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  1. /*
  2.  * G.726 ADPCM audio codec
  3.  * Copyright (c) 2004 Roman Shaposhnik
  4.  *
  5.  * This is a very straightforward rendition of the G.726
  6.  * Section 4 "Computational Details".
  7.  *
  8.  * This file is part of FFmpeg.
  9.  *
  10.  * FFmpeg is free software; you can redistribute it and/or
  11.  * modify it under the terms of the GNU Lesser General Public
  12.  * License as published by the Free Software Foundation; either
  13.  * version 2.1 of the License, or (at your option) any later version.
  14.  *
  15.  * FFmpeg is distributed in the hope that it will be useful,
  16.  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  17.  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  18.  * Lesser General Public License for more details.
  19.  *
  20.  * You should have received a copy of the GNU Lesser General Public
  21.  * License along with FFmpeg; if not, write to the Free Software
  22.  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  23.  */
  24. #include <limits.h>
  25.  
  26. #include "libavutil/avassert.h"
  27. #include "libavutil/channel_layout.h"
  28. #include "libavutil/opt.h"
  29. #include "avcodec.h"
  30. #include "internal.h"
  31. #include "get_bits.h"
  32. #include "put_bits.h"
  33.  
  34. /**
  35.  * G.726 11bit float.
  36.  * G.726 Standard uses rather odd 11bit floating point arithmentic for
  37.  * numerous occasions. It's a mystery to me why they did it this way
  38.  * instead of simply using 32bit integer arithmetic.
  39.  */
  40. typedef struct Float11 {
  41.     uint8_t sign;   /**< 1bit sign */
  42.     uint8_t exp;    /**< 4bit exponent */
  43.     uint8_t mant;   /**< 6bit mantissa */
  44. } Float11;
  45.  
  46. static inline Float11* i2f(int i, Float11* f)
  47. {
  48.     f->sign = (i < 0);
  49.     if (f->sign)
  50.         i = -i;
  51.     f->exp = av_log2_16bit(i) + !!i;
  52.     f->mant = i? (i<<6) >> f->exp : 1<<5;
  53.     return f;
  54. }
  55.  
  56. static inline int16_t mult(Float11* f1, Float11* f2)
  57. {
  58.         int res, exp;
  59.  
  60.         exp = f1->exp + f2->exp;
  61.         res = (((f1->mant * f2->mant) + 0x30) >> 4);
  62.         res = exp > 19 ? res << (exp - 19) : res >> (19 - exp);
  63.         return (f1->sign ^ f2->sign) ? -res : res;
  64. }
  65.  
  66. static inline int sgn(int value)
  67. {
  68.     return (value < 0) ? -1 : 1;
  69. }
  70.  
  71. typedef struct G726Tables {
  72.     const int* quant;         /**< quantization table */
  73.     const int16_t* iquant;    /**< inverse quantization table */
  74.     const int16_t* W;         /**< special table #1 ;-) */
  75.     const uint8_t* F;         /**< special table #2 */
  76. } G726Tables;
  77.  
  78. typedef struct G726Context {
  79.     AVClass *class;
  80.     G726Tables tbls;    /**< static tables needed for computation */
  81.  
  82.     Float11 sr[2];      /**< prev. reconstructed samples */
  83.     Float11 dq[6];      /**< prev. difference */
  84.     int a[2];           /**< second order predictor coeffs */
  85.     int b[6];           /**< sixth order predictor coeffs */
  86.     int pk[2];          /**< signs of prev. 2 sez + dq */
  87.  
  88.     int ap;             /**< scale factor control */
  89.     int yu;             /**< fast scale factor */
  90.     int yl;             /**< slow scale factor */
  91.     int dms;            /**< short average magnitude of F[i] */
  92.     int dml;            /**< long average magnitude of F[i] */
  93.     int td;             /**< tone detect */
  94.  
  95.     int se;             /**< estimated signal for the next iteration */
  96.     int sez;            /**< estimated second order prediction */
  97.     int y;              /**< quantizer scaling factor for the next iteration */
  98.     int code_size;
  99.     int little_endian;  /**< little-endian bitstream as used in aiff and Sun AU */
  100. } G726Context;
  101.  
  102. static const int quant_tbl16[] =                  /**< 16kbit/s 2bits per sample */
  103.            { 260, INT_MAX };
  104. static const int16_t iquant_tbl16[] =
  105.            { 116, 365, 365, 116 };
  106. static const int16_t W_tbl16[] =
  107.            { -22, 439, 439, -22 };
  108. static const uint8_t F_tbl16[] =
  109.            { 0, 7, 7, 0 };
  110.  
