/* * G.726 ADPCM audio codec * Copyright (c) 2004 Roman Shaposhnik * * This is a very straightforward rendition of the G.726 * Section 4 "Computational Details". * * This file is part of FFmpeg. * * FFmpeg 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. * * FFmpeg 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 FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ #include <limits.h> #include "libavutil/avassert.h" #include "libavutil/opt.h" #include "avcodec.h" #include "internal.h" #include "get_bits.h" #include "put_bits.h" /** * G.726 11bit float. * G.726 Standard uses rather odd 11bit floating point arithmentic for * numerous occasions. It's a mystery to me why they did it this way * instead of simply using 32bit integer arithmetic. */ typedef struct Float11 { uint8_t sign; /**< 1bit sign */ uint8_t exp; /**< 4bit exponent */ uint8_t mant; /**< 6bit mantissa */ } Float11; static inline Float11* i2f(int i, Float11* f) { f->sign = (i < 0); if (f->sign) i = -i; f->exp = av_log2_16bit(i) + !!i; f->mant = i? (i<<6) >> f->exp : 1<<5; return f; } static inline int16_t mult(Float11* f1, Float11* f2) { int res, exp; exp = f1->exp + f2->exp; res = (((f1->mant * f2->mant) + 0x30) >> 4); res = exp > 19 ? res << (exp - 19) : res >> (19 - exp); return (f1->sign ^ f2->sign) ? -res : res; } static inline int sgn(int value) { return (value < 0) ? -1 : 1; } typedef struct G726Tables { const int* quant; /**< quantization table */ const int16_t* iquant; /**< inverse quantization table */ const int16_t* W; /**< special table #1 ;-) */ const uint8_t* F; /**< special table #2 */ } G726Tables; typedef struct G726Context { AVClass *class; AVFrame frame; G726Tables tbls; /**< static tables needed for computation */ Float11 sr[2]; /**< prev. reconstructed samples */ Float11 dq[6]; /**< prev. difference */ int a[2]; /**< second order predictor coeffs */ int b[6]; /**< sixth order predictor coeffs */ int pk[2]; /**< signs of prev. 2 sez + dq */ int ap; /**< scale factor control */ int yu; /**< fast scale factor */ int yl; /**< slow scale factor */ int dms; /**< short average magnitude of F[i] */ int dml; /**< long average magnitude of F[i] */ int td; /**< tone detect */ int se; /**< estimated signal for the next iteration */ int sez; /**< estimated second order prediction */ int y; /**< quantizer scaling factor for the next iteration */ int code_size; } G726Context; static const int quant_tbl16[] = /**< 16kbit/s 2bits per sample */ { 260, INT_MAX }; static const int16_t iquant_tbl16[] = { 116, 365, 365, 116 }; static const int16_t W_tbl16[] = { -22, 439, 439, -22 }; static const uint8_t F_tbl16[] = { 0, 7, 7, 0 }; static const int quant_tbl24[] = /**< 24kbit/s 3bits per sample */ { 7, 217, 330, INT_MAX }; static const int16_t iquant_tbl24[] = { INT16_MIN, 135, 273, 373, 373, 273, 135, INT16_MIN }; static const int16_t W_tbl24[] = { -4, 30, 137, 582, 582, 137, 30, -4 }; static const uint8_t F_tbl24[] = { 0, 1, 2, 7, 7, 2, 1, 0 }; static const int quant_tbl32[] = /**< 32kbit/s 4bits per sample */ { -125, 79, 177, 245, 299, 348, 399, INT_MAX }; static