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1254 lines
46 KiB
1254 lines
46 KiB
/* |
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* AMR wideband decoder |
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* Copyright (c) 2010 Marcelo Galvao Povoa |
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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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* @file |
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* AMR wideband decoder |
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*/ |
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#include "libavutil/common.h" |
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#include "libavutil/lfg.h" |
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#include "avcodec.h" |
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#include "dsputil.h" |
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#include "lsp.h" |
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#include "celp_filters.h" |
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#include "acelp_filters.h" |
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#include "acelp_vectors.h" |
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#include "acelp_pitch_delay.h" |
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#define AMR_USE_16BIT_TABLES |
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#include "amr.h" |
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#include "amrwbdata.h" |
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typedef struct { |
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AVFrame avframe; ///< AVFrame for decoded samples |
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AMRWBFrame frame; ///< AMRWB parameters decoded from bitstream |
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enum Mode fr_cur_mode; ///< mode index of current frame |
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uint8_t fr_quality; ///< frame quality index (FQI) |
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float isf_cur[LP_ORDER]; ///< working ISF vector from current frame |
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float isf_q_past[LP_ORDER]; ///< quantized ISF vector of the previous frame |
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float isf_past_final[LP_ORDER]; ///< final processed ISF vector of the previous frame |
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double isp[4][LP_ORDER]; ///< ISP vectors from current frame |
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double isp_sub4_past[LP_ORDER]; ///< ISP vector for the 4th subframe of the previous frame |
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float lp_coef[4][LP_ORDER]; ///< Linear Prediction Coefficients from ISP vector |
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uint8_t base_pitch_lag; ///< integer part of pitch lag for the next relative subframe |
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uint8_t pitch_lag_int; ///< integer part of pitch lag of the previous subframe |
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float excitation_buf[AMRWB_P_DELAY_MAX + LP_ORDER + 2 + AMRWB_SFR_SIZE]; ///< current excitation and all necessary excitation history |
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float *excitation; ///< points to current excitation in excitation_buf[] |
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float pitch_vector[AMRWB_SFR_SIZE]; ///< adaptive codebook (pitch) vector for current subframe |
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float fixed_vector[AMRWB_SFR_SIZE]; ///< algebraic codebook (fixed) vector for current subframe |
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float prediction_error[4]; ///< quantified prediction errors {20log10(^gamma_gc)} for previous four subframes |
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float pitch_gain[6]; ///< quantified pitch gains for the current and previous five subframes |
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float fixed_gain[2]; ///< quantified fixed gains for the current and previous subframes |
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float tilt_coef; ///< {beta_1} related to the voicing of the previous subframe |
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float prev_sparse_fixed_gain; ///< previous fixed gain; used by anti-sparseness to determine "onset" |
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uint8_t prev_ir_filter_nr; ///< previous impulse response filter "impNr": 0 - strong, 1 - medium, 2 - none |
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float prev_tr_gain; ///< previous initial gain used by noise enhancer for threshold |
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float samples_az[LP_ORDER + AMRWB_SFR_SIZE]; ///< low-band samples and memory from synthesis at 12.8kHz |
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float samples_up[UPS_MEM_SIZE + AMRWB_SFR_SIZE]; ///< low-band samples and memory processed for upsampling |
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float samples_hb[LP_ORDER_16k + AMRWB_SFR_SIZE_16k]; ///< high-band samples and memory from synthesis at 16kHz |
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float hpf_31_mem[2], hpf_400_mem[2]; ///< previous values in the high pass filters |
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float demph_mem[1]; ///< previous value in the de-emphasis filter |
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float bpf_6_7_mem[HB_FIR_SIZE]; ///< previous values in the high-band band pass filter |
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float lpf_7_mem[HB_FIR_SIZE]; ///< previous values in the high-band low pass filter |
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AVLFG prng; ///< random number generator for white noise excitation |
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uint8_t first_frame; ///< flag active during decoding of the first frame |
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} AMRWBContext; |
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static av_cold int amrwb_decode_init(AVCodecContext *avctx) |
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{ |
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AMRWBContext *ctx = avctx->priv_data; |
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int i; |
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avctx->sample_fmt = AV_SAMPLE_FMT_FLT; |
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av_lfg_init(&ctx->prng, 1); |
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ctx->excitation = &ctx->excitation_buf[AMRWB_P_DELAY_MAX + LP_ORDER + 1]; |
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ctx->first_frame = 1; |
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for (i = 0; i < LP_ORDER; i++) |
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ctx->isf_past_final[i] = isf_init[i] * (1.0f / (1 << 15)); |
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for (i = 0; i < 4; i++) |
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ctx->prediction_error[i] = MIN_ENERGY; |
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avcodec_get_frame_defaults(&ctx->avframe); |
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avctx->coded_frame = &ctx->avframe; |
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return 0; |
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} |
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/** |
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* Decode the frame header in the "MIME/storage" format. This format |
