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https://github.com/pineappleEA/pineapple-src.git
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1212 lines
39 KiB
C
Executable File
1212 lines
39 KiB
C
Executable File
/*
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* G.723.1 compatible encoder
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* Copyright (c) Mohamed Naufal <naufal22@gmail.com>
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* G.723.1 compatible encoder
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*/
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#include <stdint.h>
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#include <string.h>
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#include "libavutil/channel_layout.h"
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#include "libavutil/common.h"
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#include "libavutil/mem.h"
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#include "libavutil/opt.h"
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#include "avcodec.h"
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#include "celp_math.h"
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#include "g723_1.h"
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#include "internal.h"
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#define BITSTREAM_WRITER_LE
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#include "put_bits.h"
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static av_cold int g723_1_encode_init(AVCodecContext *avctx)
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{
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G723_1_Context *s = avctx->priv_data;
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G723_1_ChannelContext *p = &s->ch[0];
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if (avctx->sample_rate != 8000) {
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av_log(avctx, AV_LOG_ERROR, "Only 8000Hz sample rate supported\n");
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return AVERROR(EINVAL);
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}
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if (avctx->channels != 1) {
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av_log(avctx, AV_LOG_ERROR, "Only mono supported\n");
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return AVERROR(EINVAL);
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}
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if (avctx->bit_rate == 6300) {
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p->cur_rate = RATE_6300;
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} else if (avctx->bit_rate == 5300) {
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av_log(avctx, AV_LOG_ERROR, "Use bitrate 6300 instead of 5300.\n");
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avpriv_report_missing_feature(avctx, "Bitrate 5300");
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return AVERROR_PATCHWELCOME;
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} else {
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av_log(avctx, AV_LOG_ERROR, "Bitrate not supported, use 6300\n");
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return AVERROR(EINVAL);
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}
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avctx->frame_size = 240;
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memcpy(p->prev_lsp, dc_lsp, LPC_ORDER * sizeof(int16_t));
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return 0;
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}
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/**
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* Remove DC component from the input signal.
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*
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* @param buf input signal
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* @param fir zero memory
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* @param iir pole memory
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*/
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static void highpass_filter(int16_t *buf, int16_t *fir, int *iir)
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{
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int i;
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for (i = 0; i < FRAME_LEN; i++) {
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*iir = (buf[i] << 15) + ((-*fir) << 15) + MULL2(*iir, 0x7f00);
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*fir = buf[i];
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buf[i] = av_clipl_int32((int64_t)*iir + (1 << 15)) >> 16;
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}
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}
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/**
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* Estimate autocorrelation of the input vector.
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*
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* @param buf input buffer
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* @param autocorr autocorrelation coefficients vector
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*/
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static void comp_autocorr(int16_t *buf, int16_t *autocorr)
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{
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int i, scale, temp;
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int16_t vector[LPC_FRAME];
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ff_g723_1_scale_vector(vector, buf, LPC_FRAME);
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/* Apply the Hamming window */
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for (i = 0; i < LPC_FRAME; i++)
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vector[i] = (vector[i] * hamming_window[i] + (1 << 14)) >> 15;
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/* Compute the first autocorrelation coefficient */
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temp = ff_dot_product(vector, vector, LPC_FRAME);
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/* Apply a white noise correlation factor of (1025/1024) */
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temp += temp >> 10;
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/* Normalize */
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scale = ff_g723_1_normalize_bits(temp, 31);
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autocorr[0] = av_clipl_int32((int64_t) (temp << scale) +
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(1 << 15)) >> 16;
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/* Compute the remaining coefficients */
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if (!autocorr[0]) {
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memset(autocorr + 1, 0, LPC_ORDER * sizeof(int16_t));
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} else {
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for (i = 1; i <= LPC_ORDER; i++) {
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temp = ff_dot_product(vector, vector + i, LPC_FRAME - i);
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temp = MULL2((temp << scale), binomial_window[i - 1]);
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autocorr[i] = av_clipl_int32((int64_t) temp + (1 << 15)) >> 16;
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}
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}
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}
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/**
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* Use Levinson-Durbin recursion to compute LPC coefficients from
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* autocorrelation values.
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*
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* @param lpc LPC coefficients vector
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* @param autocorr autocorrelation coefficients vector
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* @param error prediction error
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*/
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static void levinson_durbin(int16_t *lpc, int16_t *autocorr, int16_t error)
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{
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int16_t vector[LPC_ORDER];
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int16_t partial_corr;
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int i, j, temp;
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memset(lpc, 0, LPC_ORDER * sizeof(int16_t));
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for (i = 0; i < LPC_ORDER; i++) {
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/* Compute the partial correlation coefficient */
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temp = 0;
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for (j = 0; j < i; j++)
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temp -= lpc[j] * autocorr[i - j - 1];
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temp = ((autocorr[i] << 13) + temp) << 3;
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if (FFABS(temp) >= (error << 16))
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break;
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partial_corr = temp / (error << 1);
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lpc[i] = av_clipl_int32((int64_t) (partial_corr << 14) +
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(1 << 15)) >> 16;
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/* Update the prediction error */
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temp = MULL2(temp, partial_corr);
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error = av_clipl_int32((int64_t) (error << 16) - temp +
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(1 << 15)) >> 16;
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memcpy(vector, lpc, i * sizeof(int16_t));
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for (j = 0; j < i; j++) {
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temp = partial_corr * vector[i - j - 1] << 1;
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lpc[j] = av_clipl_int32((int64_t) (lpc[j] << 16) - temp +
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(1 << 15)) >> 16;
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}
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}
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}
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/**
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* Calculate LPC coefficients for the current frame.