  111. static const int quant_tbl24[] =                  /**< 24kbit/s 3bits per sample */
  112.            {  7, 217, 330, INT_MAX };
  113. static const int16_t iquant_tbl24[] =
  114.            { INT16_MIN, 135, 273, 373, 373, 273, 135, INT16_MIN };
  115. static const int16_t W_tbl24[] =
  116.            { -4,  30, 137, 582, 582, 137,  30, -4 };
  117. static const uint8_t F_tbl24[] =
  118.            { 0, 1, 2, 7, 7, 2, 1, 0 };
  119.  
  120. static const int quant_tbl32[] =                  /**< 32kbit/s 4bits per sample */
  121.            { -125,  79, 177, 245, 299, 348, 399, INT_MAX };
  122. static const int16_t iquant_tbl32[] =
  123.          { INT16_MIN,   4, 135, 213, 273, 323, 373, 425,
  124.                  425, 373, 323, 273, 213, 135,   4, INT16_MIN };
  125. static const int16_t W_tbl32[] =
  126.            { -12,  18,  41,  64, 112, 198, 355, 1122,
  127.             1122, 355, 198, 112,  64,  41,  18, -12};
  128. static const uint8_t F_tbl32[] =
  129.            { 0, 0, 0, 1, 1, 1, 3, 7, 7, 3, 1, 1, 1, 0, 0, 0 };
  130.  
  131. static const int quant_tbl40[] =                  /**< 40kbit/s 5bits per sample */
  132.            { -122, -16,  67, 138, 197, 249, 297, 338,
  133.               377, 412, 444, 474, 501, 527, 552, INT_MAX };
  134. static const int16_t iquant_tbl40[] =
  135.          { INT16_MIN, -66,  28, 104, 169, 224, 274, 318,
  136.                  358, 395, 429, 459, 488, 514, 539, 566,
  137.                  566, 539, 514, 488, 459, 429, 395, 358,
  138.                  318, 274, 224, 169, 104,  28, -66, INT16_MIN };
  139. static const int16_t W_tbl40[] =
  140.            {   14,  14,  24,  39,  40,  41,   58,  100,
  141.               141, 179, 219, 280, 358, 440,  529,  696,
  142.               696, 529, 440, 358, 280, 219,  179,  141,
  143.               100,  58,  41,  40,  39,  24,   14,   14 };
  144. static const uint8_t F_tbl40[] =
  145.            { 0, 0, 0, 0, 0, 1, 1, 1, 1, 1, 2, 3, 4, 5, 6, 6,
  146.              6, 6, 5, 4, 3, 2, 1, 1, 1, 1, 1, 0, 0, 0, 0, 0 };
  147.  
  148. static const G726Tables G726Tables_pool[] =
  149.            {{ quant_tbl16, iquant_tbl16, W_tbl16, F_tbl16 },
  150.             { quant_tbl24, iquant_tbl24, W_tbl24, F_tbl24 },
  151.             { quant_tbl32, iquant_tbl32, W_tbl32, F_tbl32 },
  152.             { quant_tbl40, iquant_tbl40, W_tbl40, F_tbl40 }};
  153.  
  154.  
  155. /**
  156.  * Para 4.2.2 page 18: Adaptive quantizer.
  157.  */
  158. static inline uint8_t quant(G726Context* c, int d)
  159. {
  160.     int sign, exp, i, dln;
  161.  
  162.     sign = i = 0;
  163.     if (d < 0) {
  164.         sign = 1;
  165.         d = -d;
  166.     }
  167.     exp = av_log2_16bit(d);
  168.     dln = ((exp<<7) + (((d<<7)>>exp)&0x7f)) - (c->y>>2);
  169.  
  170.     while (c->tbls.quant[i] < INT_MAX && c->tbls.quant[i] < dln)
  171.         ++i;
  172.  
  173.     if (sign)
  174.         i = ~i;
  175.     if (c->code_size != 2 && i == 0) /* I'm not sure this is a good idea */
  176.         i = 0xff;
  177.  
  178.     return i;
  179. }
  180.  
  181. /**
  182.  * Para 4.2.3 page 22: Inverse adaptive quantizer.
  183.  */
  184. static inline int16_t inverse_quant(G726Context* c, int i)
  185. {
  186.     int dql, dex, dqt;
  187.  
  188.     dql = c->tbls.iquant[i] + (c->y >> 2);
  189.     dex = (dql>>7) & 0xf;        /* 4bit exponent */
  190.     dqt = (1<<7) + (dql & 0x7f); /* log2 -> linear */
  191.     return (dql < 0) ? 0 : ((dqt<<dex) >> 7);
  192. }
  193.  
