Browse code
lavc: Add DSS SP decoder
Signed-off-by: Oleksij Rempel <linux@rempel-privat.de>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
Oleksij Rempel authored on 2015/02/13 16:36:16Showing 8 changed files
- Changelog
- doc/general.texi
- libavcodec/Makefile
- libavcodec/allcodecs.c
- libavcodec/avcodec.h
- libavcodec/codec_desc.c
- libavcodec/dss_sp.c
- libavcodec/version.h
| ... | ... |
@@ -820,6 +820,7 @@ following image formats are supported: |
| 820 | 820 |
@tab Used in Bink and Smacker files in many games. |
| 821 | 821 |
@item Delphine Software International CIN audio @tab @tab X |
| 822 | 822 |
@tab Codec used in Delphine Software International games. |
| 823 |
+@item Digital Speech Standard - Standard Play mode (DSS SP) @tab @tab X |
|
| 823 | 824 |
@item Discworld II BMV Audio @tab @tab X |
| 824 | 825 |
@item COOK @tab @tab X |
| 825 | 826 |
@tab All versions except 5.1 are supported. |
| ... | ... |
@@ -169,6 +169,7 @@ OBJS-$(CONFIG_DPX_DECODER) += dpx.o |
| 169 | 169 |
OBJS-$(CONFIG_DPX_ENCODER) += dpxenc.o |
| 170 | 170 |
OBJS-$(CONFIG_DSICINAUDIO_DECODER) += dsicinaudio.o |
| 171 | 171 |
OBJS-$(CONFIG_DSICINVIDEO_DECODER) += dsicinvideo.o |
| 172 |
+OBJS-$(CONFIG_DSS_SP_DECODER) += dss_sp.o |
|
| 172 | 173 |
OBJS-$(CONFIG_DVBSUB_DECODER) += dvbsubdec.o |
| 173 | 174 |
OBJS-$(CONFIG_DVBSUB_ENCODER) += dvbsub.o |
| 174 | 175 |
OBJS-$(CONFIG_DVDSUB_DECODER) += dvdsubdec.o |
| ... | ... |
@@ -308,6 +308,7 @@ void avcodec_register_all(void) |
| 308 | 308 |
REGISTER_DECODER(COOK, cook); |
| 309 | 309 |
REGISTER_DECODER(DCA, dca); |
| 310 | 310 |
REGISTER_DECODER(DSICINAUDIO, dsicinaudio); |
| 311 |
+ REGISTER_DECODER(DSS_SP, dss_sp); |
|
| 311 | 312 |
REGISTER_ENCDEC (EAC3, eac3); |
| 312 | 313 |
REGISTER_ENCDEC (FLAC, flac); |
| 313 | 314 |
REGISTER_DECODER(G723_1, g723_1); |
| ... | ... |
@@ -444,6 +444,7 @@ enum AVCodecID {
|
| 444 | 444 |
AV_CODEC_ID_METASOUND, |
| 445 | 445 |
AV_CODEC_ID_PAF_AUDIO, |
| 446 | 446 |
AV_CODEC_ID_ON2AVC, |
| 447 |
+ AV_CODEC_ID_DSS_SP, |
|
| 447 | 448 |
|
| 448 | 449 |
/* subtitle codecs */ |
| 449 | 450 |
AV_CODEC_ID_FIRST_SUBTITLE = 0x17000, ///< A dummy ID pointing at the start of subtitle codecs. |
| ... | ... |
@@ -2202,6 +2202,13 @@ static const AVCodecDescriptor codec_descriptors[] = {
|
| 2202 | 2202 |
.props = AV_CODEC_PROP_LOSSY, |
| 2203 | 2203 |
}, |
| 2204 | 2204 |
{
|
| 2205 |
+ .id = AV_CODEC_ID_DSS_SP, |
|
| 2206 |
+ .type = AVMEDIA_TYPE_AUDIO, |
|
| 2207 |
+ .name = "dss_sp", |
|
| 2208 |
+ .long_name = NULL_IF_CONFIG_SMALL("Digital Speech Standard - Standard Play mode (DSS SP)"),
|
|
| 2209 |
+ .props = AV_CODEC_PROP_LOSSY, |
|
| 2210 |
+ }, |
|
| 2211 |
+ {
|
|
| 2205 | 2212 |
.id = AV_CODEC_ID_G729, |
| 2206 | 2213 |
.type = AVMEDIA_TYPE_AUDIO, |
| 2207 | 2214 |
.name = "g729", |
| 2208 | 2215 |
new file mode 100644 |
| ... | ... |
@@ -0,0 +1,780 @@ |
| 0 |
+/* |
|
| 1 |
+ * Digital Speech Standard - Standard Play mode (DSS SP) audio decoder. |
|
| 2 |
+ * Copyright (C) 2014 Oleksij Rempel <linux@rempel-privat.de> |
|
| 3 |
+ * |
|
| 4 |
+ * This file is part of Libav. |
|
| 5 |
+ * |
|
| 6 |
+ * Libav is free software; you can redistribute it and/or |
|
| 7 |
+ * modify it under the terms of the GNU Lesser General Public |
|
| 8 |
+ * License as published by the Free Software Foundation; either |
|
| 9 |
+ * version 2.1 of the License, or (at your option) any later version. |
|
| 10 |
+ * |
|
| 11 |
+ * Libav is distributed in the hope that it will be useful, |
|
| 12 |
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of |
|
| 13 |
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
|
| 14 |
+ * Lesser General Public License for more details. |
|
| 15 |