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*
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* @param buf current frame
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* @param prev_data 2 trailing subframes of the previous frame
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* @param lpc LPC coefficients vector
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*/
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static void comp_lpc_coeff(int16_t *buf, int16_t *lpc)
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{
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int16_t autocorr[(LPC_ORDER + 1) * SUBFRAMES];
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int16_t *autocorr_ptr = autocorr;
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int16_t *lpc_ptr = lpc;
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int i, j;
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for (i = 0, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) {
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comp_autocorr(buf + i, autocorr_ptr);
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levinson_durbin(lpc_ptr, autocorr_ptr + 1, autocorr_ptr[0]);
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lpc_ptr += LPC_ORDER;
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autocorr_ptr += LPC_ORDER + 1;
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}
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}
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static void lpc2lsp(int16_t *lpc, int16_t *prev_lsp, int16_t *lsp)
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{
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int f[LPC_ORDER + 2]; ///< coefficients of the sum and difference
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///< polynomials (F1, F2) ordered as
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///< f1[0], f2[0], ...., f1[5], f2[5]
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int max, shift, cur_val, prev_val, count, p;
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int i, j;
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int64_t temp;
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/* Initialize f1[0] and f2[0] to 1 in Q25 */
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for (i = 0; i < LPC_ORDER; i++)
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lsp[i] = (lpc[i] * bandwidth_expand[i] + (1 << 14)) >> 15;
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/* Apply bandwidth expansion on the LPC coefficients */
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f[0] = f[1] = 1 << 25;
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/* Compute the remaining coefficients */
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for (i = 0; i < LPC_ORDER / 2; i++) {
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/* f1 */
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f[2 * i + 2] = -f[2 * i] - ((lsp[i] + lsp[LPC_ORDER - 1 - i]) << 12);
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/* f2 */
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f[2 * i + 3] = f[2 * i + 1] - ((lsp[i] - lsp[LPC_ORDER - 1 - i]) << 12);
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}
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/* Divide f1[5] and f2[5] by 2 for use in polynomial evaluation */
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f[LPC_ORDER] >>= 1;
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f[LPC_ORDER + 1] >>= 1;
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/* Normalize and shorten */
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max = FFABS(f[0]);
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for (i = 1; i < LPC_ORDER + 2; i++)
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max = FFMAX(max, FFABS(f[i]));
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shift = ff_g723_1_normalize_bits(max, 31);
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for (i = 0; i < LPC_ORDER + 2; i++)
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f[i] = av_clipl_int32((int64_t) (f[i] << shift) + (1 << 15)) >> 16;
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/**
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* Evaluate F1 and F2 at uniform intervals of pi/256 along the
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* unit circle and check for zero crossings.
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*/
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p = 0;
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temp = 0;
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for (i = 0; i <= LPC_ORDER / 2; i++)
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temp += f[2 * i] * cos_tab[0];
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prev_val = av_clipl_int32(temp << 1);
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count = 0;
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for (i = 1; i < COS_TBL_SIZE / 2; i++) {
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/* Evaluate */
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temp = 0;
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for (j = 0; j <= LPC_ORDER / 2; j++)
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temp += f[LPC_ORDER - 2 * j + p] * cos_tab[i * j % COS_TBL_SIZE];
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cur_val = av_clipl_int32(temp << 1);
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/* Check for sign change, indicating a zero crossing */
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if ((cur_val ^ prev_val) < 0) {
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int abs_cur = FFABS(cur_val);
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int abs_prev = FFABS(prev_val);
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int sum = abs_cur + abs_prev;
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shift = ff_g723_1_normalize_bits(sum, 31);
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sum <<= shift;
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abs_prev = abs_prev << shift >> 8;
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lsp[count++] = ((i - 1) << 7) + (abs_prev >> 1) / (sum >> 16);
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if (count == LPC_ORDER)
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break;
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/* Switch between sum and difference polynomials */
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p ^= 1;
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/* Evaluate */
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temp = 0;
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for (j = 0; j <= LPC_ORDER / 2; j++)
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temp += f[LPC_ORDER - 2 * j + p] *
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cos_tab[i * j % COS_TBL_SIZE];
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cur_val = av_clipl_int32(temp << 1);
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}
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prev_val = cur_val;
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}
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if (count != LPC_ORDER)
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memcpy(lsp, prev_lsp, LPC_ORDER * sizeof(int16_t));
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}
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/**
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* Quantize the current LSP subvector.
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*
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* @param num band number
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* @param offset offset of the current subvector in an LPC_ORDER vector
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* @param size size of the current subvector
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*/
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#define get_index(num, offset, size) \
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{ \
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int error, max = -1; \
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int16_t temp[4]; \
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int i, j; \
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\
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for (i = 0; i < LSP_CB_SIZE; i++) { \
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for (j = 0; j < size; j++){ \
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temp[j] = (weight[j + (offset)] * lsp_band##num[i][j] + \
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(1 << 14)) >> 15; \
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} \
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error = ff_g723_1_dot_product(lsp + (offset), temp, size) << 1; \
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error -= ff_g723_1_dot_product(lsp_band##num[i], temp, size); \
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if (error > max) { \
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max = error; \
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lsp_index[num] = i; \
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} \
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} \
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}
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/**
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* Vector quantize the LSP frequencies.
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*
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* @param lsp the current lsp vector
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* @param prev_lsp the previous lsp vector
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*/
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static void lsp_quantize(uint8_t *lsp_index, int16_t *lsp, int16_t *prev_lsp)
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{
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int16_t weight[LPC_ORDER];
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int16_t min, max;
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int shift, i;
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/* Calculate the VQ weighting vector */
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weight[0] = (1 << 20) / (lsp[1] - lsp[0]);
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weight[LPC_ORDER - 1] = (1 << 20) /
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(lsp[LPC_ORDER - 1] - lsp[LPC_ORDER - 2]);
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for (i = 1; i < LPC_ORDER - 1; i++) {
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min = FFMIN(lsp[i] - lsp[i - 1], lsp[i + 1] - lsp[i]);
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if (min > 0x20)
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weight[i] = (1 << 20) / min;
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else
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weight[i] = INT16_MAX;
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}
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/* Normalize */
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max = 0;
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for (i = 0; i < LPC_ORDER; i++)
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max = FFMAX(weight[i], max);
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shift = ff_g723_1_normalize_bits(max, 15);
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for (i = 0; i < LPC_ORDER; i++) {
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weight[i] <<= shift;
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}
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/* Compute the VQ target vector */
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for (i = 0; i < LPC_ORDER; i++) {
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lsp[i] -= dc_lsp[i] +
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(((prev_lsp[i] - dc_lsp[i]) * 12288 + (1 << 14)) >> 15);
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}
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get_index(0, 0, 3);
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get_index(1, 3, 3);
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get_index(2, 6, 4);
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}
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/**
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* Perform IIR filtering.
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*
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* @param fir_coef FIR coefficients
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* @param iir_coef IIR coefficients
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* @param src source vector
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* @param dest destination vector
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*/
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static void iir_filter(int16_t *fir_coef, int16_t *iir_coef,
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int16_t *src, int16_t *dest)
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{
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int m, n;
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for (m = 0; m < SUBFRAME_LEN; m++) {
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int64_t filter = 0;
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for (n = 1; n <= LPC_ORDER; n++) {
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filter -= fir_coef[n - 1] * src[m - n] -
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iir_coef[n - 1] * dest[m - n];
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}
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dest[m] = av_clipl_int32((src[m] << 16) + (filter << 3) +
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(1 << 15)) >> 16;
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}
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}
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/**
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* Apply the formant perceptual weighting filter.