const int16_t iquant_tbl32[] = { INT16_MIN, 4, 135, 213, 273, 323, 373, 425, 425, 373, 323, 273, 213, 135, 4, INT16_MIN }; static const int16_t W_tbl32[] = { -12, 18, 41, 64, 112, 198, 355, 1122, 1122, 355, 198, 112, 64, 41, 18, -12}; static const uint8_t F_tbl32[] = { 0, 0, 0, 1, 1, 1, 3, 7, 7, 3, 1, 1, 1, 0, 0, 0 }; static const int quant_tbl40[] = /**< 40kbit/s 5bits per sample */ { -122, -16, 67, 138, 197, 249, 297, 338, 377, 412, 444, 474, 501, 527, 552, INT_MAX }; static const int16_t iquant_tbl40[] = { INT16_MIN, -66, 28, 104, 169, 224, 274, 318, 358, 395, 429, 459, 488, 514, 539, 566, 566, 539, 514, 488, 459, 429, 395, 358, 318, 274, 224, 169, 104, 28, -66, INT16_MIN }; static const int16_t W_tbl40[] = { 14, 14, 24, 39, 40, 41, 58, 100, 141, 179, 219, 280, 358, 440, 529, 696, 696, 529, 440, 358, 280, 219, 179, 141, 100, 58, 41, 40, 39, 24, 14, 14 }; static const uint8_t F_tbl40[] = { 0, 0, 0, 0, 0, 1, 1, 1, 1, 1, 2, 3, 4, 5, 6, 6, 6, 6, 5, 4, 3, 2, 1, 1, 1, 1, 1, 0, 0, 0, 0, 0 }; static const G726Tables G726Tables_pool[] = {{ quant_tbl16, iquant_tbl16, W_tbl16, F_tbl16 }, { quant_tbl24, iquant_tbl24, W_tbl24, F_tbl24 }, { quant_tbl32, iquant_tbl32, W_tbl32, F_tbl32 }, { quant_tbl40, iquant_tbl40, W_tbl40, F_tbl40 }}; /** * Para 4.2.2 page 18: Adaptive quantizer. */ static inline uint8_t quant(G726Context* c, int d) { int sign, exp, i, dln; sign = i = 0; if (d < 0) { sign = 1; d = -d; } exp = av_log2_16bit(d); dln = ((exp<<7) + (((d<<7)>>exp)&0x7f)) - (c->y>>2); while (c->tbls.quant[i] < INT_MAX && c->tbls.quant[i] < dln) ++i; if (sign) i = ~i; if (c->code_size != 2 && i == 0) /* I'm not sure this is a good idea */ i = 0xff; return i; } /** * Para 4.2.3 page 22: Inverse adaptive quantizer. */ static inline int16_t inverse_quant(G726Context* c, int i) { int dql, dex, dqt; dql = c->tbls.iquant[i] + (c->y >> 2); dex = (dql>>7) & 0xf; /* 4bit exponent */ dqt = (1<<7) + (dql & 0x7f); /* log2 -> linear */ return (dql < 0) ? 0 : ((dqt<<dex) >> 7); } static int16_t g726_decode(G726Context* c, int I) { int dq, re_signal, pk0, fa1, i, tr, ylint, ylfrac, thr2, al, dq0; Float11 f; int I_sig= I >> (c->code_size - 1); dq = inverse_quant(c, I); /* Transition detect */ ylint = (c->yl >> 15); ylfrac = (c->yl >> 10) & 0x1f; thr2 = (ylint > 9) ? 0x1f << 10 : (0x20 + ylfrac) << ylint; tr= (c->td == 1 && dq > ((3*thr2)>>2)); if (I_sig) /* get the sign */ dq = -dq; re_signal = c->se + dq; /* Update second order predictor coefficient A2 and A1 */ pk0 = (c->sez + dq) ? sgn(c->sez + dq) : 0; dq0 = dq ? sgn(dq) : 0; if (tr) { c->a[0] = 0; c->a[1] = 0; for (i=0; i<6; i++) c->b[i] = 0; } else { /* This is a bit crazy, but it really is +255 not +256 */ fa1 = av_clip((-c->a[0]*c->pk[0]*pk0)>>5, -256, 255); c->a[1] += 128*pk0*c->pk[1] + fa1 - (c->a[1]>>7); c->a[1] = av_clip(c->a[1], -12288, 12288); c->a[0] += 64*3*pk0*c->pk[0] - (c->a[0] >> 8); c->a[0] = av_clip(c->a[0], -(15360 - c->a[1]), 15360 - c->a[1]); for (i=0; i<6; i++) c->b[i] += 128*dq0*sgn(-c->dq[i].sign) - (c->b[i]>>8); } /* Update Dq and Sr and Pk */ c->pk[1] = c->pk[0]; c->pk[0] = pk0 ? pk0 : 