From 165cc6fb9defcd79fd71c08167f3e8df26b058ff Mon Sep 17 00:00:00 2001 From: Vittorio Giovara <vittorio.giovara@gmail.com> Date: Mon, 23 Nov 2015 17:10:53 -0500 Subject: g723_1: Move sharable functions to a separate file Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com> --- libavcodec/Makefile | 4 +- libavcodec/g723_1.c | 267 ++++++++++++++++++++++++++++++++ libavcodec/g723_1.h | 141 ++++++++++++++++- libavcodec/g723_1dec.c | 404 +++++-------------------------------------------- 4 files changed, 443 insertions(+), 373 deletions(-) create mode 100644 libavcodec/g723_1.c diff --git a/libavcodec/Makefile b/libavcodec/Makefile index dfefab66f8..85738fa1f0 100644 --- a/libavcodec/Makefile +++ b/libavcodec/Makefile @@ -225,8 +225,8 @@ OBJS-$(CONFIG_FOURXM_DECODER) += 4xm.o OBJS-$(CONFIG_FRAPS_DECODER) += fraps.o OBJS-$(CONFIG_FRWU_DECODER) += frwu.o OBJS-$(CONFIG_G2M_DECODER) += g2meet.o elsdec.o -OBJS-$(CONFIG_G723_1_DECODER) += g723_1dec.o acelp_vectors.o \ - celp_filters.o +OBJS-$(CONFIG_G723_1_DECODER) += g723_1dec.o g723_1.o \ + acelp_vectors.o celp_filters.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/g723_1.c b/libavcodec/g723_1.c new file mode 100644 index 0000000000..af4777cc35 --- /dev/null +++ b/libavcodec/g723_1.c @@ -0,0 +1,267 @@ +/* + * G.723.1 compatible decoder + * Copyright (c) 2006 Benjamin Larsson + * Copyright (c) 2010 Mohamed Naufal Basheer + * + * 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 <stdint.h> + +#include "libavutil/common.h" + +#include "acelp_vectors.h" +#include "avcodec.h" +#include "celp_math.h" +#include "g723_1.h" + +int ff_g723_1_scale_vector(int16_t *dst, const int16_t *vector, int length) +{ + int bits, max = 0; + int i; + + for (i = 0; i < length; i++) + max |= FFABS(vector[i]); + + max = FFMIN(max, 0x7FFF); + bits = ff_g723_1_normalize_bits(max, 15); + + for (i = 0; i < length; i++) + dst[i] = vector[i] << bits >> 3; + + return bits - 3; +} + +int ff_g723_1_normalize_bits(int num, int width) +{ + return width - av_log2(num) - 1; +} + +int ff_g723_1_dot_product(const int16_t *a, const int16_t *b, int length) +{ + int i, sum = 0; + + for (i = 0; i < length; i++) { + int prod = a[i] * b[i]; + sum = av_sat_dadd32(sum, prod); + } + return sum; +} + +void ff_g723_1_get_residual(int16_t *residual, int16_t *prev_excitation, + int lag) +{ + int offset = PITCH_MAX - PITCH_ORDER / 2 - lag; + int i; + + residual[0] = prev_excitation[offset]; + residual[1] = prev_excitation[offset + 1]; + + offset += 2; + for (i = 2; i < SUBFRAME_LEN + PITCH_ORDER - 1; i++) + residual[i] = prev_excitation[offset + (i - 2) % lag]; +} + +void ff_g723_1_gen_dirac_train(int16_t *buf, int pitch_lag) +{ + int16_t vector[SUBFRAME_LEN]; + int i, j; + + memcpy(vector, buf, SUBFRAME_LEN * sizeof(*vector)); + for (i = pitch_lag; i < SUBFRAME_LEN; i += pitch_lag) { + for (j = 0; j < SUBFRAME_LEN - i; j++) + buf[i + j] += vector[j]; + } +} + +void ff_g723_1_gen_acb_excitation(int16_t *vector, int16_t *prev_excitation, + int pitch_lag, G723_1_Subframe *subfrm, + enum Rate cur_rate) +{ + int16_t residual[SUBFRAME_LEN + PITCH_ORDER - 1]; + const int16_t *cb_ptr; + int lag = pitch_lag + subfrm->ad_cb_lag - 1; + + int i; + int sum; + + ff_g723_1_get_residual(residual, prev_excitation, lag); + + /* Select quantization table */ + if (cur_rate == RATE_6300 && pitch_lag < SUBFRAME_LEN - 2) + cb_ptr = adaptive_cb_gain85; + else + cb_ptr = adaptive_cb_gain170; + + /* Calculate adaptive