From 2b2a597ec08b10a4995159b9f2572308c14dff47 Mon Sep 17 00:00:00 2001 From: "Ronald S. Bultje" Date: Sat, 18 Dec 2010 03:03:18 +0000 Subject: AMR-WB decoder, written as part of Google Summer of Code 2010 by Marcelo Galvão Póvoa , mentored by Robert Swain . MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Originally committed as revision 26051 to svn://svn.ffmpeg.org/ffmpeg/trunk --- libavcodec/amrwbdec.c | 1237 +++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 1237 insertions(+) create mode 100644 libavcodec/amrwbdec.c (limited to 'libavcodec/amrwbdec.c') diff --git a/libavcodec/amrwbdec.c b/libavcodec/amrwbdec.c new file mode 100644 index 0000000000..64ab707fd3 --- /dev/null +++ b/libavcodec/amrwbdec.c @@ -0,0 +1,1237 @@ +/* + * AMR wideband decoder + * Copyright (c) 2010 Marcelo Galvao Povoa + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A particular PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +/** + * @file + * AMR wideband decoder + */ + +#include "libavutil/lfg.h" + +#include "avcodec.h" +#include "get_bits.h" +#include "lsp.h" +#include "celp_math.h" +#include "celp_filters.h" +#include "acelp_filters.h" +#include "acelp_vectors.h" +#include "acelp_pitch_delay.h" + +#define AMR_USE_16BIT_TABLES +#include "amr.h" + +#include "amrwbdata.h" + +typedef struct { + AMRWBFrame frame; ///< AMRWB parameters decoded from bitstream + enum Mode fr_cur_mode; ///< mode index of current frame + uint8_t fr_quality; ///< frame quality index (FQI) + float isf_cur[LP_ORDER]; ///< working ISF vector from current frame + float isf_q_past[LP_ORDER]; ///< quantized ISF vector of the previous frame + float isf_past_final[LP_ORDER]; ///< final processed ISF vector of the previous frame + double isp[4][LP_ORDER]; ///< ISP vectors from current frame + double isp_sub4_past[LP_ORDER]; ///< ISP vector for the 4th subframe of the previous frame + + float lp_coef[4][LP_ORDER]; ///< Linear Prediction Coefficients from ISP vector + + uint8_t base_pitch_lag; ///< integer part of pitch lag for the next relative subframe + uint8_t pitch_lag_int; ///< integer part of pitch lag of the previous subframe + + float excitation_buf[AMRWB_P_DELAY_MAX + LP_ORDER + 2 + AMRWB_SFR_SIZE]; ///< current excitation and all necessary excitation history + float *excitation; ///< points to current excitation in excitation_buf[] + + float pitch_vector[AMRWB_SFR_SIZE]; ///< adaptive codebook (pitch) vector for current subframe + float fixed_vector[AMRWB_SFR_SIZE]; ///< algebraic codebook (fixed) vector for current subframe + + float prediction_error[4]; ///< quantified prediction errors {20log10(^gamma_gc)} for previous four subframes + float pitch_gain[6]; ///< quantified pitch gains for the current and previous five subframes + float fixed_gain[2]; ///< quantified fixed gains for the current and previous subframes + + float tilt_coef; ///< {beta_1} related to the voicing of the previous subframe + + float prev_sparse_fixed_gain; ///< previous fixed gain; used by anti-sparseness to determine "onset" + uint8_t prev_ir_filter_nr; ///< previous impulse response filter "impNr": 0 - strong, 1 - medium, 2 - none + float prev_tr_gain; ///< previous initial gain used by noise enhancer for threshold + + float samples_az[LP_ORDER + AMRWB_SFR_SIZE]; ///< low-band samples and memory from synthesis at 12.8kHz + float samples_up[UPS_MEM_SIZE + AMRWB_SFR_SIZE]; ///< low-band samples and memory processed for upsampling + float samples_hb[LP_ORDER_16k + AMRWB_SFR_SIZE_16k]; ///< high-band samples and memory from synthesis at 16kHz + + float hpf_31_mem[2], hpf_400_mem[2]; ///< previous values in the high pass filters + float demph_mem[1]; ///< previous value in the de-emphasis filter + float bpf_6_7_mem[HB_FIR_SIZE]; ///< previous values in the high-band band pass filter + float lpf_7_mem[HB_FIR_SIZE]; ///< previous values in the high-band low pass filter + + AVLFG prng; ///< random number generator for white noise excitation + uint8_t first_frame; ///< flag active during decoding of the