From 8992cb1d0d07edc33d274f6d7924ecdf6f83d994 Mon Sep 17 00:00:00 2001 From: Tim Redfern Date: Thu, 5 Sep 2013 17:57:22 +0100 Subject: making act segmenter --- ffmpeg/libavcodec/g726.c | 468 +++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 468 insertions(+) create mode 100644 ffmpeg/libavcodec/g726.c (limited to 'ffmpeg/libavcodec/g726.c') diff --git a/ffmpeg/libavcodec/g726.c b/ffmpeg/libavcodec/g726.c new file mode 100644 index 0000000..58d0468 --- /dev/null +++ b/ffmpeg/libavcodec/g726.c @@ -0,0 +1,468 @@ +/* + * 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 + +#include "libavutil/avassert.h" +#include "libavutil/channel_layout.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; + 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<> 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<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); + + /* 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; +} + +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)) < 0) + 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, { .i64 = 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, + .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; + + avctx->channels = 1; + avctx->channel_layout = AV_CH_LAYOUT_MONO; + + 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; + + return 0; +} + +static int g726_decode_frame(AVCodecContext *avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) +{ + AVFrame *frame = data; + 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 */ + frame->nb_samples = out_samples; + if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) + return ret; + samples = (int16_t *)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; + + 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 -- cgit v1.2.3