vector */ + cb_ptr += subfrm->ad_cb_gain * 20; + for (i = 0; i < SUBFRAME_LEN; i++) { + sum = ff_g723_1_dot_product(residual + i, cb_ptr, PITCH_ORDER); + vector[i] = av_sat_dadd32(1 << 15, sum) >> 16; + } +} + +/** + * Convert LSP frequencies to LPC coefficients. + * + * @param lpc buffer for LPC coefficients + */ +static void lsp2lpc(int16_t *lpc) +{ + int f1[LPC_ORDER / 2 + 1]; + int f2[LPC_ORDER / 2 + 1]; + int i, j; + + /* Calculate negative cosine */ + for (j = 0; j < LPC_ORDER; j++) { + int index = (lpc[j] >> 7) & 0x1FF; + int offset = lpc[j] & 0x7f; + int temp1 = cos_tab[index] << 16; + int temp2 = (cos_tab[index + 1] - cos_tab[index]) * + ((offset << 8) + 0x80) << 1; + + lpc[j] = -(av_sat_dadd32(1 << 15, temp1 + temp2) >> 16); + } + + /* + * Compute sum and difference polynomial coefficients + * (bitexact alternative to lsp2poly() in lsp.c) + */ + /* Initialize with values in Q28 */ + f1[0] = 1 << 28; + f1[1] = (lpc[0] << 14) + (lpc[2] << 14); + f1[2] = lpc[0] * lpc[2] + (2 << 28); + + f2[0] = 1 << 28; + f2[1] = (lpc[1] << 14) + (lpc[3] << 14); + f2[2] = lpc[1] * lpc[3] + (2 << 28); + + /* + * Calculate and scale the coefficients by 1/2 in + * each iteration for a final scaling factor of Q25 + */ + for (i = 2; i < LPC_ORDER / 2; i++) { + f1[i + 1] = f1[i - 1] + MULL2(f1[i], lpc[2 * i]); + f2[i + 1] = f2[i - 1] + MULL2(f2[i], lpc[2 * i + 1]); + + for (j = i; j >= 2; j--) { + f1[j] = MULL2(f1[j - 1], lpc[2 * i]) + + (f1[j] >> 1) + (f1[j - 2] >> 1); + f2[j] = MULL2(f2[j - 1], lpc[2 * i + 1]) + + (f2[j] >> 1) + (f2[j - 2] >> 1); + } + + f1[0] >>= 1; + f2[0] >>= 1; + f1[1] = ((lpc[2 * i] << 16 >> i) + f1[1]) >> 1; + f2[1] = ((lpc[2 * i + 1] << 16 >> i) + f2[1]) >> 1; + } + + /* Convert polynomial coefficients to LPC coefficients */ + for (i = 0; i < LPC_ORDER / 2; i++) { + int64_t ff1 = f1[i + 1] + f1[i]; + int64_t ff2 = f2[i + 1] - f2[i]; + + lpc[i] = av_clipl_int32(((ff1 + ff2) << 3) + + (1 << 15)) >> 16; + lpc[LPC_ORDER - i - 1] = av_clipl_int32(((ff1 - ff2) << 3) + + (1 << 15)) >> 16; + } +} + +void ff_g723_1_lsp_interpolate(int16_t *lpc, int16_t *cur_lsp, + int16_t *prev_lsp) +{ + int i; + int16_t *lpc_ptr = lpc; + + /* cur_lsp * 0.25 + prev_lsp * 0.75 */ + ff_acelp_weighted_vector_sum(lpc, cur_lsp, prev_lsp, + 4096, 12288, 1 << 13, 14, LPC_ORDER); + ff_acelp_weighted_vector_sum(lpc + LPC_ORDER, cur_lsp, prev_lsp, + 8192, 8192, 1 << 13, 14, LPC_ORDER); + ff_acelp_weighted_vector_sum(lpc + 2 * LPC_ORDER, cur_lsp, prev_lsp, + 12288, 4096, 1 << 13, 14, LPC_ORDER); + memcpy(lpc + 3 * LPC_ORDER, cur_lsp, LPC_ORDER * sizeof(*lpc)); + + for (i = 0; i < SUBFRAMES; i++) { + lsp2lpc(lpc_ptr); + lpc_ptr += LPC_ORDER; + } +} + +void ff_g723_1_inverse_quant(int16_t *cur_lsp, int16_t *prev_lsp, + uint8_t *lsp_index, int bad_frame) +{ + int min_dist, pred; + int i, j, temp, stable; + + /* Check for frame erasure */ + if (!bad_frame) { + min_dist = 0x100; + pred = 12288; + } else { + min_dist = 0x200; + pred = 23552; + lsp_index[0] = lsp_index[1] = lsp_index[2] = 0; + } + + /* Get the VQ table entry corresponding to the transmitted index */ + cur_lsp[0] = lsp_band0[lsp_index[0]][0]; + cur_lsp[1] = lsp_band0[lsp_index[0]][1]; + cur_lsp[2] = lsp_band0[lsp_index[0]][2]; + cur_lsp[3] = lsp_band1[lsp_index[1]][0]; + cur_lsp[4] = lsp_band1[lsp_index[1]][1]; + cur_lsp[5] = lsp_band1[lsp_index[1]][2]; + cur_lsp[6] = lsp_band2[lsp_index[2]][0]; + cur_lsp[7] = lsp_band2[lsp_index[2]][1]; + cur_lsp[8] = lsp_band2[lsp_index[2]][2]; + cur_lsp[9] = lsp_band2[lsp_index[2]][3]; + + /* Add