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* is simpler and does not carry the auxiliary frame information. |
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* |
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* @param[in] ctx The Context |
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* @param[in] buf Pointer to the input buffer |
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* |
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* @return The decoded header length in bytes |
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*/ |
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static int decode_mime_header(AMRWBContext *ctx, const uint8_t *buf) |
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{ |
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/* Decode frame header (1st octet) */ |
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ctx->fr_cur_mode = buf[0] >> 3 & 0x0F; |
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ctx->fr_quality = (buf[0] & 0x4) == 0x4; |
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return 1; |
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} |
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/** |
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* Decode quantized ISF vectors using 36-bit indexes (6K60 mode only). |
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* |
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* @param[in] ind Array of 5 indexes |
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* @param[out] isf_q Buffer for isf_q[LP_ORDER] |
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* |
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*/ |
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static void decode_isf_indices_36b(uint16_t *ind, float *isf_q) |
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{ |
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int i; |
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for (i = 0; i < 9; i++) |
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isf_q[i] = dico1_isf[ind[0]][i] * (1.0f / (1 << 15)); |
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for (i = 0; i < 7; i++) |
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isf_q[i + 9] = dico2_isf[ind[1]][i] * (1.0f / (1 << 15)); |
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for (i = 0; i < 5; i++) |
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isf_q[i] += dico21_isf_36b[ind[2]][i] * (1.0f / (1 << 15)); |
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for (i = 0; i < 4; i++) |
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isf_q[i + 5] += dico22_isf_36b[ind[3]][i] * (1.0f / (1 << 15)); |
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for (i = 0; i < 7; i++) |
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isf_q[i + 9] += dico23_isf_36b[ind[4]][i] * (1.0f / (1 << 15)); |
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} |
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/** |
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* Decode quantized ISF vectors using 46-bit indexes (except 6K60 mode). |
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* |
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* @param[in] ind Array of 7 indexes |
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* @param[out] isf_q Buffer for isf_q[LP_ORDER] |
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* |
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*/ |
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static void decode_isf_indices_46b(uint16_t *ind, float *isf_q) |
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{ |
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int i; |
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for (i = 0; i < 9; i++) |
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isf_q[i] = dico1_isf[ind[0]][i] * (1.0f / (1 << 15)); |
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for (i = 0; i < 7; i++) |
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isf_q[i + 9] = dico2_isf[ind[1]][i] * (1.0f / (1 << 15)); |
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for (i = 0; i < 3; i++) |
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isf_q[i] += dico21_isf[ind[2]][i] * (1.0f / (1 << 15)); |
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for (i = 0; i < 3; i++) |
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isf_q[i + 3] += dico22_isf[ind[3]][i] * (1.0f / (1 << 15)); |
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for (i = 0; i < 3; i++) |
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isf_q[i + 6] += dico23_isf[ind[4]][i] * (1.0f / (1 << 15)); |
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for (i = 0; i < 3; i++) |
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isf_q[i + 9] += dico24_isf[ind[5]][i] * (1.0f / (1 << 15)); |
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for (i = 0; i < 4; i++) |
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isf_q[i + 12] += dico25_isf[ind[6]][i] * (1.0f / (1 << 15)); |
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} |
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/** |
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* Apply mean and past ISF values using the prediction factor. |
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* Updates past ISF vector. |
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* |
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* @param[in,out] isf_q Current quantized ISF |
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* @param[in,out] isf_past Past quantized ISF |
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* |
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*/ |
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static void isf_add_mean_and_past(float *isf_q, float *isf_past) |
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{ |
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int i; |
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float tmp; |
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for (i = 0; i < LP_ORDER; i++) { |
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tmp = isf_q[i]; |
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isf_q[i] += isf_mean[i] * (1.0f / (1 << 15)); |
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isf_q[i] += PRED_FACTOR * isf_past[i]; |
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isf_past[i] = tmp; |
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} |
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} |
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/** |
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* Interpolate the fourth ISP vector from current and past frames |
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* to obtain an ISP vector for each subframe. |
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* |
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* @param[in,out] isp_q ISPs for each subframe |
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* @param[in] isp4_past Past ISP for subframe 4 |
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*/ |
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static void interpolate_isp(double isp_q[4][LP_ORDER], const double *isp4_past) |
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{ |
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int i, k; |
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for (k = 0; k < 3; k++) { |
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float c = isfp_inter[k]; |
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for (i = 0; i < LP_ORDER; i++) |
