On Tue, Dec 2, 2014 at 3:25 PM, Vittorio Giovara <[email protected]> wrote: > From: Oleksij Rempel <[email protected]> > > Signed-off-by: Oleksij Rempel <[email protected]> > Signed-off-by: Luca Barbato <[email protected]> > Signed-off-by: Vittorio Giovara <[email protected]> > --- > Changelog | 1 + > libavcodec/Makefile | 1 + > libavcodec/allcodecs.c | 1 + > libavcodec/avcodec.h | 1 + > libavcodec/codec_desc.c | 7 + > libavcodec/dss_sp_dec.c | 794 > ++++++++++++++++++++++++++++++++++++++++++++++++ > libavcodec/version.h | 2 +- > 7 files changed, 806 insertions(+), 1 deletion(-) > create mode 100644 libavcodec/dss_sp_dec.c > > diff --git a/Changelog b/Changelog > index 6af2e8a..49bd12e 100644 > --- a/Changelog > +++ b/Changelog > @@ -7,6 +7,7 @@ version <next>: > - avplay now exits by default at the end of playback > - XCB-based screen-grabber > - creating DASH compatible fragmented MP4, MPEG-DASH segmenting muxer > +- DSS decoder (sp/lp) > > > version 11: > diff --git a/libavcodec/Makefile b/libavcodec/Makefile > index 66aa0ce..1b1ad0c 100644 > --- a/libavcodec/Makefile > +++ b/libavcodec/Makefile > @@ -201,6 +201,7 @@ OBJS-$(CONFIG_FRWU_DECODER) += frwu.o > OBJS-$(CONFIG_G2M_DECODER) += g2meet.o mjpeg.o > OBJS-$(CONFIG_G723_1_DECODER) += g723_1.o acelp_vectors.o \ > celp_filters.o > +OBJS-$(CONFIG_DSS_SP_DECODER) += dss_sp_dec.o > OBJS-$(CONFIG_GIF_DECODER) += gifdec.o lzw.o > OBJS-$(CONFIG_GIF_ENCODER) += gif.o lzwenc.o > OBJS-$(CONFIG_GSM_DECODER) += gsmdec.o gsmdec_data.o msgsmdec.o > diff --git a/libavcodec/allcodecs.c b/libavcodec/allcodecs.c > index 6a71b2c..340ff42 100644 > --- a/libavcodec/allcodecs.c > +++ b/libavcodec/allcodecs.c > @@ -305,6 +305,7 @@ void avcodec_register_all(void) > REGISTER_DECODER(COOK, cook); > REGISTER_DECODER(DCA, dca); > REGISTER_DECODER(DSICINAUDIO, dsicinaudio); > + REGISTER_DECODER(DSS_SP, dss_sp); > REGISTER_ENCDEC (EAC3, eac3); > REGISTER_ENCDEC (FLAC, flac); > REGISTER_DECODER(G723_1, g723_1); > diff --git a/libavcodec/avcodec.h b/libavcodec/avcodec.h > index b3c6419..9ef78b3 100644 > --- a/libavcodec/avcodec.h > +++ b/libavcodec/avcodec.h > @@ -444,6 +444,7 @@ enum AVCodecID { > AV_CODEC_ID_METASOUND, > AV_CODEC_ID_PAF_AUDIO, > AV_CODEC_ID_ON2AVC, > + AV_CODEC_ID_DSS_SP, > > /* subtitle codecs */ > AV_CODEC_ID_FIRST_SUBTITLE = 0x17000, ///< A dummy ID pointing > at the start of subtitle codecs. > diff --git a/libavcodec/codec_desc.c b/libavcodec/codec_desc.c > index c21c57e..91aa975 100644 > --- a/libavcodec/codec_desc.c > +++ b/libavcodec/codec_desc.c > @@ -2202,6 +2202,13 @@ static const AVCodecDescriptor codec_descriptors[] = { > .props = AV_CODEC_PROP_LOSSY, > }, > { > + .id = AV_CODEC_ID_DSS_SP, > + .type = AVMEDIA_TYPE_AUDIO, > + .name = "dss_sp", > + .long_name = NULL_IF_CONFIG_SMALL("DSS SP"), > + .props = AV_CODEC_PROP_LOSSY, > + }, > + { > .id = AV_CODEC_ID_G729, > .type = AVMEDIA_TYPE_AUDIO, > .name = "g729", > diff --git a/libavcodec/dss_sp_dec.c b/libavcodec/dss_sp_dec.c > new file mode 100644 > index 0000000..2673985 > --- /dev/null > +++ b/libavcodec/dss_sp_dec.c > @@ -0,0 +1,794 @@ > +/* > + * dss_sp audio decoder > + * Copyright (C) 2014 Oleksij Rempel <[email protected]> > + * > + * This file is part of Libav. > + * > + * Libav is free software; you can redistribute it and/or > + * modify it under the terms of the GNU Lesser General Public > + * License as published by the Free Software Foundation; either > + * version 2.1 of the License, or (at your option) any later version. > + * > + * Libav is distributed in the hope that it will be useful, > + * but WITHOUT ANY WARRANTY; without even the implied warranty of > + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU > + * Lesser General Public License for more details. > + * > + * You should have received a copy of the GNU Lesser General Public > + * License along with Libav; if not, write to the Free Software > + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 > USA > + */ > + > +#include "libavutil/common.h" > +#include "libavutil/channel_layout.h" > +#include "libavutil/mem.h" > +#include "libavutil/opt.h" > +#include "avcodec.h" > +#include "internal.h" > +#include "get_bits.h" > + > +#define SUBFRAMES 4 > +#define PULSE_MAX 8 > +#define DSS_SP_FRAME_SIZE 42 > +#define DSS_SP_SAMPLE_COUNT (66 * SUBFRAMES) > + > +#define DSS_FORMULA(a, b, c) ((((a) << 15) + (b) * (c)) + 0x4000) >> 15 > +#define DSS_FRAME_SIZE 42 > + > +typedef struct DssSpSubframe { > + int16_t gain; > + int32_t combined_pulse_pos; > + int16_t pulse_pos[7]; > + int16_t pulse_val[7]; > +} DssSpSubframe; > + > +typedef struct DssSpFrame { > + int16_t filter_idx[14]; > + int16_t sf_adaptive_gain[SUBFRAMES]; > + int16_t pitch_lag[SUBFRAMES]; > + struct DssSpSubframe sf[SUBFRAMES]; > +} DssSpFrame; > + > +typedef struct LPCData { > + int32_t filter[14]; > +} LPCData; > + > +typedef struct DssSpContext { > + int32_t excitation[288 + 6]; > + int32_t history[187]; > + DssSpFrame fparam; > + int32_t working_buffer[SUBFRAMES][72]; > + int32_t audio_buf[15]; > + int32_t err_buf1[15]; > + LPCData lpc; > + int32_t filter[15]; > + int32_t vector_buf[72]; > + int noise_state; > + int32_t err_buf2[15]; > + > + int pulse_dec_mode; > + > + DECLARE_ALIGNED(16, uint8_t, bits)[DSS_FRAME_SIZE + > FF_INPUT_BUFFER_PADDING_SIZE]; > +} DssSpContext; > + > + > + > +/* > + * Comment from: ./libavcodec/g723_1_data.h > + * Used for the coding/decoding of the pulses positions > + * for the MP-MLQ codebook > + */ > +static const uint32_t dss_sp_combinatorial_table[PULSE_MAX][72] = { > + { 0, 0, 0, 0, 0, 0, > + 0, 0, 0, 0, 0, 0, > + 0, 0, 0, 0, 0, 0, > + 0, 0, 0, 0, 0, 0, > + 0, 0, 0, 0, 0, 0, > + 0, 0, 0, 0, 0, 0, > + 0, 0, 0, 0, 0, 0, > + 0, 0, 0, 0, 0, 0, > + 0, 0, 0, 0, 0, 0, > + 0, 0, 0, 0, 0, 0, > + 0, 0, 0, 0, 0, 0, > + 0, 0, 0, 0, 0, 0 }, > + { 0, 1, 2, 3, 4, 5, > + 6, 7, 8, 9, 10, 11, > + 12, 13, 14, 15, 16, 17, > + 18, 19, 20, 21, 22, 23, > + 24, 25, 26, 27, 28, 29, > + 30, 31, 32, 33, 34, 35, > + 36, 37, 38, 39, 40, 41, > + 42, 43, 44, 45, 46, 47, > + 48, 49, 50, 51, 52, 53, > + 54, 55, 56, 57, 58, 59, > + 60, 61, 62, 63, 64, 65, > + 66, 67, 68, 69, 70, 71 }, > + { 0, 0, 1, 3, 6, 10, > + 15, 21, 28, 36, 45, 55, > + 66, 78, 91, 105, 120, 136, > + 153, 171, 190, 210, 231, 253, > + 276, 300, 325, 351, 378, 406, > + 435, 465, 496, 528, 561, 595, > + 630, 666, 703, 741, 780, 820, > + 861, 903, 946, 990, 1035, 1081, > + 1128, 1176, 1225, 1275, 1326, 1378, > + 1431, 1485, 1540, 1596, 1653, 1711, > + 1770, 1830, 1891, 1953, 2016, 2080, > + 2145, 2211, 2278, 2346, 2415, 2485 }, > + { 0, 0, 0, 1, 4, 10, > + 20, 35, 56, 84, 