predicted vector & DC component to the previously quantized vector */ + for (i = 0; i < LPC_ORDER; i++) { + temp = ((prev_lsp[i] - dc_lsp[i]) * pred + (1 << 14)) >> 15; + cur_lsp[i] += dc_lsp[i] + temp; + } + + for (i = 0; i < LPC_ORDER; i++) { + cur_lsp[0] = FFMAX(cur_lsp[0], 0x180); + cur_lsp[LPC_ORDER - 1] = FFMIN(cur_lsp[LPC_ORDER - 1], 0x7e00); + + /* Stability check */ + for (j = 1; j < LPC_ORDER; j++) { + temp = min_dist + cur_lsp[j - 1] - cur_lsp[j]; + if (temp > 0) { + temp >>= 1; + cur_lsp[j - 1] -= temp; + cur_lsp[j] += temp; + } + } + stable = 1; + for (j = 1; j < LPC_ORDER; j++) { + temp = cur_lsp[j - 1] + min_dist - cur_lsp[j] - 4; + if (temp > 0) { + stable = 0; + break; + } + } + if (stable) + break; + } + if (!stable) + memcpy(cur_lsp, prev_lsp, LPC_ORDER * sizeof(*cur_lsp)); +} diff --git a/libavcodec/g723_1.h b/libavcodec/g723_1.h index 71e2df4ad3..391ca464a9 100644 --- a/libavcodec/g723_1.h +++ b/libavcodec/g723_1.h @@ -1,5 +1,5 @@ /* - * G.723.1 compatible decoder data tables. + * G.723.1 common header and data tables * Copyright (c) 2006 Benjamin Larsson * Copyright (c) 2010 Mohamed Naufal Basheer * @@ -22,7 +22,7 @@ /** * @file - * G.723.1 compatible decoder data tables + * G.723.1 types, functions and data tables */ #ifndef AVCODEC_G723_1_H @@ -44,6 +44,143 @@ #define GAIN_LEVELS 24 #define COS_TBL_SIZE 512 +/** + * Bitexact implementation of 2ab scaled by 1/2^16. + * + * @param a 32 bit multiplicand + * @param b 16 bit multiplier + */ +#define MULL2(a, b) \ + ((((a) >> 16) * (b) << 1) + (((a) & 0xffff) * (b) >> 15)) + +/** + * G723.1 frame types + */ +enum FrameType { + ACTIVE_FRAME, ///< Active speech + SID_FRAME, ///< Silence Insertion Descriptor frame + UNTRANSMITTED_FRAME +}; + +/** + * G723.1 rate values + */ +enum Rate { + RATE_6300, + RATE_5300 +}; + +/** + * G723.1 unpacked data subframe + */ +typedef struct G723_1_Subframe { + int ad_cb_lag; ///< adaptive codebook lag + int ad_cb_gain; + int dirac_train; + int pulse_sign; + int grid_index; + int amp_index; + int pulse_pos; +} G723_1_Subframe; + +/** + * Pitch postfilter parameters + */ +typedef struct PPFParam { + int index; ///< postfilter backward/forward lag + int16_t opt_gain; ///< optimal gain + int16_t sc_gain; ///< scaling gain +} PPFParam; + +typedef struct g723_1_context { + AVClass *class; + + G723_1_Subframe subframe[4]; + enum FrameType cur_frame_type; + enum FrameType past_frame_type; + enum Rate cur_rate; + uint8_t lsp_index[LSP_BANDS]; + int pitch_lag[2]; + int erased_frames; + + int16_t prev_lsp[LPC_ORDER]; + int16_t sid_lsp[LPC_ORDER]; + int16_t prev_excitation[PITCH_MAX]; + int16_t excitation[PITCH_MAX + FRAME_LEN + 4]; + int16_t synth_mem[LPC_ORDER]; + int16_t fir_mem[LPC_ORDER]; + int iir_mem[LPC_ORDER]; + + int random_seed; + int cng_random_seed; + int interp_index; + int interp_gain; + int sid_gain; + int cur_gain; + int reflection_coef; + int pf_gain; + int postfilter; + + int16_t audio[FRAME_LEN + LPC_ORDER + PITCH_MAX + 4]; +} G723_1_Context; + + +/** + * Scale vector contents based on the largest of their absolutes. + */ +int ff_g723_1_scale_vector(int16_t *dst, const int16_t *vector, int length); + +/** + * Calculate the number of left-shifts required for normalizing the input. + * + * @param num input number + * @param width width of the input, 16 bits(0) / 32 bits(1) + */ +int ff_g723_1_normalize_bits(int num, int