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*
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* @param flt_coef filter coefficients
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* @param unq_lpc unquantized lpc vector
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*/
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static void perceptual_filter(G723_1_ChannelContext *p, int16_t *flt_coef,
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int16_t *unq_lpc, int16_t *buf)
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{
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int16_t vector[FRAME_LEN + LPC_ORDER];
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int i, j, k, l = 0;
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memcpy(buf, p->iir_mem, sizeof(int16_t) * LPC_ORDER);
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memcpy(vector, p->fir_mem, sizeof(int16_t) * LPC_ORDER);
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memcpy(vector + LPC_ORDER, buf + LPC_ORDER, sizeof(int16_t) * FRAME_LEN);
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for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) {
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for (k = 0; k < LPC_ORDER; k++) {
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flt_coef[k + 2 * l] = (unq_lpc[k + l] * percept_flt_tbl[0][k] +
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(1 << 14)) >> 15;
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flt_coef[k + 2 * l + LPC_ORDER] = (unq_lpc[k + l] *
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percept_flt_tbl[1][k] +
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(1 << 14)) >> 15;
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}
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iir_filter(flt_coef + 2 * l, flt_coef + 2 * l + LPC_ORDER,
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vector + i, buf + i);
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l += LPC_ORDER;
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}
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memcpy(p->iir_mem, buf + FRAME_LEN, sizeof(int16_t) * LPC_ORDER);
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memcpy(p->fir_mem, vector + FRAME_LEN, sizeof(int16_t) * LPC_ORDER);
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}
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/**
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* Estimate the open loop pitch period.
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*
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* @param buf perceptually weighted speech
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* @param start estimation is carried out from this position
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*/
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static int estimate_pitch(int16_t *buf, int start)
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{
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int max_exp = 32;
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int max_ccr = 0x4000;
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int max_eng = 0x7fff;
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int index = PITCH_MIN;
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int offset = start - PITCH_MIN + 1;
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int ccr, eng, orig_eng, ccr_eng, exp;
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int diff, temp;
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int i;
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orig_eng = ff_dot_product(buf + offset, buf + offset, HALF_FRAME_LEN);
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for (i = PITCH_MIN; i <= PITCH_MAX - 3; i++) {
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offset--;
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/* Update energy and compute correlation */
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orig_eng += buf[offset] * buf[offset] -
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buf[offset + HALF_FRAME_LEN] * buf[offset + HALF_FRAME_LEN];
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ccr = ff_dot_product(buf + start, buf + offset, HALF_FRAME_LEN);
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if (ccr <= 0)
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continue;
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/* Split into mantissa and exponent to maintain precision */
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exp = ff_g723_1_normalize_bits(ccr, 31);
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ccr = av_clipl_int32((int64_t) (ccr << exp) + (1 << 15)) >> 16;
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exp <<= 1;
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ccr *= ccr;
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temp = ff_g723_1_normalize_bits(ccr, 31);
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ccr = ccr << temp >> 16;
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exp += temp;
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temp = ff_g723_1_normalize_bits(orig_eng, 31);
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eng = av_clipl_int32((int64_t) (orig_eng << temp) + (1 << 15)) >> 16;
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exp -= temp;
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if (ccr >= eng) {
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exp--;
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ccr >>= 1;
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}
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if (exp > max_exp)
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continue;
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if (exp + 1 < max_exp)
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goto update;
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/* Equalize exponents before comparison */
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if (exp + 1 == max_exp)
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temp = max_ccr >> 1;
|
|
else
|
|
temp = max_ccr;
|
|
ccr_eng = ccr * max_eng;
|
|
diff = ccr_eng - eng * temp;
|
|
if (diff > 0 && (i - index < PITCH_MIN || diff > ccr_eng >> 2)) {
|
|
update:
|
|
index = i;
|
|
max_exp = exp;
|
|
max_ccr = ccr;
|
|
max_eng = eng;
|
|
}
|
|
}
|
|
return index;
|
|
}
|
|
|
|
/**
|
|
* Compute harmonic noise filter parameters.
|
|
*
|
|
* @param buf perceptually weighted speech
|
|
* @param pitch_lag open loop pitch period
|
|
* @param hf harmonic filter parameters
|
|
*/
|
|
static void comp_harmonic_coeff(int16_t *buf, int16_t pitch_lag, HFParam *hf)
|
|
{
|
|
int ccr, eng, max_ccr, max_eng;
|
|
int exp, max, diff;
|
|
int energy[15];
|
|
int i, j;
|
|
|
|
for (i = 0, j = pitch_lag - 3; j <= pitch_lag + 3; i++, j++) {
|
|
/* Compute residual energy */
|
|
energy[i << 1] = ff_dot_product(buf - j, buf - j, SUBFRAME_LEN);
|
|
/* Compute correlation */
|
|
energy[(i << 1) + 1] = ff_dot_product(buf, buf - j, SUBFRAME_LEN);
|
|
}
|
|
|
|
/* Compute target energy */
|
|
energy[14] = ff_dot_product(buf, buf, SUBFRAME_LEN);
|
|
|
|
/* Normalize */
|
|
max = 0;
|
|
for (i = 0; i < 15; i++)
|
|
max = FFMAX(max, FFABS(energy[i]));
|
|
|
|
exp = ff_g723_1_normalize_bits(max, 31);
|
|
for (i = 0; i < 15; i++) {
|
|
energy[i] = av_clipl_int32((int64_t)(energy[i] << exp) +
|
|
(1 << 15)) >> 16;
|
|
}
|
|
|
|
hf->index = -1;
|
|
hf->gain = 0;
|
|
max_ccr = 1;
|
|
max_eng = 0x7fff;
|
|
|
|
for (i = 0; i <= 6; i++) {
|
|
eng = energy[i << 1];
|
|
ccr = energy[(i << 1) + 1];
|
|
|
|
if (ccr <= 0)
|
|
continue;
|
|
|
|
ccr = (ccr * ccr + (1 << 14)) >> 15;
|
|
diff = ccr * max_eng - eng * max_ccr;
|
|
if (diff > 0) {
|
|
max_ccr = ccr;
|
|
max_eng = eng;
|
|
hf->index = i;
|
|
}
|
|
}
|
|
|
|
if (hf->index == -1) {
|
|
hf->index = pitch_lag;
|
|
return;
|
|
}
|
|
|
|
eng = energy[14] * max_eng;
|
|
eng = (eng >> 2) + (eng >> 3);
|
|
ccr = energy[(hf->index << 1) + 1] * energy[(hf->index << 1) + 1];
|
|
if (eng < ccr) {
|
|
eng = energy[(hf->index << 1) + 1];
|
|
|
|
if (eng >= max_eng)
|
|
hf->gain = 0x2800;
|
|
else
|
|
hf->gain = ((eng << 15) / max_eng * 0x2800 + (1 << 14)) >> 15;
|
|
}
|
|
hf->index += pitch_lag - 3;
|
|
}
|
|
|
|
/**
|
|
* Apply the harmonic noise shaping filter.
