mirror of
https://github.com/FFmpeg/FFmpeg.git
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416d2f7a12
* qatar/master:
vc1: export some functions
configure: use HOSTCC_C/O in check_host_cc
configure: use AS_O setting in check_as
configure: use LD_O setting in check_ld()
Revert "dsputil: make {add/put/put_signed}_pixels_clamped() non-static."
build: Restore dependency of acelp_filters.o on celp_math.o
celp_math: Replace duplicate ff_dot_productf() by ff_scalarproduct_c()
celp_math: Move ff_cos() to the only place it is used
build: Use portable abstraction for linker/hostcc output file syntax
configure: Fix shared library creation for OpenBSD
vp56: Don't use DECLARE_ALIGN on a typedef name
mss1: move code that will be reused by MSS2 decoder into separate file
mss1: merge decode_intra() and decode_inter()
avprobe: Get rid of ugly casts in the options table
vf_hqdn3d: Remove a duplicate inline declaration
Conflicts:
Makefile
configure
ffprobe.c
libavcodec/Makefile
libavcodec/amrnbdec.c
libavcodec/amrwbdec.c
libavcodec/celp_math.c
libavcodec/celp_math.h
libavcodec/dsputil.c
libavcodec/lsp.c
libavcodec/mss1.c
libavcodec/ra288.c
libavcodec/vc1dec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
1275 lines
47 KiB
C
1275 lines
47 KiB
C
/*
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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 FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A particular PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* 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 "celp_math.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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#include "mips/amrwbdec_mips.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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ACELPFContext acelpf_ctx; ///< context for filters for ACELP-based codecs
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ACELPVContext acelpv_ctx; ///< context for vector operations for ACELP-based codecs
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CELPFContext celpf_ctx; ///< context for filters for CELP-based codecs
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CELPMContext celpm_ctx; ///< context for fixed point math operations
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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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ff_acelp_filter_init(&ctx->acelpf_ctx);
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ff_acelp_vectors_init(&ctx->acelpv_ctx);
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ff_celp_filter_init(&ctx->celpf_ctx);
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ff_celp_math_init(&ctx->celpm_ctx);
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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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ctx->acelpf_ctx.acelp_interpolatef(exc,
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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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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 */
|
|
half_4p = BIT_POS(code, 4*m - 3) << (m - 1); // which has 4 pulses
|
|
subhalf_2p = BIT_POS(code, 2*m - 3) << (m - 2);
|
|
|
|
decode_2p_track(out, BIT_STR(code, 0, 2*m - 3),
|
|
m - 2, off + half_4p + subhalf_2p);
|
|
decode_2p_track(out + 2, BIT_STR(code, 2*m - 2, 2*m - 1),
|
|
m - 1, off + half_4p);
|
|
break;
|
|
case 1: /* 1 pulse in A, 3 pulses in B */
|
|
decode_1p_track(out, BIT_STR(code, 3*m - 2, m),
|
|
m - 1, off);
|
|
decode_3p_track(out + 1, BIT_STR(code, 0, 3*m - 2),
|
|
m - 1, off + b_offset);
|
|
break;
|
|
case 2: /* 2 pulses in each half */
|
|
decode_2p_track(out, BIT_STR(code, 2*m - 1, 2*m - 1),
|
|
m - 1, off);
|
|