width); + +int ff_g723_1_dot_product(const int16_t *a, const int16_t *b, int length); + +/** + * Get delayed contribution from the previous excitation vector. + */ +void ff_g723_1_get_residual(int16_t *residual, int16_t *prev_excitation, + int lag); + +/** + * Generate a train of dirac functions with period as pitch lag. + */ +void ff_g723_1_gen_dirac_train(int16_t *buf, int pitch_lag); + + +/** + * Generate adaptive codebook excitation. + */ +void ff_g723_1_gen_acb_excitation(int16_t *vector, int16_t *prev_excitation, + int pitch_lag, G723_1_Subframe *subfrm, + enum Rate cur_rate); +/** + * Quantize LSP frequencies by interpolation and convert them to + * the corresponding LPC coefficients. + * + * @param lpc buffer for LPC coefficients + * @param cur_lsp the current LSP vector + * @param prev_lsp the previous LSP vector + */ +void ff_g723_1_lsp_interpolate(int16_t *lpc, int16_t *cur_lsp, + int16_t *prev_lsp); + +/** + * Perform inverse quantization of LSP frequencies. + * + * @param cur_lsp the current LSP vector + * @param prev_lsp the previous LSP vector + * @param lsp_index VQ indices + * @param bad_frame bad frame flag + */ +void ff_g723_1_inverse_quant(int16_t *cur_lsp, int16_t *prev_lsp, + uint8_t *lsp_index, int bad_frame); + + static const uint8_t frame_size[4] = { 24, 20, 4, 1 }; /* Postfilter gain weighting factors scaled by 2^15 */ diff --git a/libavcodec/g723_1dec.c b/libavcodec/g723_1dec.c index dc05ed2121..99043169fe 100644 --- a/libavcodec/g723_1dec.c +++ b/libavcodec/g723_1dec.c @@ -38,74 +38,6 @@ #define CNG_RANDOM_SEED 12345 -/** - * G723.1 frame types - */ -enum FrameType { - ACTIVE_FRAME, ///< Active speech - SID_FRAME, ///< Silence Insertion Descriptor frame - UNTRANSMITTED_FRAME -}; - -enum Rate { - RATE_6300, - RATE_5300 -}; - -/** - * G723.1 unpacked data subframe - */ -typedef struct G723_1_Subframe { - int ad_cb_lag; ///< adaptive codebook lag - int ad_cb_gain; - int dirac_train; - int pulse_sign; - int grid_index; - int amp_index; - int pulse_pos; -} G723_1_Subframe; - -/** - * Pitch postfilter parameters - */ -typedef struct PPFParam { - int index; ///< postfilter backward/forward lag - int16_t opt_gain; ///< optimal gain - int16_t sc_gain; ///< scaling gain -} PPFParam; - -typedef struct g723_1_context { - AVClass *class; - - G723_1_Subframe subframe[4]; - enum FrameType cur_frame_type; - enum FrameType past_frame_type; - enum Rate cur_rate; - uint8_t lsp_index[LSP_BANDS]; - int pitch_lag[2]; - int erased_frames; - - int16_t prev_lsp[LPC_ORDER]; - int16_t sid_lsp[LPC_ORDER]; - int16_t prev_excitation[PITCH_MAX]; - int16_t excitation[PITCH_MAX + FRAME_LEN + 4]; - int16_t synth_mem[LPC_ORDER]; - int16_t fir_mem[LPC_ORDER]; - int iir_mem[LPC_ORDER]; - - int random_seed; - int cng_random_seed; - int interp_index; - int interp_gain; - int sid_gain; - int cur_gain; - int reflection_coef; - int pf_gain; - int postfilter; - - int16_t audio[FRAME_LEN + LPC_ORDER + PITCH_MAX + 4]; -} G723_1_Context; - static av_cold int g723_1_decode_init(AVCodecContext *avctx) { G723_1_Context *p = avctx->priv_data; @@ -262,108 +194,6 @@ static int16_t square_root(int val) return res; } -/** - * Calculate the number of left-shifts required for normalizing the input. - * - * @param num input number - * @param width width of the input, 16 bits(0) / 32 bits(1) - */ -static int normalize_bits(int num, int width) -{ - return width - av_log2(num) - 1; -} - -/** - * Scale vector contents based on the