|
|
*
|
|
* @param hf filter parameters
|
|
*/
|
|
static void harmonic_filter(HFParam *hf, const int16_t *src, int16_t *dest)
|
|
{
|
|
int i;
|
|
|
|
for (i = 0; i < SUBFRAME_LEN; i++) {
|
|
int64_t temp = hf->gain * src[i - hf->index] << 1;
|
|
dest[i] = av_clipl_int32((src[i] << 16) - temp + (1 << 15)) >> 16;
|
|
}
|
|
}
|
|
|
|
static void harmonic_noise_sub(HFParam *hf, const int16_t *src, int16_t *dest)
|
|
{
|
|
int i;
|
|
for (i = 0; i < SUBFRAME_LEN; i++) {
|
|
int64_t temp = hf->gain * src[i - hf->index] << 1;
|
|
dest[i] = av_clipl_int32(((dest[i] - src[i]) << 16) + temp +
|
|
(1 << 15)) >> 16;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* Combined synthesis and formant perceptual weighting filer.
|
|
*
|
|
* @param qnt_lpc quantized lpc coefficients
|
|
* @param perf_lpc perceptual filter coefficients
|
|
* @param perf_fir perceptual filter fir memory
|
|
* @param perf_iir perceptual filter iir memory
|
|
* @param scale the filter output will be scaled by 2^scale
|
|
*/
|
|
static void synth_percept_filter(int16_t *qnt_lpc, int16_t *perf_lpc,
|
|
int16_t *perf_fir, int16_t *perf_iir,
|
|
const int16_t *src, int16_t *dest, int scale)
|
|
{
|
|
int i, j;
|
|
int16_t buf_16[SUBFRAME_LEN + LPC_ORDER];
|
|
int64_t buf[SUBFRAME_LEN];
|
|
|
|
int16_t *bptr_16 = buf_16 + LPC_ORDER;
|
|
|
|
memcpy(buf_16, perf_fir, sizeof(int16_t) * LPC_ORDER);
|
|
memcpy(dest - LPC_ORDER, perf_iir, sizeof(int16_t) * LPC_ORDER);
|
|
|
|
for (i = 0; i < SUBFRAME_LEN; i++) {
|
|
int64_t temp = 0;
|
|
for (j = 1; j <= LPC_ORDER; j++)
|
|
temp -= qnt_lpc[j - 1] * bptr_16[i - j];
|
|
|
|
buf[i] = (src[i] << 15) + (temp << 3);
|
|
bptr_16[i] = av_clipl_int32(buf[i] + (1 << 15)) >> 16;
|
|
}
|
|
|
|
for (i = 0; i < SUBFRAME_LEN; i++) {
|
|
int64_t fir = 0, iir = 0;
|
|
for (j = 1; j <= LPC_ORDER; j++) {
|
|
fir -= perf_lpc[j - 1] * bptr_16[i - j];
|
|
iir += perf_lpc[j + LPC_ORDER - 1] * dest[i - j];
|
|
}
|
|
dest[i] = av_clipl_int32(((buf[i] + (fir << 3)) << scale) + (iir << 3) +
|
|
(1 << 15)) >> 16;
|
|
}
|
|
memcpy(perf_fir, buf_16 + SUBFRAME_LEN, sizeof(int16_t) * LPC_ORDER);
|
|
memcpy(perf_iir, dest + SUBFRAME_LEN - LPC_ORDER,
|
|
sizeof(int16_t) * LPC_ORDER);
|
|
}
|
|
|
|
/**
|
|
* Compute the adaptive codebook contribution.
|
|
*
|
|
* @param buf input signal
|
|
* @param index the current subframe index
|
|
*/
|
|
static void acb_search(G723_1_ChannelContext *p, int16_t *residual,
|
|
int16_t *impulse_resp, const int16_t *buf,
|
|
int index)
|
|
{
|
|
int16_t flt_buf[PITCH_ORDER][SUBFRAME_LEN];
|
|
|
|
const int16_t *cb_tbl = adaptive_cb_gain85;
|
|
|
|
int ccr_buf[PITCH_ORDER * SUBFRAMES << 2];
|
|
|
|
int pitch_lag = p->pitch_lag[index >> 1];
|
|
int acb_lag = 1;
|
|
int acb_gain = 0;
|
|
int odd_frame = index & 1;
|
|
int iter = 3 + odd_frame;
|
|
int count = 0;
|
|
int tbl_size = 85;
|
|
|
|
int i, j, k, l, max;
|
|
int64_t temp;
|
|
|
|
if (!odd_frame) {
|
|
if (pitch_lag == PITCH_MIN)
|
|
pitch_lag++;
|
|
else
|
|
pitch_lag = FFMIN(pitch_lag, PITCH_MAX - 5);
|
|
}
|
|
|
|
for (i = 0; i < iter; i++) {
|
|
ff_g723_1_get_residual(residual, p->prev_excitation, pitch_lag + i - 1);
|
|
|
|
for (j = 0; j < SUBFRAME_LEN; j++) {
|
|
temp = 0;
|
|
for (k = 0; k <= j; k++)
|
|
temp += residual[PITCH_ORDER - 1 + k] * impulse_resp[j - k];
|
|
flt_buf[PITCH_ORDER - 1][j] = av_clipl_int32((temp << 1) +
|
|
(1 << 15)) >> 16;
|
|
}
|
|
|
|
for (j = PITCH_ORDER - 2; j >= 0; j--) {
|
|
flt_buf[j][0] = ((residual[j] << 13) + (1 << 14)) >> 15;
|
|
for (k = 1; k < SUBFRAME_LEN; k++) {
|
|
temp = (flt_buf[j + 1][k - 1] << 15) +
|
|
residual[j] * impulse_resp[k];
|
|
flt_buf[j][k] = av_clipl_int32((temp << 1) + (1 << 15)) >> 16;
|
|
}
|
|
}
|
|
|
|
/* Compute crosscorrelation with the signal */
|
|
for (j = 0; j < PITCH_ORDER; j++) {
|
|
temp = ff_dot_product(buf, flt_buf[j], SUBFRAME_LEN);
|
|
ccr_buf[count++] = av_clipl_int32(temp << 1);
|
|
}
|
|
|
|
/* Compute energies */
|
|
for (j = 0; j < PITCH_ORDER; j++) {
|
|
ccr_buf[count++] = ff_g723_1_dot_product(flt_buf[j], flt_buf[j],
|
|
SUBFRAME_LEN);
|
|
}
|
|
|
|