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isp_q[k][i] = (1.0 - c) * isp4_past[i] + c * isp_q[3][i]; |
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} |
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} |
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/** |
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* Decode an adaptive codebook index into pitch lag (except 6k60, 8k85 modes). |
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* Calculate integer lag and fractional lag always using 1/4 resolution. |
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* In 1st and 3rd subframes the index is relative to last subframe integer lag. |
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* |
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* @param[out] lag_int Decoded integer pitch lag |
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* @param[out] lag_frac Decoded fractional pitch lag |
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* @param[in] pitch_index Adaptive codebook pitch index |
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* @param[in,out] base_lag_int Base integer lag used in relative subframes |
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* @param[in] subframe Current subframe index (0 to 3) |
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*/ |
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static void decode_pitch_lag_high(int *lag_int, int *lag_frac, int pitch_index, |
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uint8_t *base_lag_int, int subframe) |
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{ |
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if (subframe == 0 || subframe == 2) { |
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if (pitch_index < 376) { |
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*lag_int = (pitch_index + 137) >> 2; |
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*lag_frac = pitch_index - (*lag_int << 2) + 136; |
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} else if (pitch_index < 440) { |
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*lag_int = (pitch_index + 257 - 376) >> 1; |
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*lag_frac = (pitch_index - (*lag_int << 1) + 256 - 376) << 1; |
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/* the actual resolution is 1/2 but expressed as 1/4 */ |
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} else { |
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*lag_int = pitch_index - 280; |
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*lag_frac = 0; |
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} |
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/* minimum lag for next subframe */ |
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*base_lag_int = av_clip(*lag_int - 8 - (*lag_frac < 0), |
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AMRWB_P_DELAY_MIN, AMRWB_P_DELAY_MAX - 15); |
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// XXX: the spec states clearly that *base_lag_int should be |
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// the nearest integer to *lag_int (minus 8), but the ref code |
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// actually always uses its floor, I'm following the latter |
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} else { |
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*lag_int = (pitch_index + 1) >> 2; |
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*lag_frac = pitch_index - (*lag_int << 2); |
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*lag_int += *base_lag_int; |
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} |
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} |
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/** |
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* Decode an adaptive codebook index into pitch lag for 8k85 and 6k60 modes. |
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* The description is analogous to decode_pitch_lag_high, but in 6k60 the |
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* relative index is used for all subframes except the first. |
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*/ |
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static void decode_pitch_lag_low(int *lag_int, int *lag_frac, int pitch_index, |
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uint8_t *base_lag_int, int subframe, enum Mode mode) |
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{ |
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if (subframe == 0 || (subframe == 2 && mode != MODE_6k60)) { |
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if (pitch_index < 116) { |
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*lag_int = (pitch_index + 69) >> 1; |
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*lag_frac = (pitch_index - (*lag_int << 1) + 68) << 1; |
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} else { |
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*lag_int = pitch_index - 24; |
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*lag_frac = 0; |
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} |
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// XXX: same problem as before |
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*base_lag_int = av_clip(*lag_int - 8 - (*lag_frac < 0), |
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AMRWB_P_DELAY_MIN, AMRWB_P_DELAY_MAX - 15); |
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} else { |
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*lag_int = (pitch_index + 1) >> 1; |
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*lag_frac = (pitch_index - (*lag_int << 1)) << 1; |
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*lag_int += *base_lag_int; |
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} |
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} |
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/** |
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* Find the pitch vector by interpolating the past excitation at the |
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* pitch delay, which is obtained in this function. |
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* |
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* @param[in,out] ctx The context |
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* @param[in] amr_subframe Current subframe data |
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* @param[in] subframe Current subframe index (0 to 3) |
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*/ |
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static void decode_pitch_vector(AMRWBContext *ctx, |
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const AMRWBSubFrame *amr_subframe, |
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const int subframe) |
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{ |
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int pitch_lag_int, pitch_lag_frac; |
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int i; |
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float *exc = ctx->excitation; |
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enum Mode mode = ctx->fr_cur_mode; |
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if (mode <= MODE_8k85) { |
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decode_pitch_lag_low(&pitch_lag_int, &pitch_lag_frac, amr_subframe->adap, |
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&ctx->base_pitch_lag, subframe, mode); |