  194. static int16_t g726_decode(G726Context* c, int I)
  195. {
  196.     int dq, re_signal, pk0, fa1, i, tr, ylint, ylfrac, thr2, al, dq0;
  197.     Float11 f;
  198.     int I_sig= I >> (c->code_size - 1);
  199.  
  200.     dq = inverse_quant(c, I);
  201.  
  202.     /* Transition detect */
  203.     ylint = (c->yl >> 15);
  204.     ylfrac = (c->yl >> 10) & 0x1f;
  205.     thr2 = (ylint > 9) ? 0x1f << 10 : (0x20 + ylfrac) << ylint;
  206.     tr= (c->td == 1 && dq > ((3*thr2)>>2));
  207.  
  208.     if (I_sig)  /* get the sign */
  209.         dq = -dq;
  210.     re_signal = c->se + dq;
  211.  
  212.     /* Update second order predictor coefficient A2 and A1 */
  213.     pk0 = (c->sez + dq) ? sgn(c->sez + dq) : 0;
  214.     dq0 = dq ? sgn(dq) : 0;
  215.     if (tr) {
  216.         c->a[0] = 0;
  217.         c->a[1] = 0;
  218.         for (i=0; i<6; i++)
  219.             c->b[i] = 0;
  220.     } else {
  221.         /* This is a bit crazy, but it really is +255 not +256 */
  222.         fa1 = av_clip((-c->a[0]*c->pk[0]*pk0)>>5, -256, 255);
  223.  
  224.         c->a[1] += 128*pk0*c->pk[1] + fa1 - (c->a[1]>>7);
  225.         c->a[1] = av_clip(c->a[1], -12288, 12288);
  226.         c->a[0] += 64*3*pk0*c->pk[0] - (c->a[0] >> 8);
  227.         c->a[0] = av_clip(c->a[0], -(15360 - c->a[1]), 15360 - c->a[1]);
  228.  
  229.         for (i=0; i<6; i++)
  230.             c->b[i] += 128*dq0*sgn(-c->dq[i].sign) - (c->b[i]>>8);
  231.     }
  232.  
  233.     /* Update Dq and Sr and Pk */
  234.     c->pk[1] = c->pk[0];
  235.     c->pk[0] = pk0 ? pk0 : 1;
  236.     c->sr[1] = c->sr[0];
  237.     i2f(re_signal, &c->sr[0]);
  238.     for (i=5; i>0; i--)
  239.         c->dq[i] = c->dq[i-1];
  240.     i2f(dq, &c->dq[0]);
  241.     c->dq[0].sign = I_sig; /* Isn't it crazy ?!?! */
  242.  
  243.     c->td = c->a[1] < -11776;
  244.  
  245.     /* Update Ap */
  246.     c->dms += (c->tbls.F[I]<<4) + ((- c->dms) >> 5);
  247.     c->dml += (c->tbls.F[I]<<4) + ((- c->dml) >> 7);
  248.     if (tr)
  249.         c->ap = 256;
  250.     else {
  251.         c->ap += (-c->ap) >> 4;
  252.         if (c->y <= 1535 || c->td || abs((c->dms << 2) - c->dml) >= (c->dml >> 3))
  253.             c->ap += 0x20;
  254.     }
  255.  
  256.     /* Update Yu and Yl */
  257.     c->yu = av_clip(c->y + c->tbls.W[I] + ((-c->y)>>5), 544, 5120);
  258.     c->yl += c->yu + ((-c->yl)>>6);
  259.  
  260.     /* Next iteration for Y */
  261.     al = (c->ap >= 256) ? 1<<6 : c->ap >> 2;
  262.     c->y = (c->yl + (c->yu - (c->yl>>6))*al) >> 6;
  263.  
  264.     /* Next iteration for SE and SEZ */
  265.     c->se = 0;
  266.     for (i=0; i<6; i++)
  267.         c->se += mult(i2f(c->b[i] >> 2, &f), &c->dq[i]);
  268.     c->sez = c->se >> 1;
  269.     for (i=0; i<2; i++)
  270.         c->se += mult(i2f(c->a[i] >> 2, &f), &c->sr[i]);
  271.     c->se >>= 1;
  272.  
  273.     return av_clip(re_signal << 2, -0xffff, 0xffff);
  274. }
  275.  
  276. static av_cold int g726_reset(G726Context *c)
  277. {
  278.     int i;
  279.  