+ * |
|
| 16 |
+ * You should have received a copy of the GNU Lesser General Public |
|
| 17 |
+ * License along with Libav; if not, write to the Free Software |
|
| 18 |
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
|
| 19 |
+ */ |
|
| 20 |
+ |
|
| 21 |
+#include "libavutil/channel_layout.h" |
|
| 22 |
+#include "libavutil/common.h" |
|
| 23 |
+#include "libavutil/mem.h" |
|
| 24 |
+#include "libavutil/opt.h" |
|
| 25 |
+ |
|
| 26 |
+#include "avcodec.h" |
|
| 27 |
+#include "get_bits.h" |
|
| 28 |
+#include "internal.h" |
|
| 29 |
+ |
|
| 30 |
+#define SUBFRAMES 4 |
|
| 31 |
+#define PULSE_MAX 8 |
|
| 32 |
+ |
|
| 33 |
+#define DSS_SP_FRAME_SIZE 42 |
|
| 34 |
+#define DSS_SP_SAMPLE_COUNT (66 * SUBFRAMES) |
|
| 35 |
+#define DSS_SP_FORMULA(a, b, c) ((((a) << 15) + (b) * (c)) + 0x4000) >> 15 |
|
| 36 |
+ |
|
| 37 |
+typedef struct DssSpSubframe {
|
|
| 38 |
+ int16_t gain; |
|
| 39 |
+ int32_t combined_pulse_pos; |
|
| 40 |
+ int16_t pulse_pos[7]; |
|
| 41 |
+ int16_t pulse_val[7]; |
|
| 42 |
+} DssSpSubframe; |
|
| 43 |
+ |
|
| 44 |
+typedef struct DssSpFrame {
|
|
| 45 |
+ int16_t filter_idx[14]; |
|
| 46 |
+ int16_t sf_adaptive_gain[SUBFRAMES]; |
|
| 47 |
+ int16_t pitch_lag[SUBFRAMES]; |
|
| 48 |
+ struct DssSpSubframe sf[SUBFRAMES]; |
|
| 49 |
+} DssSpFrame; |
|
| 50 |
+ |
|
| 51 |
+typedef struct DssSpContext {
|
|
| 52 |
+ int32_t excitation[288 + 6]; |
|
| 53 |
+ int32_t history[187]; |
|
| 54 |
+ DssSpFrame fparam; |
|
| 55 |
+ int32_t working_buffer[SUBFRAMES][72]; |
|
| 56 |
+ int32_t audio_buf[15]; |
|
| 57 |
+ int32_t err_buf1[15]; |
|
| 58 |
+ int32_t lpc_filter[14]; |
|
| 59 |
+ int32_t filter[15]; |
|
| 60 |
+ int32_t vector_buf[72]; |
|
| 61 |
+ int noise_state; |
|
| 62 |
+ int32_t err_buf2[15]; |
|
| 63 |
+ |
|
| 64 |
+ int pulse_dec_mode; |
|
| 65 |
+ |
|
| 66 |
+ DECLARE_ALIGNED(16, uint8_t, bits)[DSS_SP_FRAME_SIZE + |
|
| 67 |
+ FF_INPUT_BUFFER_PADDING_SIZE]; |
|
| 68 |
+} DssSpContext; |
|
| 69 |
+ |
|
| 70 |
+/* |
|
| 71 |
+ * Used for the coding/decoding of the pulse positions for the MP-MLQ codebook. |
|
| 72 |
+ */ |
|
| 73 |
+static const uint32_t dss_sp_combinatorial_table[PULSE_MAX][72] = {
|
|
| 74 |
+ { 0, 0, 0, 0, 0, 0,
|
|
| 75 |
+ 0, 0, 0, 0, 0, 0, |
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+ 0, 0, 0, 0, 0, 0, |
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| 81 |
+ 0, 0, 0, 0, 0, 0, |
|
| 82 |
+ 0, 0, 0, 0, 0, 0, |
|
| 83 |
+ 0, 0, 0, 0, 0, 0, |
|
| 84 |
+ 0, 0, 0, 0, 0, 0, |
|
| 85 |
+ 0, 0, 0, 0, 0, 0 }, |
|
| 86 |
+ { 0, 1, 2, 3, 4, 5,
|
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| 87 |
+ 6, 7, 8, 9, 10, 11, |
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|
| 170 |
+}; |
|
| 171 |
+ |
|
| 172 |
+static const int16_t dss_sp_filter_cb[14][32] = {
|
|
| 173 |
+ { -32653, -32587, -32515, -32438, -32341, -32216, -32062, -31881,
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+ -2834, 1819, 6544, 11260, 16050, 20220, 24774, 28120 }, |
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+ |
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+ { -27503, -24509, -20644, -17496, -14187, -11277, -8420, -5595,
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+ -3013, -624, 1711, 3880, 5844, 7774, 9739, 11592, |
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+ 24142, 25249, 26275, 27300, 28359, 29249, 30118, 31183 }, |
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+ |
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+ { -27827, -24208, -20943, -17781, -14843, -11848, -9066, -6297,
|
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+ -3660, -910, 1918, 5025, 8223, 11649, 15086, 18423, |
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+ 0, 0, 0, 0, 0, 0, 0, 0, |
