->pkt);
305 if (s->pkt->size < 4) {
306 av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
307 ret = AVERROR_INVALIDDATA;
311 ret = bitstream_init8(bc, s->pkt->data, s->pkt->size);
315 /* skip reported size */
316 bitstream_skip(bc, 32);
319 /* get output buffer */
320 frame->nb_samples = s->frame_len;
321 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
322 av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
326 if (decode_block(s, (float **)frame->extended_data,
327 avctx->codec->id == AV_CODEC_ID_BINKAUDIO_DCT)) {
328 av_log(avctx, AV_LOG_ERROR, "Incomplete packet\n");
329 return AVERROR_INVALIDDATA;
331 get_bits_align32(bc);
332 if (!bitstream_bits_left(bc)) {
333 memset(bc, 0, sizeof(*bc));
334 av_packet_unref(s->pkt);
337 frame->nb_samples = s->block_size / avctx->channels;
341 av_packet_unref(s->pkt);
345 AVCodec ff_binkaudio_rdft_decoder = {
346 .name = "binkaudio_rdft",
347 .long_name = NULL_IF_CONFIG_SMALL("Bink Audio (RDFT)"),
348 .type = AVMEDIA_TYPE_AUDIO,
349 .id = AV_CODEC_ID_BINKAUDIO_RDFT,
350 .priv_data_size = sizeof(BinkAudioContext),
353 .receive_frame = binkaudio_receive_frame,
354 .capabilities = AV_CODEC_CAP_DELAY | AV_CODEC_CAP_DR1,
357 AVCodec ff_binkaudio_dct_decoder = {
358 .name = "binkaudio_dct",
359 .long_name = NULL_IF_CONFIG_SMALL("Bink Audio (DCT)"),
360 .type = AVMEDIA_TYPE_AUDIO,
361 .id = AV_CODEC_ID_BINKAUDIO_DCT,
362 .priv_data_size = sizeof(BinkAudioContext),
365 .receive_frame = binkaudio_receive_frame,
366 .capabilities = AV_CODEC_CAP_DELAY | AV_CODEC_CAP_DR1,