1; c->sr[1] = c->sr[0]; i2f(re_signal, &c->sr[0]); for (i=5; i>0; i--) c->dq[i] = c->dq[i-1]; i2f(dq, &c->dq[0]); c->dq[0].sign = I_sig; /* Isn't it crazy ?!?! */ c->td = c->a[1] < -11776; /* Update Ap */ c->dms += (c->tbls.F[I]<<4) + ((- c->dms) >> 5); c->dml += (c->tbls.F[I]<<4) + ((- c->dml) >> 7); if (tr) c->ap = 256; else { c->ap += (-c->ap) >> 4; if (c->y <= 1535 || c->td || abs((c->dms << 2) - c->dml) >= (c->dml >> 3)) c->ap += 0x20; } /* Update Yu and Yl */ c->yu = av_clip(c->y + c->tbls.W[I] + ((-c->y)>>5), 544, 5120); c->yl += c->yu + ((-c->yl)>>6); /* Next iteration for Y */ al = (c->ap >= 256) ? 1<<6 : c->ap >> 2; c->y = (c->yl + (c->yu - (c->yl>>6))*al) >> 6; /* Next iteration for SE and SEZ */ c->se = 0; for (i=0; i<6; i++) c->se += mult(i2f(c->b[i] >> 2, &f), &c->dq[i]); c->sez = c->se >> 1; for (i=0; i<2; i++) c->se += mult(i2f(c->a[i] >> 2, &f), &c->sr[i]); c->se >>= 1; return av_clip(re_signal << 2, -0xffff, 0xffff); } static av_cold int g726_reset(G726Context *c) { int i; c->tbls = G726Tables_pool[c->code_size - 2]; for (i=0; i<2; i++) { c->sr[i].mant = 1<<5; c->pk[i] = 1; } for (i=0; i<6; i++) { c->dq[i].mant = 1<<5; } c->yu = 544; c->yl = 34816; c->y = 544; return 0; } #if CONFIG_ADPCM_G726_ENCODER static int16_t g726_encode(G726Context* c, int16_t sig) { uint8_t i; i = quant(c, sig/4 - c->se) & ((1<<c->code_size) - 1); g726_decode(c, i); return i; } /* Interfacing to the libavcodec */ static av_cold int g726_encode_init(AVCodecContext *avctx) { G726Context* c = avctx->priv_data; if (avctx->strict_std_compliance > FF_COMPLIANCE_UNOFFICIAL && avctx->sample_rate != 8000) { av_log(avctx, AV_LOG_ERROR, "Sample rates other than 8kHz are not " "allowed when the compliance level is higher than unofficial. " "Resample or reduce the compliance level.\n"); return AVERROR(EINVAL); } av_assert0(avctx->sample_rate > 0); if(avctx->channels != 1){ av_log(avctx, AV_LOG_ERROR, "Only mono is supported\n"); return AVERROR(EINVAL); } if (avctx->bit_rate) c->code_size = (avctx->bit_rate + avctx->sample_rate/2) / avctx->sample_rate; c->code_size = av_clip(c->code_size, 2, 5); avctx->bit_rate = c->code_size * avctx->sample_rate; avctx->bits_per_coded_sample = c->code_size; g726_reset(c); #if FF_API_OLD_ENCODE_AUDIO avctx->coded_frame = avcodec_alloc_frame(); if (!avctx->coded_frame) return AVERROR(ENOMEM); avctx->coded_frame->key_frame = 1; #endif /* select a frame size that will end on a byte boundary and have a size of approximately 1024 bytes */ avctx->frame_size = ((int[]){ 4096, 2736, 2048, 1640 })[c->code_size - 2]; return 0; } #if FF_API_OLD_ENCODE_AUDIO static av_cold int g726_encode_close(AVCodecContext *avctx) { av_freep(&avctx->coded_frame); return 0; } #endif static int g726_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr) { G726Context *c = avctx->priv_data; const int16_t *samples = (const int16_t *)frame->data[0]; PutBitContext pb; int i, ret, out_size; out_size = (frame->nb_samples * c->code_size + 7) / 8; if ((ret = ff_alloc_packet2(avctx, avpkt, out_size))) return ret; init_put_bits(&pb, avpkt->data, avpkt->size); for (i = 0; i < frame->nb_samples; i++) put_bits(&pb, c->code_size, g726_encode(c, *samples++)); flush_put_bits(&pb); avpkt->size = out_size; *got_packet_ptr = 1; return 0; } #define OFFSET(x) offsetof(G726Context, x) #define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM static const AVOption options[] = { { "code_size", "Bits per code", OFFSET(code_size), AV_OPT_TYPE_INT, { 4 }, 2, 5, AE }, { NULL }, }; static const AVClass class = { .class_name = "g726", .item_name = av_default_item_name, .option = options, .version = LIBAVUTIL_VERSION_INT, }; static const AVCodecDefault defaults[] = { { "b", "0" }, { NULL }, }; AVCodec ff_adpcm_g726_encoder = { .name = "g726", .type = AVMEDIA_TYPE_AUDIO, .id = AV_CODEC_ID_ADPCM_G726, .priv_data_size = sizeof(G726Context), .init = g726_encode_init, .encode2 = g726_encode_frame, #if FF_API_OLD_ENCODE_AUDIO .close = g726_encode_close, #endif .capabilities = CODEC_CAP_SMALL_LAST_FRAME, .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE }, .long_name = NULL_IF_CONFIG_SMALL("G.726 ADPCM"), .priv_class = &class, .defaults = defaults, }; #endif #if CONFIG_ADPCM_G726_DECODER static av_cold int g726_decode_init(AVCodecContext *avctx) { G726Context* c = avctx->priv_data; if (avctx->strict_std_compliance >= FF_COMPLIANCE_STRICT && avctx->sample_rate != 8000) { av_log(avctx, AV_LOG_ERROR, "Only 8kHz sample rate is allowed when " "the compliance level is strict. Reduce the compliance level " "if you wish to decode the stream anyway.\n"); return AVERROR(EINVAL); } if(avctx->channels != 1){ av_log(avctx, AV_LOG_ERROR, "Only mono is supported\n"); return AVERROR(EINVAL); } c->code_size = avctx->bits_per_coded_sample; if (c->code_size < 2 || c->code_size > 5) { av_log(avctx, AV_LOG_ERROR, "Invalid number of bits %d\n", c->code_size); return AVERROR(EINVAL); } g726_reset(c); avctx->sample_fmt = AV_SAMPLE_FMT_S16; avcodec_get_frame_defaults(&c->frame); avctx->coded_frame = &c->frame; return 0; } static int g726_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; G726Context *c = avctx->priv_data; int16_t *samples; GetBitContext gb; int out_samples, ret; out_samples = buf_size * 8 / c->code_size; /* get output buffer */ c->frame.nb_samples = out_samples; if ((ret = avctx->get_buffer(avctx, &c->frame)) < 0) { av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); return ret; } samples = (int16_t *)c->frame.data[0]; init_get_bits(&gb, buf, buf_size * 8); while (out_samples--) *samples++ = g726_decode(c, get_bits(&gb, c->code_size)); if (get_bits_left(&gb) > 0) av_log(avctx, AV_LOG_ERROR, "Frame invalidly split, missing parser?\n"); *got_frame_ptr = 1; *(AVFrame *)data = c->frame; return buf_size; } static void g726_decode_flush(AVCodecContext *avctx) { G726Context *c = avctx->priv_data; g726_reset(c); } AVCodec ff_adpcm_g726_decoder = { .name = "g726", .type = AVMEDIA_TYPE_AUDIO, .id = AV_CODEC_ID_ADPCM_G726, .priv_data_size = sizeof(G726Context), .init = g726_decode_init, .decode = g726_decode_frame, .flush = g726_decode_flush, .capabilities = CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("G.726 ADPCM"), }; #endif