  280.     c->tbls = G726Tables_pool[c->code_size - 2];
  281.     for (i=0; i<2; i++) {
  282.         c->sr[i].mant = 1<<5;
  283.         c->pk[i] = 1;
  284.     }
  285.     for (i=0; i<6; i++) {
  286.         c->dq[i].mant = 1<<5;
  287.     }
  288.     c->yu = 544;
  289.     c->yl = 34816;
  290.  
  291.     c->y = 544;
  292.  
  293.     return 0;
  294. }
  295.  
  296. #if CONFIG_ADPCM_G726_ENCODER
  297. static int16_t g726_encode(G726Context* c, int16_t sig)
  298. {
  299.     uint8_t i;
  300.  
  301.     i = quant(c, sig/4 - c->se) & ((1<<c->code_size) - 1);
  302.     g726_decode(c, i);
  303.     return i;
  304. }
  305.  
  306. /* Interfacing to the libavcodec */
  307.  
  308. static av_cold int g726_encode_init(AVCodecContext *avctx)
  309. {
  310.     G726Context* c = avctx->priv_data;
  311.  
  312.     if (avctx->strict_std_compliance > FF_COMPLIANCE_UNOFFICIAL &&
  313.         avctx->sample_rate != 8000) {
  314.         av_log(avctx, AV_LOG_ERROR, "Sample rates other than 8kHz are not "
  315.                "allowed when the compliance level is higher than unofficial. "
  316.                "Resample or reduce the compliance level.\n");
  317.         return AVERROR(EINVAL);
  318.     }
  319.     av_assert0(avctx->sample_rate > 0);
  320.  
  321.     if(avctx->channels != 1){
  322.         av_log(avctx, AV_LOG_ERROR, "Only mono is supported\n");
  323.         return AVERROR(EINVAL);
  324.     }
  325.  
  326.     if (avctx->bit_rate)
  327.         c->code_size = (avctx->bit_rate + avctx->sample_rate/2) / avctx->sample_rate;
  328.  
  329.     c->code_size = av_clip(c->code_size, 2, 5);
  330.     avctx->bit_rate = c->code_size * avctx->sample_rate;
  331.     avctx->bits_per_coded_sample = c->code_size;
  332.  
  333.     g726_reset(c);
  334.  
  335.     /* select a frame size that will end on a byte boundary and have a size of
  336.        approximately 1024 bytes */
  337.     avctx->frame_size = ((int[]){ 4096, 2736, 2048, 1640 })[c->code_size - 2];
  338.  
  339.     return 0;
  340. }
  341.  
  342. static int g726_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
  343.                              const AVFrame *frame, int *got_packet_ptr)
  344. {
  345.     G726Context *c = avctx->priv_data;
  346.     const int16_t *samples = (const int16_t *)frame->data[0];
  347.     PutBitContext pb;
  348.     int i, ret, out_size;
  349.  
  350.     out_size = (frame->nb_samples * c->code_size + 7) / 8;
  351.     if ((ret = ff_alloc_packet2(avctx, avpkt, out_size)) < 0)
  352.         return ret;
  353.     init_put_bits(&pb, avpkt->data, avpkt->size);
  354.  
  355.     for (i = 0; i < frame->nb_samples; i++)
  356.         put_bits(&pb, c->code_size, g726_encode(c, *samples++));
  357.  
  358.     flush_put_bits(&pb);
  359.  
  360.     avpkt->size = out_size;
  361.     *got_packet_ptr = 1;
  362.     return 0;
  363. }
  364.  
  365. #define OFFSET(x) offsetof(G726Context, x)
  366. #define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
  367. static const AVOption options[] = {
  368.     { "code_size", "Bits per code", OFFSET(code_size), AV_OPT_TYPE_INT, { .i64 = 4 }, 2, 5, AE },
  369.     { NULL },
  370. };
  371.  
  372. static const AVClass g726_class = {
  373.     .class_name = "g726",
  374.     .item_name  = av_default_item_name,
  375.     .option     = options,
  376.     .version    = LIBAVUTIL_VERSION_INT,
  377. };
  378.  
  379. static const AVCodecDefault defaults[] = {
  380.     { "b", "0" },
  381.     { NULL },
  382. };
  383.  