largest of their absolutes. - */ -static int scale_vector(int16_t *dst, const int16_t *vector, int length) -{ - int bits, max = 0; - int i; - - - for (i = 0; i < length; i++) - max |= FFABS(vector[i]); - - max = FFMIN(max, 0x7FFF); - bits = normalize_bits(max, 15); - - for (i = 0; i < length; i++) - dst[i] = vector[i] << bits >> 3; - - return bits - 3; -} - -/** - * Perform inverse quantization of LSP frequencies. - * - * @param cur_lsp the current LSP vector - * @param prev_lsp the previous LSP vector - * @param lsp_index VQ indices - * @param bad_frame bad frame flag - */ -static void inverse_quant(int16_t *cur_lsp, int16_t *prev_lsp, - uint8_t *lsp_index, int bad_frame) -{ - int min_dist, pred; - int i, j, temp, stable; - - /* Check for frame erasure */ - if (!bad_frame) { - min_dist = 0x100; - pred = 12288; - } else { - min_dist = 0x200; - pred = 23552; - lsp_index[0] = lsp_index[1] = lsp_index[2] = 0; - } - - /* Get the VQ table entry corresponding to the transmitted index */ - cur_lsp[0] = lsp_band0[lsp_index[0]][0]; - cur_lsp[1] = lsp_band0[lsp_index[0]][1]; - cur_lsp[2] = lsp_band0[lsp_index[0]][2]; - cur_lsp[3] = lsp_band1[lsp_index[1]][0]; - cur_lsp[4] = lsp_band1[lsp_index[1]][1]; - cur_lsp[5] = lsp_band1[lsp_index[1]][2]; - cur_lsp[6] = lsp_band2[lsp_index[2]][0]; - cur_lsp[7] = lsp_band2[lsp_index[2]][1]; - cur_lsp[8] = lsp_band2[lsp_index[2]][2]; - cur_lsp[9] = lsp_band2[lsp_index[2]][3]; - - /* Add predicted vector & DC component to the previously quantized vector */ - for (i = 0; i < LPC_ORDER; i++) { - temp = ((prev_lsp[i] - dc_lsp[i]) * pred + (1 << 14)) >> 15; - cur_lsp[i] += dc_lsp[i] + temp; - } - - for (i = 0; i < LPC_ORDER; i++) { - cur_lsp[0] = FFMAX(cur_lsp[0], 0x180); - cur_lsp[LPC_ORDER - 1] = FFMIN(cur_lsp[LPC_ORDER - 1], 0x7e00); - - /* Stability check */ - for (j = 1; j < LPC_ORDER; j++) { - temp = min_dist + cur_lsp[j - 1] - cur_lsp[j]; - if (temp > 0) { - temp >>= 1; - cur_lsp[j - 1] -= temp; - cur_lsp[j] += temp; - } - } - stable = 1; - for (j = 1; j < LPC_ORDER; j++) { - temp = cur_lsp[j - 1] + min_dist - cur_lsp[j] - 4; - if (temp > 0) { - stable = 0; - break; - } - } - if (stable) - break; - } - if (!stable) - memcpy(cur_lsp, prev_lsp, LPC_ORDER * sizeof(*cur_lsp)); -} - /** * Bitexact implementation of 2ab scaled by 1/2^16. * @@ -373,116 +203,6 @@ static void inverse_quant(int16_t *cur_lsp, int16_t *prev_lsp, #define MULL2(a, b) \ ((((a) >> 16) * (b) << 1) + (((a) & 0xffff) * (b) >> 15)) -/** - * Convert LSP frequencies to LPC coefficients. - * - * @param lpc buffer for LPC coefficients - */ -static void lsp2lpc(int16_t *lpc) -{ - int f1[LPC_ORDER / 2 + 1]; - int f2[LPC_ORDER / 2 + 1]; - int i, j; - - /* Calculate negative cosine */ - for (j = 0; j < LPC_ORDER; j++) { - int index = (lpc[j] >> 7) & 0x1FF; - int offset = lpc[j] & 0x7f; - int temp1 = cos_tab[index] << 16; - int temp2 = (cos_tab[index + 1] - cos_tab[index]) * - ((offset << 8) + 0x80) << 1; - - lpc[j] = -(av_sat_dadd32(1 << 15, temp1 + temp2) >> 16); - } - - /* - * Compute sum and difference polynomial coefficients - * (bitexact alternative to lsp2poly() in lsp.c) - */ - /* Initialize with values in Q28 */ - f1[0] = 1 << 28; - f1[1] = (lpc[0] << 14) + (lpc[2] << 14); - f1[2] = lpc[0] * lpc[2] + (2 << 28); - - f2[0] = 1 << 28; - f2[1] = (lpc[1] << 14) + (lpc[3] << 14); - f2[2] = lpc[1] * lpc[3] + (2 << 28); - - /* - * Calculate and scale the coefficients by 1/2 in - * each iteration for a final scaling