120, 165, > + 220, 286, 364, 455, 560, 680, > + 816, 969, 1140, 1330, 1540, 1771, > + 2024, 2300, 2600, 2925, 3276, 3654, > + 4060, 4495, 4960, 5456, 5984, 6545, > + 7140, 7770, 8436, 9139, 9880, 10660, > + 11480, 12341, 13244, 14190, 15180, 16215, > + 17296, 18424, 19600, 20825, 22100, 23426, > + 24804, 26235, 27720, 29260, 30856, 32509, > + 34220, 35990, 37820, 39711, 41664, 43680, > + 45760, 47905, 50116, 52394, 54740, 57155 }, > + { 0, 0, 0, 0, 1, 5, > + 15, 35, 70, 126, 210, 330, > + 495, 715, 1001, 1365, 1820, 2380, > + 3060, 3876, 4845, 5985, 7315, 8855, > + 10626, 12650, 14950, 17550, 20475, 23751, > + 27405, 31465, 35960, 40920, 46376, 52360, > + 58905, 66045, 73815, 82251, 91390, 101270, > + 111930, 123410, 135751, 148995, 163185, 178365, > + 194580, 211876, 230300, 249900, 270725, 292825, > + 316251, 341055, 367290, 395010, 424270, 455126, > + 487635, 521855, 557845, 595665, 635376, 677040, > + 720720, 766480, 814385, 864501, 916895, 971635 }, > + { 0, 0, 0, 0, 0, 1, > + 6, 21, 56, 126, 252, 462, > + 792, 1287, 2002, 3003, 4368, 6188, > + 8568, 11628, 15504, 20349, 26334, 33649, > + 42504, 53130, 65780, 80730, 98280, 118755, > + 142506, 169911, 201376, 237336, 278256, 324632, > + 376992, 435897, 501942, 575757, 658008, 749398, > + 850668, 962598, 1086008, 1221759, 1370754, 1533939, > + 1712304, 1906884, 2118760, 2349060, 2598960, 2869685, > + 3162510, 3478761, 3819816, 4187106, 4582116, 5006386, > + 5461512, 5949147, 6471002, 7028847, 7624512, 8259888, > + 8936928, 9657648, 10424128, 11238513, 12103014, 13019909 }, > + { 0, 0, 0, 0, 0, 0, > + 1, 7, 28, 84, 210, 462, > + 924, 1716, 3003, 5005, 8008, 12376, > + 18564, 27132, 38760, 54264, 74613, 100947, > + 134596, 177100, 230230, 296010, 376740, 475020, > + 593775, 736281, 906192, 1107568, 1344904, 1623160, > + 1947792, 2324784, 2760681, 3262623, 3838380, 4496388, > + 5245786, 6096454, 7059052, 8145060, 9366819, 10737573, > + 12271512, 13983816, 15890700, 18009460, 20358520, 22957480, > + 25827165, 28989675, 32468436, 36288252, 40475358, 45057474, > + 50063860, 55525372, 61474519, 67945521, 74974368, 82598880, > + 90858768, 99795696, 109453344, 119877472, 131115985, 143218999 }, > + { 0, 0, 0, 0, 0, 0, > + 0, 1, 8, 36, 120, 330, > + 792, 1716, 3432, 6435, 11440, 19448, > + 31824, 50388, 77520, 116280, 170544, 245157, > + 346104, 480700, 657800, 888030, 1184040, 1560780, > + 2035800, 2629575, 3365856, 4272048, 5379616, 6724520, > + 8347680, 10295472, 12620256, 15380937, 18643560, 22481940, > + 26978328, 32224114, 38320568, 45379620, 53524680, 62891499, > + 73629072, 85900584, 99884400, 115775100, 133784560, 154143080, > + 177100560, 202927725, 231917400, 264385836, 300674088, 341149446, > + 386206920, 436270780, 491796152, 553270671, 621216192, 696190560, > + 778789440, 869648208, 969443904, 1078897248, 1198774720, 1329890705 }, > +}; > + > +static const int16_t dss_sp_filter_cb[14][32] = { > + { -32653, -32587, -32515, -32438, -32341, -32216, -32062, -31881, > + -31665, -31398, -31080, -30724, -30299, -29813, -29248, -28572, > + -27674, -26439, -24666, -22466, -19433, -16133, -12218, -7783, > + -2834, 1819, 6544, 11260, 16050, 20220, 24774, 28120 }, > + > + { -27503, -24509, -20644, -17496, -14187, -11277, -8420, -5595, > + -3013, -624, 1711, 3880, 5844, 7774, 9739, 11592, > + 13364, 