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+ |
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+ 0, 0, 0, 0, 0, 0, 0, 0 }, |
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+ |
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+ { -11099, -7014, -3855, -1025, 1680, 4544, 7807, 11932,
|
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+ 0, 0, 0, 0, 0, 0, 0, 0, |
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+ 0, 0, 0, 0, 0, 0, 0, 0, |
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+ 0, 0, 0, 0, 0, 0, 0, 0 }, |
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+ |
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+ { -9060, -4570, -1381, 1419, 4034, 6728, 9865, 14149,
|
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+ 0, 0, 0, 0, 0, 0, 0, 0, |
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+ 0, 0, 0, 0, 0, 0, 0, 0, |
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+ 0, 0, 0, 0, 0, 0, 0, 0 }, |
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+ |
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+ { -12450, -7985, -4596, -1734, 961, 3629, 6865, 11142,
|
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+ 0, 0, 0, 0, 0, 0, 0, 0, |
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+ 0, 0, 0, 0, 0, 0, 0, 0, |
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+ 0, 0, 0, 0, 0, 0, 0, 0 }, |
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+ |
|
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+ { -11831, -7404, -4010, -1096, 1606, 4291, 7386, 11482,
|
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+ 0, 0, 0, 0, 0, 0, 0, 0, |
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+ 0, 0, 0, 0, 0, 0, 0, 0, |
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+ 0, 0, 0, 0, 0, 0, 0, 0 }, |
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+ |
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+ { -13404, -9250, -5995, -3312, -890, 1594, 4464, 8198,
|
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+ 0, 0, 0, 0, 0, 0, 0, 0, |
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+ 0, 0, 0, 0, 0, 0, 0, 0, |
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+ 0, 0, 0, 0, 0, 0, 0, 0 }, |
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+ |
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+ { -11239, -7220, -4040, -1406, 971, 3321, 6006, 9697,
|
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+ 0, 0, 0, 0, 0, 0, 0, 0, |
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+ 0, 0, 0, 0, 0, 0, 0, 0, |
|
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+ 0, 0, 0, 0, 0, 0, 0, 0 }, |
|
| 242 |
+}; |
|
| 243 |
+ |
|
| 244 |
+static const uint16_t dss_sp_fixed_cb_gain[64] = {
|
|
| 245 |
+ 0, 4, 8, 13, 17, 22, 26, 31, |
|
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+ 35, 40, 44, 48, 53, 58, 63, 69, |
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|
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|
| 250 |
+ 646, 706, 771, 843, 922, 1007, 1101, 1204, |
|
| 251 |
+ 1316, 1438, 1572, 1719, 1879, 2053, 2244, 2453, |
|
| 252 |
+ 2682, 2931, 3204, 3502, 3828, 4184, 4574, 5000, |
|
| 253 |
+}; |
|
| 254 |
+ |
|
| 255 |
+static const int16_t dss_sp_pulse_val[8] = {
|
|
| 256 |
+ -31182, -22273, -13364, -4455, 4455, 13364, 22273, 31182 |
|
| 257 |
+}; |
|
| 258 |
+ |
|
| 259 |
+static const uint16_t binary_decreasing_array[] = {
|
|
| 260 |
+ 32767, 16384, 8192, 4096, 2048, 1024, 512, 256, |
|
| 261 |
+ 128, 64, 32, 16, 8, 4, 2, |
|
| 262 |
+}; |
|
| 263 |
+ |
|
| 264 |
+static const uint16_t dss_sp_unc_decreasing_array[] = {
|
|
| 265 |
+ 32767, 26214, 20972, 16777, 13422, 10737, 8590, 6872, |