  384. AVCodec ff_adpcm_g726_encoder = {
  385.     .name           = "g726",
  386.     .long_name      = NULL_IF_CONFIG_SMALL("G.726 ADPCM"),
  387.     .type           = AVMEDIA_TYPE_AUDIO,
  388.     .id             = AV_CODEC_ID_ADPCM_G726,
  389.     .priv_data_size = sizeof(G726Context),
  390.     .init           = g726_encode_init,
  391.     .encode2        = g726_encode_frame,
  392.     .capabilities   = CODEC_CAP_SMALL_LAST_FRAME,
  393.     .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
  394.                                                      AV_SAMPLE_FMT_NONE },
  395.     .priv_class     = &g726_class,
  396.     .defaults       = defaults,
  397. };
  398. #endif
  399.  
  400. #if CONFIG_ADPCM_G726_DECODER || CONFIG_ADPCM_G726LE_DECODER
  401. static av_cold int g726_decode_init(AVCodecContext *avctx)
  402. {
  403.     G726Context* c = avctx->priv_data;
  404.  
  405.     if(avctx->channels > 1){
  406.         avpriv_request_sample(avctx, "Decoding more than one channel");
  407.         return AVERROR_PATCHWELCOME;
  408.     }
  409.     avctx->channels       = 1;
  410.     avctx->channel_layout = AV_CH_LAYOUT_MONO;
  411.  
  412.     c->little_endian = !strcmp(avctx->codec->name, "g726le");
  413.  
  414.     c->code_size = avctx->bits_per_coded_sample;
  415.     if (c->code_size < 2 || c->code_size > 5) {
  416.         av_log(avctx, AV_LOG_ERROR, "Invalid number of bits %d\n", c->code_size);
  417.         return AVERROR(EINVAL);
  418.     }
  419.     g726_reset(c);
  420.  
  421.     avctx->sample_fmt = AV_SAMPLE_FMT_S16;
  422.  
  423.     return 0;
  424. }
  425.  
  426. static int g726_decode_frame(AVCodecContext *avctx, void *data,
  427.                              int *got_frame_ptr, AVPacket *avpkt)
  428. {
  429.     AVFrame *frame     = data;
  430.     const uint8_t *buf = avpkt->data;
  431.     int buf_size = avpkt->size;
  432.     G726Context *c = avctx->priv_data;
  433.     int16_t *samples;
  434.     GetBitContext gb;
  435.     int out_samples, ret;
  436.  
  437.     out_samples = buf_size * 8 / c->code_size;
  438.  
  439.     /* get output buffer */
  440.     frame->nb_samples = out_samples;
  441.     if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
  442.         return ret;
  443.     samples = (int16_t *)frame->data[0];
  444.  
  445.     init_get_bits(&gb, buf, buf_size * 8);
  446.  
  447.     while (out_samples--)
  448.         *samples++ = g726_decode(c, c->little_endian ?
  449.                                     get_bits_le(&gb, c->code_size) :
  450.                                     get_bits(&gb, c->code_size));
  451.  
  452.     if (get_bits_left(&gb) > 0)
  453.         av_log(avctx, AV_LOG_ERROR, "Frame invalidly split, missing parser?\n");
  454.  
  455.     *got_frame_ptr = 1;
  456.  
  457.     return buf_size;
  458. }
  459.  
  460. static void g726_decode_flush(AVCodecContext *avctx)
  461. {
  462.     G726Context *c = avctx->priv_data;
  463.     g726_reset(c);
  464. }
  465. #endif
  466.  
  467. #if CONFIG_ADPCM_G726_DECODER
  468. AVCodec ff_adpcm_g726_decoder = {
  469.     .name           = "g726",
  470.     .long_name      = NULL_IF_CONFIG_SMALL("G.726 ADPCM"),
  471.     .type           = AVMEDIA_TYPE_AUDIO,
  472.     .id             = AV_CODEC_ID_ADPCM_G726,
  473.     .priv_data_size = sizeof(G726Context),
  474.     .init           = g726_decode_init,
  475.     .decode         = g726_decode_frame,
  476.     .flush          = g726_decode_flush,
  477.     .capabilities   = CODEC_CAP_DR1,
  478. };
  479. #endif
  480.  
  481. #if CONFIG_ADPCM_G726LE_DECODER
  482. AVCodec ff_adpcm_g726le_decoder = {
  483.     .name           = "g726le",
  484.     .type           = AVMEDIA_TYPE_AUDIO,
  485.     .id             = AV_CODEC_ID_ADPCM_G726LE,
  486.     .priv_data_size = sizeof(G726Context),
  487.     .init           = g726_decode_init,
  488.     .decode         = g726_decode_frame,
  489.     .flush          = g726_decode_flush,
  490.     .capabilities   = CODEC_CAP_DR1,
  491.     .long_name      = NULL_IF_CONFIG_SMALL("G.726 ADPCM little-endian"),
  492. };
  493. #endif
  494.