14903, 16426, 17900, 19250, 20586, 21803, 23006, > + 24142, 25249, 26275, 27300, 28359, 29249, 30118, 31183 }, > + > + { -27827, -24208, -20943, -17781, -14843, -11848, -9066, -6297, > + -3660, -910, 1918, 5025, 8223, 11649, 15086, 18423, > + 0, 0, 0, 0, 0, 0, 0, 0, > + 0, 0, 0, 0, 0, 0, 0, 0 }, > + > + { -17128, -11975, -8270, -5123, -2296, 183, 2503, 4707, > + 6798, 8945, 11045, 13239, 15528, 18248, 21115, 24785, > + 0, 0, 0, 0, 0, 0, 0, 0, > + 0, 0, 0, 0, 0, 0, 0, 0 }, > + > + { -21557, -17280, -14286, -11644, -9268, -7087, -4939, -2831, > + -691, 1407, 3536, 5721, 8125, 10677, 13721, 17731, > + 0, 0, 0, 0, 0, 0, 0, 0, > + 0, 0, 0, 0, 0, 0, 0, 0 }, > + > + { -15030, -10377, -7034, -4327, -1900, 364, 2458, 4450, > + 6422, 8374, 10374, 12486, 14714, 16997, 19626, 22954, > + 0, 0, 0, 0, 0, 0, 0, 0, > + 0, 0, 0, 0, 0, 0, 0, 0 }, > + > + { -16155, -12362, -9698, -7460, -5258, -3359, -1547, 219, > + 1916, 3599, 5299, 6994, 8963, 11226, 13716, 16982, > + 0, 0, 0, 0, 0, 0, 0, 0, > + 0, 0, 0, 0, 0, 0, 0, 0 }, > + > + { -14742, -9848, -6921, -4648, -2769, -1065, 499, 2083, > + 3633, 5219, 6857, 8580, 10410, 12672, 15561, 20101, > + 0, 0, 0, 0, 0, 0, 0, 0, > + 0, 0, 0, 0, 0, 0, 0, 0 }, > + > + { -11099, -7014, -3855, -1025, 1680, 4544, 7807, 11932, > + 0, 0, 0, 0, 0, 0, 0, 0, > + 0, 0, 0, 0, 0, 0, 0, 0, > + 0, 0, 0, 0, 0, 0, 0, 0 }, > + > + { -9060, -4570, -1381, 1419, 4034, 6728, 9865, 14149, > + 0, 0, 0, 0, 0, 0, 0, 0, > + 0, 0, 0, 0, 0, 0, 0, 0, > + 0, 0, 0, 0, 0, 0, 0, 0 }, > + > + { -12450, -7985, -4596, -1734, 961, 3629, 6865, 11142, > + 0, 0, 0, 0, 0, 0, 0, 0, > + 0, 0, 0, 0, 0, 0, 0, 0, > + 0, 0, 0, 0, 0, 0, 0, 0 }, > + > + { -11831, -7404, -4010, -1096, 1606, 4291, 7386, 11482, > + 0, 0, 0, 0, 0, 0, 0, 0, > + 0, 0, 0, 0, 0, 0, 0, 0, > + 0, 0, 0, 0, 0, 0, 0, 0 }, > + > + { -13404, -9250, -5995, -3312, -890, 1594, 4464, 8198, > + 0, 0, 0, 0, 0, 0, 0, 0, > + 0, 0, 0, 0, 0, 0, 0, 0, > + 0, 0, 0, 0, 0, 0, 0, 0 }, > + > + { -11239, -7220, -4040, -1406, 971, 3321, 6006, 9697, > + 0, 0, 0, 0, 0, 0, 0, 0, > + 0, 0, 0, 0, 0, 0, 0, 0, > + 0, 0, 0, 0, 0, 0, 0, 0 }, > +}; > + > +static const int16_t dss_sp_fixed_cb_gain[64] = { > + 0, 4, 8, 13, 17, 22, 26, 31, > + 35, 40, 44, 48, 53, 58, 63, 69, > + 76, 83, 91, 99, 109, 119, 130, 142, > + 155, 170, 185, 203, 222, 242, 265, 290, > + 317, 346, 378, 414, 452, 494, 540, 591, > + 646, 706, 771, 843, 922, 1007, 1101, 1204, > + 1316, 1438, 1572, 1719, 1879, 2053, 2244, 2453, > + 2682, 2931, 3204, 3502, 3828, 4184, 4574, 5000, > +}; > + > +static const int16_t dss_sp_pulse_val[8] = { > + -31182, -22273, -13364, -4455, 4455, 13364, 22273, 31182 > +}; > + > +static const int32_t binary_decreasing_array[] = { > + 32767, 16384, 8192, 4096, 2048, 1024, 512, 256, > + 128, 64, 32, 16, 8, 4, 2, > +}; > + > +static const int32_t dss_sp_unc_decreasing_array[] = { > + 32767, 26214, 20972, 16777, 13422, 10737, 8590, 6872, > + 5498, 4398, 3518, 2815, 2252, 1801, 1441, > +}; > + > +static const int16_t dss_sp_adaptive_gain[] = { > + 102, 231, 360, 488, 617, 746, 875, 1004, > + 1133, 1261, 1390, 1519, 1648, 1777, 1905, 2034, > + 2163, 2292, 2421, 2550, 2678, 2807, 2936, 3065, > + 3194, 3323, 3451, 3580, 3709, 3838, 3967, 4096, > +}; > + > +static const int32_t dss_sp_sinc[67] = { > + 262, 293, 323, 348, 356, 336, 269, 