factor of Q25 - */ - for (i = 2; i < LPC_ORDER / 2; i++) { - f1[i + 1] = f1[i - 1] + MULL2(f1[i], lpc[2 * i]); - f2[i + 1] = f2[i - 1] + MULL2(f2[i], lpc[2 * i + 1]); - - for (j = i; j >= 2; j--) { - f1[j] = MULL2(f1[j - 1], lpc[2 * i]) + - (f1[j] >> 1) + (f1[j - 2] >> 1); - f2[j] = MULL2(f2[j - 1], lpc[2 * i + 1]) + - (f2[j] >> 1) + (f2[j - 2] >> 1); - } - - f1[0] >>= 1; - f2[0] >>= 1; - f1[1] = ((lpc[2 * i] << 16 >> i) + f1[1]) >> 1; - f2[1] = ((lpc[2 * i + 1] << 16 >> i) + f2[1]) >> 1; - } - - /* Convert polynomial coefficients to LPC coefficients */ - for (i = 0; i < LPC_ORDER / 2; i++) { - int64_t ff1 = f1[i + 1] + f1[i]; - int64_t ff2 = f2[i + 1] - f2[i]; - - lpc[i] = av_clipl_int32(((ff1 + ff2) << 3) + (1 << 15)) >> 16; - lpc[LPC_ORDER - i - 1] = av_clipl_int32(((ff1 - ff2) << 3) + - (1 << 15)) >> 16; - } -} - -/** - * Quantize LSP frequencies by interpolation and convert them to - * the corresponding LPC coefficients. - * - * @param lpc buffer for LPC coefficients - * @param cur_lsp the current LSP vector - * @param prev_lsp the previous LSP vector - */ -static void lsp_interpolate(int16_t *lpc, int16_t *cur_lsp, int16_t *prev_lsp) -{ - int i; - int16_t *lpc_ptr = lpc; - - /* cur_lsp * 0.25 + prev_lsp * 0.75 */ - ff_acelp_weighted_vector_sum(lpc, cur_lsp, prev_lsp, - 4096, 12288, 1 << 13, 14, LPC_ORDER); - ff_acelp_weighted_vector_sum(lpc + LPC_ORDER, cur_lsp, prev_lsp, - 8192, 8192, 1 << 13, 14, LPC_ORDER); - ff_acelp_weighted_vector_sum(lpc + 2 * LPC_ORDER, cur_lsp, prev_lsp, - 12288, 4096, 1 << 13, 14, LPC_ORDER); - memcpy(lpc + 3 * LPC_ORDER, cur_lsp, LPC_ORDER * sizeof(*lpc)); - - for (i = 0; i < SUBFRAMES; i++) { - lsp2lpc(lpc_ptr); - lpc_ptr += LPC_ORDER; - } -} - -/** - * Generate a train of dirac functions with period as pitch lag. - */ -static void gen_dirac_train(int16_t *buf, int pitch_lag) -{ - int16_t vector[SUBFRAME_LEN]; - int i, j; - - memcpy(vector, buf, SUBFRAME_LEN * sizeof(*vector)); - for (i = pitch_lag; i < SUBFRAME_LEN; i += pitch_lag) { - for (j = 0; j < SUBFRAME_LEN - i; j++) - buf[i + j] += vector[j]; - } -} - /** * Generate fixed codebook excitation vector. * @@ -522,7 +242,7 @@ static void gen_fcb_excitation(int16_t *vector, G723_1_Subframe *subfrm, break; } if (subfrm->dirac_train == 1) - gen_dirac_train(vector, pitch_lag); + ff_g723_1_gen_dirac_train(vector, pitch_lag); } else { /* 5300 bps */ int cb_gain = fixed_cb_gain[subfrm->amp_index]; int cb_shift = subfrm->grid_index; @@ -549,63 +269,6 @@ static void gen_fcb_excitation(int16_t *vector, G723_1_Subframe *subfrm, } } -/** - * Get delayed contribution from the previous excitation vector. - */ -static void get_residual(int16_t *residual, int16_t *prev_excitation, int lag) -{ - int offset = PITCH_MAX - PITCH_ORDER / 2 - lag; - int i; - - residual[0] = prev_excitation[offset]; - residual[1] = prev_excitation[offset + 1]; - - offset += 2; - for (i = 2; i < SUBFRAME_LEN + PITCH_ORDER - 1; i++) - residual[i] = prev_excitation[offset + (i - 2) % lag]; -} - -static int dot_product(const int16_t *a, const int16_t *b, int length) -{ - int i, sum = 0; - - for (i = 0; i < length; i++) { - int prod = a[i] * b[i]; - sum = av_sat_dadd32(sum, prod); - } - return sum; -} - -/** - * Generate adaptive codebook excitation. - */ -static void gen_acb_excitation(int16_t *vector, int16_t *prev_excitation, - int pitch_lag, G723_1_Subframe *subfrm, - enum Rate cur_rate) -{ - int16_t residual[SUBFRAME_LEN + PITCH_ORDER - 1]; - const