|
| 266 |
+ 5498, 4398, 3518, 2815, 2252, 1801, 1441, |
|
| 267 |
+}; |
|
| 268 |
+ |
|
| 269 |
+static const uint16_t dss_sp_adaptive_gain[] = {
|
|
| 270 |
+ 102, 231, 360, 488, 617, 746, 875, 1004, |
|
| 271 |
+ 1133, 1261, 1390, 1519, 1648, 1777, 1905, 2034, |
|
| 272 |
+ 2163, 2292, 2421, 2550, 2678, 2807, 2936, 3065, |
|
| 273 |
+ 3194, 3323, 3451, 3580, 3709, 3838, 3967, 4096, |
|
| 274 |
+}; |
|
| 275 |
+ |
|
| 276 |
+static const int32_t dss_sp_sinc[67] = {
|
|
| 277 |
+ 262, 293, 323, 348, 356, 336, 269, 139, |
|
| 278 |
+ -67, -358, -733, -1178, -1668, -2162, -2607, -2940, |
|
| 279 |
+ -3090, -2986, -2562, -1760, -541, 1110, 3187, 5651, |
|
| 280 |
+ 8435, 11446, 14568, 17670, 20611, 23251, 25460, 27125, |
|
| 281 |
+ 28160, 28512, 28160, |
|
| 282 |
+ 27125, 25460, 23251, 20611, 17670, 14568, 11446, 8435, |
|
| 283 |
+ 5651, 3187, 1110, -541, -1760, -2562, -2986, -3090, |
|
| 284 |
+ -2940, -2607, -2162, -1668, -1178, -733, -358, -67, |
|
| 285 |
+ 139, 269, 336, 356, 348, 323, 293, 262, |
|
| 286 |
+}; |
|
| 287 |
+ |
|
| 288 |
+static av_cold int dss_sp_decode_init(AVCodecContext *avctx) |
|
| 289 |
+{
|
|
| 290 |
+ DssSpContext *p = avctx->priv_data; |
|
| 291 |
+ avctx->channel_layout = AV_CH_LAYOUT_MONO; |
|
| 292 |
+ avctx->sample_fmt = AV_SAMPLE_FMT_S16; |
|
| 293 |
+ avctx->channels = 1; |
|
| 294 |
+ avctx->sample_rate = 11025; |
|
| 295 |
+ |
|
| 296 |
+ memset(p->history, 0, sizeof(p->history)); |
|
| 297 |
+ p->pulse_dec_mode = 1; |
|
| 298 |
+ |
|
| 299 |
+ return 0; |
|
| 300 |
+} |
|
| 301 |
+ |
|
| 302 |
+static void dss_sp_unpack_coeffs(DssSpContext *p, const uint8_t *src) |
|
| 303 |
+{
|
|
| 304 |
+ GetBitContext gb; |
|
| 305 |
+ DssSpFrame *fparam = &p->fparam; |
|
| 306 |
+ int i; |
|
| 307 |
+ int subframe_idx; |
|
| 308 |
+ uint32_t combined_pitch; |
|
| 309 |
+ uint32_t tmp; |
|
| 310 |
+ uint32_t pitch_lag; |
|
| 311 |
+ |
|
| 312 |
+ for (i = 0; i < DSS_SP_FRAME_SIZE; i += 2) {
|
|
| 313 |
+ p->bits[i] = src[i + 1]; |
|
| 314 |
+ p->bits[i + 1] = src[i]; |
|
| 315 |
+ } |
|
| 316 |
+ |
|
| 317 |
+ init_get_bits(&gb, p->bits, DSS_SP_FRAME_SIZE * 8); |
|
| 318 |
+ |
|
| 319 |
+ for (i = 0; i < 2; i++) |
|
| 320 |
+ fparam->filter_idx[i] = get_bits(&gb, 5); |
|
| 321 |
+ for (; i < 8; i++) |
|
| 322 |
+ fparam->filter_idx[i] = get_bits(&gb, 4); |
|
| 323 |
+ for (; i < 14; i++) |
|
| 324 |
+ fparam->filter_idx[i] = get_bits(&gb, 3); |
|
| 325 |
+ |
|
| 326 |
+ for (subframe_idx = 0; subframe_idx < 4; subframe_idx++) {
|
|
| 327 |
+ fparam->sf_adaptive_gain[subframe_idx] = get_bits(&gb, 5); |
|
| 328 |
+ |
|
| 329 |
+ fparam->sf[subframe_idx].combined_pulse_pos = get_bits_long(&gb, 31); |
|
| 330 |
+ |
|
| 331 |
+ fparam->sf[subframe_idx].gain = get_bits(&gb, 6); |
|
| 332 |
+ |
|
| 333 |
+ for (i = 0; i < 7; i++) |
|
| 334 |
+ fparam->sf[subframe_idx].pulse_val[i] = get_bits(&gb, 3); |
|
| 335 |
+ } |
|
| 336 |
+ |
|
| 337 |
+ for (subframe_idx = 0; subframe_idx < 4; subframe_idx++) {
|
|
| 338 |
+ unsigned int C72_binomials[PULSE_MAX] = {
|
|
| 339 |
+ 72, 2556, 59640, 1028790, 13991544, 156238908, 1473109704, |
|
| 340 |
+ 3379081753 |
|
| 341 |
+ }; |
|
| 342 |
+ unsigned int combined_pulse_pos = |
|
| 343 |
+ fparam->sf[subframe_idx].combined_pulse_pos; |
|
| 344 |
+ int index = 6; |
|
| 345 |
+ |
|
| 346 |
+ if (combined_pulse_pos < C72_binomials[PULSE_MAX - 1]) {
|
|
| 347 |
+ if (p->pulse_dec_mode) {
|
|
| 348 |
+ int pulse, pulse_idx; |
|
| 349 |
+ pulse = PULSE_MAX - 1; |
|
| 350 |
+ pulse_idx = 71; |
|
| 351 |
+ combined_pulse_pos = |
|
| 352 |
+ fparam->sf[subframe_idx].combined_pulse_pos; |
|
| 353 |
+ |
|
| 354 |
+ /* this part seems to be close to g723.1 gen_fcb_excitation() |
|