for (j = 1; j < PITCH_ORDER; j++) {
|
|
for (k = 0; k < j; k++) {
|
|
temp = ff_dot_product(flt_buf[j], flt_buf[k], SUBFRAME_LEN);
|
|
ccr_buf[count++] = av_clipl_int32(temp << 2);
|
|
}
|
|
}
|
|
}
|
|
|
|
/* Normalize and shorten */
|
|
max = 0;
|
|
for (i = 0; i < 20 * iter; i++)
|
|
max = FFMAX(max, FFABS(ccr_buf[i]));
|
|
|
|
temp = ff_g723_1_normalize_bits(max, 31);
|
|
|
|
for (i = 0; i < 20 * iter; i++)
|
|
ccr_buf[i] = av_clipl_int32((int64_t) (ccr_buf[i] << temp) +
|
|
(1 << 15)) >> 16;
|
|
|
|
max = 0;
|
|
for (i = 0; i < iter; i++) {
|
|
/* Select quantization table */
|
|
if (!odd_frame && pitch_lag + i - 1 >= SUBFRAME_LEN - 2 ||
|
|
odd_frame && pitch_lag >= SUBFRAME_LEN - 2) {
|
|
cb_tbl = adaptive_cb_gain170;
|
|
tbl_size = 170;
|
|
}
|
|
|
|
for (j = 0, k = 0; j < tbl_size; j++, k += 20) {
|
|
temp = 0;
|
|
for (l = 0; l < 20; l++)
|
|
temp += ccr_buf[20 * i + l] * cb_tbl[k + l];
|
|
temp = av_clipl_int32(temp);
|
|
|
|
if (temp > max) {
|
|
max = temp;
|
|
acb_gain = j;
|
|
acb_lag = i;
|
|
}
|
|
}
|
|
}
|
|
|
|
if (!odd_frame) {
|
|
pitch_lag += acb_lag - 1;
|
|
acb_lag = 1;
|
|
}
|
|
|
|
p->pitch_lag[index >> 1] = pitch_lag;
|
|
p->subframe[index].ad_cb_lag = acb_lag;
|
|
p->subframe[index].ad_cb_gain = acb_gain;
|
|
}
|
|
|
|
/**
|
|
* Subtract the adaptive codebook contribution from the input
|
|
* to obtain the residual.
|
|
*
|
|
* @param buf target vector
|
|
*/
|
|
static void sub_acb_contrib(const int16_t *residual, const int16_t *impulse_resp,
|
|
int16_t *buf)
|
|
{
|
|
int i, j;
|
|
/* Subtract adaptive CB contribution to obtain the residual */
|
|
for (i = 0; i < SUBFRAME_LEN; i++) {
|
|
int64_t temp = buf[i] << 14;
|
|
for (j = 0; j <= i; j++)
|
|
temp -= residual[j] * impulse_resp[i - j];
|
|
|
|
buf[i] = av_clipl_int32((temp << 2) + (1 << 15)) >> 16;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* Quantize the residual signal using the fixed codebook (MP-MLQ).
|
|
*
|
|
* @param optim optimized fixed codebook parameters
|
|
* @param buf excitation vector
|
|
*/
|
|
static void get_fcb_param(FCBParam *optim, int16_t *impulse_resp,
|
|
int16_t *buf, int pulse_cnt, int pitch_lag)
|
|
{
|
|
FCBParam param;
|
|
int16_t impulse_r[SUBFRAME_LEN];
|
|
int16_t temp_corr[SUBFRAME_LEN];
|
|
int16_t impulse_corr[SUBFRAME_LEN];
|
|
|
|
int ccr1[SUBFRAME_LEN];
|
|
int ccr2[SUBFRAME_LEN];
|
|
int amp, err, max, max_amp_index, min, scale, i, j, k, l;
|
|
|
|
int64_t temp;
|
|
|
|
/* Update impulse response */
|
|
memcpy(impulse_r, impulse_resp, sizeof(int16_t) * SUBFRAME_LEN);
|
|
param.dirac_train = 0;
|
|
if (pitch_lag < SUBFRAME_LEN - 2) {
|
|
param.dirac_train = 1;
|
|
ff_g723_1_gen_dirac_train(impulse_r, pitch_lag);
|
|
}
|
|
|
|
for (i = 0; i < SUBFRAME_LEN; i++)
|
|
temp_corr[i] = impulse_r[i] >> 1;
|
|
|
|
/* Compute impulse response autocorrelation */
|
|
temp = ff_g723_1_dot_product(temp_corr, temp_corr, SUBFRAME_LEN);
|
|
|
|
scale = ff_g723_1_normalize_bits(temp, 31);
|
|
impulse_corr[0] = av_clipl_int32((temp << scale) + (1 << 15)) >> 16;
|
|
|
|
for (i = 1; i < SUBFRAME_LEN; i++) {
|
|
temp = ff_g723_1_dot_product(temp_corr + i, temp_corr,
|
|
SUBFRAME_LEN - i);
|
|
impulse_corr[i] = av_clipl_int32((temp << scale) + (1 << 15)) >> 16;
|
|
}
|
|
|
|
/* Compute crosscorrelation of impulse response with residual signal */
|
|
scale -= 4;
|
|
for (i = 0; i < SUBFRAME_LEN; i++) {
|
|
temp = ff_g723_1_dot_product(buf + i, impulse_r, SUBFRAME_LEN - i);
|
|
if (scale < 0)
|
|
ccr1[i] = temp >> -scale;
|
|
else
|
|
ccr1[i] = av_clipl_int32(temp << scale);
|
|
}
|
|
|
|
/* Search loop */
|
|
for (i = 0; i < GRID_SIZE; i++) {
|
|
/* Maximize the crosscorrelation */
|
|
max = 0;
|
|
for (j = i; j < SUBFRAME_LEN; j += GRID_SIZE) {
|
|
temp = FFABS(ccr1[j]);
|
|
if (temp >= max) {
|
|
max = temp;
|
|
param.pulse_pos[0] = j;
|
|
}
|
|
}
|
|
|
|
/* Quantize the gain (max crosscorrelation/impulse_corr[0]) */
|
|
amp = max;
|
|
min = 1 << 30;
|
|
max_amp_index = GAIN_LEVELS - 2;
|
|
for (j = max_amp_index; j >= 2; j--) {
|
|
temp = av_clipl_int32((int64_t) fixed_cb_gain[j] *
|
|