int16_t *cb_ptr; - int lag = pitch_lag + subfrm->ad_cb_lag - 1; - - int i; - int sum; - - get_residual(residual, prev_excitation, lag); - - /* Select quantization table */ - if (cur_rate == RATE_6300 && pitch_lag < SUBFRAME_LEN - 2) - cb_ptr = adaptive_cb_gain85; - else - cb_ptr = adaptive_cb_gain170; - - /* Calculate adaptive vector */ - cb_ptr += subfrm->ad_cb_gain * 20; - for (i = 0; i < SUBFRAME_LEN; i++) { - sum = dot_product(residual + i, cb_ptr, PITCH_ORDER); - vector[i] = av_sat_dadd32(1 << 15, sum) >> 16; - } -} - /** * Estimate maximum auto-correlation around pitch lag. * @@ -629,7 +292,7 @@ static int autocorr_max(const int16_t *buf, int offset, int *ccr_max, limit = pitch_lag + 3; for (i = pitch_lag - 3; i <= limit; i++) { - ccr = dot_product(buf, buf + dir * i, length); + ccr = ff_g723_1_dot_product(buf, buf + dir * i, length); if (ccr > *ccr_max) { *ccr_max = ccr; @@ -728,22 +391,24 @@ static void comp_ppf_coeff(G723_1_Context *p, int offset, int pitch_lag, return; /* Compute target energy */ - energy[0] = dot_product(buf, buf, SUBFRAME_LEN); + energy[0] = ff_g723_1_dot_product(buf, buf, SUBFRAME_LEN); /* Compute forward residual energy */ if (fwd_lag) - energy[2] = dot_product(buf + fwd_lag, buf + fwd_lag, SUBFRAME_LEN); + energy[2] = ff_g723_1_dot_product(buf + fwd_lag, buf + fwd_lag, + SUBFRAME_LEN); /* Compute backward residual energy */ if (back_lag) - energy[4] = dot_product(buf - back_lag, buf - back_lag, SUBFRAME_LEN); + energy[4] = ff_g723_1_dot_product(buf - back_lag, buf - back_lag, + SUBFRAME_LEN); /* Normalize and shorten */ temp1 = 0; for (i = 0; i < 5; i++) temp1 = FFMAX(energy[i], temp1); - scale = normalize_bits(temp1, 31); + scale = ff_g723_1_normalize_bits(temp1, 31); for (i = 0; i < 5; i++) energy[i] = (energy[i] << scale) >> 16; @@ -789,7 +454,7 @@ static int comp_interp_index(G723_1_Context *p, int pitch_lag, int index, ccr, tgt_eng, best_eng, temp; - *scale = scale_vector(buf, p->excitation, FRAME_LEN + PITCH_MAX); + *scale = ff_g723_1_scale_vector(buf, p->excitation, FRAME_LEN + PITCH_MAX); buf += offset; /* Compute maximum backward cross-correlation */ @@ -798,14 +463,15 @@ static int comp_interp_index(G723_1_Context *p, int pitch_lag, ccr = av_sat_add32(ccr, 1 << 15) >> 16; /* Compute target energy */ - tgt_eng = dot_product(buf, buf, SUBFRAME_LEN * 2); + tgt_eng = ff_g723_1_dot_product(buf, buf, SUBFRAME_LEN * 2); *exc_eng = av_sat_add32(tgt_eng, 1 << 15) >> 16; if (ccr <= 0) return 0; /* Compute best energy */ - best_eng = dot_product(buf - index, buf - index, SUBFRAME_LEN * 2); + best_eng = ff_g723_1_dot_product(buf - index, buf - index, + SUBFRAME_LEN * 2); best_eng = av_sat_add32(best_eng, 1 << 15) >> 16; temp = best_eng * *exc_eng >> 3; @@ -853,8 +519,8 @@ static void residual_interp(int16_t *buf, int16_t *out, int lag, * @param src source vector * @param dest destination vector */ -static inline void iir_filter(int16_t *fir_coef, int16_t *iir_coef, - int16_t *src, int *dest) +static void iir_filter(int16_t *fir_coef, int16_t *iir_coef, + int16_t *src, int *dest) { int m, n; @@ -890,8 +556,8 @@ static void gain_scale(G723_1_Context *p, int16_t * buf, int energy) } if (num && denom) { - bits1 = normalize_bits(num, 31); - bits2 = normalize_bits(denom, 31); + bits1 = ff_g723_1_normalize_bits(num, 31); + bits2 = ff_g723_1_normalize_bits(denom, 31); num = num << bits1 >> 1; denom <<= bits2; @@ -936,8 +602,7 @@ static void formant_postfilter(G723_1_Context *p, int16_t *lpc, filter_coef[1][k] = (-lpc[k] * postfilter_tbl[1][k] + (1 << 14)) >> 15; } - iir_filter(filter_coef[0], filter_coef[1], buf + i, - filter_signal + i); + iir_filter(filter_coef[0], filter_coef[1], buf + i, filter_signal + i); lpc += LPC_ORDER; } @@ -953,11 +618,11 @@ static void formant_postfilter(G723_1_Context *p, int16_t *lpc, int scale, energy; /* Normalize */ - scale = scale_vector(dst, buf, SUBFRAME_LEN); + scale = ff_g723_1_scale_vector(dst, buf, SUBFRAME_LEN); /* Compute auto correlation coefficients */ - auto_corr[0] = dot_product(dst, dst + 1, SUBFRAME_LEN - 1); - auto_corr[1] = dot_product(dst, dst, SUBFRAME_LEN); + auto_corr[0] = ff_g723_1_dot_product(dst, dst + 1, SUBFRAME_LEN - 1); + auto_corr[1] = ff_g723_1_dot_product(dst, dst, SUBFRAME_LEN); /* Compute reflection coefficient */ temp = auto_corr[1] >> 16; @@ -1104,13 +769,13 @@ static void generate_noise(G723_1_Context *p) memcpy(vector_ptr, p->prev_excitation, PITCH_MAX * sizeof(*p->excitation)); for (i = 0; i < SUBFRAMES; i += 2) { - gen_acb_excitation(vector_ptr, vector_ptr, - p->pitch_lag[i >> 1], &p->subframe[i], - p->cur_rate); - gen_acb_excitation(vector_ptr + SUBFRAME_LEN, - vector_ptr + SUBFRAME_LEN, - p->pitch_lag[i >> 1], &p->subframe[i + 1], - p->cur_rate); + ff_g723_1_gen_acb_excitation(vector_ptr, vector_ptr, + p->pitch_lag[i >> 1], &p->subframe[i], + p->cur_rate); + ff_g723_1_gen_acb_excitation(vector_ptr + SUBFRAME_LEN, + vector_ptr + SUBFRAME_LEN, + p->pitch_lag[i >> 1], &p->subframe[i + 1], + p->cur_rate); t = 0; for (j = 0; j < SUBFRAME_LEN * 2; j++) @@ -1231,8 +896,8 @@ static int g723_1_decode_frame(AVCodecContext *avctx, void *data, else if (p->erased_frames != 3) p->erased_frames++; - inverse_quant(cur_lsp, p->prev_lsp, p->lsp_index, bad_frame); - lsp_interpolate(lpc, cur_lsp, p->prev_lsp); + ff_g723_1_inverse_quant(cur_lsp, p->prev_lsp, p->lsp_index, bad_frame); + ff_g723_1_lsp_interpolate(lpc, cur_lsp, p->prev_lsp); /* Save the lsp_vector for the next frame */ memcpy(p->prev_lsp, cur_lsp, LPC_ORDER * sizeof(*p->prev_lsp)); @@ -1249,9 +914,10 @@ static int g723_1_decode_frame(AVCodecContext *avctx, void *data, for (i = 0; i < SUBFRAMES; i++) { gen_fcb_excitation(vector_ptr, &p->subframe[i], p->cur_rate, p->pitch_lag[i >> 1], i); - gen_acb_excitation(acb_vector, &p->excitation[SUBFRAME_LEN * i], - p->pitch_lag[i >> 1], &p->subframe[i], - p->cur_rate); + ff_g723_1_gen_acb_excitation(acb_vector, + &p->excitation[SUBFRAME_LEN * i], + p->pitch_lag[i >> 1], + &p->subframe[i], p->cur_rate); /* Get the total excitation */ for (j = 0; j < SUBFRAME_LEN; j++) { int v = av_clip_int16(vector_ptr[j] << 1); @@ -1312,7 +978,7 @@ static int g723_1_decode_frame(AVCodecContext *avctx, void *data, } else { if (p->cur_frame_type == SID_FRAME) { p->sid_gain = sid_gain_to_lsp_index(p->subframe[0].amp_index); - inverse_quant(p->sid_lsp, p->prev_lsp, p->lsp_index, 0); + ff_g723_1_inverse_quant(p->sid_lsp, p->prev_lsp, p->lsp_index, 0); } else if (p->past_frame_type == ACTIVE_FRAME) { p->sid_gain = estimate_sid_gain(p); } @@ -1322,7 +988,7 @@ static int g723_1_decode_frame(AVCodecContext *avctx, void *data, else p->cur_gain = (p->cur_gain * 7 + p->sid_gain) >> 3; generate_noise(p); - lsp_interpolate(lpc, p->sid_lsp, p->prev_lsp); + ff_g723_1_lsp_interpolate(lpc, p->sid_lsp, p->prev_lsp); /* Save the lsp_vector for the next frame */ memcpy(p->prev_lsp, p->sid_lsp, LPC_ORDER * sizeof(*p->prev_lsp)); } -- cgit v1.2.3