| 355 |
+ * RATE_6300 */ |
|
| 356 |
+ |
|
| 357 |
+ /* TODO: what is 7? size of subframe? */ |
|
| 358 |
+ for (i = 0; i < 7; i++) {
|
|
| 359 |
+ for (; |
|
| 360 |
+ combined_pulse_pos < |
|
| 361 |
+ dss_sp_combinatorial_table[pulse][pulse_idx]; |
|
| 362 |
+ --pulse_idx) |
|
| 363 |
+ ; |
|
| 364 |
+ combined_pulse_pos -= |
|
| 365 |
+ dss_sp_combinatorial_table[pulse][pulse_idx]; |
|
| 366 |
+ pulse--; |
|
| 367 |
+ fparam->sf[subframe_idx].pulse_pos[i] = pulse_idx; |
|
| 368 |
+ } |
|
| 369 |
+ } |
|
| 370 |
+ } else {
|
|
| 371 |
+ p->pulse_dec_mode = 0; |
|
| 372 |
+ |
|
| 373 |
+ /* why do we need this? */ |
|
| 374 |
+ fparam->sf[subframe_idx].pulse_pos[6] = 0; |
|
| 375 |
+ |
|
| 376 |
+ for (i = 71; i >= 0; i--) {
|
|
| 377 |
+ if (C72_binomials[index] <= combined_pulse_pos) {
|
|
| 378 |
+ combined_pulse_pos -= C72_binomials[index]; |
|
| 379 |
+ |
|
| 380 |
+ fparam->sf[subframe_idx].pulse_pos[(index ^ 7) - 1] = i; |
|
| 381 |
+ |
|
| 382 |
+ if (!index) |
|
| 383 |
+ break; |
|
| 384 |
+ --index; |
|
| 385 |
+ } |
|
| 386 |
+ --C72_binomials[0]; |
|
| 387 |
+ if (index) {
|
|
| 388 |
+ int a; |
|
| 389 |
+ for (a = 0; a < index; a++) |
|
| 390 |
+ C72_binomials[a + 1] -= C72_binomials[a]; |
|
| 391 |
+ } |
|
| 392 |
+ } |
|
| 393 |
+ } |
|
| 394 |
+ } |
|
| 395 |
+ |
|
| 396 |
+ combined_pitch = get_bits(&gb, 24); |
|
| 397 |
+ |
|
| 398 |
+ fparam->pitch_lag[0] = (combined_pitch % 151) + 36; |
|
| 399 |
+ |
|
| 400 |
+ combined_pitch /= 151; |
|
| 401 |
+ |
|
| 402 |
+ for (i = 1; i < SUBFRAMES; i++) {
|
|
| 403 |
+ fparam->pitch_lag[i] = combined_pitch % 48; |
|
| 404 |
+ combined_pitch /= 48; |
|
| 405 |
+ } |
|
| 406 |
+ |
|
| 407 |
+ pitch_lag = fparam->pitch_lag[0]; |
|
| 408 |
+ for (i = 1; i < SUBFRAMES; i++) {
|
|
| 409 |
+ if (pitch_lag > 162) {
|
|
| 410 |
+ fparam->pitch_lag[i] += 162 - 23; |
|
| 411 |
+ } else {
|
|
| 412 |
+ tmp = pitch_lag - 23; |
|
| 413 |
+ if (tmp < 36) |
|
| 414 |
+ tmp = 36; |
|
| 415 |
+ fparam->pitch_lag[i] += tmp; |
|
| 416 |
+ } |
|
| 417 |
+ pitch_lag = fparam->pitch_lag[i]; |
|
| 418 |
+ } |
|
| 419 |
+} |
|
| 420 |
+ |
|
| 421 |
+static void dss_sp_unpack_filter(DssSpContext *p) |
|
| 422 |
+{
|
|
| 423 |
+ int i; |
|
| 424 |
+ |
|
| 425 |
+ for (i = 0; i < 14; i++) |
|
| 426 |
+ p->lpc_filter[i] = dss_sp_filter_cb[i][p->fparam.filter_idx[i]]; |
|
| 427 |
+} |
|
| 428 |
+ |
|
| 429 |
+static void dss_sp_convert_coeffs(int32_t *lpc_filter, int32_t *coeffs) |
|
| 430 |
+{
|
|
| 431 |
+ int a, a_plus, i; |
|
| 432 |
+ |
|
| 433 |
+ coeffs[0] = 0x2000; |
|
| 434 |
+ for (a = 0; a < 14; a++) {
|
|
| 435 |
+ a_plus = a + 1; |
|
| 436 |
+ coeffs[a_plus] = lpc_filter[a] >> 2; |
|
| 437 |
+ if (a_plus / 2 >= 1) {
|
|
| 438 |
+ for (i = 1; i <= a_plus / 2; i++) {
|
|
| 439 |
+ int coeff_1, coeff_2, tmp; |
|
| 440 |
+ |
|
| 441 |
+ coeff_1 = coeffs[i]; |
|
| 442 |
+ coeff_2 = coeffs[a_plus - i]; |
|
| 443 |
+ |
|
| 444 |
+ tmp = DSS_SP_FORMULA(coeff_1, lpc_filter[a], coeff_2); |
|
| 445 |
+ coeffs[i] = av_clip_int16(tmp); |
|
| 446 |
+ |
|
| 447 |
+ tmp = DSS_SP_FORMULA(coeff_2, lpc_filter[a], coeff_1); |
|
| 448 |
+ coeffs[a_plus - i] = av_clip_int16(tmp); |
|
| 449 |
+ } |
|
| 450 |
+ } |
|
| 451 |
+ } |
|
| 452 |
+} |
|
| 453 |
+ |
|
| 454 |
+static void dss_sp_add_pulses(int32_t *vector_buf, |
|
| 455 |
+ const struct DssSpSubframe *sf) |
|
| 456 |
+{
|
|
| 457 |
+ int i; |
|
| 458 |
+ |
|
| 459 |
+ for (i = 0; i < 7; i++) |
|
| 460 |
+ vector_buf[sf->pulse_pos[i]] += (dss_sp_fixed_cb_gain[sf->gain] * |
|
| 461 |
+ dss_sp_pulse_val[sf->pulse_val[i]] + |
|
| 462 |
+ 0x4000) >> 15; |
|
| 463 |
+} |
|
| 464 |
+ |
|
| 465 |
+static void dss_sp_gen_exc(int32_t *vector, int32_t *prev_exc, |