decode_2p_track(out + 2, BIT_STR(code, 0, 2*m - 1),
|
|
m - 1, off + b_offset);
|
|
break;
|
|
case 3: /* 3 pulses in A, 1 pulse in B */
|
|
decode_3p_track(out, BIT_STR(code, m, 3*m - 2),
|
|
m - 1, off);
|
|
decode_1p_track(out + 3, BIT_STR(code, 0, m),
|
|
m - 1, off + b_offset);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void decode_5p_track(int *out, int code, int m, int off) ///code: 5m bits
|
|
{
|
|
int half_3p = BIT_POS(code, 5*m - 1) << (m - 1);
|
|
|
|
decode_3p_track(out, BIT_STR(code, 2*m + 1, 3*m - 2),
|
|
m - 1, off + half_3p);
|
|
|
|
decode_2p_track(out + 3, BIT_STR(code, 0, 2*m + 1), m, off);
|
|
}
|
|
|
|
static void decode_6p_track(int *out, int code, int m, int off) ///code: 6m-2 bits
|
|
{
|
|
int b_offset = 1 << (m - 1);
|
|
/* which half has more pulses in cases 0 to 2 */
|
|
int half_more = BIT_POS(code, 6*m - 5) << (m - 1);
|
|
int half_other = b_offset - half_more;
|
|
|
|
switch (BIT_STR(code, 6*m - 4, 2)) { /* case ID (2 bits) */
|
|
case 0: /* 0 pulses in A, 6 pulses in B or vice versa */
|
|
decode_1p_track(out, BIT_STR(code, 0, m),
|
|
m - 1, off + half_more);
|
|
decode_5p_track(out + 1, BIT_STR(code, m, 5*m - 5),
|
|
m - 1, off + half_more);
|
|
break;
|
|
case 1: /* 1 pulse in A, 5 pulses in B or vice versa */
|
|
decode_1p_track(out, BIT_STR(code, 0, m),
|
|
m - 1, off + half_other);
|
|
decode_5p_track(out + 1, BIT_STR(code, m, 5*m - 5),
|
|
m - 1, off + half_more);
|
|
break;
|
|
case 2: /* 2 pulses in A, 4 pulses in B or vice versa */
|
|
decode_2p_track(out, BIT_STR(code, 0, 2*m - 1),
|
|
m - 1, off + half_other);
|
|
decode_4p_track(out + 2, BIT_STR(code, 2*m - 1, 4*m - 4),
|
|
m - 1, off + half_more);
|
|
break;
|
|
case 3: /* 3 pulses in A, 3 pulses in B */
|
|
decode_3p_track(out, BIT_STR(code, 3*m - 2, 3*m - 2),
|
|
m - 1, off);
|
|
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
|
|
* @param[in] ctx The context
|
|
*/
|
|
// 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,
|
|
CELPMContext *ctx)
|
|
{
|
|
double p_ener = (double) ctx->dot_productf(p_vector, p_vector,
|
|
AMRWB_SFR_SIZE) *
|
|
p_gain * p_gain;
|
|
double f_ener = (double) ctx->dot_productf(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)
|
|
{
|
|
ctx->acelpv_ctx.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 = ctx->celpm_ctx.dot_productf(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);
|
|
}
|
|
|
|
ctx->celpf_ctx.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
|
|
* @param[in] ctx The context
|
|
*/
|
|
static void upsample_5_4(float *out, const float *in, int o_size, CELPMContext *ctx)
|
|
{
|
|
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] = ctx->dot_productf(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 = ctx->celpm_ctx.dot_productf(synth, synth + 1, AMRWB_SFR_SIZE - 1) /
|
|
ctx->celpm_ctx.dot_productf(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 = ctx->celpm_ctx.dot_productf(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];
|
|
|
|
ctx->acelpv_ctx.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);
|
|
}
|
|
|
|
ctx->celpf_ctx.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
|
|
*/
|
|
|
|
#ifndef hb_fir_filter
|
|
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));
|
|
}
|
|
#endif /* hb_fir_filter */
|
|
|
|
/**
|
|
* 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 -1;
|
|
}
|
|
|
|
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,
|
|
ctx->celpm_ctx.dot_productf(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->celpm_ctx);
|
|
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);
|
|
|
|
ctx->acelpf_ctx.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, &ctx->celpm_ctx);
|
|
|
|
/* High frequency band (6.4 - 7.0 kHz) generation part */
|
|
ctx->acelpf_ctx.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 },
|
|
};
|