first frame +} AMRWBContext; + +static av_cold int amrwb_decode_init(AVCodecContext *avctx) +{ + AMRWBContext *ctx = avctx->priv_data; + int i; + + avctx->sample_fmt = SAMPLE_FMT_FLT; + + av_lfg_init(&ctx->prng, 1); + + ctx->excitation = &ctx->excitation_buf[AMRWB_P_DELAY_MAX + LP_ORDER + 1]; + ctx->first_frame = 1; + + for (i = 0; i < LP_ORDER; i++) + ctx->isf_past_final[i] = isf_init[i] * (1.0f / (1 << 15)); + + for (i = 0; i < 4; i++) + ctx->prediction_error[i] = MIN_ENERGY; + + return 0; +} + +/** + * Decode the frame header in the "MIME/storage" format. This format + * is simpler and does not carry the auxiliary information of the frame + * + * @param[in] ctx The Context + * @param[in] buf Pointer to the input buffer + * + * @return The decoded header length in bytes + */ +static int decode_mime_header(AMRWBContext *ctx, const uint8_t *buf) +{ + GetBitContext gb; + init_get_bits(&gb, buf, 8); + + /* Decode frame header (1st octet) */ + skip_bits(&gb, 1); // padding bit + ctx->fr_cur_mode = get_bits(&gb, 4); + ctx->fr_quality = get_bits1(&gb); + skip_bits(&gb, 2); // padding bits + + return 1; +} + +/** + * Decodes quantized ISF vectors using 36-bit indexes (6K60 mode only) + * + * @param[in] ind Array of 5 indexes + * @param[out] isf_q Buffer for isf_q[LP_ORDER] + * + */ +static void decode_isf_indices_36b(uint16_t *ind, float *isf_q) +{ + int i; + + for (i = 0; i < 9; i++) + isf_q[i] = dico1_isf[ind[0]][i] * (1.0f / (1 << 15)); + + for (i = 0; i < 7; i++) + isf_q[i + 9] = dico2_isf[ind[1]][i] * (1.0f / (1 << 15)); + + for (i = 0; i < 5; i++) + isf_q[i] += dico21_isf_36b[ind[2]][i] * (1.0f / (1 << 15)); + + for (i = 0; i < 4; i++) + isf_q[i + 5] += dico22_isf_36b[ind[3]][i] * (1.0f / (1 << 15)); + + for (i = 0; i < 7; i++) + isf_q[i + 9] += dico23_isf_36b[ind[4]][i] * (1.0f / (1 << 15)); +} + +/** + * Decodes quantized ISF vectors using 46-bit indexes (except 6K60 mode) + * + * @param[in] ind Array of 7 indexes + * @param[out] isf_q Buffer for isf_q[LP_ORDER] + * + */ +static void decode_isf_indices_46b(uint16_t *ind, float *isf_q) +{ + int i; + + for (i = 0; i < 9; i++) + isf_q[i] = dico1_isf[ind[0]][i] * (1.0f / (1 << 15)); + + for (i = 0; i < 7; i++) + isf_q[i + 9] = dico2_isf[ind[1]][i] * (1.0f / (1 << 15)); + + for (i = 0; i < 3; i++) + isf_q[i] += dico21_isf[ind[2]][i] * (1.0f / (1 << 15)); + + for (i = 0; i < 3; i++) + isf_q[i + 3] += dico22_isf[ind[3]][i] * (1.0f / (1 << 15)); + + for (i = 0; i < 3; i++) + isf_q[i + 6] += dico23_isf[ind[4]][i] * (1.0f / (1 << 15)); + + for (i = 0; i < 3; i++) + isf_q[i + 9] += dico24_isf[ind[5]][i] * (1.0f / (1 << 15)); + + for (i = 0; i < 4; i++) + isf_q[i + 12] += dico25_isf[ind[6]][i] * (1.0f / (1 << 15)); +} + +/** + * Apply mean and past ISF values using the prediction factor + * Updates past ISF vector + * + * @param[in,out] isf_q Current quantized ISF + * @param[in,out] isf_past Past quantized ISF + * + */ +static void isf_add_mean_and_past(float *isf_q, float *isf_past) +{ + int i; + float tmp; + + for (i = 0; i < LP_ORDER; i++) { + tmp = isf_q[i]; + isf_q[i] += isf_mean[i] * (1.0f / (1 << 15)); + isf_q[i] += PRED_FACTOR * isf_past[i]; + isf_past[i] = tmp; + } +} + +/** + * Interpolate the fourth ISP vector from current and past frames + * to obtain a ISP vector for each subframe + * + * @param[in,out] isp_q ISPs for each subframe + * @param[in] isp4_past Past ISP for subframe 4 + */ +static void interpolate_isp(double isp_q[4][LP_ORDER], const double *isp4_past) +{ + int i, k; + + for (k = 0; k < 3; k++) { + float c = isfp_inter[k]; + for (i = 0; i < LP_ORDER; i++) + isp_q[k][i] = (1.0 - c) * isp4_past[i] + c * isp_q[3][i]; + } +} + +/** + * Decode an adaptive codebook index into pitch lag (except 6k60, 8k85 modes) + * Calculate integer lag and fractional lag always using 1/4 resolution + * In 1st and 3rd subframes the index is relative to last subframe integer lag + * + * @param[out] lag_int Decoded integer pitch lag + * @param[out] lag_frac Decoded fractional pitch lag + * @param[in] pitch_index