|
| 466 |
+ int pitch_lag, int gain) |
|
| 467 |
+{
|
|
| 468 |
+ int i; |
|
| 469 |
+ |
|
| 470 |
+ /* do we actually need this check? we can use just [a3 - i % a3] |
|
| 471 |
+ * for both cases */ |
|
| 472 |
+ if (pitch_lag < 72) |
|
| 473 |
+ for (i = 0; i < 72; i++) |
|
| 474 |
+ vector[i] = prev_exc[pitch_lag - i % pitch_lag]; |
|
| 475 |
+ else |
|
| 476 |
+ for (i = 0; i < 72; i++) |
|
| 477 |
+ vector[i] = prev_exc[pitch_lag - i]; |
|
| 478 |
+ |
|
| 479 |
+ for (i = 0; i < 72; i++) {
|
|
| 480 |
+ int tmp = gain * vector[i] >> 11; |
|
| 481 |
+ vector[i] = av_clip_int16(tmp); |
|
| 482 |
+ } |
|
| 483 |
+} |
|
| 484 |
+ |
|
| 485 |
+static void dss_sp_scale_vector(int32_t *vec, int bits, int size) |
|
| 486 |
+{
|
|
| 487 |
+ int i; |
|
| 488 |
+ |
|
| 489 |
+ if (bits < 0) |
|
| 490 |
+ for (i = 0; i < size; i++) |
|
| 491 |
+ vec[i] = vec[i] >> -bits; |
|
| 492 |
+ else |
|
| 493 |
+ for (i = 0; i < size; i++) |
|
| 494 |
+ vec[i] = vec[i] << bits; |
|
| 495 |
+} |
|
| 496 |
+ |
|
| 497 |
+static void dss_sp_update_buf(int32_t *hist, int32_t *vector) |
|
| 498 |
+{
|
|
| 499 |
+ int i; |
|
| 500 |
+ |
|
| 501 |
+ for (i = 114; i > 0; i--) |
|
| 502 |
+ vector[i + 72] = vector[i]; |
|
| 503 |
+ |
|
| 504 |
+ for (i = 0; i < 72; i++) |
|
| 505 |
+ vector[72 - i] = hist[i]; |
|
| 506 |
+} |
|
| 507 |
+ |
|
| 508 |
+static void dss_sp_shift_sq_sub(const int32_t *filter_buf, |
|
| 509 |
+ int32_t *error_buf, int32_t *dst) |
|
| 510 |
+{
|
|
| 511 |
+ int a; |
|
| 512 |
+ |
|
| 513 |
+ for (a = 0; a < 72; a++) {
|
|
| 514 |
+ int i, tmp; |
|
| 515 |
+ |
|
| 516 |
+ tmp = dst[a] * filter_buf[0]; |
|
| 517 |
+ |
|
| 518 |
+ for (i = 14; i > 0; i--) |
|
| 519 |
+ tmp -= error_buf[i] * filter_buf[i]; |
|
| 520 |
+ |
|
| 521 |
+ for (i = 14; i > 0; i--) |
|
| 522 |
+ error_buf[i] = error_buf[i - 1]; |
|
| 523 |
+ |
|
| 524 |
+ tmp = (tmp + 4096) >> 13; |
|
| 525 |
+ |
|
| 526 |
+ error_buf[1] = tmp; |
|
| 527 |
+ |
|
| 528 |
+ dst[a] = av_clip_int16(tmp); |
|
| 529 |
+ } |
|
| 530 |
+} |
|
| 531 |
+ |
|
| 532 |
+static void dss_sp_shift_sq_add(const int32_t *filter_buf, int32_t *audio_buf, |
|
| 533 |
+ int32_t *dst) |
|
| 534 |
+{
|
|
| 535 |
+ int a; |
|
| 536 |
+ |
|
| 537 |
+ for (a = 0; a < 72; a++) {
|
|
| 538 |
+ int i, tmp = 0; |
|
| 539 |
+ |
|
| 540 |
+ audio_buf[0] = dst[a]; |
|
| 541 |
+ |
|
| 542 |
+ for (i = 14; i >= 0; i--) |
|
| 543 |
+ tmp += audio_buf[i] * filter_buf[i]; |
|
| 544 |
+ |
|
| 545 |
+ for (i = 14; i > 0; i--) |
|
| 546 |
+ audio_buf[i] = audio_buf[i - 1]; |
|
| 547 |
+ |
|
| 548 |
+ tmp = (tmp + 4096) >> 13; |
|
| 549 |
+ |
|
| 550 |
+ dst[a] = av_clip_int16(tmp); |
|
| 551 |
+ } |
|
| 552 |
+} |
|
| 553 |
+ |
|
| 554 |
+static void dss_sp_vec_mult(const int32_t *src, int32_t *dst, |
|
| 555 |
+ const int16_t *mult) |
|
| 556 |
+{
|
|
| 557 |
+ int i; |
|
| 558 |
+ |
|
| 559 |
+ dst[0] = src[0]; |
|
| 560 |
+ |
|
| 561 |
+ for (i = 1; i < 15; i++) |
|
| 562 |
+ dst[i] = (src[i] * mult[i] + 0x4000) >> 15; |
|
| 563 |
+} |
|
| 564 |
+ |
|
| 565 |
+static int dss_sp_get_normalize_bits(int32_t *vector_buf, int16_t size) |
|
| 566 |
+{
|
|
| 567 |
+ unsigned int val; |
|
| 568 |
+ int max_val; |
|
| 569 |
+ int i; |
|
| 570 |
+ |
|
| 571 |
+ val = 1; |
|
| 572 |
+ for (i = 0; i < size; i++) |
|
| 573 |
+ val |= FFABS(vector_buf[i]); |
|
| 574 |
+ |
|
| 575 |
+ for (max_val = 0; val <= 0x4000; ++max_val) |
|
| 576 |
+ val *= 2; |
|
| 577 |
+ return max_val; |
|
| 578 |
+} |
|
| 579 |
+ |
|
| 580 |
+static int dss_sp_vector_sum(DssSpContext *p, int size) |
|
| 581 |
+{
|
|
| 582 |
+ int i, sum = 0; |
|
| 583 |
+ for (i = 0; i < size; i++) |
|
| 584 |
+ sum += FFABS(p->vector_buf[i]); |
|
| 585 |
+ return sum; |
