X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Fsipr.c;h=c9fccb2d3add8833c1c7d08a76ab36f484f449d8;hb=c2d3f561072132044114588a5f56b8e1974a2af7;hp=45bd71be2ccb0cf8be685378c105d21d798d5664;hpb=1c3c129b8fd0101a14d89db3a613c357ab8979bf;p=ffmpeg diff --git a/libavcodec/sipr.c b/libavcodec/sipr.c index 45bd71be2cc..c9fccb2d3ad 100644 --- a/libavcodec/sipr.c +++ b/libavcodec/sipr.c @@ -4,26 +4,28 @@ * Copyright (c) 2008 Vladimir Voroshilov * Copyright (c) 2009 Vitor Sessak * - * This file is part of FFmpeg. + * This file is part of Libav. * - * FFmpeg is free software; you can redistribute it and/or + * Libav 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, + * Libav 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 + * License along with Libav; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ #include #include +#include +#include "libavutil/mathematics.h" #include "avcodec.h" #define ALT_BITSTREAM_READER_LE #include "get_bits.h" @@ -36,28 +38,11 @@ #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 - #define MAX_SUBFRAME_COUNT 5 +#include "sipr.h" #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; @@ -67,6 +52,7 @@ typedef struct { /* bitstream parameters */ uint8_t number_of_fc_indexes; + uint8_t ma_predictor_bits; ///< size in bits of the switched MA predictor /** size in bits of the i-th stage vector of quantizer */ uint8_t vq_indexes_bits[5]; @@ -80,6 +66,22 @@ typedef struct { } SiprModeParam; static const SiprModeParam modes[MODE_COUNT] = { + [MODE_16k] = { + .mode_name = "16k", + .bits_per_frame = 160, + .subframe_count = SUBFRAME_COUNT_16k, + .frames_per_packet = 1, + .pitch_sharp_factor = 0.00, + + .number_of_fc_indexes = 10, + .ma_predictor_bits = 1, + .vq_indexes_bits = {7, 8, 7, 7, 7}, + .pitch_delay_bits = {9, 6}, + .gp_index_bits = 4, + .fc_index_bits = {4, 5, 4, 5, 4, 5, 4, 5, 4, 5}, + .gc_index_bits = 5 + }, + [MODE_8k5] = { .mode_name = "8k5", .bits_per_frame = 152, @@ -88,6 +90,7 @@ static const SiprModeParam modes[MODE_COUNT] = { .pitch_sharp_factor = 0.8, .number_of_fc_indexes = 3, + .ma_predictor_bits = 0, .vq_indexes_bits = {6, 7, 7, 7, 5}, .pitch_delay_bits = {8, 5, 5}, .gp_index_bits = 0, @@ -103,6 +106,7 @@ static const SiprModeParam modes[MODE_COUNT] = { .pitch_sharp_factor = 0.8, .number_of_fc_indexes = 3, + .ma_predictor_bits = 0, .vq_indexes_bits = {6, 7, 7, 7, 5}, .pitch_delay_bits = {8, 5, 5}, .gp_index_bits = 0, @@ -118,6 +122,7 @@ static const SiprModeParam modes[MODE_COUNT] = { .pitch_sharp_factor = 0.85, .number_of_fc_indexes = 1, + .ma_predictor_bits = 0, .vq_indexes_bits = {6, 7, 7, 7, 5}, .pitch_delay_bits = {8, 5, 8, 5, 5}, .gp_index_bits = 0, @@ -126,40 +131,12 @@ static const SiprModeParam modes[MODE_COUNT] = { } }; -typedef struct { - AVCodecContext *avctx; - DSPContext dsp; - - 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; - +const float ff_pow_0_5[] = { + 1.0/(1 << 1), 1.0/(1 << 2), 1.0/(1 << 3), 1.0/(1 << 4), + 1.0/(1 << 5), 1.0/(1 << 6), 1.0/(1 << 7), 1.0/(1 << 8), + 1.0/(1 << 9), 1.0/(1 << 10), 1.0/(1 << 11), 1.0/(1 << 12), + 1.0/(1 << 13), 1.0/(1 << 14), 1.0/(1 << 15), 1.0/(1 << 16) +}; static void dequant(float *out, const int *idx, const float *cbs[]) { @@ -208,7 +185,7 @@ static void pitch_sharpening(int pitch_lag_int, float beta, } /** - * Extracts decoding parameters from the input bitstream. + * Extract decoding parameters from the input bitstream. * @param parms parameters structure * @param pgb pointer to initialized GetBitContext structure */ @@ -217,6 +194,8 @@ static void decode_parameters(SiprParameters* parms, GetBitContext *pgb, { int i, j; + parms->ma_pred_switch = get_bits(pgb, p->ma_predictor_bits); + for (i = 0; i < 5; i++) parms->vq_indexes[i] = get_bits(pgb, p->vq_indexes_bits[i]); @@ -231,32 +210,6 @@ static void decode_parameters(SiprParameters* parms, GetBitContext *pgb, } } -static void lsp2lpc_sipr(const double *lsp, float *Az) -{ - int lp_half_order = LP_FILTER_ORDER >> 1; - double buf[(LP_FILTER_ORDER >> 1) + 1]; - double pa[(LP_FILTER_ORDER >> 1) + 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) { @@ -269,14 +222,14 @@ static void sipr_decode_lp(float *lsfnew, const float *lsfold, float *Az, for (j = 0; j < LP_FILTER_ORDER; j++) lsfint[j] = lsfold[j] * (1 - t) + t * lsfnew[j]; - lsp2lpc_sipr(lsfint, Az); + ff_amrwb_lsp2lpc(lsfint, Az, LP_FILTER_ORDER); Az += LP_FILTER_ORDER; t += t0; } } /** - * Evaluates the adaptative impulse response. + * Evaluate the adaptive impulse response. */ static void eval_ir(const float *Az, int pitch_lag, float *freq, float pitch_sharp_factor) @@ -298,7 +251,7 @@ static void eval_ir(const float *Az, int pitch_lag, float *freq, } /** - * Evaluates the convolution of a vector with a sparse vector. + * Evaluate the convolution of a vector with a sparse vector. */ static void convolute_with_sparse(float *out, const AMRFixed *pulses, const float *shape, int length) @@ -324,7 +277,7 @@ static void postfilter_5k0(SiprContext *ctx, const float *lpc, float *samples) 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]; + lpc_n[i] = lpc[i] * ff_pow_0_5 [i]; }; memcpy(pole_out - LP_FILTER_ORDER, ctx->postfilter_mem, @@ -463,7 +416,7 @@ static void decode_frame(SiprContext *ctx, SiprParameters *params, 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), + 34 - 15.0/(0.05*M_LN10/M_LN2), pred); ff_weighted_vector_sumf(excitation, excitation, fixed_vector, @@ -501,25 +454,23 @@ static void decode_frame(SiprContext *ctx, SiprParameters *params, 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); + ff_adaptive_gain_control(&synth[i * SUBFR_SIZE], + &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, + memmove(ctx->excitation, excitation - PITCH_DELAY_MAX - L_INTERPOL, (PITCH_DELAY_MAX + L_INTERPOL) * sizeof(float)); - ff_acelp_apply_order_2_transfer_function(synth, + ff_acelp_apply_order_2_transfer_function(out_data, 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) @@ -534,21 +485,16 @@ static av_cold int sipr_decoder_init(AVCodecContext * avctx) av_log(avctx, AV_LOG_DEBUG, "Mode: %s\n", modes[ctx->mode].mode_name); + if (ctx->mode == MODE_16k) + ff_sipr_init_16k(ctx); + 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); + avctx->sample_fmt = AV_SAMPLE_FMT_FLT; return 0; } @@ -562,6 +508,7 @@ static int sipr_decode_frame(AVCodecContext *avctx, void *datap, const SiprModeParam *mode_par = &modes[ctx->mode]; GetBitContext gb; float *data = datap; + int subframe_size = ctx->mode == MODE_16k ? L_SUBFR_16k : SUBFR_SIZE; int i; ctx->avctx = avctx; @@ -573,7 +520,7 @@ static int sipr_decode_frame(AVCodecContext *avctx, void *datap, *data_size = 0; return -1; } - if (*data_size < SUBFR_SIZE * mode_par->subframe_count * sizeof(float)) { + if (*data_size < subframe_size * mode_par->subframe_count * sizeof(float)) { av_log(avctx, AV_LOG_ERROR, "Error processing packet: output buffer (%d) too small\n", *data_size); @@ -586,25 +533,27 @@ static int sipr_decode_frame(AVCodecContext *avctx, void *datap, for (i = 0; i < mode_par->frames_per_packet; i++) { decode_parameters(&parm, &gb, mode_par); - decode_frame(ctx, &parm, data); - data += SUBFR_SIZE * mode_par->subframe_count; + if (ctx->mode == MODE_16k) + ff_sipr_decode_frame_16k(ctx, &parm, data); + else + decode_frame(ctx, &parm, data); + + data += subframe_size * mode_par->subframe_count; } - *data_size = mode_par->frames_per_packet * SUBFR_SIZE * + *data_size = mode_par->frames_per_packet * subframe_size * mode_par->subframe_count * sizeof(float); return mode_par->bits_per_frame >> 3; -}; +} -AVCodec sipr_decoder = { - "sipr", - CODEC_TYPE_AUDIO, - CODEC_ID_SIPR, - sizeof(SiprContext), - sipr_decoder_init, - NULL, - NULL, - sipr_decode_frame, +AVCodec ff_sipr_decoder = { + .name = "sipr", + .type = AVMEDIA_TYPE_AUDIO, + .id = CODEC_ID_SIPR, + .priv_data_size = sizeof(SiprContext), + .init = sipr_decoder_init, + .decode = sipr_decode_frame, .long_name = NULL_IF_CONFIG_SMALL("RealAudio SIPR / ACELP.NET"), };