[FFmpeg-cvslog] g723_1: Move sharable functions to a separate file
Vittorio Giovara
git at videolan.org
Mon Dec 7 15:27:53 CET 2015
ffmpeg | branch: master | Vittorio Giovara <vittorio.giovara at gmail.com> | Mon Nov 23 17:10:53 2015 -0500| [165cc6fb9defcd79fd71c08167f3e8df26b058ff] | committer: Vittorio Giovara
g723_1: Move sharable functions to a separate file
Signed-off-by: Vittorio Giovara <vittorio.giovara at gmail.com>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=165cc6fb9defcd79fd71c08167f3e8df26b058ff
---
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(-)
diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index dfefab6..85738fa 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 0000000..af4777c
--- /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 71e2df4..391ca46 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 dc05ed2..9904316 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;
@@ -263,108 +195,6 @@ static int16_t square_root(int val)
}
/**
- * 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.
*
* @param a 32 bit multiplicand
@@ -374,116 +204,6 @@ static void inverse_quant(int16_t *cur_lsp, int16_t *prev_lsp,
((((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.
*
* @param vector decoded 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;
@@ -550,63 +270,6 @@ 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.
*
* @param buf buffer with offset applied
@@ -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));
}
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