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} else |
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decode_pitch_lag_high(&pitch_lag_int, &pitch_lag_frac, amr_subframe->adap, |
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&ctx->base_pitch_lag, subframe); |
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ctx->pitch_lag_int = pitch_lag_int; |
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pitch_lag_int += pitch_lag_frac > 0; |
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/* Calculate the pitch vector by interpolating the past excitation at the |
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pitch lag using a hamming windowed sinc function */ |
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ff_acelp_interpolatef(exc, exc + 1 - pitch_lag_int, |
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ac_inter, 4, |
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pitch_lag_frac + (pitch_lag_frac > 0 ? 0 : 4), |
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LP_ORDER, AMRWB_SFR_SIZE + 1); |
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/* Check which pitch signal path should be used |
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* 6k60 and 8k85 modes have the ltp flag set to 0 */ |
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if (amr_subframe->ltp) { |
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memcpy(ctx->pitch_vector, exc, AMRWB_SFR_SIZE * sizeof(float)); |
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} else { |
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for (i = 0; i < AMRWB_SFR_SIZE; i++) |
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ctx->pitch_vector[i] = 0.18 * exc[i - 1] + 0.64 * exc[i] + |
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0.18 * exc[i + 1]; |
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memcpy(exc, ctx->pitch_vector, AMRWB_SFR_SIZE * sizeof(float)); |
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} |
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} |
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/** Get x bits in the index interval [lsb,lsb+len-1] inclusive */ |
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#define BIT_STR(x,lsb,len) (((x) >> (lsb)) & ((1 << (len)) - 1)) |
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/** Get the bit at specified position */ |
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#define BIT_POS(x, p) (((x) >> (p)) & 1) |
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/** |
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* The next six functions decode_[i]p_track decode exactly i pulses |
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* positions and amplitudes (-1 or 1) in a subframe track using |
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* an encoded pulse indexing (TS 26.190 section 5.8.2). |
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* |
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* The results are given in out[], in which a negative number means |
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* amplitude -1 and vice versa (i.e., ampl(x) = x / abs(x) ). |
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* |
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* @param[out] out Output buffer (writes i elements) |
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* @param[in] code Pulse index (no. of bits varies, see below) |
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* @param[in] m (log2) Number of potential positions |
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* @param[in] off Offset for decoded positions |
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*/ |
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static inline void decode_1p_track(int *out, int code, int m, int off) |
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{ |
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int pos = BIT_STR(code, 0, m) + off; ///code: m+1 bits |
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out[0] = BIT_POS(code, m) ? -pos : pos; |
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} |
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static inline void decode_2p_track(int *out, int code, int m, int off) ///code: 2m+1 bits |
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{ |
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int pos0 = BIT_STR(code, m, m) + off; |
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int pos1 = BIT_STR(code, 0, m) + off; |
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out[0] = BIT_POS(code, 2*m) ? -pos0 : pos0; |
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out[1] = BIT_POS(code, 2*m) ? -pos1 : pos1; |
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out[1] = pos0 > pos1 ? -out[1] : out[1]; |
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} |
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static void decode_3p_track(int *out, int code, int m, int off) ///code: 3m+1 bits |
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{ |
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int half_2p = BIT_POS(code, 2*m - 1) << (m - 1); |
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decode_2p_track(out, BIT_STR(code, 0, 2*m - 1), |
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m - 1, off + half_2p); |
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decode_1p_track(out + 2, BIT_STR(code, 2*m, m + 1), m, off); |
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} |
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static void decode_4p_track(int *out, int code, int m, int off) ///code: 4m bits |
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{ |
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int half_4p, subhalf_2p; |
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int b_offset = 1 << (m - 1); |
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|
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switch (BIT_STR(code, 4*m - 2, 2)) { /* case ID (2 bits) */ |
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case 0: /* 0 pulses in A, 4 pulses in B or vice versa */ |
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half_4p = BIT_POS(code, 4*m - 3) << (m - 1); // which has 4 pulses |
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subhalf_2p = BIT_POS(code, 2*m - 3) << (m - 2); |
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decode_2p_track(out, BIT_STR(code, 0, 2*m - 3), |
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m - 2, off + half_4p + subhalf_2p); |
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decode_2p_track(out + 2, BIT_STR(code, 2*m - 2, 2*m - 1), |
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m - 1, off + half_4p); |
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break; |
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case 1: /* 1 pulse in A, 3 pulses in B */ |
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decode_1p_track(out, BIT_STR(code, 3*m - 2, m), |
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m - 1, off); |
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decode_3p_track(out + 1, BIT_STR(code, 0, 3*m - 2), |
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m - 1, off + b_offset); |
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break; |
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case 2: /* 2 pulses in each half */ |
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decode_2p_track(out, BIT_STR(code, 2*m - 1, 2*m - 1), |
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m - 1, off); |