impulse_corr[0] << 1);
|
|
temp = FFABS(temp - amp);
|
|
if (temp < min) {
|
|
min = temp;
|
|
max_amp_index = j;
|
|
}
|
|
}
|
|
|
|
max_amp_index--;
|
|
/* Select additional gain values */
|
|
for (j = 1; j < 5; j++) {
|
|
for (k = i; k < SUBFRAME_LEN; k += GRID_SIZE) {
|
|
temp_corr[k] = 0;
|
|
ccr2[k] = ccr1[k];
|
|
}
|
|
param.amp_index = max_amp_index + j - 2;
|
|
amp = fixed_cb_gain[param.amp_index];
|
|
|
|
param.pulse_sign[0] = (ccr2[param.pulse_pos[0]] < 0) ? -amp : amp;
|
|
temp_corr[param.pulse_pos[0]] = 1;
|
|
|
|
for (k = 1; k < pulse_cnt; k++) {
|
|
max = INT_MIN;
|
|
for (l = i; l < SUBFRAME_LEN; l += GRID_SIZE) {
|
|
if (temp_corr[l])
|
|
continue;
|
|
temp = impulse_corr[FFABS(l - param.pulse_pos[k - 1])];
|
|
temp = av_clipl_int32((int64_t) temp *
|
|
param.pulse_sign[k - 1] << 1);
|
|
ccr2[l] -= temp;
|
|
temp = FFABS(ccr2[l]);
|
|
if (temp > max) {
|
|
max = temp;
|
|
param.pulse_pos[k] = l;
|
|
}
|
|
}
|
|
|
|
param.pulse_sign[k] = (ccr2[param.pulse_pos[k]] < 0) ?
|
|
-amp : amp;
|
|
temp_corr[param.pulse_pos[k]] = 1;
|
|
}
|
|
|
|
/* Create the error vector */
|
|
memset(temp_corr, 0, sizeof(int16_t) * SUBFRAME_LEN);
|
|
|
|
for (k = 0; k < pulse_cnt; k++)
|
|
temp_corr[param.pulse_pos[k]] = param.pulse_sign[k];
|
|
|
|
for (k = SUBFRAME_LEN - 1; k >= 0; k--) {
|
|
temp = 0;
|
|
for (l = 0; l <= k; l++) {
|
|
int prod = av_clipl_int32((int64_t) temp_corr[l] *
|
|
impulse_r[k - l] << 1);
|
|
temp = av_clipl_int32(temp + prod);
|
|
}
|
|
temp_corr[k] = temp << 2 >> 16;
|
|
}
|
|
|
|
/* Compute square of error */
|
|
err = 0;
|
|
for (k = 0; k < SUBFRAME_LEN; k++) {
|
|
int64_t prod;
|
|
prod = av_clipl_int32((int64_t) buf[k] * temp_corr[k] << 1);
|
|
err = av_clipl_int32(err - prod);
|
|
prod = av_clipl_int32((int64_t) temp_corr[k] * temp_corr[k]);
|
|
err = av_clipl_int32(err + prod);
|
|
}
|
|
|
|
/* Minimize */
|
|
if (err < optim->min_err) {
|
|
optim->min_err = err;
|
|
optim->grid_index = i;
|
|
optim->amp_index = param.amp_index;
|
|
optim->dirac_train = param.dirac_train;
|
|
|
|
for (k = 0; k < pulse_cnt; k++) {
|
|
optim->pulse_sign[k] = param.pulse_sign[k];
|
|
optim->pulse_pos[k] = param.pulse_pos[k];
|
|
}
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
/**
|
|
* Encode the pulse position and gain of the current subframe.
|
|
*
|
|
* @param optim optimized fixed CB parameters
|
|
* @param buf excitation vector
|
|
*/
|
|
static void pack_fcb_param(G723_1_Subframe *subfrm, FCBParam *optim,
|
|
int16_t *buf, int pulse_cnt)
|
|
{
|
|
int i, j;
|
|
|
|
j = PULSE_MAX - pulse_cnt;
|
|
|
|
subfrm->pulse_sign = 0;
|
|
subfrm->pulse_pos = 0;
|
|
|
|
for (i = 0; i < SUBFRAME_LEN >> 1; i++) {
|
|
int val = buf[optim->grid_index + (i << 1)];
|
|
if (!val) {
|
|
subfrm->pulse_pos += combinatorial_table[j][i];
|
|
} else {
|
|
subfrm->pulse_sign <<= 1;
|
|
if (val < 0)
|
|
subfrm->pulse_sign++;
|
|
j++;
|
|
|
|
if (j == PULSE_MAX)
|
|
break;
|
|
}
|
|
}
|
|
subfrm->amp_index = optim->amp_index;
|
|
subfrm->grid_index = optim->grid_index;
|
|
subfrm->dirac_train = optim->dirac_train;
|
|
}
|
|
|
|
/**
|
|
* Compute the fixed codebook excitation.
|
|
*
|
|
* @param buf target vector
|
|
* @param impulse_resp impulse response of the combined filter
|
|
*/
|
|
static void fcb_search(G723_1_ChannelContext *p, int16_t *impulse_resp,
|
|
int16_t *buf, int index)
|
|
{
|
|
FCBParam optim;
|
|
int pulse_cnt = pulses[index];
|
|
int i;
|
|
|
|
optim.min_err = 1 << 30;
|
|
get_fcb_param(&optim, impulse_resp, buf, pulse_cnt, SUBFRAME_LEN);
|
|
|
|
if (p->pitch_lag[index >> 1] < SUBFRAME_LEN - 2) {
|
|
get_fcb_param(&optim, impulse_resp, buf, pulse_cnt,
|
|
p->pitch_lag[index >> 1]);
|
|
}
|
|
|
|
/* Reconstruct the excitation */
|
|
memset(buf, 0, sizeof(int16_t) * SUBFRAME_LEN);
|
|
for (i = 0; i < pulse_cnt; i++)
|
|
buf[optim.pulse_pos[i]] = optim.pulse_sign[i];
|
|
|
|
pack_fcb_param(&p->subframe[index], &optim, buf, pulse_cnt);
|
|
|
|
if (optim.dirac_train)
|
|
ff_g723_1_gen_dirac_train(buf, p->pitch_lag[index >> 1]);
|
|
}
|
|
|
|
/**
|
|
* Pack the frame parameters into output bitstream.
|
|
*
|
|
* @param frame output buffer
|
|
* @param size size of the buffer
|
|
*/
|
|
static int pack_bitstream(G723_1_ChannelContext *p, AVPacket *avpkt)
|
|
{
|
|
PutBitContext pb;
|
|
int info_bits = 0;
|
|
int i, temp;
|
|
|
|
init_put_bits(&pb, avpkt->data, avpkt->size);
|
|
|
|
put_bits(&pb, 2, info_bits);
|
|
|
|
put_bits(&pb, 8, p->lsp_index[2]);
|
|
put_bits(&pb, 8, p->lsp_index[1]);
|
|
put_bits(&pb, 8, p->lsp_index[0]);
|
|
|
|
put_bits(&pb, 7, p->pitch_lag[0] - PITCH_MIN);
|
|
put_bits(&pb, 2, p->subframe[1].ad_cb_lag);
|
|
put_bits(&pb, 7, p->pitch_lag[1] - PITCH_MIN);
|
|
put_bits(&pb, 2, p->subframe[3].ad_cb_lag);
|
|
|
|
/* Write 12 bit combined gain */
|
|
for (i = 0; i < SUBFRAMES; i++) {
|
|
temp = p->subframe[i].ad_cb_gain * GAIN_LEVELS +
|
|
p->subframe[i].amp_index;
|
|
if (p->cur_rate == RATE_6300)
|
|
temp += p->subframe[i].dirac_train << 11;
|
|
put_bits(&pb, 12, temp);
|
|
}
|
|
|
|
put_bits(&pb, 1, p->subframe[0].grid_index);
|
|
put_bits(&pb, 1, p->subframe[1].grid_index);
|
|
put_bits(&pb, 1, p->subframe[2].grid_index);
|
|
put_bits(&pb, 1, p->subframe[3].grid_index);
|
|
|
|
if (p->cur_rate == RATE_6300) {
|
|
skip_put_bits(&pb, 1); /* reserved bit */
|
|
|
|
/* Write 13 bit combined position index */
|
|
temp = (p->subframe[0].pulse_pos >> 16) * 810 +
|
|
(p->subframe[1].pulse_pos >> 14) * 90 +
|
|
(p->subframe[2].pulse_pos >> 16) * 9 +
|
|
(p->subframe[3].pulse_pos >> 14);
|
|
put_bits(&pb, 13, temp);
|
|
|
|
put_bits(&pb, 16, p->subframe[0].pulse_pos & 0xffff);
|
|
put_bits(&pb, 14, p->subframe[1].pulse_pos & 0x3fff);
|
|
put_bits(&pb, 16, p->subframe[2].pulse_pos & 0xffff);
|
|
put_bits(&pb, 14, p->subframe[3].pulse_pos & 0x3fff);
|
|
|
|
put_bits(&pb, 6, p->subframe[0].pulse_sign);
|
|
put_bits(&pb, 5, p->subframe[1].pulse_sign);
|
|
put_bits(&pb, 6, p->subframe[2].pulse_sign);
|
|
put_bits(&pb, 5, p->subframe[3].pulse_sign);
|
|
}
|
|
|
|
flush_put_bits(&pb);
|
|
return frame_size[info_bits];
|
|
}
|
|
|
|
static int g723_1_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
|
|
const AVFrame *frame, int *got_packet_ptr)
|
|
{
|
|
G723_1_Context *s = avctx->priv_data;
|
|
G723_1_ChannelContext *p = &s->ch[0];
|
|
int16_t unq_lpc[LPC_ORDER * SUBFRAMES];
|
|
int16_t qnt_lpc[LPC_ORDER * SUBFRAMES];
|
|
int16_t cur_lsp[LPC_ORDER];
|
|
int16_t weighted_lpc[LPC_ORDER * SUBFRAMES << 1];
|
|
int16_t vector[FRAME_LEN + PITCH_MAX];
|
|
int offset, ret, i, j;
|
|
int16_t *in, *start;
|
|
HFParam hf[4];
|
|
|
|
/* duplicate input */
|
|
start = in = av_malloc(frame->nb_samples * sizeof(int16_t));
|
|
if (!in)
|
|
return AVERROR(ENOMEM);
|
|
memcpy(in, frame->data[0], frame->nb_samples * sizeof(int16_t));
|
|
|
|
highpass_filter(in, &p->hpf_fir_mem, &p->hpf_iir_mem);
|
|
|
|
memcpy(vector, p->prev_data, HALF_FRAME_LEN * sizeof(int16_t));
|
|
memcpy(vector + HALF_FRAME_LEN, in, FRAME_LEN * sizeof(int16_t));
|
|
|
|
comp_lpc_coeff(vector, unq_lpc);
|
|
lpc2lsp(&unq_lpc[LPC_ORDER * 3], p->prev_lsp, cur_lsp);
|
|
lsp_quantize(p->lsp_index, cur_lsp, p->prev_lsp);
|
|
|
|
/* Update memory */
|
|
memcpy(vector + LPC_ORDER, p->prev_data + SUBFRAME_LEN,
|
|
sizeof(int16_t) * SUBFRAME_LEN);
|
|
memcpy(vector + LPC_ORDER + SUBFRAME_LEN, in,
|
|
sizeof(int16_t) * (HALF_FRAME_LEN + SUBFRAME_LEN));
|
|
memcpy(p->prev_data, in + HALF_FRAME_LEN,
|
|
sizeof(int16_t) * HALF_FRAME_LEN);
|
|
memcpy(in, vector + LPC_ORDER, sizeof(int16_t) * FRAME_LEN);
|
|
|
|
perceptual_filter(p, weighted_lpc, unq_lpc, vector);
|
|
|
|
memcpy(in, vector + LPC_ORDER, sizeof(int16_t) * FRAME_LEN);
|
|
memcpy(vector, p->prev_weight_sig, sizeof(int16_t) * PITCH_MAX);
|
|
memcpy(vector + PITCH_MAX, in, sizeof(int16_t) * FRAME_LEN);
|
|
|
|
ff_g723_1_scale_vector(vector, vector, FRAME_LEN + PITCH_MAX);
|
|
|
|
p->pitch_lag[0] = estimate_pitch(vector, PITCH_MAX);
|
|
p->pitch_lag[1] = estimate_pitch(vector, PITCH_MAX + HALF_FRAME_LEN);
|
|
|
|
for (i = PITCH_MAX, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
|
|
comp_harmonic_coeff(vector + i, p->pitch_lag[j >> 1], hf + j);
|
|
|
|
memcpy(vector, p->prev_weight_sig, sizeof(int16_t) * PITCH_MAX);
|
|
memcpy(vector + PITCH_MAX, in, sizeof(int16_t) * FRAME_LEN);
|
|
memcpy(p->prev_weight_sig, vector + FRAME_LEN, sizeof(int16_t) * PITCH_MAX);
|
|
|
|
for (i = 0, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
|
|
harmonic_filter(hf + j, vector + PITCH_MAX + i, in + i);
|
|
|
|
ff_g723_1_inverse_quant(cur_lsp, p->prev_lsp, p->lsp_index, 0);
|
|
ff_g723_1_lsp_interpolate(qnt_lpc, cur_lsp, p->prev_lsp);
|
|
|
|
memcpy(p->prev_lsp, cur_lsp, sizeof(int16_t) * LPC_ORDER);
|
|
|
|
offset = 0;
|
|
for (i = 0; i < SUBFRAMES; i++) {
|
|
int16_t impulse_resp[SUBFRAME_LEN];
|
|
int16_t residual[SUBFRAME_LEN + PITCH_ORDER - 1];
|
|
int16_t flt_in[SUBFRAME_LEN];
|
|
int16_t zero[LPC_ORDER], fir[LPC_ORDER], iir[LPC_ORDER];
|
|
|
|
/**
|
|
* Compute the combined impulse response of the synthesis filter,
|
|
* formant perceptual weighting filter and harmonic noise shaping filter
|
|
*/
|
|
memset(zero, 0, sizeof(int16_t) * LPC_ORDER);
|
|
memset(vector, 0, sizeof(int16_t) * PITCH_MAX);
|
|
memset(flt_in, 0, sizeof(int16_t) * SUBFRAME_LEN);
|
|
|
|
flt_in[0] = 1 << 13; /* Unit impulse */
|
|
synth_percept_filter(qnt_lpc + offset, weighted_lpc + (offset << 1),
|
|
zero, zero, flt_in, vector + PITCH_MAX, 1);
|
|
harmonic_filter(hf + i, vector + PITCH_MAX, impulse_resp);
|
|
|
|
/* Compute the combined zero input response */
|
|
flt_in[0] = 0;
|
|
memcpy(fir, p->perf_fir_mem, sizeof(int16_t) * LPC_ORDER);
|
|
memcpy(iir, p->perf_iir_mem, sizeof(int16_t) * LPC_ORDER);
|
|
|
|
synth_percept_filter(qnt_lpc + offset, weighted_lpc + (offset << 1),
|
|
fir, iir, flt_in, vector + PITCH_MAX, 0);
|
|
memcpy(vector, p->harmonic_mem, sizeof(int16_t) * PITCH_MAX);
|
|
harmonic_noise_sub(hf + i, vector + PITCH_MAX, in);
|
|
|
|
acb_search(p, residual, impulse_resp, in, i);
|
|
ff_g723_1_gen_acb_excitation(residual, p->prev_excitation,
|
|
p->pitch_lag[i >> 1], &p->subframe[i],
|
|
p->cur_rate);
|
|
sub_acb_contrib(residual, impulse_resp, in);
|
|
|
|
fcb_search(p, impulse_resp, in, i);
|
|
|
|
/* Reconstruct the excitation */
|
|
ff_g723_1_gen_acb_excitation(impulse_resp, p->prev_excitation,
|
|
p->pitch_lag[i >> 1], &p->subframe[i],
|
|
RATE_6300);
|
|
|
|
memmove(p->prev_excitation, p->prev_excitation + SUBFRAME_LEN,
|
|
sizeof(int16_t) * (PITCH_MAX - SUBFRAME_LEN));
|
|
for (j = 0; j < SUBFRAME_LEN; j++)
|
|
in[j] = av_clip_int16((in[j] << 1) + impulse_resp[j]);
|
|
memcpy(p->prev_excitation + PITCH_MAX - SUBFRAME_LEN, in,
|
|
sizeof(int16_t) * SUBFRAME_LEN);
|
|
|
|
/* Update filter memories */
|
|
synth_percept_filter(qnt_lpc + offset, weighted_lpc + (offset << 1),
|
|
p->perf_fir_mem, p->perf_iir_mem,
|
|
in, vector + PITCH_MAX, 0);
|
|
memmove(p->harmonic_mem, p->harmonic_mem + SUBFRAME_LEN,
|
|
sizeof(int16_t) * (PITCH_MAX - SUBFRAME_LEN));
|
|
memcpy(p->harmonic_mem + PITCH_MAX - SUBFRAME_LEN, vector + PITCH_MAX,
|
|
sizeof(int16_t) * SUBFRAME_LEN);
|
|
|
|
in += SUBFRAME_LEN;
|
|
offset += LPC_ORDER;
|
|
}
|
|
|
|
av_free(start);
|
|
|
|
if ((ret = ff_alloc_packet2(avctx, avpkt, 24, 0)) < 0)
|
|
return ret;
|
|
|
|
*got_packet_ptr = 1;
|
|
avpkt->size = pack_bitstream(p, avpkt);
|
|
return 0;
|
|
}
|
|
|
|
static const AVCodecDefault defaults[] = {
|
|
{ "b", "6300" },
|
|
{ NULL },
|
|
};
|
|
|
|
AVCodec ff_g723_1_encoder = {
|
|
.name = "g723_1",
|
|
.long_name = NULL_IF_CONFIG_SMALL("G.723.1"),
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
.id = AV_CODEC_ID_G723_1,
|
|
.priv_data_size = sizeof(G723_1_Context),
|
|
.init = g723_1_encode_init,
|
|
.encode2 = g723_1_encode_frame,
|
|
.defaults = defaults,
|
|
.sample_fmts = (const enum AVSampleFormat[]) {
|
|
AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
|
|
},
|
|
};
|