|
| 586 |
+} |
|
| 587 |
+ |
|
| 588 |
+static void dss_sp_sf_synthesis(DssSpContext *p, int32_t lpc_filter, |
|
| 589 |
+ int32_t *dst, int size) |
|
| 590 |
+{
|
|
| 591 |
+ int32_t tmp_buf[15]; |
|
| 592 |
+ int32_t noise[72]; |
|
| 593 |
+ int bias, vsum_2 = 0, vsum_1 = 0, v36, normalize_bits; |
|
| 594 |
+ int i, tmp; |
|
| 595 |
+ |
|
| 596 |
+ if (size > 0) {
|
|
| 597 |
+ vsum_1 = dss_sp_vector_sum(p, size); |
|
| 598 |
+ |
|
| 599 |
+ if (vsum_1 > 0xFFFFF) |
|
| 600 |
+ vsum_1 = 0xFFFFF; |
|
| 601 |
+ } |
|
| 602 |
+ |
|
| 603 |
+ normalize_bits = dss_sp_get_normalize_bits(p->vector_buf, size); |
|
| 604 |
+ |
|
| 605 |
+ dss_sp_scale_vector(p->vector_buf, normalize_bits - 3, size); |
|
| 606 |
+ dss_sp_scale_vector(p->audio_buf, normalize_bits, 15); |
|
| 607 |
+ dss_sp_scale_vector(p->err_buf1, normalize_bits, 15); |
|
| 608 |
+ |
|
| 609 |
+ v36 = p->err_buf1[1]; |
|
| 610 |
+ |
|
| 611 |
+ dss_sp_vec_mult(p->filter, tmp_buf, binary_decreasing_array); |
|
| 612 |
+ dss_sp_shift_sq_add(tmp_buf, p->audio_buf, p->vector_buf); |
|
| 613 |
+ |
|
| 614 |
+ dss_sp_vec_mult(p->filter, tmp_buf, dss_sp_unc_decreasing_array); |
|
| 615 |
+ dss_sp_shift_sq_sub(tmp_buf, p->err_buf1, p->vector_buf); |
|
| 616 |
+ |
|
| 617 |
+ /* lpc_filter can be negative */ |
|
| 618 |
+ lpc_filter = lpc_filter >> 1; |
|
| 619 |
+ if (lpc_filter >= 0) |
|
| 620 |
+ lpc_filter = 0; |
|
| 621 |
+ |
|
| 622 |
+ if (size > 1) {
|
|
| 623 |
+ for (i = size - 1; i > 0; i--) {
|
|
| 624 |
+ tmp = DSS_SP_FORMULA(p->vector_buf[i], lpc_filter, |
|
| 625 |
+ p->vector_buf[i - 1]); |
|
| 626 |
+ p->vector_buf[i] = av_clip_int16(tmp); |
|
| 627 |
+ } |
|
| 628 |
+ } |
|
| 629 |
+ |
|
| 630 |
+ tmp = DSS_SP_FORMULA(p->vector_buf[0], lpc_filter, v36); |
|
| 631 |
+ p->vector_buf[0] = av_clip_int16(tmp); |
|
| 632 |
+ |
|
| 633 |
+ dss_sp_scale_vector(p->vector_buf, -normalize_bits, size); |
|
| 634 |
+ dss_sp_scale_vector(p->audio_buf, -normalize_bits, 15); |
|
| 635 |
+ dss_sp_scale_vector(p->err_buf1, -normalize_bits, 15); |
|
| 636 |
+ |
|
| 637 |
+ if (size > 0) |
|
| 638 |
+ vsum_2 = dss_sp_vector_sum(p, size); |
|
| 639 |
+ |
|
| 640 |
+ if (vsum_2 >= 0x40) |
|
| 641 |
+ tmp = (vsum_1 << 11) / vsum_2; |
|
| 642 |
+ else |
|
| 643 |
+ tmp = 1; |
|
| 644 |
+ |
|
| 645 |
+ bias = 409 * tmp >> 15 << 15; |
|
| 646 |
+ tmp = (bias + 32358 * p->noise_state) >> 15; |
|
| 647 |
+ noise[0] = av_clip_int16(tmp); |
|
| 648 |
+ |
|
| 649 |
+ for (i = 1; i < size; i++) {
|
|
| 650 |
+ tmp = (bias + 32358 * noise[i - 1]) >> 15; |
|
| 651 |
+ noise[i] = av_clip_int16(tmp); |
|
| 652 |
+ } |
|
| 653 |
+ |
|
| 654 |
+ p->noise_state = noise[size - 1]; |
|
| 655 |
+ for (i = 0; i < size; i++) {
|
|
| 656 |
+ tmp = (p->vector_buf[i] * noise[i]) >> 11; |
|
| 657 |
+ dst[i] = av_clip_int16(tmp); |
|
| 658 |
+ } |
|
| 659 |
+} |
|
| 660 |
+ |
|
| 661 |
+static void dss_sp_update_state(DssSpContext *p, int32_t *dst) |
|
| 662 |
+{
|
|
| 663 |
+ int i, offset = 6, counter = 0, a = 0; |
|
| 664 |
+ |
|
| 665 |
+ for (i = 0; i < 6; i++) |
|
| 666 |
+ p->excitation[i] = p->excitation[288 + i]; |
|
| 667 |
+ |
|
| 668 |
+ for (i = 0; i < 72 * SUBFRAMES; i++) |
|
| 669 |
+ p->excitation[6 + i] = dst[i]; |
|
| 670 |
+ |
|
| 671 |
+ do {
|
|
| 672 |
+ int tmp = 0; |
|
| 673 |
+ |
|
| 674 |
+ for (i = 0; i < 6; i++) |
|
| 675 |
+ tmp += p->excitation[offset--] * dss_sp_sinc[a + i * 11]; |
|
| 676 |
+ |
|
| 677 |
+ offset += 7; |
|
| 678 |
+ |
|
| 679 |
+ tmp >>= 15; |
|
| 680 |
+ dst[counter] = av_clip_int16(tmp); |
|
| 681 |
+ |
|
| 682 |
+ counter++; |
|
| 683 |
+ |
|
| 684 |
+ a = (a + 1) % 11; |
|
| 685 |
+ if (!a) |
|
| 686 |
+ offset++; |
|
| 687 |
+ } while (offset < FF_ARRAY_ELEMS(p->excitation)); |