139, > + -67, -358, -733, -1178, -1668, -2162, -2607, -2940, > + -3090, -2986, -2562, -1760, -541, 1110, 3187, 5651, > + 8435, 11446, 14568, 17670, 20611, 23251, 25460, 27125, > + 28160, 28512, 28160, > + 27125, 25460, 23251, 20611, 17670, 14568, 11446, 8435, > + 5651, 3187, 1110, -541, -1760, -2562, -2986, -3090, > + -2940, -2607, -2162, -1668, -1178, -733, -358, -67, > + 139, 269, 336, 356, 348, 323, 293, 262, > +}; > + > +static av_cold int dss_sp_decode_init(AVCodecContext *avctx) > +{ > + DssSpContext *p = avctx->priv_data; > + avctx->channel_layout = AV_CH_LAYOUT_MONO; > + avctx->sample_fmt = AV_SAMPLE_FMT_S16; > + avctx->channels = 1; > + avctx->sample_rate = 11025; > + > + memset(p->history, 0, sizeof(p->history)); > + p->pulse_dec_mode = 1; > + > + return 0; > +} > + > +static void dss_sp_unpack_coeffs(DssSpContext *p, const uint8_t *src) > +{ > + GetBitContext gb; > + DssSpFrame *fparam = &p->fparam; > + int i; > + int subframe_idx; > + uint32_t combined_pitch; > + uint32_t tmp; > + uint32_t pitch_lag; > + > + for (i = 0; i < DSS_FRAME_SIZE; i += 2) { > + p->bits[i] = src[i + 1]; > + p->bits[i + 1] = src[i]; > + } > + > + init_get_bits(&gb, p->bits, DSS_FRAME_SIZE * 8); > + > + for (i = 0; i < 2; i++) > + fparam->filter_idx[i] = get_bits(&gb, 5); > + for (; i < 8; i++) > + fparam->filter_idx[i] = get_bits(&gb, 4); > + for (; i < 14; i++) > + fparam->filter_idx[i] = get_bits(&gb, 3); > + > + for (subframe_idx = 0; subframe_idx < 4; subframe_idx++) { > + fparam->sf_adaptive_gain[subframe_idx] = get_bits(&gb, 5); > + > + fparam->sf[subframe_idx].combined_pulse_pos = get_bits_long(&gb, 31); > + > + fparam->sf[subframe_idx].gain = get_bits(&gb, 6); > + > + for (i = 0; i < 7; i++) > + fparam->sf[subframe_idx].pulse_val[i] = get_bits(&gb, 3); > + } > + > + for (subframe_idx = 0; subframe_idx < 4; subframe_idx++) { > + unsigned int C72_binomials[PULSE_MAX] = { > + 72, 2556, 59640, 1028790, 13991544, 156238908, 1473109704, > + 3379081753 > + }; > + unsigned int combined_pulse_pos = > + fparam->sf[subframe_idx].combined_pulse_pos; > + int index = 6; > + > + if (combined_pulse_pos < C72_binomials[PULSE_MAX - 1]) { > + if (p->pulse_dec_mode) { > + int pulse, pulse_idx; > + pulse = PULSE_MAX - 1; > + pulse_idx = 71; > + combined_pulse_pos = > + fparam->sf[subframe_idx].combined_pulse_pos; > + > + /* this part seems to be close to g723.1 gen_fcb_excitation() > + * RATE_6300 */ > + > + /* TODO: what is 7? size of subframe? */ > + for (i = 0; i < 7; i++) { > + for (; > + combined_pulse_pos < > + dss_sp_combinatorial_table[pulse][pulse_idx]; > + --pulse_idx) > + ; > + combined_pulse_pos -= > + dss_sp_combinatorial_table[pulse][pulse_idx]; > + pulse--; > + fparam->sf[subframe_idx].pulse_pos[i] = pulse_idx; > + } > + } > + } else { > + p->pulse_dec_mode = 0; > + > + /* why do we need this? */ > + fparam->sf[subframe_idx].pulse_pos[6] = 0; > + > + for (i = 71; i >= 0; i--) { > + if (C72_binomials[index] <= combined_pulse_pos) { > + combined_pulse_pos -= C72_binomials[index]; > + > + fparam->sf[subframe_idx].pulse_pos[(index ^ 7) - 1] = i; > + > + if (!index) > + break; > + --index; > + } > + --C72_binomials[0]; > + if (index) { > + int a; > + for (a = 0; a < index; a++) > + C72_binomials[a + 1] -= C72_binomials[a]; > + } > + } > + } > + } > + > + combined_pitch = get_bits(&gb, 24); > + > + fparam->pitch_lag[0] = (combined_pitch % 151) + 36; > + > + combined_pitch /= 151; > + > + for (i = 1; i < SUBFRAMES; i++) { > + fparam->pitch_lag[i] = combined_pitch % 48; > + combined_pitch /= 48; > + } > + > + pitch_lag = fparam->pitch_lag[0]; > + for (i = 1; i < SUBFRAMES; i++) { > + if (pitch_lag > 162) { > + fparam->pitch_lag[i] += 162 - 23; > + } else { > + tmp = pitch_lag - 23; > + if (tmp < 36) > + tmp = 36; > + fparam->pitch_lag[i] += tmp; > + } > + pitch_lag = fparam->pitch_lag[i]; > + } > +} > + > +static void dss_sp_unpack_filter(DssSpContext *p) > +{ > + int i; > + > + for (i = 0; i < 14; i++) > + p->lpc.filter[i] = dss_sp_filter_cb[i][p->fparam.filter_idx[i]]; > +} > + > +static void dss_sp_convert_coeffs(LPCData *lpc, > + int32_t *coeffs) > +{ > + int a, a_plus, i; > + > + coeffs[0] = 0x2000; > + for (a = 0; a < 14; a++) { > + a_plus = a + 1; > + coeffs[a_plus] = lpc->filter[a] >> 2; > + if (a_plus / 2 >= 1) { > + for (i = 1; i <= a_plus / 2; i++) { > + int coeff_1, coeff_2, tmp; > + > + coeff_1 = coeffs[i]; > + coeff_2 = coeffs[a_plus - i]; > + > + tmp = DSS_FORMULA(coeff_1, lpc->filter[a], coeff_2); > + coeffs[i] = av_clip_int16(tmp); > + > + tmp = DSS_FORMULA(coeff_2, lpc->filter[a], coeff_1); > + coeffs[a_plus - i] = av_clip_int16(tmp); > + } > + } > + } > +} > + > +static void dss_sp_add_pulses(int32_t *vector_buf, > + const struct DssSpSubframe *sf) > +{ > + int i; > + > + for (i = 0; i < 7; i++) > + vector_buf[sf->pulse_pos[i]] += (dss_sp_fixed_cb_gain[sf->gain] * > + dss_sp_pulse_val[sf->pulse_val[i]] + > + 0x4000) >> 15; > +} > + > +static void dss_sp_gen_exc(int32_t *vector, int32_t *prev_exc, > + int pitch_lag, int gain) > +{ > + int i; > + > + /* do we actually need this check? we can use just [a3 - i % a3] > + * for both cases */ > + if (pitch_lag < 72) > + for (i = 0; i < 72; i++) > + vector[i] = prev_exc[pitch_lag - i % pitch_lag]; > + else > + for (i = 0; i < 72; i++) > + vector[i] = prev_exc[pitch_lag - i]; > + > + for (i = 0; i < 72; i++) { > + int tmp = gain * vector[i] >> 11; > + vector[i] = av_clip_int16(tmp); > + } > +} > + > +static void dss_sp_scale_vector(int32_t *vec, int bits, int size) > +{ > + int i; > + > + if (bits < 0) > + for (i = 0; i < size; i++) > + vec[i] = vec[i] >> -bits; > + else > + for (i = 0; i < size; i++) > + vec[i] = vec[i] << bits; > +} > + > +static void dss_sp_update_buf(int32_t *hist, int32_t *vector) > +{ > + int i; > + > + for (i = 114; i > 0; i--) > + vector[i + 72] = vector[i]; > + > + for (i = 0; i < 72; i++) > + vector[72 - i] = hist[i]; > +} > + > +static void dss_sp_shift_sq_sub(const int32_t *filter_buf, > + int32_t *error_buf, int32_t *dst) > +{ > + int a; > + > + for (a = 0; a < 72; a++) { > + int i, tmp; > + > + tmp = dst[a] * filter_buf[0]; > + > + for (i = 14; i > 0; i--) > + tmp -= error_buf[i] * filter_buf[i]; > + > + for (i = 14; i > 0; i--) > + error_buf[i] = error_buf[i - 1]; > + > + tmp = (tmp + 4096) >> 13; > + > + error_buf[1] = tmp; > + > + dst[a] = av_clip_int16(tmp); > + } > +} > + > +static void dss_sp_shift_sq_add(const int32_t *filter_buf, int32_t > *audio_buf, > + int32_t *dst) > +{ > + int a; > + > + for (a = 0; a < 72; a++) { > + int i, tmp = 0; > + > + audio_buf[0] = dst[a]; > + > + for (i = 14; i >= 0; i--) > + tmp += audio_buf[i] * filter_buf[i]; > + > + for (i = 14; i > 0; i--) > + audio_buf[i] = audio_buf[i - 1]; > + > + tmp = (tmp + 4096) >> 13; > + > + dst[a] = av_clip_int16(tmp); > + } > +} > + > +static void dss_sp_vec_mult(const int32_t *src, int32_t *dst, > + const int32_t *mult) > +{ > + int i; > + > + dst[0] = src[0]; > + > + for (i = 1; i < 15; i++) > + dst[i] = (src[i] * mult[i] + 0x4000) >> 15; > +} > + > +static int dss_sp_get_normalize_bits(int32_t *vector_buf, int16_t size) > +{ > + unsigned int val; > + int max_val; > + int i; > + > + val = 1; > + for (i = 0; i < size; i++) > + val |= FFABS(vector_buf[i]); > + > + for (max_val = 0; val <= 0x4000; ++max_val) > + val *= 2; > + return max_val; > +} > + > +static int dss_sp_vector_sum(DssSpContext *p, int size) > +{ > + int i, sum = 0; > + for (i = 0; i < size; i++) > + sum += FFABS(p->vector_buf[i]); > + return sum; > +} > + > +static void dss_sp_sf_synthesis(DssSpContext *p, int32_t lpc_filter, > + int32_t *dst, int size) > +{ > + int32_t tmp_buf[15]; > + int32_t noise[72]; > + int bias, vsum_2 = 0, vsum_1 = 0, v36, normalize_bits; > + int i, tmp; > + > + if (size > 0) { > + vsum_1 = dss_sp_vector_sum(p, size); > + > + if (vsum_1 > 0xFFFFF) > + vsum_1 = 0xFFFFF; > + } > + > + normalize_bits = dss_sp_get_normalize_bits(p->vector_buf, size); > + > + dss_sp_scale_vector(p->vector_buf, normalize_bits - 3, size); > + dss_sp_scale_vector(p->audio_buf, normalize_bits, 15); > + dss_sp_scale_vector(p->err_buf1, normalize_bits, 15); > + > + v36 = p->err_buf1[1]; > + > + dss_sp_vec_mult(p->filter, tmp_buf, > + binary_decreasing_array); > + dss_sp_shift_sq_add(tmp_buf, p->audio_buf, > + p->vector_buf); > + > + dss_sp_vec_mult(p->filter, tmp_buf, > + dss_sp_unc_decreasing_array); > + dss_sp_shift_sq_sub(tmp_buf, > + p->err_buf1, p->vector_buf); > + > + /* lpc_filter can be negative */ > + lpc_filter = lpc_filter >> 1; > + if (lpc_filter >= 0) > + lpc_filter = 0; > + > + if (size > 1) { > + for (i = size - 1; i > 0; i--) { > + tmp = DSS_FORMULA(p->vector_buf[i], lpc_filter, > + p->vector_buf[i - 1]); > + p->vector_buf[i] = av_clip_int16(tmp); > + } > + } > + > + tmp = DSS_FORMULA(p->vector_buf[0], lpc_filter, v36); > + p->vector_buf[0] = av_clip_int16(tmp); > + > + dss_sp_scale_vector(p->vector_buf, -normalize_bits, size); > + dss_sp_scale_vector(p->audio_buf, -normalize_bits, 15); > + dss_sp_scale_vector(p->err_buf1, -normalize_bits, 15); > + > + if (size > 0) > + vsum_2 = dss_sp_vector_sum(p, size); > + > + if (vsum_2 >= 0x40) > + tmp = (vsum_1 << 11) / vsum_2; > + else > + tmp = 1; > + > + bias = 409 * tmp >> 15 << 15; > + tmp = (bias + 32358 * p->noise_state) >> 15; > + noise[0] = av_clip_int16(tmp); > + > + for (i = 1; i < size; i++) { > + tmp = (bias + 32358 * noise[i - 1]) >> 15; > + noise[i] = av_clip_int16(tmp); > + } > + > + p->noise_state = noise[size - 1]; > + for (i = 0; i < size; i++) { > + tmp = (p->vector_buf[i] * noise[i]) >> 11; > + dst[i] = av_clip_int16(tmp); > + } > +} > + > +static void dss_sp_update_state(DssSpContext *p, int32_t *dst) > +{ > + int i, offset, counter, a = 0; > + > + for (i = 0; i < 6; i++) > + p->excitation[i] = p->excitation[288 + i]; > + > + for (i = 0; i < 72 * SUBFRAMES; i++) > + p->excitation[6 + i] = dst[i]; > + > + offset = 6; > + counter = 0; > + do { > + int tmp = 0; > + > + for (i = 0; i < 6; i++) > + tmp += p->excitation[offset--] * dss_sp_sinc[a + i * 11]; > + > + offset += 7; > + > + tmp >>= 15; > + dst[counter] = av_clip_int16(tmp); > + > + counter++; > + > + a = (a + 1) % 11; > + if (!a) > + offset++; > + } while (offset < FF_ARRAY_ELEMS(p->excitation)); > +} > + > +static void dss_sp_32to16bit(int16_t *dst, int32_t *src, int size) > +{ > + int i; > + > + for (i = 0; i < size; i++) > + dst[i] = av_clip_int16(src[i]); > +} > + > +static int dss_sp_decode_one_frame(DssSpContext *p, > + int16_t *abuf_dst, const uint8_t > *abuff_src) > +{ > + int i, j; > + > + dss_sp_unpack_coeffs(p, abuff_src); > + > + dss_sp_unpack_filter(p); > + > + dss_sp_convert_coeffs(&p->lpc, p->filter); > + > + for (j = 0; j < SUBFRAMES; j++) { > + dss_sp_gen_exc(p->vector_buf, p->history, > + p->fparam.pitch_lag[j], > + dss_sp_adaptive_gain[p->fparam.sf_adaptive_gain[j]]); > + > + dss_sp_add_pulses(p->vector_buf, &p->fparam.sf[j]); > + > + dss_sp_update_buf(p->vector_buf, p->history); > + > + for (i = 0; i < 72; i++) > + p->vector_buf[i] = p->history[72 - i]; > + > + dss_sp_shift_sq_sub(p->filter, > + p->err_buf2, p->vector_buf); > + > + dss_sp_sf_synthesis(p, p->lpc.filter[0], > + &p->working_buffer[j][0], 72); > + } > + > + dss_sp_update_state(p, &p->working_buffer[0][0]); > + > + dss_sp_32to16bit(abuf_dst, > + &p->working_buffer[0][0], 264); > + return 0; > +} > + > +static int dss_sp_decode_frame(AVCodecContext *avctx, void *data, > + int *got_frame_ptr, AVPacket *avpkt) > +{ > + DssSpContext *p = avctx->priv_data; > + AVFrame *frame = data; > + const uint8_t *buf = avpkt->data; > + int buf_size = avpkt->size; > + > + int16_t *out; > + int ret; > + > + if (buf_size < DSS_SP_FRAME_SIZE) { > + if (buf_size) > + av_log(avctx, AV_LOG_WARNING, > + "Expected %d bytes, got %d - skipping packet\n", > + DSS_SP_FRAME_SIZE, buf_size); > + *got_frame_ptr = 0; > + return AVERROR_INVALIDDATA; > + } > + > + frame->nb_samples = DSS_SP_SAMPLE_COUNT; > + if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) { > + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); > + return ret; > + } > + > + out = (int16_t *)frame->data[0]; > + > + dss_sp_decode_one_frame(p, out, buf); > + > + *got_frame_ptr = 1; > + > + return DSS_SP_FRAME_SIZE; > +} > + > +AVCodec ff_dss_sp_decoder = { > + .name = "dss_sp", > + .long_name = NULL_IF_CONFIG_SMALL("DSS SP"), > + .type = AVMEDIA_TYPE_AUDIO, > + .id = AV_CODEC_ID_DSS_SP, > + .priv_data_size = sizeof(DssSpContext), > + .init = dss_sp_decode_init, > + .decode = dss_sp_decode_frame, > + .capabilities = CODEC_CAP_DR1, > +}; > diff --git a/libavcodec/version.h b/libavcodec/version.h > index fd774a7..f51c9d7 100644 > --- a/libavcodec/version.h > +++ b/libavcodec/version.h > @@ -29,7 +29,7 @@ > #include "libavutil/version.h" > > #define LIBAVCODEC_VERSION_MAJOR 56 > -#define LIBAVCODEC_VERSION_MINOR 6 > +#define LIBAVCODEC_VERSION_MINOR 7 > #define LIBAVCODEC_VERSION_MICRO 0 > > #define LIBAVCODEC_VERSION_INT AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \ > -- > 1.9.3 (Apple Git-50) >
Does anyone have any more comments on the set? -- Vittorio _______________________________________________ libav-devel mailing list [email protected] https://lists.libav.org/mailman/listinfo/libav-devel
