Mercurial > libavcodec.hg
diff sipr.c @ 10836:f20f9b76f4c8 libavcodec
SIPR decoder for modes 8k5, 6k5 and 5k0.
Patch by Vladimir Voroshilov and myself.
| author | vitor |
|---|---|
| date | Sun, 10 Jan 2010 21:38:15 +0000 |
| parents | |
| children | c06d95ce2db0 |
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--- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/sipr.c Sun Jan 10 21:38:15 2010 +0000 @@ -0,0 +1,608 @@ +/* + * SIPR / ACELP.NET decoder + * + * Copyright (c) 2008 Vladimir Voroshilov + * Copyright (c) 2009 Vitor Sessak + * + * 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 + */ + +#include <math.h> +#include <stdint.h> + +#include "avcodec.h" +#define ALT_BITSTREAM_READER_LE +#include "get_bits.h" +#include "dsputil.h" + +#include "lsp.h" +#include "celp_math.h" +#include "acelp_vectors.h" +#include "acelp_pitch_delay.h" +#include "acelp_filters.h" +#include "celp_filters.h" + +#define LSFQ_DIFF_MIN (0.0125 * M_PI) + +#define LP_FILTER_ORDER 10 + +/** Number of past samples needed for excitation interpolation */ +#define L_INTERPOL (LP_FILTER_ORDER + 1) + +/** Subframe size for all modes except 16k */ +#define SUBFR_SIZE 48 + +#include "siprdata.h" + +typedef enum { + MODE_16k, + MODE_8k5, + MODE_6k5, + MODE_5k0, + MODE_COUNT +} SiprMode; + +typedef struct { + const char *mode_name; + uint16_t bits_per_frame; + uint8_t subframe_count; + uint8_t frames_per_packet; + float pitch_sharp_factor; + + /* bitstream parameters */ + uint8_t number_of_fc_indexes; + + /** size in bits of the i-th stage vector of quantizer */ + uint8_t vq_indexes_bits[5]; + + /** size in bits of the adaptive-codebook index for every subframe */ + uint8_t pitch_delay_bits[5]; + + uint8_t gp_index_bits; + uint8_t fc_index_bits[10]; ///< size in bits of the fixed codebook indexes + uint8_t gc_index_bits; ///< size in bits of the gain codebook indexes +} SiprModeParam; + +static const SiprModeParam modes[MODE_COUNT] = { + [MODE_8k5] = { + .mode_name = "8k5", + .bits_per_frame = 152, + .subframe_count = 3, + .frames_per_packet = 1, + .pitch_sharp_factor = 0.8, + + .number_of_fc_indexes = 3, + .vq_indexes_bits = {6, 7, 7, 7, 5}, + .pitch_delay_bits = {8, 5, 5}, + .gp_index_bits = 0, + .fc_index_bits = {9, 9, 9}, + .gc_index_bits = 7 + }, + + [MODE_6k5] = { + .mode_name = "6k5", + .bits_per_frame = 232, + .subframe_count = 3, + .frames_per_packet = 2, + .pitch_sharp_factor = 0.8, + + .number_of_fc_indexes = 3, + .vq_indexes_bits = {6, 7, 7, 7, 5}, + .pitch_delay_bits = {8, 5, 5}, + .gp_index_bits = 0, + .fc_index_bits = {5, 5, 5}, + .gc_index_bits = 7 + }, + + [MODE_5k0] = { + .mode_name = "5k0", + .bits_per_frame = 296, + .subframe_count = 5, + .frames_per_packet = 2, + .pitch_sharp_factor = 0.85, + + .number_of_fc_indexes = 1, + .vq_indexes_bits = {6, 7, 7, 7, 5}, + .pitch_delay_bits = {8, 5, 8, 5, 5}, + .gp_index_bits = 0, + .fc_index_bits = {10}, + .gc_index_bits = 7 + } +}; + +typedef struct { + AVCodecContext *avctx; + DSPContext dsp; + + SiprModeParam m; + SiprMode mode; + + float past_pitch_gain; + float lsf_history[LP_FILTER_ORDER]; + + float excitation[L_INTERPOL + PITCH_DELAY_MAX + 5*SUBFR_SIZE]; + + DECLARE_ALIGNED_16(float, synth_buf[LP_FILTER_ORDER + 5*SUBFR_SIZE + 6]); + + float lsp_history[LP_FILTER_ORDER]; + float gain_mem; + float energy_history[4]; + float highpass_filt_mem[2]; + float postfilter_mem[PITCH_DELAY_MAX + LP_FILTER_ORDER]; + + /* 5k0 */ + float tilt_mem; + float postfilter_agc; + float postfilter_mem5k0[PITCH_DELAY_MAX + LP_FILTER_ORDER]; + float postfilter_syn5k0[LP_FILTER_ORDER + SUBFR_SIZE*5]; +} SiprContext; + +typedef struct { + int vq_indexes[5]; + int pitch_delay[5]; ///< pitch delay + int gp_index[5]; ///< adaptive-codebook gain indexes + int16_t fc_indexes[5][10]; ///< fixed-codebook indexes + int gc_index[5]; ///< fixed-codebook gain indexes +} SiprParameters; + + +static void dequant(float *out, const int *idx, const float *cbs[]) +{ + int i; + int stride = 2; + int num_vec = 5; + + for (i = 0; i < num_vec; i++) + memcpy(out + stride*i, cbs[i] + stride*idx[i], stride*sizeof(float)); + +} + +static void lsf_decode_fp(float *lsfnew, float *lsf_history, + const SiprParameters *parm) +{ + int i; + float lsf_tmp[LP_FILTER_ORDER]; + + dequant(lsf_tmp, parm->vq_indexes, lsf_codebooks); + + for (i = 0; i < LP_FILTER_ORDER; i++) + lsfnew[i] = lsf_history[i] * 0.33 + lsf_tmp[i] + mean_lsf[i]; + + ff_sort_nearly_sorted_floats(lsfnew, LP_FILTER_ORDER - 1); + + /* Note that a minimum distance is not enforced between the last value and + the previous one, contrary to what is done in ff_acelp_reorder_lsf() */ + ff_set_min_dist_lsf(lsfnew, LSFQ_DIFF_MIN, LP_FILTER_ORDER - 1); + lsfnew[9] = FFMIN(lsfnew[LP_FILTER_ORDER - 1], 1.3 * M_PI); + + memcpy(lsf_history, lsf_tmp, LP_FILTER_ORDER * sizeof(*lsf_history)); + + for (i = 0; i < LP_FILTER_ORDER - 1; i++) + lsfnew[i] = cos(lsfnew[i]); + lsfnew[LP_FILTER_ORDER - 1] *= 6.153848 / M_PI; +} + +/** Apply pitch lag to the fixed vector (AMR section 6.1.2). */ +static void pitch_sharpening(int pitch_lag_int, float beta, + float *fixed_vector) +{ + int i; + + for (i = pitch_lag_int; i < SUBFR_SIZE; i++) + fixed_vector[i] += beta * fixed_vector[i - pitch_lag_int]; +} + +/** + * Extracts decoding parameters from the input bitstream. + * @param parms parameters structure + * @param pgb pointer to initialized GetBitContext structure + */ +static void decode_parameters(SiprParameters* parms, GetBitContext *pgb, + const SiprModeParam *p) +{ + int i, j; + + for (i = 0; i < 5; i++) + parms->vq_indexes[i] = get_bits(pgb, p->vq_indexes_bits[i]); + + for (i = 0; i < p->subframe_count; i++) { + parms->pitch_delay[i] = get_bits(pgb, p->pitch_delay_bits[i]); + parms->gp_index[i] = get_bits(pgb, p->gp_index_bits); + + for (j = 0; j < p->number_of_fc_indexes; j++) + parms->fc_indexes[i][j] = get_bits(pgb, p->fc_index_bits[j]); + + parms->gc_index[i] = get_bits(pgb, p->gc_index_bits); + } +} + +static void lsp2lpc_sipr(const double *lsp, float *Az) +{ + int lp_half_order = LP_FILTER_ORDER >> 1; + double buf[lp_half_order + 1]; + double pa[lp_half_order + 1]; + double *qa = buf + 1; + int i,j; + + qa[-1] = 0.0; + + ff_lsp2polyf(lsp , pa, lp_half_order ); + ff_lsp2polyf(lsp + 1, qa, lp_half_order - 1); + + for (i = 1, j = LP_FILTER_ORDER - 1; i < lp_half_order; i++, j--) { + double paf = pa[i] * (1 + lsp[LP_FILTER_ORDER - 1]); + double qaf = (qa[i] - qa[i-2]) * (1 - lsp[LP_FILTER_ORDER - 1]); + Az[i-1] = (paf + qaf) * 0.5; + Az[j-1] = (paf - qaf) * 0.5; + } + + Az[lp_half_order - 1] = (1.0 + lsp[LP_FILTER_ORDER - 1]) * + pa[lp_half_order] * 0.5; + + Az[LP_FILTER_ORDER - 1] = lsp[LP_FILTER_ORDER - 1]; +} + +static void sipr_decode_lp(float *lsfnew, const float *lsfold, float *Az, + int num_subfr) +{ + double lsfint[LP_FILTER_ORDER]; + int i,j; + float t, t0 = 1.0 / num_subfr; + + t = t0 * 0.5; + for (i = 0; i < num_subfr; i++) { + for (j = 0; j < LP_FILTER_ORDER; j++) + lsfint[j] = lsfold[j] * (1 - t) + t * lsfnew[j]; + + lsp2lpc_sipr(lsfint, Az); + Az += LP_FILTER_ORDER; + t += t0; + } +} + +/** + * Evaluates the adaptative impulse response. + */ +static void eval_ir(const float *Az, int pitch_lag, float *freq, + float pitch_sharp_factor) +{ + float tmp1[SUBFR_SIZE+1], tmp2[LP_FILTER_ORDER+1]; + int i; + + tmp1[0] = 1.; + for (i = 0; i < LP_FILTER_ORDER; i++) { + tmp1[i+1] = Az[i] * ff_pow_0_55[i]; + tmp2[i ] = Az[i] * ff_pow_0_7 [i]; + } + memset(tmp1 + 11, 0, 37 * sizeof(float)); + + ff_celp_lp_synthesis_filterf(freq, tmp2, tmp1, SUBFR_SIZE, + LP_FILTER_ORDER); + + pitch_sharpening(pitch_lag, pitch_sharp_factor, freq); +} + +/** + * Evaluates the convolution of a vector with a sparse vector. + */ +static void convolute_with_sparse(float *out, const AMRFixed *pulses, + const float *shape, int length) +{ + int i, j; + + memset(out, 0, length*sizeof(float)); + for (i = 0; i < pulses->n; i++) + for (j = pulses->x[i]; j < length; j++) + out[j] += pulses->y[i] * shape[j - pulses->x[i]]; +} + +/** + * Apply postfilter, very similar to AMR one. + */ +static void postfilter_5k0(SiprContext *ctx, const float *lpc, float *samples) +{ + float buf[SUBFR_SIZE + LP_FILTER_ORDER]; + float *pole_out = buf + LP_FILTER_ORDER; + float lpc_n[LP_FILTER_ORDER]; + float lpc_d[LP_FILTER_ORDER]; + int i; + + for (i = 0; i < LP_FILTER_ORDER; i++) { + lpc_d[i] = lpc[i] * ff_pow_0_75[i]; + lpc_n[i] = lpc[i] * pow_0_5 [i]; + }; + + memcpy(pole_out - LP_FILTER_ORDER, ctx->postfilter_mem, + LP_FILTER_ORDER*sizeof(float)); + + ff_celp_lp_synthesis_filterf(pole_out, lpc_d, samples, SUBFR_SIZE, + LP_FILTER_ORDER); + + memcpy(ctx->postfilter_mem, pole_out + SUBFR_SIZE - LP_FILTER_ORDER, + LP_FILTER_ORDER*sizeof(float)); + + ff_tilt_compensation(&ctx->tilt_mem, 0.4, pole_out, SUBFR_SIZE); + + memcpy(pole_out - LP_FILTER_ORDER, ctx->postfilter_mem5k0, + LP_FILTER_ORDER*sizeof(*pole_out)); + + memcpy(ctx->postfilter_mem5k0, pole_out + SUBFR_SIZE - LP_FILTER_ORDER, + LP_FILTER_ORDER*sizeof(*pole_out)); + + ff_celp_lp_zero_synthesis_filterf(samples, lpc_n, pole_out, SUBFR_SIZE, + LP_FILTER_ORDER); + +} + +static void decode_fixed_sparse(AMRFixed *fixed_sparse, const int16_t *pulses, + SiprMode mode, int low_gain) +{ + int i; + + switch (mode) { + case MODE_6k5: + for (i = 0; i < 3; i++) { + fixed_sparse->x[i] = 3 * (pulses[i] & 0xf) + i; + fixed_sparse->y[i] = pulses[i] & 0x10 ? -1 : 1; + } + fixed_sparse->n = 3; + break; + case MODE_8k5: + for (i = 0; i < 3; i++) { + fixed_sparse->x[2*i ] = 3 * ((pulses[i] >> 4) & 0xf) + i; + fixed_sparse->x[2*i + 1] = 3 * ( pulses[i] & 0xf) + i; + + fixed_sparse->y[2*i ] = (pulses[i] & 0x100) ? -1.0: 1.0; + + fixed_sparse->y[2*i + 1] = + (fixed_sparse->x[2*i + 1] < fixed_sparse->x[2*i]) ? + -fixed_sparse->y[2*i ] : fixed_sparse->y[2*i]; + } + + fixed_sparse->n = 6; + break; + case MODE_5k0: + default: + if (low_gain) { + int offset = (pulses[0] & 0x200) ? 2 : 0; + int val = pulses[0]; + + for (i = 0; i < 3; i++) { + int index = (val & 0x7) * 6 + 4 - i*2; + + fixed_sparse->y[i] = (offset + index) & 0x3 ? -1 : 1; + fixed_sparse->x[i] = index; + + val >>= 3; + } + fixed_sparse->n = 3; + } else { + int pulse_subset = (pulses[0] >> 8) & 1; + + fixed_sparse->x[0] = ((pulses[0] >> 4) & 15) * 3 + pulse_subset; + fixed_sparse->x[1] = ( pulses[0] & 15) * 3 + pulse_subset + 1; + + fixed_sparse->y[0] = pulses[0] & 0x200 ? -1 : 1; + fixed_sparse->y[1] = -fixed_sparse->y[0]; + fixed_sparse->n = 2; + } + break; + } +} + +static void decode_frame(SiprContext *ctx, SiprParameters *params, + float *out_data) +{ + int i, j; + int frame_size = ctx->m.subframe_count * SUBFR_SIZE; + float Az[LP_FILTER_ORDER * ctx->m.subframe_count]; + float *excitation; + float ir_buf[SUBFR_SIZE + LP_FILTER_ORDER]; + float lsf_new[LP_FILTER_ORDER]; + float *impulse_response = ir_buf + LP_FILTER_ORDER; + float *synth = ctx->synth_buf + 16; // 16 instead of LP_FILTER_ORDER for + // memory alignment + int t0_first = 0; + AMRFixed fixed_cb; + + memset(ir_buf, 0, LP_FILTER_ORDER * sizeof(float)); + lsf_decode_fp(lsf_new, ctx->lsf_history, params); + + sipr_decode_lp(lsf_new, ctx->lsp_history, Az, ctx->m.subframe_count); + + memcpy(ctx->lsp_history, lsf_new, LP_FILTER_ORDER * sizeof(float)); + + excitation = ctx->excitation + PITCH_DELAY_MAX + L_INTERPOL; + + for (i = 0; i < ctx->m.subframe_count; i++) { + float *pAz = Az + i*LP_FILTER_ORDER; + float fixed_vector[SUBFR_SIZE]; + int T0,T0_frac; + float pitch_gain, gain_code, avg_energy; + + ff_decode_pitch_lag(&T0, &T0_frac, params->pitch_delay[i], t0_first, i, + ctx->mode == MODE_5k0, 6); + + if (i == 0 || (i == 2 && ctx->mode == MODE_5k0)) + t0_first = T0; + + ff_acelp_interpolatef(excitation, excitation - T0 + (T0_frac <= 0), + ff_b60_sinc, 6, + 2 * ((2 + T0_frac)%3 + 1), LP_FILTER_ORDER, + SUBFR_SIZE); + + decode_fixed_sparse(&fixed_cb, params->fc_indexes[i], ctx->mode, + ctx->past_pitch_gain < 0.8); + + eval_ir(pAz, T0, impulse_response, ctx->m.pitch_sharp_factor); + + convolute_with_sparse(fixed_vector, &fixed_cb, impulse_response, + SUBFR_SIZE); + + avg_energy = + (0.01 + ff_dot_productf(fixed_vector, fixed_vector, SUBFR_SIZE))/ + SUBFR_SIZE; + + ctx->past_pitch_gain = pitch_gain = gain_cb[params->gc_index[i]][0]; + + gain_code = ff_amr_set_fixed_gain(gain_cb[params->gc_index[i]][1], + avg_energy, ctx->energy_history, + 34 - 15.0/(log2f(10.0) * 0.05), + pred); + + ff_weighted_vector_sumf(excitation, excitation, fixed_vector, + pitch_gain, gain_code, SUBFR_SIZE); + + pitch_gain *= 0.5 * pitch_gain; + pitch_gain = FFMIN(pitch_gain, 0.4); + + ctx->gain_mem = 0.7 * ctx->gain_mem + 0.3 * pitch_gain; + ctx->gain_mem = FFMIN(ctx->gain_mem, pitch_gain); + gain_code *= ctx->gain_mem; + + for (j = 0; j < SUBFR_SIZE; j++) + fixed_vector[j] = excitation[j] - gain_code * fixed_vector[j]; + + if (ctx->mode == MODE_5k0) { + postfilter_5k0(ctx, pAz, fixed_vector); + + ff_celp_lp_synthesis_filterf(ctx->postfilter_syn5k0 + LP_FILTER_ORDER + i*SUBFR_SIZE, + pAz, excitation, SUBFR_SIZE, + LP_FILTER_ORDER); + } + + ff_celp_lp_synthesis_filterf(synth + i*SUBFR_SIZE, pAz, fixed_vector, + SUBFR_SIZE, LP_FILTER_ORDER); + + excitation += SUBFR_SIZE; + } + + memcpy(synth - LP_FILTER_ORDER, synth + frame_size - LP_FILTER_ORDER, + LP_FILTER_ORDER * sizeof(float)); + + if (ctx->mode == MODE_5k0) { + for (i = 0; i < ctx->m.subframe_count; i++) { + float energy = ff_dot_productf(ctx->postfilter_syn5k0 + LP_FILTER_ORDER + i*SUBFR_SIZE, + ctx->postfilter_syn5k0 + LP_FILTER_ORDER + i*SUBFR_SIZE, + SUBFR_SIZE); + ff_adaptative_gain_control(&synth[i * SUBFR_SIZE], energy, + SUBFR_SIZE, 0.9, &ctx->postfilter_agc); + } + + memcpy(ctx->postfilter_syn5k0, ctx->postfilter_syn5k0 + frame_size, + LP_FILTER_ORDER*sizeof(float)); + } + memcpy(ctx->excitation, excitation - PITCH_DELAY_MAX - L_INTERPOL, + (PITCH_DELAY_MAX + L_INTERPOL) * sizeof(float)); + + ff_acelp_apply_order_2_transfer_function(synth, + (const float[2]) {-1.99997 , 1.000000000}, + (const float[2]) {-1.93307352, 0.935891986}, + 0.939805806, + ctx->highpass_filt_mem, + frame_size); + + ctx->dsp.vector_clipf(out_data, synth, -1, 32767./(1<<15), frame_size); + +} + +static av_cold int sipr_decoder_init(AVCodecContext * avctx) +{ + SiprContext *ctx = avctx->priv_data; + int i; + + if (avctx->bit_rate > 12200) ctx->mode = MODE_16k; + else if (avctx->bit_rate > 7500 ) ctx->mode = MODE_8k5; + else if (avctx->bit_rate > 5750 ) ctx->mode = MODE_6k5; + else ctx->mode = MODE_5k0; + + ctx->m = modes[ctx->mode]; + av_log(avctx, AV_LOG_DEBUG, "Mode: %s\n", ctx->m.mode_name); + + for (i = 0; i < LP_FILTER_ORDER; i++) + ctx->lsp_history[i] = cos((i+1) * M_PI / (LP_FILTER_ORDER + 1)); + + for (i = 0; i < 4; i++) + ctx->energy_history[i] = -14; + + avctx->sample_fmt = SAMPLE_FMT_FLT; + + if (ctx->mode == MODE_16k) { + av_log(avctx, AV_LOG_ERROR, "decoding 16kbps SIPR files is not " + "supported yet.\n"); + return -1; + } + + dsputil_init(&ctx->dsp, avctx); + + return 0; +} + +static int sipr_decode_frame(AVCodecContext *avctx, void *datap, + int *data_size, AVPacket *avpkt) +{ + SiprContext *ctx = avctx->priv_data; + const uint8_t *buf=avpkt->data; + SiprParameters parm; + GetBitContext gb; + float *data = datap; + int i; + + ctx->avctx = avctx; + if (avpkt->size < (ctx->m.bits_per_frame >> 3)) { + av_log(avctx, AV_LOG_ERROR, + "Error processing packet: packet size (%d) too small\n", + avpkt->size); + + *data_size = 0; + return -1; + } + if (*data_size < SUBFR_SIZE * ctx->m.subframe_count * sizeof(float)) { + av_log(avctx, AV_LOG_ERROR, + "Error processing packet: output buffer (%d) too small\n", + *data_size); + + *data_size = 0; + return -1; + } + + init_get_bits(&gb, buf, ctx->m.bits_per_frame); + + for (i = 0; i < ctx->m.frames_per_packet; i++) { + decode_parameters(&parm, &gb, &ctx->m); + decode_frame(ctx, &parm, data); + + data += SUBFR_SIZE * ctx->m.subframe_count; + } + + *data_size = ctx->m.frames_per_packet * SUBFR_SIZE * + ctx->m.subframe_count * sizeof(float); + + return ctx->m.bits_per_frame >> 3; +}; + +AVCodec sipr_decoder = { + "sipr", + CODEC_TYPE_AUDIO, + CODEC_ID_SIPR, + sizeof(SiprContext), + sipr_decoder_init, + NULL, + NULL, + sipr_decode_frame, + .long_name = NULL_IF_CONFIG_SMALL("RealAudio SIPR / ACELP.NET"), +};