Adaptive codebook pitch index + * @param[in,out] base_lag_int Base integer lag used in relative subframes + * @param[in] subframe Current subframe index (0 to 3) + */ +static void decode_pitch_lag_high(int *lag_int, int *lag_frac, int pitch_index, + uint8_t *base_lag_int, int subframe) +{ + if (subframe == 0 || subframe == 2) { + if (pitch_index < 376) { + *lag_int = (pitch_index + 137) >> 2; + *lag_frac = pitch_index - (*lag_int << 2) + 136; + } else if (pitch_index < 440) { + *lag_int = (pitch_index + 257 - 376) >> 1; + *lag_frac = (pitch_index - (*lag_int << 1) + 256 - 376) << 1; + /* the actual resolution is 1/2 but expressed as 1/4 */ + } else { + *lag_int = pitch_index - 280; + *lag_frac = 0; + } + /* minimum lag for next subframe */ + *base_lag_int = av_clip(*lag_int - 8 - (*lag_frac < 0), + AMRWB_P_DELAY_MIN, AMRWB_P_DELAY_MAX - 15); + // XXX: the spec states clearly that *base_lag_int should be + // the nearest integer to *lag_int (minus 8), but the ref code + // actually always uses its floor, I'm following the latter + } else { + *lag_int = (pitch_index + 1) >> 2; + *lag_frac = pitch_index - (*lag_int << 2); + *lag_int += *base_lag_int; + } +} + +/** + * Decode a adaptive codebook index into pitch lag for 8k85 and 6k60 modes + * Description is analogous to decode_pitch_lag_high, but in 6k60 relative + * index is used for all subframes except the first + */ +static void decode_pitch_lag_low(int *lag_int, int *lag_frac, int pitch_index, + uint8_t *base_lag_int, int subframe, enum Mode mode) +{ + if (subframe == 0 || (subframe == 2 && mode != MODE_6k60)) { + if (pitch_index < 116) { + *lag_int = (pitch_index + 69) >> 1; + *lag_frac = (pitch_index - (*lag_int << 1) + 68) << 1; + } else { + *lag_int = pitch_index - 24; + *lag_frac = 0; + } + // XXX: same problem as before + *base_lag_int = av_clip(*lag_int - 8 - (*lag_frac < 0), + AMRWB_P_DELAY_MIN, AMRWB_P_DELAY_MAX - 15); + } else { + *lag_int = (pitch_index + 1) >> 1; + *lag_frac = (pitch_index - (*lag_int << 1)) << 1; + *lag_int += *base_lag_int; + } +} + +/** + * Find the pitch vector by interpolating the past excitation at the + * pitch delay, which is obtained in this function + * + * @param[in,out] ctx The context + * @param[in] amr_subframe Current subframe data + * @param[in] subframe Current subframe index (0 to 3) + */ +static void decode_pitch_vector(AMRWBContext *ctx, + const AMRWBSubFrame *amr_subframe, + const int subframe) +{ + int pitch_lag_int, pitch_lag_frac; + int i; + float *exc = ctx->excitation; + enum Mode mode = ctx->fr_cur_mode; + + if (mode <= MODE_8k85) { + decode_pitch_lag_low(&pitch_lag_int, &pitch_lag_frac, amr_subframe->adap, + &ctx->base_pitch_lag, subframe, mode); + } else + decode_pitch_lag_high(&pitch_lag_int, &pitch_lag_frac, amr_subframe->adap, + &ctx->base_pitch_lag, subframe); + + ctx->pitch_lag_int = pitch_lag_int; + pitch_lag_int += pitch_lag_frac > 0; + + /* Calculate the pitch vector by interpolating the past excitation at the + pitch lag using a hamming windowed sinc function */ + ff_acelp_interpolatef(exc, exc + 1 - pitch_lag_int, + ac_inter, 4, + pitch_lag_frac + (pitch_lag_frac > 0 ? 0 : 4), + LP_ORDER, AMRWB_SFR_SIZE + 1); + + /* Check which pitch signal path should be used + * 6k60 and 8k85 modes have the ltp flag set to 0 */ + if (amr_subframe->ltp) { + memcpy(ctx->pitch_vector, exc, AMRWB_SFR_SIZE * sizeof(float)); + } else { + for (i = 0; i < AMRWB_SFR_SIZE; i++) + ctx->pitch_vector[i] = 0.18 * exc[i - 1] + 0.64 * exc[i] + + 0.18 * exc[i + 1]; + memcpy(exc, ctx->pitch_vector, AMRWB_SFR_SIZE * sizeof(float)); + } +} + +/** Get x bits in the index interval [lsb,lsb+len-1] inclusive */ +#define BIT_STR(x,lsb,len) (((x) >> (lsb)) & ((1 << (len)) - 1)) + +/** Get the bit at specified position */ +#define BIT_POS(x, p) (((x) >> (p)) & 1) + +/** + * The next six functions decode_[i]p_track decode exactly i pulses + * positions and amplitudes (-1 or 1) in a subframe track using + * an encoded pulse indexing (TS 26.190 section 5.8.2) + * + * The results are given in out[], in which a negative number means + * amplitude -1 and vice versa (i.e., ampl(x) = x / abs(x) ) + * + * @param[out] out Output buffer (writes i elements) + * @param[in] code Pulse index (no. of bits varies, see below) + * @param[in] m (log2) Number of potential positions + * @param[in] off Offset for decoded positions + */ +static inline void decode_1p_track(int *out, int code, int m, int off) +{ + int pos = BIT_STR(code, 0, m) + off; ///code: m+1 bits + + out[0] = BIT_POS(code, m) ? -pos : pos; +} + +static inline void decode_2p_track(int *out, int code, int m, int off) ///code: 2m+1 bits +{ + int pos0 = BIT_STR(code, m, m) + off; + int pos1 = BIT_STR(code, 0, m) + off; + + out[0] = BIT_POS(code, 2*m) ? -pos0 : pos0; + out[1] = BIT_POS(code, 2*m) ? -pos1 : pos1; + out[1] = pos0 > pos1 ? -out[1] : out[1]; +} + +static void decode_3p_track(int *out, int code, int m, int off) ///code: 3m+1 bits +{ + int half_2p = BIT_POS(code, 2*m - 1) << (m - 1); + + decode_2p_track(out, BIT_STR(code, 0, 2*m - 1), + m - 1, off + half_2p); + decode_1p_track(out + 2, BIT_STR(code, 2*m, m + 1), m, off); +} + +static void decode_4p_track(int *out, int code, int m, int off) ///code: 4m bits +{ + int half_4p, subhalf_2p; + int b_offset = 1 << (m - 1); + + switch (BIT_STR(code, 4*m - 2, 2)) { /* case ID (2 bits) */ + 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 + */ +// 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_dot_productf(p_vector, p_vector, + AMRWB_SFR_SIZE) * p_gain * p_gain; + double f_ener = (double) ff_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) +{ + 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_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); + } + + 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_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 = ff_dot_productf(synth, synth + 1, AMRWB_SFR_SIZE - 1) / + ff_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 = ff_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] out Buffer for extrapolated isf + * @param[in] isf Input isf vector + */ +static void extrapolate_isf(float out[LP_ORDER_16k], float isf[LP_ORDER]) +{ + 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; + + memcpy(out, isf, (LP_ORDER - 1) * sizeof(float)); + out[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++) + out[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 + (out[2] - out[3] - out[4]) / 6.0; + scale = 0.5 * (FFMIN(est, 7600) - out[LP_ORDER - 2]) / + (out[LP_ORDER_16k - 2] - out[LP_ORDER - 2]); + + for (i = LP_ORDER - 1; i < LP_ORDER_16k - 1; i++) + diff_hi[i] = scale * (out[i] - out[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++) + out[i] = out[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++) + out[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); + + 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 to high-band samples a 15th order filter + * 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 *data_size, + 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 = data; + 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; + + header_size = decode_mime_header(ctx, buf); + 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); + *data_size = 0; + return buf_size; + } + + 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); + + if (ctx->fr_cur_mode >= MODE_SID) + 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, + ff_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->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)); + + /* report how many samples we got */ + *data_size = 4 * AMRWB_SFR_SIZE_16k * sizeof(float); + + return expected_fr_size; +} + +AVCodec amrwb_decoder = { + .name = "amrwb", + .type = CODEC_TYPE_AUDIO, + .id = CODEC_ID_AMR_WB, + .priv_data_size = sizeof(AMRWBContext), + .init = amrwb_decode_init, + .decode = amrwb_decode_frame, + .long_name = NULL_IF_CONFIG_SMALL("Adaptive Multi-Rate WideBand"), + .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_FLT,SAMPLE_FMT_NONE}, +}; -- cgit v1.2.3