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decode_2p_track(out + 2, BIT_STR(code, 0, 2*m - 1), |
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m - 1, off + b_offset); |
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break; |
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case 3: /* 3 pulses in A, 1 pulse in B */ |
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decode_3p_track(out, BIT_STR(code, m, 3*m - 2), |
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m - 1, off); |
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decode_1p_track(out + 3, BIT_STR(code, 0, m), |
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m - 1, off + b_offset); |
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break; |
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} |
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} |
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static void decode_5p_track(int *out, int code, int m, int off) ///code: 5m bits |
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{ |
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int half_3p = BIT_POS(code, 5*m - 1) << (m - 1); |
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|
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decode_3p_track(out, BIT_STR(code, 2*m + 1, 3*m - 2), |
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m - 1, off + half_3p); |
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|
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decode_2p_track(out + 3, BIT_STR(code, 0, 2*m + 1), m, off); |
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} |
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|
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static void decode_6p_track(int *out, int code, int m, int off) ///code: 6m-2 bits |
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{ |
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int b_offset = 1 << (m - 1); |
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/* which half has more pulses in cases 0 to 2 */ |
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int half_more = BIT_POS(code, 6*m - 5) << (m - 1); |
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int half_other = b_offset - half_more; |
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|
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switch (BIT_STR(code, 6*m - 4, 2)) { /* case ID (2 bits) */ |
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case 0: /* 0 pulses in A, 6 pulses in B or vice versa */ |
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decode_1p_track(out, BIT_STR(code, 0, m), |
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m - 1, off + half_more); |
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decode_5p_track(out + 1, BIT_STR(code, m, 5*m - 5), |
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m - 1, off + half_more); |
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break; |
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case 1: /* 1 pulse in A, 5 pulses in B or vice versa */ |
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decode_1p_track(out, BIT_STR(code, 0, m), |
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m - 1, off + half_other); |
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decode_5p_track(out + 1, BIT_STR(code, m, 5*m - 5), |
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m - 1, off + half_more); |
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break; |
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case 2: /* 2 pulses in A, 4 pulses in B or vice versa */ |
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decode_2p_track(out, BIT_STR(code, 0, 2*m - 1), |
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m - 1, off + half_other); |
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decode_4p_track(out + 2, BIT_STR(code, 2*m - 1, 4*m - 4), |
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m - 1, off + half_more); |
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break; |
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case 3: /* 3 pulses in A, 3 pulses in B */ |
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decode_3p_track(out, BIT_STR(code, 3*m - 2, 3*m - 2), |
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m - 1, off); |
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decode_3p_track(out + 3, BIT_STR(code, 0, 3*m - 2), |
|
m - 1, off + b_offset); |
|
break; |
|
} |
|
} |
|
|
|
/** |
|
* Decode the algebraic codebook index to pulse positions and signs, |
|
* then construct the algebraic codebook vector. |
|
* |
|
* @param[out] fixed_vector Buffer for the fixed codebook excitation |
|
* @param[in] pulse_hi MSBs part of the pulse index array (higher modes only) |
|
* @param[in] pulse_lo LSBs part of the pulse index array |
|
* @param[in] mode Mode of the current frame |
|
*/ |
|
static void decode_fixed_vector(float *fixed_vector, const uint16_t *pulse_hi, |
|
const uint16_t *pulse_lo, const enum Mode mode) |
|
{ |
|
/* sig_pos stores for each track the decoded pulse position indexes |
|
* (1-based) multiplied by its corresponding amplitude (+1 or -1) */ |
|
int sig_pos[4][6]; |
|
int spacing = (mode == MODE_6k60) ? 2 : 4; |
|
int i, j; |
|
|
|
switch (mode) { |
|
case MODE_6k60: |
|
for (i = 0; i < 2; i++) |
|
decode_1p_track(sig_pos[i], pulse_lo[i], 5, 1); |
|
break; |
|
case MODE_8k85: |
|
for (i = 0; i < 4; i++) |
|
decode_1p_track(sig_pos[i], pulse_lo[i], 4, 1); |
|
break; |
|
case MODE_12k65: |
|
for (i = 0; i < 4; i++) |
|
decode_2p_track(sig_pos[i], pulse_lo[i], 4, 1); |
|
break; |
|
case MODE_14k25: |
|
for (i = 0; i < 2; i++) |
|
decode_3p_track(sig_pos[i], pulse_lo[i], 4, 1); |
|
for (i = 2; i < 4; i++) |
|
decode_2p_track(sig_pos[i], pulse_lo[i], 4, 1); |
|
break; |
|
case MODE_15k85: |
|
for (i = 0; i < 4; i++) |
|
decode_3p_track(sig_pos[i], pulse_lo[i], 4, 1); |
|
break; |
|
case MODE_18k25: |
|
for (i = 0; i < 4; i++) |
|
decode_4p_track(sig_pos[i], (int) pulse_lo[i] + |
|
((int) pulse_hi[i] << 14), 4, 1); |
|
break; |
|
case MODE_19k85: |
|
for (i = 0; i < 2; i++) |
|
decode_5p_track(sig_pos[i], (int) pulse_lo[i] + |
|
((int) pulse_hi[i] << 10), 4, 1); |
|
for (i = 2; i < 4; i++) |
|
decode_4p_track(sig_pos[i], (int) pulse_lo[i] + |
|
((int) pulse_hi[i] << 14), 4, 1); |
|
break; |
|
case MODE_23k05: |
|
case MODE_23k85: |
|
for (i = 0; i < 4; i++) |
|
decode_6p_track(sig_pos[i], (int) pulse_lo[i] + |
|
((int) pulse_hi[i] << 11), 4, 1); |
|
break; |
|
} |
|
|
|
memset(fixed_vector, 0, sizeof(float) * AMRWB_SFR_SIZE); |
|
|
|
for (i = 0; i < 4; i++) |
|
for (j = 0; j < pulses_nb_per_mode_tr[mode][i]; j++) { |
|
int pos = (FFABS(sig_pos[i][j]) - 1) * spacing + i; |
|
|
|
fixed_vector[pos] += sig_pos[i][j] < 0 ? -1.0 : 1.0; |
|
} |
|
} |
|
|
|
/** |
|
* Decode pitch gain and fixed gain correction factor. |
|
* |
|
* @param[in] vq_gain Vector-quantized index for gains |
|
* @param[in] mode Mode of the current frame |
|
* @param[out] fixed_gain_factor Decoded fixed gain correction factor |
|
* @param[out] pitch_gain Decoded pitch gain |
|
*/ |
|
static void decode_gains(const uint8_t vq_gain, const enum Mode mode, |
|
float *fixed_gain_factor, float *pitch_gain) |
|
{ |
|
const int16_t *gains = (mode <= MODE_8k85 ? qua_gain_6b[vq_gain] : |
|
qua_gain_7b[vq_gain]); |
|
|
|
*pitch_gain = gains[0] * (1.0f / (1 << 14)); |
|
*fixed_gain_factor = gains[1] * (1.0f / (1 << 11)); |
|
} |
|
|
|
/** |
|
* Apply pitch sharpening filters to the fixed codebook vector. |
|
* |
|
* @param[in] ctx The context |
|
* @param[in,out] fixed_vector Fixed codebook excitation |
|
*/ |
|
// XXX: Spec states this procedure should be applied when the pitch |
|
// lag is less than 64, but this checking seems absent in reference and AMR-NB |
|
static void pitch_sharpening(AMRWBContext *ctx, float *fixed_vector) |
|
{ |
|
int i; |
|
|
|
/* Tilt part */ |
|
for (i = AMRWB_SFR_SIZE - 1; i != 0; i--) |
|
fixed_vector[i] -= fixed_vector[i - 1] * ctx->tilt_coef; |
|
|
|
/* Periodicity enhancement part */ |
|
for (i = ctx->pitch_lag_int; i < AMRWB_SFR_SIZE; i++) |
|
fixed_vector[i] += fixed_vector[i - ctx->pitch_lag_int] * 0.85; |
|
} |
|
|
|
/** |
|
* Calculate the voicing factor (-1.0 = unvoiced to 1.0 = voiced). |
|
* |
|
* @param[in] p_vector, f_vector Pitch and fixed excitation vectors |
|
* @param[in] p_gain, f_gain Pitch and fixed gains |
|
*/ |
|
// XXX: There is something wrong with the precision here! The magnitudes |
|
// of the energies are not correct. Please check the reference code carefully |
|
static float voice_factor(float *p_vector, float p_gain, |
|
float *f_vector, float f_gain) |
|
{ |
|
double p_ener = (double) ff_scalarproduct_float_c(p_vector, p_vector, |
|
AMRWB_SFR_SIZE) * |
|
p_gain * p_gain; |
|
double f_ener = (double) ff_scalarproduct_float_c(f_vector, f_vector, |
|
AMRWB_SFR_SIZE) * |
|
f_gain * f_gain; |
|
|
|
return (p_ener - f_ener) / (p_ener + f_ener); |
|
} |
|
|
|
/** |
|
* Reduce fixed vector sparseness by smoothing with one of three IR filters, |
|
* also known as "adaptive phase dispersion". |
|
* |
|
* @param[in] ctx The context |
|
* @param[in,out] fixed_vector Unfiltered fixed vector |
|
* @param[out] buf Space for modified vector if necessary |
|
* |
|
* @return The potentially overwritten filtered fixed vector address |
|
*/ |
|
static float *anti_sparseness(AMRWBContext *ctx, |
|
float *fixed_vector, float *buf) |
|
{ |
|
int ir_filter_nr; |
|
|
|
if (ctx->fr_cur_mode > MODE_8k85) // no filtering in higher modes |
|
return fixed_vector; |
|
|
|
if (ctx->pitch_gain[0] < 0.6) { |
|
ir_filter_nr = 0; // strong filtering |
|
} else if (ctx->pitch_gain[0] < 0.9) { |
|
ir_filter_nr = 1; // medium filtering |
|
} else |
|
ir_filter_nr = 2; // no filtering |
|
|
|
/* detect 'onset' */ |
|
if (ctx->fixed_gain[0] > 3.0 * ctx->fixed_gain[1]) { |
|
if (ir_filter_nr < 2) |
|
ir_filter_nr++; |
|
} else { |
|
int i, count = 0; |
|
|
|
for (i = 0; i < 6; i++) |
|
if (ctx->pitch_gain[i] < 0.6) |
|
count++; |
|
|
|
if (count > 2) |
|
ir_filter_nr = 0; |
|
|
|
if (ir_filter_nr > ctx->prev_ir_filter_nr + 1) |
|
ir_filter_nr--; |
|
} |
|
|
|
/* update ir filter strength history */ |
|
ctx->prev_ir_filter_nr = ir_filter_nr; |
|
|
|
ir_filter_nr += (ctx->fr_cur_mode == MODE_8k85); |
|
|
|
if (ir_filter_nr < 2) { |
|
int i; |
|
const float *coef = ir_filters_lookup[ir_filter_nr]; |
|
|
|
/* Circular convolution code in the reference |
|
* decoder was modified to avoid using one |
|
* extra array. The filtered vector is given by: |
|
* |
|
* c2(n) = sum(i,0,len-1){ c(i) * coef( (n - i + len) % len ) } |
|
*/ |
|
|
|
memset(buf, 0, sizeof(float) * AMRWB_SFR_SIZE); |
|
for (i = 0; i < AMRWB_SFR_SIZE; i++) |
|
if (fixed_vector[i]) |
|
ff_celp_circ_addf(buf, buf, coef, i, fixed_vector[i], |
|
AMRWB_SFR_SIZE); |
|
fixed_vector = buf; |
|
} |
|
|
|
return fixed_vector; |
|
} |
|
|
|
/** |
|
* Calculate a stability factor {teta} based on distance between |
|
* current and past isf. A value of 1 shows maximum signal stability. |
|
*/ |
|
static float stability_factor(const float *isf, const float *isf_past) |
|
{ |
|
int i; |
|
float acc = 0.0; |
|
|
|
for (i = 0; i < LP_ORDER - 1; i++) |
|
acc += (isf[i] - isf_past[i]) * (isf[i] - isf_past[i]); |
|
|
|
// XXX: This part is not so clear from the reference code |
|
// the result is more accurate changing the "/ 256" to "* 512" |
|
return FFMAX(0.0, 1.25 - acc * 0.8 * 512); |
|
} |
|
|
|
/** |
|
* Apply a non-linear fixed gain smoothing in order to reduce |
|
* fluctuation in the energy of excitation. |
|
* |
|
* @param[in] fixed_gain Unsmoothed fixed gain |
|
* @param[in,out] prev_tr_gain Previous threshold gain (updated) |
|
* @param[in] voice_fac Frame voicing factor |
|
* @param[in] stab_fac Frame stability factor |
|
* |
|
* @return The smoothed gain |
|
*/ |
|
static float noise_enhancer(float fixed_gain, float *prev_tr_gain, |
|
float voice_fac, float stab_fac) |
|
{ |
|
float sm_fac = 0.5 * (1 - voice_fac) * stab_fac; |
|
float g0; |
|
|
|
// XXX: the following fixed-point constants used to in(de)crement |
|
// gain by 1.5dB were taken from the reference code, maybe it could |
|
// be simpler |
|
if (fixed_gain < *prev_tr_gain) { |
|
g0 = FFMIN(*prev_tr_gain, fixed_gain + fixed_gain * |
|
(6226 * (1.0f / (1 << 15)))); // +1.5 dB |
|
} else |
|
g0 = FFMAX(*prev_tr_gain, fixed_gain * |
|
(27536 * (1.0f / (1 << 15)))); // -1.5 dB |
|
|
|
*prev_tr_gain = g0; // update next frame threshold |
|
|
|
return sm_fac * g0 + (1 - sm_fac) * fixed_gain; |
|
} |
|
|
|
/** |
|
* Filter the fixed_vector to emphasize the higher frequencies. |
|
* |
|
* @param[in,out] fixed_vector Fixed codebook vector |
|
* @param[in] voice_fac Frame voicing factor |
|
*/ |
|
static void pitch_enhancer(float *fixed_vector, float voice_fac) |
|
{ |
|
int i; |
|
float cpe = 0.125 * (1 + voice_fac); |
|
float last = fixed_vector[0]; // holds c(i - 1) |
|
|
|
fixed_vector[0] -= cpe * fixed_vector[1]; |
|
|
|
for (i = 1; i < AMRWB_SFR_SIZE - 1; i++) { |
|
float cur = fixed_vector[i]; |
|
|
|
fixed_vector[i] -= cpe * (last + fixed_vector[i + 1]); |
|
last = cur; |
|
} |
|
|
|
fixed_vector[AMRWB_SFR_SIZE - 1] -= cpe * last; |
|
} |
|
|
|
/** |
|
* Conduct 16th order linear predictive coding synthesis from excitation. |
|
* |
|
* @param[in] ctx Pointer to the AMRWBContext |
|
* @param[in] lpc Pointer to the LPC coefficients |
|
* @param[out] excitation Buffer for synthesis final excitation |
|
* @param[in] fixed_gain Fixed codebook gain for synthesis |
|
* @param[in] fixed_vector Algebraic codebook vector |
|
* @param[in,out] samples Pointer to the output samples and memory |
|
*/ |
|
static void synthesis(AMRWBContext *ctx, float *lpc, float *excitation, |
|
float fixed_gain, const float *fixed_vector, |
|
float *samples) |
|
{ |
|
ff_weighted_vector_sumf(excitation, ctx->pitch_vector, fixed_vector, |
|
ctx->pitch_gain[0], fixed_gain, AMRWB_SFR_SIZE); |
|
|
|
/* emphasize pitch vector contribution in low bitrate modes */ |
|
if (ctx->pitch_gain[0] > 0.5 && ctx->fr_cur_mode <= MODE_8k85) { |
|
int i; |
|
float energy = ff_scalarproduct_float_c(excitation, excitation, |
|
AMRWB_SFR_SIZE); |
|
|
|
// XXX: Weird part in both ref code and spec. A unknown parameter |
|
// {beta} seems to be identical to the current pitch gain |
|
float pitch_factor = 0.25 * ctx->pitch_gain[0] * ctx->pitch_gain[0]; |
|
|
|
for (i = 0; i < AMRWB_SFR_SIZE; i++) |
|
excitation[i] += pitch_factor * ctx->pitch_vector[i]; |
|
|
|
ff_scale_vector_to_given_sum_of_squares(excitation, excitation, |
|
energy, AMRWB_SFR_SIZE); |
|
} |
|
|
|
ff_celp_lp_synthesis_filterf(samples, lpc, excitation, |
|
AMRWB_SFR_SIZE, LP_ORDER); |
|
} |
|
|
|
/** |
|
* Apply to synthesis a de-emphasis filter of the form: |
|
* H(z) = 1 / (1 - m * z^-1) |
|
* |
|
* @param[out] out Output buffer |
|
* @param[in] in Input samples array with in[-1] |
|
* @param[in] m Filter coefficient |
|
* @param[in,out] mem State from last filtering |
|
*/ |
|
static void de_emphasis(float *out, float *in, float m, float mem[1]) |
|
{ |
|
int i; |
|
|
|
out[0] = in[0] + m * mem[0]; |
|
|
|
for (i = 1; i < AMRWB_SFR_SIZE; i++) |
|
out[i] = in[i] + out[i - 1] * m; |
|
|
|
mem[0] = out[AMRWB_SFR_SIZE - 1]; |
|
} |
|
|
|
/** |
|
* Upsample a signal by 5/4 ratio (from 12.8kHz to 16kHz) using |
|
* a FIR interpolation filter. Uses past data from before *in address. |
|
* |
|
* @param[out] out Buffer for interpolated signal |
|
* @param[in] in Current signal data (length 0.8*o_size) |
|
* @param[in] o_size Output signal length |
|
*/ |
|
static void upsample_5_4(float *out, const float *in, int o_size) |
|
{ |
|
const float *in0 = in - UPS_FIR_SIZE + 1; |
|
int i, j, k; |
|
int int_part = 0, frac_part; |
|
|
|
i = 0; |
|
for (j = 0; j < o_size / 5; j++) { |
|
out[i] = in[int_part]; |
|
frac_part = 4; |
|
i++; |
|
|
|
for (k = 1; k < 5; k++) { |
|
out[i] = ff_scalarproduct_float_c(in0 + int_part, |
|
upsample_fir[4 - frac_part], |
|
UPS_MEM_SIZE); |
|
int_part++; |
|
frac_part--; |
|
i++; |
|
} |
|
} |
|
} |
|
|
|
/** |
|
* Calculate the high-band gain based on encoded index (23k85 mode) or |
|
* on the low-band speech signal and the Voice Activity Detection flag. |
|
* |
|
* @param[in] ctx The context |
|
* @param[in] synth LB speech synthesis at 12.8k |
|
* @param[in] hb_idx Gain index for mode 23k85 only |
|
* @param[in] vad VAD flag for the frame |
|
*/ |
|
static float find_hb_gain(AMRWBContext *ctx, const float *synth, |
|
uint16_t hb_idx, uint8_t vad) |
|
{ |
|
int wsp = (vad > 0); |
|
float tilt; |
|
|
|
if (ctx->fr_cur_mode == MODE_23k85) |
|
return qua_hb_gain[hb_idx] * (1.0f / (1 << 14)); |
|
|
|
tilt = ff_scalarproduct_float_c(synth, synth + 1, AMRWB_SFR_SIZE - 1) / |
|
ff_scalarproduct_float_c(synth, synth, AMRWB_SFR_SIZE); |
|
|
|
/* return gain bounded by [0.1, 1.0] */ |
|
return av_clipf((1.0 - FFMAX(0.0, tilt)) * (1.25 - 0.25 * wsp), 0.1, 1.0); |
|
} |
|
|
|
/** |
|
* Generate the high-band excitation with the same energy from the lower |
|
* one and scaled by the given gain. |
|
* |
|
* @param[in] ctx The context |
|
* @param[out] hb_exc Buffer for the excitation |
|
* @param[in] synth_exc Low-band excitation used for synthesis |
|
* @param[in] hb_gain Wanted excitation gain |
|
*/ |
|
static void scaled_hb_excitation(AMRWBContext *ctx, float *hb_exc, |
|
const float *synth_exc, float hb_gain) |
|
{ |
|
int i; |
|
float energy = ff_scalarproduct_float_c(synth_exc, synth_exc, AMRWB_SFR_SIZE); |
|
|
|
/* Generate a white-noise excitation */ |
|
for (i = 0; i < AMRWB_SFR_SIZE_16k; i++) |
|
hb_exc[i] = 32768.0 - (uint16_t) av_lfg_get(&ctx->prng); |
|
|
|
ff_scale_vector_to_given_sum_of_squares(hb_exc, hb_exc, |
|
energy * hb_gain * hb_gain, |
|
AMRWB_SFR_SIZE_16k); |
|
} |
|
|
|
/** |
|
* Calculate the auto-correlation for the ISF difference vector. |
|
*/ |
|
static float auto_correlation(float *diff_isf, float mean, int lag) |
|
{ |
|
int i; |
|
float sum = 0.0; |
|
|
|
for (i = 7; i < LP_ORDER - 2; i++) { |
|
float prod = (diff_isf[i] - mean) * (diff_isf[i - lag] - mean); |
|
sum += prod * prod; |
|
} |
|
return sum; |
|
} |
|
|
|
/** |
|
* Extrapolate a ISF vector to the 16kHz range (20th order LP) |
|
* used at mode 6k60 LP filter for the high frequency band. |
|
* |
|
* @param[out] isf Buffer for extrapolated isf; contains LP_ORDER |
|
* values on input |
|
*/ |
|
static void extrapolate_isf(float isf[LP_ORDER_16k]) |
|
{ |
|
float diff_isf[LP_ORDER - 2], diff_mean; |
|
float *diff_hi = diff_isf - LP_ORDER + 1; // diff array for extrapolated indexes |
|
float corr_lag[3]; |
|
float est, scale; |
|
int i, i_max_corr; |
|
|
|
isf[LP_ORDER_16k - 1] = isf[LP_ORDER - 1]; |
|
|
|
/* Calculate the difference vector */ |
|
for (i = 0; i < LP_ORDER - 2; i++) |
|
diff_isf[i] = isf[i + 1] - isf[i]; |
|
|
|
diff_mean = 0.0; |
|
for (i = 2; i < LP_ORDER - 2; i++) |
|
diff_mean += diff_isf[i] * (1.0f / (LP_ORDER - 4)); |
|
|
|
/* Find which is the maximum autocorrelation */ |
|
i_max_corr = 0; |
|
for (i = 0; i < 3; i++) { |
|
corr_lag[i] = auto_correlation(diff_isf, diff_mean, i + 2); |
|
|
|
if (corr_lag[i] > corr_lag[i_max_corr]) |
|
i_max_corr = i; |
|
} |
|
i_max_corr++; |
|
|
|
for (i = LP_ORDER - 1; i < LP_ORDER_16k - 1; i++) |
|
isf[i] = isf[i - 1] + isf[i - 1 - i_max_corr] |
|
- isf[i - 2 - i_max_corr]; |
|
|
|
/* Calculate an estimate for ISF(18) and scale ISF based on the error */ |
|
est = 7965 + (isf[2] - isf[3] - isf[4]) / 6.0; |
|
scale = 0.5 * (FFMIN(est, 7600) - isf[LP_ORDER - 2]) / |
|
(isf[LP_ORDER_16k - 2] - isf[LP_ORDER - 2]); |
|
|
|
for (i = LP_ORDER - 1; i < LP_ORDER_16k - 1; i++) |
|
diff_hi[i] = scale * (isf[i] - isf[i - 1]); |
|
|
|
/* Stability insurance */ |
|
for (i = LP_ORDER; i < LP_ORDER_16k - 1; i++) |
|
if (diff_hi[i] + diff_hi[i - 1] < 5.0) { |
|
if (diff_hi[i] > diff_hi[i - 1]) { |
|
diff_hi[i - 1] = 5.0 - diff_hi[i]; |
|
} else |
|
diff_hi[i] = 5.0 - diff_hi[i - 1]; |
|
} |
|
|
|
for (i = LP_ORDER - 1; i < LP_ORDER_16k - 1; i++) |
|
isf[i] = isf[i - 1] + diff_hi[i] * (1.0f / (1 << 15)); |
|
|
|
/* Scale the ISF vector for 16000 Hz */ |
|
for (i = 0; i < LP_ORDER_16k - 1; i++) |
|
isf[i] *= 0.8; |
|
} |
|
|
|
/** |
|
* Spectral expand the LP coefficients using the equation: |
|
* y[i] = x[i] * (gamma ** i) |
|
* |
|
* @param[out] out Output buffer (may use input array) |
|
* @param[in] lpc LP coefficients array |
|
* @param[in] gamma Weighting factor |
|
* @param[in] size LP array size |
|
*/ |
|
static void lpc_weighting(float *out, const float *lpc, float gamma, int size) |
|
{ |
|
int i; |
|
float fac = gamma; |
|
|
|
for (i = 0; i < size; i++) { |
|
out[i] = lpc[i] * fac; |
|
fac *= gamma; |
|
} |
|
} |
|
|
|
/** |
|
* Conduct 20th order linear predictive coding synthesis for the high |
|
* frequency band excitation at 16kHz. |
|
* |
|
* @param[in] ctx The context |
|
* @param[in] subframe Current subframe index (0 to 3) |
|
* @param[in,out] samples Pointer to the output speech samples |
|
* @param[in] exc Generated white-noise scaled excitation |
|
* @param[in] isf Current frame isf vector |
|
* @param[in] isf_past Past frame final isf vector |
|
*/ |
|
static void hb_synthesis(AMRWBContext *ctx, int subframe, float *samples, |
|
const float *exc, const float *isf, const float *isf_past) |
|
{ |
|
float hb_lpc[LP_ORDER_16k]; |
|
enum Mode mode = ctx->fr_cur_mode; |
|
|
|
if (mode == MODE_6k60) { |
|
float e_isf[LP_ORDER_16k]; // ISF vector for extrapolation |
|
double e_isp[LP_ORDER_16k]; |
|
|
|
ff_weighted_vector_sumf(e_isf, isf_past, isf, isfp_inter[subframe], |
|
1.0 - isfp_inter[subframe], LP_ORDER); |
|
|
|
extrapolate_isf(e_isf); |
|
|
|
e_isf[LP_ORDER_16k - 1] *= 2.0; |
|
ff_acelp_lsf2lspd(e_isp, e_isf, LP_ORDER_16k); |
|
ff_amrwb_lsp2lpc(e_isp, hb_lpc, LP_ORDER_16k); |
|
|
|
lpc_weighting(hb_lpc, hb_lpc, 0.9, LP_ORDER_16k); |
|
} else { |
|
lpc_weighting(hb_lpc, ctx->lp_coef[subframe], 0.6, LP_ORDER); |
|
} |
|
|
|
ff_celp_lp_synthesis_filterf(samples, hb_lpc, exc, AMRWB_SFR_SIZE_16k, |
|
(mode == MODE_6k60) ? LP_ORDER_16k : LP_ORDER); |
|
} |
|
|
|
/** |
|
* Apply a 15th order filter to high-band samples. |
|
* The filter characteristic depends on the given coefficients. |
|
* |
|
* @param[out] out Buffer for filtered output |
|
* @param[in] fir_coef Filter coefficients |
|
* @param[in,out] mem State from last filtering (updated) |
|
* @param[in] in Input speech data (high-band) |
|
* |
|
* @remark It is safe to pass the same array in in and out parameters |
|
*/ |
|
static void hb_fir_filter(float *out, const float fir_coef[HB_FIR_SIZE + 1], |
|
float mem[HB_FIR_SIZE], const float *in) |
|
{ |
|
int i, j; |
|
float data[AMRWB_SFR_SIZE_16k + HB_FIR_SIZE]; // past and current samples |
|
|
|
memcpy(data, mem, HB_FIR_SIZE * sizeof(float)); |
|
memcpy(data + HB_FIR_SIZE, in, AMRWB_SFR_SIZE_16k * sizeof(float)); |
|
|
|
for (i = 0; i < AMRWB_SFR_SIZE_16k; i++) { |
|
out[i] = 0.0; |
|
for (j = 0; j <= HB_FIR_SIZE; j++) |
|
out[i] += data[i + j] * fir_coef[j]; |
|
} |
|
|
|
memcpy(mem, data + AMRWB_SFR_SIZE_16k, HB_FIR_SIZE * sizeof(float)); |
|
} |
|
|
|
/** |
|
* Update context state before the next subframe. |
|
*/ |
|
static void update_sub_state(AMRWBContext *ctx) |
|
{ |
|
memmove(&ctx->excitation_buf[0], &ctx->excitation_buf[AMRWB_SFR_SIZE], |
|
(AMRWB_P_DELAY_MAX + LP_ORDER + 1) * sizeof(float)); |
|
|
|
memmove(&ctx->pitch_gain[1], &ctx->pitch_gain[0], 5 * sizeof(float)); |
|
memmove(&ctx->fixed_gain[1], &ctx->fixed_gain[0], 1 * sizeof(float)); |
|
|
|
memmove(&ctx->samples_az[0], &ctx->samples_az[AMRWB_SFR_SIZE], |
|
LP_ORDER * sizeof(float)); |
|
memmove(&ctx->samples_up[0], &ctx->samples_up[AMRWB_SFR_SIZE], |
|
UPS_MEM_SIZE * sizeof(float)); |
|
memmove(&ctx->samples_hb[0], &ctx->samples_hb[AMRWB_SFR_SIZE_16k], |
|
LP_ORDER_16k * sizeof(float)); |
|
} |
|
|
|
static int amrwb_decode_frame(AVCodecContext *avctx, void *data, |
|
int *got_frame_ptr, AVPacket *avpkt) |
|
{ |
|
AMRWBContext *ctx = avctx->priv_data; |
|
AMRWBFrame *cf = &ctx->frame; |
|
const uint8_t *buf = avpkt->data; |
|
int buf_size = avpkt->size; |
|
int expected_fr_size, header_size; |
|
float *buf_out; |
|
float spare_vector[AMRWB_SFR_SIZE]; // extra stack space to hold result from anti-sparseness processing |
|
float fixed_gain_factor; // fixed gain correction factor (gamma) |
|
float *synth_fixed_vector; // pointer to the fixed vector that synthesis should use |
|
float synth_fixed_gain; // the fixed gain that synthesis should use |
|
float voice_fac, stab_fac; // parameters used for gain smoothing |
|
float synth_exc[AMRWB_SFR_SIZE]; // post-processed excitation for synthesis |
|
float hb_exc[AMRWB_SFR_SIZE_16k]; // excitation for the high frequency band |
|
float hb_samples[AMRWB_SFR_SIZE_16k]; // filtered high-band samples from synthesis |
|
float hb_gain; |
|
int sub, i, ret; |
|
|
|
/* get output buffer */ |
|
ctx->avframe.nb_samples = 4 * AMRWB_SFR_SIZE_16k; |
|
if ((ret = avctx->get_buffer(avctx, &ctx->avframe)) < 0) { |
|
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); |
|
return ret; |
|
} |
|
buf_out = (float *)ctx->avframe.data[0]; |
|
|
|
header_size = decode_mime_header(ctx, buf); |
|
if (ctx->fr_cur_mode > MODE_SID) { |
|
av_log(avctx, AV_LOG_ERROR, |
|
"Invalid mode %d\n", ctx->fr_cur_mode); |
|
return AVERROR_INVALIDDATA; |
|
} |
|
expected_fr_size = ((cf_sizes_wb[ctx->fr_cur_mode] + 7) >> 3) + 1; |
|
|
|
if (buf_size < expected_fr_size) { |
|
av_log(avctx, AV_LOG_ERROR, |
|
"Frame too small (%d bytes). Truncated file?\n", buf_size); |
|
*got_frame_ptr = 0; |
|
return AVERROR_INVALIDDATA; |
|
} |
|
|
|
if (!ctx->fr_quality || ctx->fr_cur_mode > MODE_SID) |
|
av_log(avctx, AV_LOG_ERROR, "Encountered a bad or corrupted frame\n"); |
|
|
|
if (ctx->fr_cur_mode == MODE_SID) { /* Comfort noise frame */ |
|
av_log_missing_feature(avctx, "SID mode", 1); |
|
return AVERROR_PATCHWELCOME; |
|
} |
|
|
|
ff_amr_bit_reorder((uint16_t *) &ctx->frame, sizeof(AMRWBFrame), |
|
buf + header_size, amr_bit_orderings_by_mode[ctx->fr_cur_mode]); |
|
|
|
/* Decode the quantized ISF vector */ |
|
if (ctx->fr_cur_mode == MODE_6k60) { |
|
decode_isf_indices_36b(cf->isp_id, ctx->isf_cur); |
|
} else { |
|
decode_isf_indices_46b(cf->isp_id, ctx->isf_cur); |
|
} |
|
|
|
isf_add_mean_and_past(ctx->isf_cur, ctx->isf_q_past); |
|
ff_set_min_dist_lsf(ctx->isf_cur, MIN_ISF_SPACING, LP_ORDER - 1); |
|
|
|
stab_fac = stability_factor(ctx->isf_cur, ctx->isf_past_final); |
|
|
|
ctx->isf_cur[LP_ORDER - 1] *= 2.0; |
|
ff_acelp_lsf2lspd(ctx->isp[3], ctx->isf_cur, LP_ORDER); |
|
|
|
/* Generate a ISP vector for each subframe */ |
|
if (ctx->first_frame) { |
|
ctx->first_frame = 0; |
|
memcpy(ctx->isp_sub4_past, ctx->isp[3], LP_ORDER * sizeof(double)); |
|
} |
|
interpolate_isp(ctx->isp, ctx->isp_sub4_past); |
|
|
|
for (sub = 0; sub < 4; sub++) |
|
ff_amrwb_lsp2lpc(ctx->isp[sub], ctx->lp_coef[sub], LP_ORDER); |
|
|
|
for (sub = 0; sub < 4; sub++) { |
|
const AMRWBSubFrame *cur_subframe = &cf->subframe[sub]; |
|
float *sub_buf = buf_out + sub * AMRWB_SFR_SIZE_16k; |
|
|
|
/* Decode adaptive codebook (pitch vector) */ |
|
decode_pitch_vector(ctx, cur_subframe, sub); |
|
/* Decode innovative codebook (fixed vector) */ |
|
decode_fixed_vector(ctx->fixed_vector, cur_subframe->pul_ih, |
|
cur_subframe->pul_il, ctx->fr_cur_mode); |
|
|
|
pitch_sharpening(ctx, ctx->fixed_vector); |
|
|
|
decode_gains(cur_subframe->vq_gain, ctx->fr_cur_mode, |
|
&fixed_gain_factor, &ctx->pitch_gain[0]); |
|
|
|
ctx->fixed_gain[0] = |
|
ff_amr_set_fixed_gain(fixed_gain_factor, |
|
ff_scalarproduct_float_c(ctx->fixed_vector, |
|
ctx->fixed_vector, |
|
AMRWB_SFR_SIZE) / |
|
AMRWB_SFR_SIZE, |
|
ctx->prediction_error, |
|
ENERGY_MEAN, energy_pred_fac); |
|
|
|
/* Calculate voice factor and store tilt for next subframe */ |
|
voice_fac = voice_factor(ctx->pitch_vector, ctx->pitch_gain[0], |
|
ctx->fixed_vector, ctx->fixed_gain[0]); |
|
ctx->tilt_coef = voice_fac * 0.25 + 0.25; |
|
|
|
/* Construct current excitation */ |
|
for (i = 0; i < AMRWB_SFR_SIZE; i++) { |
|
ctx->excitation[i] *= ctx->pitch_gain[0]; |
|
ctx->excitation[i] += ctx->fixed_gain[0] * ctx->fixed_vector[i]; |
|
ctx->excitation[i] = truncf(ctx->excitation[i]); |
|
} |
|
|
|
/* Post-processing of excitation elements */ |
|
synth_fixed_gain = noise_enhancer(ctx->fixed_gain[0], &ctx->prev_tr_gain, |
|
voice_fac, stab_fac); |
|
|
|
synth_fixed_vector = anti_sparseness(ctx, ctx->fixed_vector, |
|
spare_vector); |
|
|
|
pitch_enhancer(synth_fixed_vector, voice_fac); |
|
|
|
synthesis(ctx, ctx->lp_coef[sub], synth_exc, synth_fixed_gain, |
|
synth_fixed_vector, &ctx->samples_az[LP_ORDER]); |
|
|
|
/* Synthesis speech post-processing */ |
|
de_emphasis(&ctx->samples_up[UPS_MEM_SIZE], |
|
&ctx->samples_az[LP_ORDER], PREEMPH_FAC, ctx->demph_mem); |
|
|
|
ff_acelp_apply_order_2_transfer_function(&ctx->samples_up[UPS_MEM_SIZE], |
|
&ctx->samples_up[UPS_MEM_SIZE], hpf_zeros, hpf_31_poles, |
|
hpf_31_gain, ctx->hpf_31_mem, AMRWB_SFR_SIZE); |
|
|
|
upsample_5_4(sub_buf, &ctx->samples_up[UPS_FIR_SIZE], |
|
AMRWB_SFR_SIZE_16k); |
|
|
|
/* High frequency band (6.4 - 7.0 kHz) generation part */ |
|
ff_acelp_apply_order_2_transfer_function(hb_samples, |
|
&ctx->samples_up[UPS_MEM_SIZE], hpf_zeros, hpf_400_poles, |
|
hpf_400_gain, ctx->hpf_400_mem, AMRWB_SFR_SIZE); |
|
|
|
hb_gain = find_hb_gain(ctx, hb_samples, |
|
cur_subframe->hb_gain, cf->vad); |
|
|
|
scaled_hb_excitation(ctx, hb_exc, synth_exc, hb_gain); |
|
|
|
hb_synthesis(ctx, sub, &ctx->samples_hb[LP_ORDER_16k], |
|
hb_exc, ctx->isf_cur, ctx->isf_past_final); |
|
|
|
/* High-band post-processing filters */ |
|
hb_fir_filter(hb_samples, bpf_6_7_coef, ctx->bpf_6_7_mem, |
|
&ctx->samples_hb[LP_ORDER_16k]); |
|
|
|
if (ctx->fr_cur_mode == MODE_23k85) |
|
hb_fir_filter(hb_samples, lpf_7_coef, ctx->lpf_7_mem, |
|
hb_samples); |
|
|
|
/* Add the low and high frequency bands */ |
|
for (i = 0; i < AMRWB_SFR_SIZE_16k; i++) |
|
sub_buf[i] = (sub_buf[i] + hb_samples[i]) * (1.0f / (1 << 15)); |
|
|
|
/* Update buffers and history */ |
|
update_sub_state(ctx); |
|
} |
|
|
|
/* update state for next frame */ |
|
memcpy(ctx->isp_sub4_past, ctx->isp[3], LP_ORDER * sizeof(ctx->isp[3][0])); |
|
memcpy(ctx->isf_past_final, ctx->isf_cur, LP_ORDER * sizeof(float)); |
|
|
|
*got_frame_ptr = 1; |
|
*(AVFrame *)data = ctx->avframe; |
|
|
|
return expected_fr_size; |
|
} |
|
|
|
AVCodec ff_amrwb_decoder = { |
|
.name = "amrwb", |
|
.type = AVMEDIA_TYPE_AUDIO, |
|
.id = AV_CODEC_ID_AMR_WB, |
|
.priv_data_size = sizeof(AMRWBContext), |
|
.init = amrwb_decode_init, |
|
.decode = amrwb_decode_frame, |
|
.capabilities = CODEC_CAP_DR1, |
|
.long_name = NULL_IF_CONFIG_SMALL("AMR-WB (Adaptive Multi-Rate WideBand)"), |
|
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT, |
|
AV_SAMPLE_FMT_NONE }, |
|
};
|
|
|