|
| 688 |
+} |
|
| 689 |
+ |
|
| 690 |
+static void dss_sp_32to16bit(int16_t *dst, int32_t *src, int size) |
|
| 691 |
+{
|
|
| 692 |
+ int i; |
|
| 693 |
+ |
|
| 694 |
+ for (i = 0; i < size; i++) |
|
| 695 |
+ dst[i] = av_clip_int16(src[i]); |
|
| 696 |
+} |
|
| 697 |
+ |
|
| 698 |
+static int dss_sp_decode_one_frame(DssSpContext *p, |
|
| 699 |
+ int16_t *abuf_dst, const uint8_t *abuf_src) |
|
| 700 |
+{
|
|
| 701 |
+ int i, j; |
|
| 702 |
+ |
|
| 703 |
+ dss_sp_unpack_coeffs(p, abuf_src); |
|
| 704 |
+ |
|
| 705 |
+ dss_sp_unpack_filter(p); |
|
| 706 |
+ |
|
| 707 |
+ dss_sp_convert_coeffs(p->lpc_filter, p->filter); |
|
| 708 |
+ |
|
| 709 |
+ for (j = 0; j < SUBFRAMES; j++) {
|
|
| 710 |
+ dss_sp_gen_exc(p->vector_buf, p->history, |
|
| 711 |
+ p->fparam.pitch_lag[j], |
|
| 712 |
+ dss_sp_adaptive_gain[p->fparam.sf_adaptive_gain[j]]); |
|
| 713 |
+ |
|
| 714 |
+ dss_sp_add_pulses(p->vector_buf, &p->fparam.sf[j]); |
|
| 715 |
+ |
|
| 716 |
+ dss_sp_update_buf(p->vector_buf, p->history); |
|
| 717 |
+ |
|
| 718 |
+ for (i = 0; i < 72; i++) |
|
| 719 |
+ p->vector_buf[i] = p->history[72 - i]; |
|
| 720 |
+ |
|
| 721 |
+ dss_sp_shift_sq_sub(p->filter, |
|
| 722 |
+ p->err_buf2, p->vector_buf); |
|
| 723 |
+ |
|
| 724 |
+ dss_sp_sf_synthesis(p, p->lpc_filter[0], |
|
| 725 |
+ &p->working_buffer[j][0], 72); |
|
| 726 |
+ } |
|
| 727 |
+ |
|
| 728 |
+ dss_sp_update_state(p, &p->working_buffer[0][0]); |
|
| 729 |
+ |
|
| 730 |
+ dss_sp_32to16bit(abuf_dst, |
|
| 731 |
+ &p->working_buffer[0][0], 264); |
|
| 732 |
+ return 0; |
|
| 733 |
+} |
|
| 734 |
+ |
|
| 735 |
+static int dss_sp_decode_frame(AVCodecContext *avctx, void *data, |
|
| 736 |
+ int *got_frame_ptr, AVPacket *avpkt) |
|
| 737 |
+{
|
|
| 738 |
+ DssSpContext *p = avctx->priv_data; |
|
| 739 |
+ AVFrame *frame = data; |
|
| 740 |
+ const uint8_t *buf = avpkt->data; |
|
| 741 |
+ int buf_size = avpkt->size; |
|
| 742 |
+ |
|
| 743 |
+ int16_t *out; |
|
| 744 |
+ int ret; |
|
| 745 |
+ |
|
| 746 |
+ if (buf_size < DSS_SP_FRAME_SIZE) {
|
|
| 747 |
+ if (buf_size) |
|
| 748 |
+ av_log(avctx, AV_LOG_WARNING, |
|
| 749 |
+ "Expected %d bytes, got %d - skipping packet.\n", |
|
| 750 |
+ DSS_SP_FRAME_SIZE, buf_size); |
|
| 751 |
+ *got_frame_ptr = 0; |
|
| 752 |
+ return AVERROR_INVALIDDATA; |
|
| 753 |
+ } |
|
| 754 |
+ |
|
| 755 |
+ frame->nb_samples = DSS_SP_SAMPLE_COUNT; |
|
| 756 |
+ if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
|
|
| 757 |
+ av_log(avctx, AV_LOG_ERROR, "get_buffer() failed.\n"); |
|
| 758 |
+ return ret; |
|
| 759 |
+ } |
|
| 760 |
+ |
|
| 761 |
+ out = (int16_t *)frame->data[0]; |
|
| 762 |
+ |
|
| 763 |
+ dss_sp_decode_one_frame(p, out, buf); |
|
| 764 |
+ |
|
| 765 |
+ *got_frame_ptr = 1; |
|
| 766 |
+ |
|
| 767 |
+ return DSS_SP_FRAME_SIZE; |
|
| 768 |
+} |
|
| 769 |
+ |
|
| 770 |
+AVCodec ff_dss_sp_decoder = {
|
|
| 771 |
+ .name = "DSS SP", |
|
| 772 |
+ .long_name = NULL_IF_CONFIG_SMALL("Digital Speech Standard - Standard Play mode (DSS SP)"),
|
|
| 773 |
+ .type = AVMEDIA_TYPE_AUDIO, |
|
| 774 |
+ .id = AV_CODEC_ID_DSS_SP, |
|
| 775 |
+ .priv_data_size = sizeof(DssSpContext), |
|
| 776 |
+ .init = dss_sp_decode_init, |
|
| 777 |
+ .decode = dss_sp_decode_frame, |
|
| 778 |
+ .capabilities = CODEC_CAP_DR1, |
|
| 779 |
+}; |
| ... | ... |
@@ -29,7 +29,7 @@ |
| 29 | 29 |
#include "libavutil/version.h" |
| 30 | 30 |
|
| 31 | 31 |
#define LIBAVCODEC_VERSION_MAJOR 56 |
| 32 |
-#define LIBAVCODEC_VERSION_MINOR 14 |
|
| 32 |
+#define LIBAVCODEC_VERSION_MINOR 15 |
|
| 33 | 33 |
#define LIBAVCODEC_VERSION_MICRO 0 |
| 34 | 34 |
|
| 35 | 35 |
#define LIBAVCODEC_VERSION_INT AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \ |