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1381 lines
42 KiB
1381 lines
42 KiB
/* |
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* G.723.1 compatible decoder |
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* Copyright (c) 2006 Benjamin Larsson |
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* Copyright (c) 2010 Mohamed Naufal Basheer |
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* |
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* This file is part of Libav. |
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* |
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* Libav 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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* Libav 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 Libav; 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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/** |
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* @file |
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* G.723.1 compatible decoder |
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*/ |
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|
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#define BITSTREAM_READER_LE |
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#include "libavutil/audioconvert.h" |
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#include "libavutil/lzo.h" |
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#include "libavutil/opt.h" |
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#include "avcodec.h" |
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#include "get_bits.h" |
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#include "acelp_vectors.h" |
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#include "celp_filters.h" |
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#include "g723_1_data.h" |
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|
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#define CNG_RANDOM_SEED 12345 |
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|
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/** |
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* G723.1 frame types |
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*/ |
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enum FrameType { |
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ACTIVE_FRAME, ///< Active speech |
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SID_FRAME, ///< Silence Insertion Descriptor frame |
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UNTRANSMITTED_FRAME |
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}; |
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|
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enum Rate { |
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RATE_6300, |
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RATE_5300 |
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}; |
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|
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/** |
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* G723.1 unpacked data subframe |
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*/ |
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typedef struct { |
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int ad_cb_lag; ///< adaptive codebook lag |
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int ad_cb_gain; |
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int dirac_train; |
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int pulse_sign; |
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int grid_index; |
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int amp_index; |
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int pulse_pos; |
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} G723_1_Subframe; |
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|
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/** |
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* Pitch postfilter parameters |
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*/ |
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typedef struct { |
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int index; ///< postfilter backward/forward lag |
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int16_t opt_gain; ///< optimal gain |
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int16_t sc_gain; ///< scaling gain |
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} PPFParam; |
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|
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typedef struct g723_1_context { |
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AVClass *class; |
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AVFrame frame; |
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|
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G723_1_Subframe subframe[4]; |
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enum FrameType cur_frame_type; |
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enum FrameType past_frame_type; |
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enum Rate cur_rate; |
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uint8_t lsp_index[LSP_BANDS]; |
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int pitch_lag[2]; |
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int erased_frames; |
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|
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int16_t prev_lsp[LPC_ORDER]; |
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int16_t sid_lsp[LPC_ORDER]; |
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int16_t prev_excitation[PITCH_MAX]; |
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int16_t excitation[PITCH_MAX + FRAME_LEN + 4]; |
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int16_t synth_mem[LPC_ORDER]; |
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int16_t fir_mem[LPC_ORDER]; |
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int iir_mem[LPC_ORDER]; |
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|
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int random_seed; |
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int cng_random_seed; |
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int interp_index; |
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int interp_gain; |
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int sid_gain; |
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int cur_gain; |
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int reflection_coef; |
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int pf_gain; |
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int postfilter; |
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int16_t audio[FRAME_LEN + LPC_ORDER + PITCH_MAX + 4]; |
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} G723_1_Context; |
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|
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static av_cold int g723_1_decode_init(AVCodecContext *avctx) |
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{ |
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G723_1_Context *p = avctx->priv_data; |
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|
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avctx->channel_layout = AV_CH_LAYOUT_MONO; |
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avctx->sample_fmt = AV_SAMPLE_FMT_S16; |
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avctx->channels = 1; |
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avctx->sample_rate = 8000; |
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p->pf_gain = 1 << 12; |
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|
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avcodec_get_frame_defaults(&p->frame); |
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avctx->coded_frame = &p->frame; |
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memcpy(p->prev_lsp, dc_lsp, LPC_ORDER * sizeof(*p->prev_lsp)); |
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memcpy(p->sid_lsp, dc_lsp, LPC_ORDER * sizeof(*p->sid_lsp)); |
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|
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p->cng_random_seed = CNG_RANDOM_SEED; |
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p->past_frame_type = SID_FRAME; |
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return 0; |
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} |
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|
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/** |
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* Unpack the frame into parameters. |
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* |
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* @param p the context |
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* @param buf pointer to the input buffer |
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* @param buf_size size of the input buffer |
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*/ |
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static int unpack_bitstream(G723_1_Context *p, const uint8_t *buf, |
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int buf_size) |
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{ |
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GetBitContext gb; |
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int ad_cb_len; |
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int temp, info_bits, i; |
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|
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init_get_bits(&gb, buf, buf_size * 8); |
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|
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/* Extract frame type and rate info */ |
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info_bits = get_bits(&gb, 2); |
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|
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if (info_bits == 3) { |
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p->cur_frame_type = UNTRANSMITTED_FRAME; |
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return 0; |
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} |
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|
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/* Extract 24 bit lsp indices, 8 bit for each band */ |
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p->lsp_index[2] = get_bits(&gb, 8); |
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p->lsp_index[1] = get_bits(&gb, 8); |
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p->lsp_index[0] = get_bits(&gb, 8); |
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|
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if (info_bits == 2) { |
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p->cur_frame_type = SID_FRAME; |
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p->subframe[0].amp_index = get_bits(&gb, 6); |
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return 0; |
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} |
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|
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/* Extract the info common to both rates */ |
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p->cur_rate = info_bits ? RATE_5300 : RATE_6300; |
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p->cur_frame_type = ACTIVE_FRAME; |
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|
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p->pitch_lag[0] = get_bits(&gb, 7); |
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if (p->pitch_lag[0] > 123) /* test if forbidden code */ |
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return -1; |
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p->pitch_lag[0] += PITCH_MIN; |
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p->subframe[1].ad_cb_lag = get_bits(&gb, 2); |
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p->pitch_lag[1] = get_bits(&gb, 7); |
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if (p->pitch_lag[1] > 123) |
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return -1; |
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p->pitch_lag[1] += PITCH_MIN; |
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p->subframe[3].ad_cb_lag = get_bits(&gb, 2); |
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p->subframe[0].ad_cb_lag = 1; |
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p->subframe[2].ad_cb_lag = 1; |
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|
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for (i = 0; i < SUBFRAMES; i++) { |
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/* Extract combined gain */ |
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temp = get_bits(&gb, 12); |
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ad_cb_len = 170; |
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p->subframe[i].dirac_train = 0; |
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if (p->cur_rate == RATE_6300 && p->pitch_lag[i >> 1] < SUBFRAME_LEN - 2) { |
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p->subframe[i].dirac_train = temp >> 11; |
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temp &= 0x7FF; |
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ad_cb_len = 85; |
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} |
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p->subframe[i].ad_cb_gain = FASTDIV(temp, GAIN_LEVELS); |
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if (p->subframe[i].ad_cb_gain < ad_cb_len) { |
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p->subframe[i].amp_index = temp - p->subframe[i].ad_cb_gain * |
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GAIN_LEVELS; |
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} else { |
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return -1; |
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} |
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} |
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p->subframe[0].grid_index = get_bits(&gb, 1); |
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p->subframe[1].grid_index = get_bits(&gb, 1); |
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p->subframe[2].grid_index = get_bits(&gb, 1); |
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p->subframe[3].grid_index = get_bits(&gb, 1); |
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|
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if (p->cur_rate == RATE_6300) { |
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skip_bits(&gb, 1); /* skip reserved bit */ |
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|
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/* Compute pulse_pos index using the 13-bit combined position index */ |
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temp = get_bits(&gb, 13); |
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p->subframe[0].pulse_pos = temp / 810; |
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temp -= p->subframe[0].pulse_pos * 810; |
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p->subframe[1].pulse_pos = FASTDIV(temp, 90); |
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temp -= p->subframe[1].pulse_pos * 90; |
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p->subframe[2].pulse_pos = FASTDIV(temp, 9); |
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p->subframe[3].pulse_pos = temp - p->subframe[2].pulse_pos * 9; |
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p->subframe[0].pulse_pos = (p->subframe[0].pulse_pos << 16) + |
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get_bits(&gb, 16); |
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p->subframe[1].pulse_pos = (p->subframe[1].pulse_pos << 14) + |
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get_bits(&gb, 14); |
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p->subframe[2].pulse_pos = (p->subframe[2].pulse_pos << 16) + |
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get_bits(&gb, 16); |
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p->subframe[3].pulse_pos = (p->subframe[3].pulse_pos << 14) + |
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get_bits(&gb, 14); |
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|
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p->subframe[0].pulse_sign = get_bits(&gb, 6); |
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p->subframe[1].pulse_sign = get_bits(&gb, 5); |
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p->subframe[2].pulse_sign = get_bits(&gb, 6); |
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p->subframe[3].pulse_sign = get_bits(&gb, 5); |
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} else { /* 5300 bps */ |
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p->subframe[0].pulse_pos = get_bits(&gb, 12); |
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p->subframe[1].pulse_pos = get_bits(&gb, 12); |
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p->subframe[2].pulse_pos = get_bits(&gb, 12); |
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p->subframe[3].pulse_pos = get_bits(&gb, 12); |
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|
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p->subframe[0].pulse_sign = get_bits(&gb, 4); |
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p->subframe[1].pulse_sign = get_bits(&gb, 4); |
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p->subframe[2].pulse_sign = get_bits(&gb, 4); |
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p->subframe[3].pulse_sign = get_bits(&gb, 4); |
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} |
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|
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return 0; |
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} |
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|
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/** |
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* Bitexact implementation of sqrt(val/2). |
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*/ |
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static int16_t square_root(int val) |
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{ |
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int16_t res = 0; |
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int16_t exp = 0x4000; |
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int i; |
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for (i = 0; i < 14; i ++) { |
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int res_exp = res + exp; |
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if (val >= res_exp * res_exp << 1) |
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res += exp; |
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exp >>= 1; |
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} |
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return res; |
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} |
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|
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/** |
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* Calculate the number of left-shifts required for normalizing the input. |
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* |
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* @param num input number |
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* @param width width of the input, 16 bits(0) / 32 bits(1) |
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*/ |
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static int normalize_bits(int num, int width) |
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{ |
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return width - av_log2(num) - 1; |
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} |
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|
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/** |
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* Scale vector contents based on the largest of their absolutes. |
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*/ |
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static int scale_vector(int16_t *dst, const int16_t *vector, int length) |
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{ |
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int bits, max = 0; |
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int i; |
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for (i = 0; i < length; i++) |
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max |= FFABS(vector[i]); |
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max = FFMIN(max, 0x7FFF); |
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bits = normalize_bits(max, 15); |
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for (i = 0; i < length; i++) |
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dst[i] = vector[i] << bits >> 3; |
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return bits - 3; |
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} |
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|
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/** |
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* Perform inverse quantization of LSP frequencies. |
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* |
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* @param cur_lsp the current LSP vector |
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* @param prev_lsp the previous LSP vector |
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* @param lsp_index VQ indices |
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* @param bad_frame bad frame flag |
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*/ |
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static void inverse_quant(int16_t *cur_lsp, int16_t *prev_lsp, |
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uint8_t *lsp_index, int bad_frame) |
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{ |
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int min_dist, pred; |
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int i, j, temp, stable; |
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|
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/* Check for frame erasure */ |
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if (!bad_frame) { |
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min_dist = 0x100; |
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pred = 12288; |
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} else { |
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min_dist = 0x200; |
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pred = 23552; |
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lsp_index[0] = lsp_index[1] = lsp_index[2] = 0; |
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} |
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|
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/* Get the VQ table entry corresponding to the transmitted index */ |
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cur_lsp[0] = lsp_band0[lsp_index[0]][0]; |
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cur_lsp[1] = lsp_band0[lsp_index[0]][1]; |
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cur_lsp[2] = lsp_band0[lsp_index[0]][2]; |
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cur_lsp[3] = lsp_band1[lsp_index[1]][0]; |
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cur_lsp[4] = lsp_band1[lsp_index[1]][1]; |
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cur_lsp[5] = lsp_band1[lsp_index[1]][2]; |
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cur_lsp[6] = lsp_band2[lsp_index[2]][0]; |
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cur_lsp[7] = lsp_band2[lsp_index[2]][1]; |
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cur_lsp[8] = lsp_band2[lsp_index[2]][2]; |
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cur_lsp[9] = lsp_band2[lsp_index[2]][3]; |
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|
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/* Add predicted vector & DC component to the previously quantized vector */ |
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for (i = 0; i < LPC_ORDER; i++) { |
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temp = ((prev_lsp[i] - dc_lsp[i]) * pred + (1 << 14)) >> 15; |
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cur_lsp[i] += dc_lsp[i] + temp; |
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} |
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|
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for (i = 0; i < LPC_ORDER; i++) { |
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cur_lsp[0] = FFMAX(cur_lsp[0], 0x180); |
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cur_lsp[LPC_ORDER - 1] = FFMIN(cur_lsp[LPC_ORDER - 1], 0x7e00); |
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|
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/* Stability check */ |
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for (j = 1; j < LPC_ORDER; j++) { |
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temp = min_dist + cur_lsp[j - 1] - cur_lsp[j]; |
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if (temp > 0) { |
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temp >>= 1; |
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cur_lsp[j - 1] -= temp; |
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cur_lsp[j] += temp; |
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} |
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} |
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stable = 1; |
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for (j = 1; j < LPC_ORDER; j++) { |
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temp = cur_lsp[j - 1] + min_dist - cur_lsp[j] - 4; |
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if (temp > 0) { |
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stable = 0; |
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break; |
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} |
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} |
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if (stable) |
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break; |
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} |
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if (!stable) |
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memcpy(cur_lsp, prev_lsp, LPC_ORDER * sizeof(*cur_lsp)); |
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} |
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|
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/** |
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* Bitexact implementation of 2ab scaled by 1/2^16. |
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* |
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* @param a 32 bit multiplicand |
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* @param b 16 bit multiplier |
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*/ |
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#define MULL2(a, b) \ |
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((((a) >> 16) * (b) << 1) + (((a) & 0xffff) * (b) >> 15)) |
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|
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/** |
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* Convert LSP frequencies to LPC coefficients. |
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* |
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* @param lpc buffer for LPC coefficients |
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*/ |
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static void lsp2lpc(int16_t *lpc) |
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{ |
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int f1[LPC_ORDER / 2 + 1]; |
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int f2[LPC_ORDER / 2 + 1]; |
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int i, j; |
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|
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/* Calculate negative cosine */ |
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for (j = 0; j < LPC_ORDER; j++) { |
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int index = lpc[j] >> 7; |
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int offset = lpc[j] & 0x7f; |
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int temp1 = cos_tab[index] << 16; |
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int temp2 = (cos_tab[index + 1] - cos_tab[index]) * |
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((offset << 8) + 0x80) << 1; |
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|
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lpc[j] = -(av_sat_dadd32(1 << 15, temp1 + temp2) >> 16); |
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} |
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|
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/* |
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* Compute sum and difference polynomial coefficients |
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* (bitexact alternative to lsp2poly() in lsp.c) |
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*/ |
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/* Initialize with values in Q28 */ |
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f1[0] = 1 << 28; |
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f1[1] = (lpc[0] << 14) + (lpc[2] << 14); |
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f1[2] = lpc[0] * lpc[2] + (2 << 28); |
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|
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f2[0] = 1 << 28; |
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f2[1] = (lpc[1] << 14) + (lpc[3] << 14); |
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f2[2] = lpc[1] * lpc[3] + (2 << 28); |
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|
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/* |
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* Calculate and scale the coefficients by 1/2 in |
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* each iteration for a final scaling factor of Q25 |
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*/ |
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for (i = 2; i < LPC_ORDER / 2; i++) { |
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f1[i + 1] = f1[i - 1] + MULL2(f1[i], lpc[2 * i]); |
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f2[i + 1] = f2[i - 1] + MULL2(f2[i], lpc[2 * i + 1]); |
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|
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for (j = i; j >= 2; j--) { |
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f1[j] = MULL2(f1[j - 1], lpc[2 * i]) + |
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(f1[j] >> 1) + (f1[j - 2] >> 1); |
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f2[j] = MULL2(f2[j - 1], lpc[2 * i + 1]) + |
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(f2[j] >> 1) + (f2[j - 2] >> 1); |
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} |
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|
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f1[0] >>= 1; |
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f2[0] >>= 1; |
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f1[1] = ((lpc[2 * i] << 16 >> i) + f1[1]) >> 1; |
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f2[1] = ((lpc[2 * i + 1] << 16 >> i) + f2[1]) >> 1; |
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} |
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|
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/* Convert polynomial coefficients to LPC coefficients */ |
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for (i = 0; i < LPC_ORDER / 2; i++) { |
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int64_t ff1 = f1[i + 1] + f1[i]; |
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int64_t ff2 = f2[i + 1] - f2[i]; |
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|
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lpc[i] = av_clipl_int32(((ff1 + ff2) << 3) + (1 << 15)) >> 16; |
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lpc[LPC_ORDER - i - 1] = av_clipl_int32(((ff1 - ff2) << 3) + |
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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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* Quantize LSP frequencies by interpolation and convert them to |
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* the corresponding LPC coefficients. |
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* |
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* @param lpc buffer for LPC coefficients |
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* @param cur_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_interpolate(int16_t *lpc, int16_t *cur_lsp, int16_t *prev_lsp) |
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{ |
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int i; |
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int16_t *lpc_ptr = lpc; |
|
|
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/* cur_lsp * 0.25 + prev_lsp * 0.75 */ |
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ff_acelp_weighted_vector_sum(lpc, cur_lsp, prev_lsp, |
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4096, 12288, 1 << 13, 14, LPC_ORDER); |
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ff_acelp_weighted_vector_sum(lpc + LPC_ORDER, cur_lsp, prev_lsp, |
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8192, 8192, 1 << 13, 14, LPC_ORDER); |
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ff_acelp_weighted_vector_sum(lpc + 2 * LPC_ORDER, cur_lsp, prev_lsp, |
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12288, 4096, 1 << 13, 14, LPC_ORDER); |
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memcpy(lpc + 3 * LPC_ORDER, cur_lsp, LPC_ORDER * sizeof(*lpc)); |
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|
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for (i = 0; i < SUBFRAMES; i++) { |
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lsp2lpc(lpc_ptr); |
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lpc_ptr += LPC_ORDER; |
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} |
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} |
|
|
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/** |
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* Generate a train of dirac functions with period as pitch lag. |
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*/ |
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static void gen_dirac_train(int16_t *buf, int pitch_lag) |
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{ |
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int16_t vector[SUBFRAME_LEN]; |
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int i, j; |
|
|
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memcpy(vector, buf, SUBFRAME_LEN * sizeof(*vector)); |
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for (i = pitch_lag; i < SUBFRAME_LEN; i += pitch_lag) { |
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for (j = 0; j < SUBFRAME_LEN - i; j++) |
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buf[i + j] += vector[j]; |
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} |
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} |
|
|
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/** |
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* Generate fixed codebook excitation vector. |
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* |
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* @param vector decoded excitation vector |
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* @param subfrm current subframe |
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* @param cur_rate current bitrate |
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* @param pitch_lag closed loop pitch lag |
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* @param index current subframe index |
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*/ |
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static void gen_fcb_excitation(int16_t *vector, G723_1_Subframe *subfrm, |
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enum Rate cur_rate, int pitch_lag, int index) |
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{ |
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int temp, i, j; |
|
|
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memset(vector, 0, SUBFRAME_LEN * sizeof(*vector)); |
|
|
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if (cur_rate == RATE_6300) { |
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if (subfrm->pulse_pos >= max_pos[index]) |
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return; |
|
|
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/* Decode amplitudes and positions */ |
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j = PULSE_MAX - pulses[index]; |
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temp = subfrm->pulse_pos; |
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for (i = 0; i < SUBFRAME_LEN / GRID_SIZE; i++) { |
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temp -= combinatorial_table[j][i]; |
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if (temp >= 0) |
|
continue; |
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temp += combinatorial_table[j++][i]; |
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if (subfrm->pulse_sign & (1 << (PULSE_MAX - j))) { |
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vector[subfrm->grid_index + GRID_SIZE * i] = |
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-fixed_cb_gain[subfrm->amp_index]; |
|
} else { |
|
vector[subfrm->grid_index + GRID_SIZE * i] = |
|
fixed_cb_gain[subfrm->amp_index]; |
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} |
|
if (j == PULSE_MAX) |
|
break; |
|
} |
|
if (subfrm->dirac_train == 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; |
|
int cb_sign = subfrm->pulse_sign; |
|
int cb_pos = subfrm->pulse_pos; |
|
int offset, beta, lag; |
|
|
|
for (i = 0; i < 8; i += 2) { |
|
offset = ((cb_pos & 7) << 3) + cb_shift + i; |
|
vector[offset] = (cb_sign & 1) ? cb_gain : -cb_gain; |
|
cb_pos >>= 3; |
|
cb_sign >>= 1; |
|
} |
|
|
|
/* Enhance harmonic components */ |
|
lag = pitch_contrib[subfrm->ad_cb_gain << 1] + pitch_lag + |
|
subfrm->ad_cb_lag - 1; |
|
beta = pitch_contrib[(subfrm->ad_cb_gain << 1) + 1]; |
|
|
|
if (lag < SUBFRAME_LEN - 2) { |
|
for (i = lag; i < SUBFRAME_LEN; i++) |
|
vector[i] += beta * vector[i - lag] >> 15; |
|
} |
|
} |
|
} |
|
|
|
/** |
|
* 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 |
|
* @param offset offset of the excitation vector |
|
* @param ccr_max pointer to the maximum auto-correlation |
|
* @param pitch_lag decoded pitch lag |
|
* @param length length of autocorrelation |
|
* @param dir forward lag(1) / backward lag(-1) |
|
*/ |
|
static int autocorr_max(const int16_t *buf, int offset, int *ccr_max, |
|
int pitch_lag, int length, int dir) |
|
{ |
|
int limit, ccr, lag = 0; |
|
int i; |
|
|
|
pitch_lag = FFMIN(PITCH_MAX - 3, pitch_lag); |
|
if (dir > 0) |
|
limit = FFMIN(FRAME_LEN + PITCH_MAX - offset - length, pitch_lag + 3); |
|
else |
|
limit = pitch_lag + 3; |
|
|
|
for (i = pitch_lag - 3; i <= limit; i++) { |
|
ccr = dot_product(buf, buf + dir * i, length); |
|
|
|
if (ccr > *ccr_max) { |
|
*ccr_max = ccr; |
|
lag = i; |
|
} |
|
} |
|
return lag; |
|
} |
|
|
|
/** |
|
* Calculate pitch postfilter optimal and scaling gains. |
|
* |
|
* @param lag pitch postfilter forward/backward lag |
|
* @param ppf pitch postfilter parameters |
|
* @param cur_rate current bitrate |
|
* @param tgt_eng target energy |
|
* @param ccr cross-correlation |
|
* @param res_eng residual energy |
|
*/ |
|
static void comp_ppf_gains(int lag, PPFParam *ppf, enum Rate cur_rate, |
|
int tgt_eng, int ccr, int res_eng) |
|
{ |
|
int pf_residual; /* square of postfiltered residual */ |
|
int temp1, temp2; |
|
|
|
ppf->index = lag; |
|
|
|
temp1 = tgt_eng * res_eng >> 1; |
|
temp2 = ccr * ccr << 1; |
|
|
|
if (temp2 > temp1) { |
|
if (ccr >= res_eng) { |
|
ppf->opt_gain = ppf_gain_weight[cur_rate]; |
|
} else { |
|
ppf->opt_gain = (ccr << 15) / res_eng * |
|
ppf_gain_weight[cur_rate] >> 15; |
|
} |
|
/* pf_res^2 = tgt_eng + 2*ccr*gain + res_eng*gain^2 */ |
|
temp1 = (tgt_eng << 15) + (ccr * ppf->opt_gain << 1); |
|
temp2 = (ppf->opt_gain * ppf->opt_gain >> 15) * res_eng; |
|
pf_residual = av_sat_add32(temp1, temp2 + (1 << 15)) >> 16; |
|
|
|
if (tgt_eng >= pf_residual << 1) { |
|
temp1 = 0x7fff; |
|
} else { |
|
temp1 = (tgt_eng << 14) / pf_residual; |
|
} |
|
|
|
/* scaling_gain = sqrt(tgt_eng/pf_res^2) */ |
|
ppf->sc_gain = square_root(temp1 << 16); |
|
} else { |
|
ppf->opt_gain = 0; |
|
ppf->sc_gain = 0x7fff; |
|
} |
|
|
|
ppf->opt_gain = av_clip_int16(ppf->opt_gain * ppf->sc_gain >> 15); |
|
} |
|
|
|
/** |
|
* Calculate pitch postfilter parameters. |
|
* |
|
* @param p the context |
|
* @param offset offset of the excitation vector |
|
* @param pitch_lag decoded pitch lag |
|
* @param ppf pitch postfilter parameters |
|
* @param cur_rate current bitrate |
|
*/ |
|
static void comp_ppf_coeff(G723_1_Context *p, int offset, int pitch_lag, |
|
PPFParam *ppf, enum Rate cur_rate) |
|
{ |
|
|
|
int16_t scale; |
|
int i; |
|
int temp1, temp2; |
|
|
|
/* |
|
* 0 - target energy |
|
* 1 - forward cross-correlation |
|
* 2 - forward residual energy |
|
* 3 - backward cross-correlation |
|
* 4 - backward residual energy |
|
*/ |
|
int energy[5] = {0, 0, 0, 0, 0}; |
|
int16_t *buf = p->audio + LPC_ORDER + offset; |
|
int fwd_lag = autocorr_max(buf, offset, &energy[1], pitch_lag, |
|
SUBFRAME_LEN, 1); |
|
int back_lag = autocorr_max(buf, offset, &energy[3], pitch_lag, |
|
SUBFRAME_LEN, -1); |
|
|
|
ppf->index = 0; |
|
ppf->opt_gain = 0; |
|
ppf->sc_gain = 0x7fff; |
|
|
|
/* Case 0, Section 3.6 */ |
|
if (!back_lag && !fwd_lag) |
|
return; |
|
|
|
/* Compute target energy */ |
|
energy[0] = 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); |
|
|
|
/* Compute backward residual energy */ |
|
if (back_lag) |
|
energy[4] = 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); |
|
for (i = 0; i < 5; i++) |
|
energy[i] = (energy[i] << scale) >> 16; |
|
|
|
if (fwd_lag && !back_lag) { /* Case 1 */ |
|
comp_ppf_gains(fwd_lag, ppf, cur_rate, energy[0], energy[1], |
|
energy[2]); |
|
} else if (!fwd_lag) { /* Case 2 */ |
|
comp_ppf_gains(-back_lag, ppf, cur_rate, energy[0], energy[3], |
|
energy[4]); |
|
} else { /* Case 3 */ |
|
|
|
/* |
|
* Select the largest of energy[1]^2/energy[2] |
|
* and energy[3]^2/energy[4] |
|
*/ |
|
temp1 = energy[4] * ((energy[1] * energy[1] + (1 << 14)) >> 15); |
|
temp2 = energy[2] * ((energy[3] * energy[3] + (1 << 14)) >> 15); |
|
if (temp1 >= temp2) { |
|
comp_ppf_gains(fwd_lag, ppf, cur_rate, energy[0], energy[1], |
|
energy[2]); |
|
} else { |
|
comp_ppf_gains(-back_lag, ppf, cur_rate, energy[0], energy[3], |
|
energy[4]); |
|
} |
|
} |
|
} |
|
|
|
/** |
|
* Classify frames as voiced/unvoiced. |
|
* |
|
* @param p the context |
|
* @param pitch_lag decoded pitch_lag |
|
* @param exc_eng excitation energy estimation |
|
* @param scale scaling factor of exc_eng |
|
* |
|
* @return residual interpolation index if voiced, 0 otherwise |
|
*/ |
|
static int comp_interp_index(G723_1_Context *p, int pitch_lag, |
|
int *exc_eng, int *scale) |
|
{ |
|
int offset = PITCH_MAX + 2 * SUBFRAME_LEN; |
|
int16_t *buf = p->audio + LPC_ORDER; |
|
|
|
int index, ccr, tgt_eng, best_eng, temp; |
|
|
|
*scale = scale_vector(buf, p->excitation, FRAME_LEN + PITCH_MAX); |
|
buf += offset; |
|
|
|
/* Compute maximum backward cross-correlation */ |
|
ccr = 0; |
|
index = autocorr_max(buf, offset, &ccr, pitch_lag, SUBFRAME_LEN * 2, -1); |
|
ccr = av_sat_add32(ccr, 1 << 15) >> 16; |
|
|
|
/* Compute target energy */ |
|
tgt_eng = 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 = av_sat_add32(best_eng, 1 << 15) >> 16; |
|
|
|
temp = best_eng * *exc_eng >> 3; |
|
|
|
if (temp < ccr * ccr) |
|
return index; |
|
else |
|
return 0; |
|
} |
|
|
|
/** |
|
* Peform residual interpolation based on frame classification. |
|
* |
|
* @param buf decoded excitation vector |
|
* @param out output vector |
|
* @param lag decoded pitch lag |
|
* @param gain interpolated gain |
|
* @param rseed seed for random number generator |
|
*/ |
|
static void residual_interp(int16_t *buf, int16_t *out, int lag, |
|
int gain, int *rseed) |
|
{ |
|
int i; |
|
if (lag) { /* Voiced */ |
|
int16_t *vector_ptr = buf + PITCH_MAX; |
|
/* Attenuate */ |
|
for (i = 0; i < lag; i++) |
|
out[i] = vector_ptr[i - lag] * 3 >> 2; |
|
av_memcpy_backptr((uint8_t*)(out + lag), lag * sizeof(*out), |
|
(FRAME_LEN - lag) * sizeof(*out)); |
|
} else { /* Unvoiced */ |
|
for (i = 0; i < FRAME_LEN; i++) { |
|
*rseed = *rseed * 521 + 259; |
|
out[i] = gain * *rseed >> 15; |
|
} |
|
memset(buf, 0, (FRAME_LEN + PITCH_MAX) * sizeof(*buf)); |
|
} |
|
} |
|
|
|
/** |
|
* Perform IIR filtering. |
|
* |
|
* @param fir_coef FIR coefficients |
|
* @param iir_coef IIR coefficients |
|
* @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) |
|
{ |
|
int m, n; |
|
|
|
for (m = 0; m < SUBFRAME_LEN; m++) { |
|
int64_t filter = 0; |
|
for (n = 1; n <= LPC_ORDER; n++) { |
|
filter -= fir_coef[n - 1] * src[m - n] - |
|
iir_coef[n - 1] * (dest[m - n] >> 16); |
|
} |
|
|
|
dest[m] = av_clipl_int32((src[m] << 16) + (filter << 3) + (1 << 15)); |
|
} |
|
} |
|
|
|
/** |
|
* Adjust gain of postfiltered signal. |
|
* |
|
* @param p the context |
|
* @param buf postfiltered output vector |
|
* @param energy input energy coefficient |
|
*/ |
|
static void gain_scale(G723_1_Context *p, int16_t * buf, int energy) |
|
{ |
|
int num, denom, gain, bits1, bits2; |
|
int i; |
|
|
|
num = energy; |
|
denom = 0; |
|
for (i = 0; i < SUBFRAME_LEN; i++) { |
|
int temp = buf[i] >> 2; |
|
temp *= temp; |
|
denom = av_sat_dadd32(denom, temp); |
|
} |
|
|
|
if (num && denom) { |
|
bits1 = normalize_bits(num, 31); |
|
bits2 = normalize_bits(denom, 31); |
|
num = num << bits1 >> 1; |
|
denom <<= bits2; |
|
|
|
bits2 = 5 + bits1 - bits2; |
|
bits2 = FFMAX(0, bits2); |
|
|
|
gain = (num >> 1) / (denom >> 16); |
|
gain = square_root(gain << 16 >> bits2); |
|
} else { |
|
gain = 1 << 12; |
|
} |
|
|
|
for (i = 0; i < SUBFRAME_LEN; i++) { |
|
p->pf_gain = (15 * p->pf_gain + gain + (1 << 3)) >> 4; |
|
buf[i] = av_clip_int16((buf[i] * (p->pf_gain + (p->pf_gain >> 4)) + |
|
(1 << 10)) >> 11); |
|
} |
|
} |
|
|
|
/** |
|
* Perform formant filtering. |
|
* |
|
* @param p the context |
|
* @param lpc quantized lpc coefficients |
|
* @param buf input buffer |
|
* @param dst output buffer |
|
*/ |
|
static void formant_postfilter(G723_1_Context *p, int16_t *lpc, |
|
int16_t *buf, int16_t *dst) |
|
{ |
|
int16_t filter_coef[2][LPC_ORDER]; |
|
int filter_signal[LPC_ORDER + FRAME_LEN], *signal_ptr; |
|
int i, j, k; |
|
|
|
memcpy(buf, p->fir_mem, LPC_ORDER * sizeof(*buf)); |
|
memcpy(filter_signal, p->iir_mem, LPC_ORDER * sizeof(*filter_signal)); |
|
|
|
for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) { |
|
for (k = 0; k < LPC_ORDER; k++) { |
|
filter_coef[0][k] = (-lpc[k] * postfilter_tbl[0][k] + |
|
(1 << 14)) >> 15; |
|
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); |
|
lpc += LPC_ORDER; |
|
} |
|
|
|
memcpy(p->fir_mem, buf + FRAME_LEN, LPC_ORDER * sizeof(*p->fir_mem)); |
|
memcpy(p->iir_mem, filter_signal + FRAME_LEN, |
|
LPC_ORDER * sizeof(*p->iir_mem)); |
|
|
|
buf += LPC_ORDER; |
|
signal_ptr = filter_signal + LPC_ORDER; |
|
for (i = 0; i < SUBFRAMES; i++) { |
|
int temp; |
|
int auto_corr[2]; |
|
int scale, energy; |
|
|
|
/* Normalize */ |
|
scale = 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); |
|
|
|
/* Compute reflection coefficient */ |
|
temp = auto_corr[1] >> 16; |
|
if (temp) { |
|
temp = (auto_corr[0] >> 2) / temp; |
|
} |
|
p->reflection_coef = (3 * p->reflection_coef + temp + 2) >> 2; |
|
temp = -p->reflection_coef >> 1 & ~3; |
|
|
|
/* Compensation filter */ |
|
for (j = 0; j < SUBFRAME_LEN; j++) { |
|
dst[j] = av_sat_dadd32(signal_ptr[j], |
|
(signal_ptr[j - 1] >> 16) * temp) >> 16; |
|
} |
|
|
|
/* Compute normalized signal energy */ |
|
temp = 2 * scale + 4; |
|
if (temp < 0) { |
|
energy = av_clipl_int32((int64_t)auto_corr[1] << -temp); |
|
} else |
|
energy = auto_corr[1] >> temp; |
|
|
|
gain_scale(p, dst, energy); |
|
|
|
buf += SUBFRAME_LEN; |
|
signal_ptr += SUBFRAME_LEN; |
|
dst += SUBFRAME_LEN; |
|
} |
|
} |
|
|
|
static int sid_gain_to_lsp_index(int gain) |
|
{ |
|
if (gain < 0x10) |
|
return gain << 6; |
|
else if (gain < 0x20) |
|
return gain - 8 << 7; |
|
else |
|
return gain - 20 << 8; |
|
} |
|
|
|
static inline int cng_rand(int *state, int base) |
|
{ |
|
*state = (*state * 521 + 259) & 0xFFFF; |
|
return (*state & 0x7FFF) * base >> 15; |
|
} |
|
|
|
static int estimate_sid_gain(G723_1_Context *p) |
|
{ |
|
int i, shift, seg, seg2, t, val, val_add, x, y; |
|
|
|
shift = 16 - p->cur_gain * 2; |
|
if (shift > 0) |
|
t = p->sid_gain << shift; |
|
else |
|
t = p->sid_gain >> -shift; |
|
x = t * cng_filt[0] >> 16; |
|
|
|
if (x >= cng_bseg[2]) |
|
return 0x3F; |
|
|
|
if (x >= cng_bseg[1]) { |
|
shift = 4; |
|
seg = 3; |
|
} else { |
|
shift = 3; |
|
seg = (x >= cng_bseg[0]); |
|
} |
|
seg2 = FFMIN(seg, 3); |
|
|
|
val = 1 << shift; |
|
val_add = val >> 1; |
|
for (i = 0; i < shift; i++) { |
|
t = seg * 32 + (val << seg2); |
|
t *= t; |
|
if (x >= t) |
|
val += val_add; |
|
else |
|
val -= val_add; |
|
val_add >>= 1; |
|
} |
|
|
|
t = seg * 32 + (val << seg2); |
|
y = t * t - x; |
|
if (y <= 0) { |
|
t = seg * 32 + (val + 1 << seg2); |
|
t = t * t - x; |
|
val = (seg2 - 1 << 4) + val; |
|
if (t >= y) |
|
val++; |
|
} else { |
|
t = seg * 32 + (val - 1 << seg2); |
|
t = t * t - x; |
|
val = (seg2 - 1 << 4) + val; |
|
if (t >= y) |
|
val--; |
|
} |
|
|
|
return val; |
|
} |
|
|
|
static void generate_noise(G723_1_Context *p) |
|
{ |
|
int i, j, idx, t; |
|
int off[SUBFRAMES]; |
|
int signs[SUBFRAMES / 2 * 11], pos[SUBFRAMES / 2 * 11]; |
|
int tmp[SUBFRAME_LEN * 2]; |
|
int16_t *vector_ptr; |
|
int64_t sum; |
|
int b0, c, delta, x, shift; |
|
|
|
p->pitch_lag[0] = cng_rand(&p->cng_random_seed, 21) + 123; |
|
p->pitch_lag[1] = cng_rand(&p->cng_random_seed, 19) + 123; |
|
|
|
for (i = 0; i < SUBFRAMES; i++) { |
|
p->subframe[i].ad_cb_gain = cng_rand(&p->cng_random_seed, 50) + 1; |
|
p->subframe[i].ad_cb_lag = cng_adaptive_cb_lag[i]; |
|
} |
|
|
|
for (i = 0; i < SUBFRAMES / 2; i++) { |
|
t = cng_rand(&p->cng_random_seed, 1 << 13); |
|
off[i * 2] = t & 1; |
|
off[i * 2 + 1] = ((t >> 1) & 1) + SUBFRAME_LEN; |
|
t >>= 2; |
|
for (j = 0; j < 11; j++) { |
|
signs[i * 11 + j] = (t & 1) * 2 - 1 << 14; |
|
t >>= 1; |
|
} |
|
} |
|
|
|
idx = 0; |
|
for (i = 0; i < SUBFRAMES; i++) { |
|
for (j = 0; j < SUBFRAME_LEN / 2; j++) |
|
tmp[j] = j; |
|
t = SUBFRAME_LEN / 2; |
|
for (j = 0; j < pulses[i]; j++, idx++) { |
|
int idx2 = cng_rand(&p->cng_random_seed, t); |
|
|
|
pos[idx] = tmp[idx2] * 2 + off[i]; |
|
tmp[idx2] = tmp[--t]; |
|
} |
|
} |
|
|
|
vector_ptr = p->audio + LPC_ORDER; |
|
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); |
|
|
|
t = 0; |
|
for (j = 0; j < SUBFRAME_LEN * 2; j++) |
|
t |= FFABS(vector_ptr[j]); |
|
t = FFMIN(t, 0x7FFF); |
|
if (!t) { |
|
shift = 0; |
|
} else { |
|
shift = -10 + av_log2(t); |
|
if (shift < -2) |
|
shift = -2; |
|
} |
|
sum = 0; |
|
if (shift < 0) { |
|
for (j = 0; j < SUBFRAME_LEN * 2; j++) { |
|
t = vector_ptr[j] << -shift; |
|
sum += t * t; |
|
tmp[j] = t; |
|
} |
|
} else { |
|
for (j = 0; j < SUBFRAME_LEN * 2; j++) { |
|
t = vector_ptr[j] >> shift; |
|
sum += t * t; |
|
tmp[j] = t; |
|
} |
|
} |
|
|
|
b0 = 0; |
|
for (j = 0; j < 11; j++) |
|
b0 += tmp[pos[(i / 2) * 11 + j]] * signs[(i / 2) * 11 + j]; |
|
b0 = b0 * 2 * 2979LL + (1 << 29) >> 30; // approximated division by 11 |
|
|
|
c = p->cur_gain * (p->cur_gain * SUBFRAME_LEN >> 5); |
|
if (shift * 2 + 3 >= 0) |
|
c >>= shift * 2 + 3; |
|
else |
|
c <<= -(shift * 2 + 3); |
|
c = (av_clipl_int32(sum << 1) - c) * 2979LL >> 15; |
|
|
|
delta = b0 * b0 * 2 - c; |
|
if (delta <= 0) { |
|
x = -b0; |
|
} else { |
|
delta = square_root(delta); |
|
x = delta - b0; |
|
t = delta + b0; |
|
if (FFABS(t) < FFABS(x)) |
|
x = -t; |
|
} |
|
shift++; |
|
if (shift < 0) |
|
x >>= -shift; |
|
else |
|
x <<= shift; |
|
x = av_clip(x, -10000, 10000); |
|
|
|
for (j = 0; j < 11; j++) { |
|
idx = (i / 2) * 11 + j; |
|
vector_ptr[pos[idx]] = av_clip_int16(vector_ptr[pos[idx]] + |
|
(x * signs[idx] >> 15)); |
|
} |
|
|
|
/* copy decoded data to serve as a history for the next decoded subframes */ |
|
memcpy(vector_ptr + PITCH_MAX, vector_ptr, |
|
sizeof(*vector_ptr) * SUBFRAME_LEN * 2); |
|
vector_ptr += SUBFRAME_LEN * 2; |
|
} |
|
/* Save the excitation for the next frame */ |
|
memcpy(p->prev_excitation, p->audio + LPC_ORDER + FRAME_LEN, |
|
PITCH_MAX * sizeof(*p->excitation)); |
|
} |
|
|
|
static int g723_1_decode_frame(AVCodecContext *avctx, void *data, |
|
int *got_frame_ptr, AVPacket *avpkt) |
|
{ |
|
G723_1_Context *p = avctx->priv_data; |
|
const uint8_t *buf = avpkt->data; |
|
int buf_size = avpkt->size; |
|
int dec_mode = buf[0] & 3; |
|
|
|
PPFParam ppf[SUBFRAMES]; |
|
int16_t cur_lsp[LPC_ORDER]; |
|
int16_t lpc[SUBFRAMES * LPC_ORDER]; |
|
int16_t acb_vector[SUBFRAME_LEN]; |
|
int16_t *out; |
|
int bad_frame = 0, i, j, ret; |
|
int16_t *audio = p->audio; |
|
|
|
if (buf_size < frame_size[dec_mode]) { |
|
if (buf_size) |
|
av_log(avctx, AV_LOG_WARNING, |
|
"Expected %d bytes, got %d - skipping packet\n", |
|
frame_size[dec_mode], buf_size); |
|
*got_frame_ptr = 0; |
|
return buf_size; |
|
} |
|
|
|
if (unpack_bitstream(p, buf, buf_size) < 0) { |
|
bad_frame = 1; |
|
if (p->past_frame_type == ACTIVE_FRAME) |
|
p->cur_frame_type = ACTIVE_FRAME; |
|
else |
|
p->cur_frame_type = UNTRANSMITTED_FRAME; |
|
} |
|
|
|
p->frame.nb_samples = FRAME_LEN; |
|
if ((ret = avctx->get_buffer(avctx, &p->frame)) < 0) { |
|
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); |
|
return ret; |
|
} |
|
|
|
out = (int16_t *)p->frame.data[0]; |
|
|
|
if (p->cur_frame_type == ACTIVE_FRAME) { |
|
if (!bad_frame) |
|
p->erased_frames = 0; |
|
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); |
|
|
|
/* Save the lsp_vector for the next frame */ |
|
memcpy(p->prev_lsp, cur_lsp, LPC_ORDER * sizeof(*p->prev_lsp)); |
|
|
|
/* Generate the excitation for the frame */ |
|
memcpy(p->excitation, p->prev_excitation, |
|
PITCH_MAX * sizeof(*p->excitation)); |
|
if (!p->erased_frames) { |
|
int16_t *vector_ptr = p->excitation + PITCH_MAX; |
|
|
|
/* Update interpolation gain memory */ |
|
p->interp_gain = fixed_cb_gain[(p->subframe[2].amp_index + |
|
p->subframe[3].amp_index) >> 1]; |
|
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); |
|
/* Get the total excitation */ |
|
for (j = 0; j < SUBFRAME_LEN; j++) { |
|
int v = av_clip_int16(vector_ptr[j] << 1); |
|
vector_ptr[j] = av_clip_int16(v + acb_vector[j]); |
|
} |
|
vector_ptr += SUBFRAME_LEN; |
|
} |
|
|
|
vector_ptr = p->excitation + PITCH_MAX; |
|
|
|
p->interp_index = comp_interp_index(p, p->pitch_lag[1], |
|
&p->sid_gain, &p->cur_gain); |
|
|
|
/* Peform pitch postfiltering */ |
|
if (p->postfilter) { |
|
i = PITCH_MAX; |
|
for (j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) |
|
comp_ppf_coeff(p, i, p->pitch_lag[j >> 1], |
|
ppf + j, p->cur_rate); |
|
|
|
for (i = 0, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) |
|
ff_acelp_weighted_vector_sum(p->audio + LPC_ORDER + i, |
|
vector_ptr + i, |
|
vector_ptr + i + ppf[j].index, |
|
ppf[j].sc_gain, |
|
ppf[j].opt_gain, |
|
1 << 14, 15, SUBFRAME_LEN); |
|
} else { |
|
audio = vector_ptr - LPC_ORDER; |
|
} |
|
|
|
/* Save the excitation for the next frame */ |
|
memcpy(p->prev_excitation, p->excitation + FRAME_LEN, |
|
PITCH_MAX * sizeof(*p->excitation)); |
|
} else { |
|
p->interp_gain = (p->interp_gain * 3 + 2) >> 2; |
|
if (p->erased_frames == 3) { |
|
/* Mute output */ |
|
memset(p->excitation, 0, |
|
(FRAME_LEN + PITCH_MAX) * sizeof(*p->excitation)); |
|
memset(p->prev_excitation, 0, |
|
PITCH_MAX * sizeof(*p->excitation)); |
|
memset(p->frame.data[0], 0, |
|
(FRAME_LEN + LPC_ORDER) * sizeof(int16_t)); |
|
} else { |
|
int16_t *buf = p->audio + LPC_ORDER; |
|
|
|
/* Regenerate frame */ |
|
residual_interp(p->excitation, buf, p->interp_index, |
|
p->interp_gain, &p->random_seed); |
|
|
|
/* Save the excitation for the next frame */ |
|
memcpy(p->prev_excitation, buf + (FRAME_LEN - PITCH_MAX), |
|
PITCH_MAX * sizeof(*p->excitation)); |
|
} |
|
} |
|
p->cng_random_seed = CNG_RANDOM_SEED; |
|
} 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); |
|
} else if (p->past_frame_type == ACTIVE_FRAME) { |
|
p->sid_gain = estimate_sid_gain(p); |
|
} |
|
|
|
if (p->past_frame_type == ACTIVE_FRAME) |
|
p->cur_gain = p->sid_gain; |
|
else |
|
p->cur_gain = (p->cur_gain * 7 + p->sid_gain) >> 3; |
|
generate_noise(p); |
|
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)); |
|
} |
|
|
|
p->past_frame_type = p->cur_frame_type; |
|
|
|
memcpy(p->audio, p->synth_mem, LPC_ORDER * sizeof(*p->audio)); |
|
for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) |
|
ff_celp_lp_synthesis_filter(p->audio + i, &lpc[j * LPC_ORDER], |
|
audio + i, SUBFRAME_LEN, LPC_ORDER, |
|
0, 1, 1 << 12); |
|
memcpy(p->synth_mem, p->audio + FRAME_LEN, LPC_ORDER * sizeof(*p->audio)); |
|
|
|
if (p->postfilter) { |
|
formant_postfilter(p, lpc, p->audio, out); |
|
} else { // if output is not postfiltered it should be scaled by 2 |
|
for (i = 0; i < FRAME_LEN; i++) |
|
out[i] = av_clip_int16(p->audio[LPC_ORDER + i] << 1); |
|
} |
|
|
|
*got_frame_ptr = 1; |
|
*(AVFrame *)data = p->frame; |
|
|
|
return frame_size[dec_mode]; |
|
} |
|
|
|
#define OFFSET(x) offsetof(G723_1_Context, x) |
|
#define AD AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_DECODING_PARAM |
|
|
|
static const AVOption options[] = { |
|
{ "postfilter", "postfilter on/off", OFFSET(postfilter), AV_OPT_TYPE_INT, |
|
{ .i64 = 1 }, 0, 1, AD }, |
|
{ NULL } |
|
}; |
|
|
|
|
|
static const AVClass g723_1dec_class = { |
|
.class_name = "G.723.1 decoder", |
|
.item_name = av_default_item_name, |
|
.option = options, |
|
.version = LIBAVUTIL_VERSION_INT, |
|
}; |
|
|
|
AVCodec ff_g723_1_decoder = { |
|
.name = "g723_1", |
|
.type = AVMEDIA_TYPE_AUDIO, |
|
.id = AV_CODEC_ID_G723_1, |
|
.priv_data_size = sizeof(G723_1_Context), |
|
.init = g723_1_decode_init, |
|
.decode = g723_1_decode_frame, |
|
.long_name = NULL_IF_CONFIG_SMALL("G.723.1"), |
|
.capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1, |
|
.priv_class = &g723_1dec_class, |
|
};
|
|
|