This is an automated email from the git hooks/post-receive script. It was generated because a ref change was pushed to the repository containing the project "AMR-WB decoder". The branch, master has been updated via 01100290fe3b220e4fbbe6ec16f7f85652d25092 (commit) from 1c399e4d298648f1ad8a70f50507c41fa7b21a62 (commit) Those revisions listed above that are new to this repository have not appeared on any other notification email; so we list those revisions in full, below. - Log ----------------------------------------------------------------- commit 01100290fe3b220e4fbbe6ec16f7f85652d25092 Author: Marcelo Povoa <marspeoplester@gmail.com> Date: Thu Jul 29 17:41:24 2010 -0300 Write the high-band synthesis process for all modes diff --git a/libavcodec/amrwbdata.h b/libavcodec/amrwbdata.h index ce5fb63..76a1a60 100644 --- a/libavcodec/amrwbdata.h +++ b/libavcodec/amrwbdata.h @@ -1621,6 +1621,9 @@ static const int16_t isf_init[LP_ORDER] = { 9216, 10240, 11264, 12288, 13312, 14336, 15360, 3840 }; +/* ISF/ISP interpolation coefficients for each subframe */ +static const float isfp_inter[4] = { 0.45, 0.8, 0.96, 1.0 }; + /* Coefficients for FIR interpolation of excitation vector * at pitch lag resulting the adaptive codebook vector */ static const float ac_inter[65] = { diff --git a/libavcodec/amrwbdec.c b/libavcodec/amrwbdec.c index 6833f23..7f6152f 100644 --- a/libavcodec/amrwbdec.c +++ b/libavcodec/amrwbdec.c @@ -74,6 +74,7 @@ typedef struct { float samples_az[LP_ORDER + AMRWB_SUBFRAME_SIZE]; ///< lower band samples from synthesis at 12.8kHz float samples_up[UPS_MEM_SIZE + AMRWB_SUBFRAME_SIZE]; ///< lower band samples processed for upsampling at 12.8kHz + float samples_hb[LP_ORDER_16k + AMRWB_SFR_SIZE_OUT]; ///< higher band samples from synthesis at 16kHz float hpf_31_mem[4], hpf_400_mem[4]; ///< previous values in the high-pass filters float demph_mem[1]; ///< previous value in the de-emphasis filter @@ -298,17 +299,14 @@ static void isf_set_min_dist(float *isf, float min_spacing, int size) { */ static void interpolate_isp(double isp_q[4][LP_ORDER], const double *isp4_past) { - int i; + int i, k; // XXX: Did not use ff_weighted_vector_sumf because using double - for (i = 0; i < LP_ORDER; i++) - isp_q[0][i] = 0.55 * isp4_past[i] + 0.45 * isp_q[3][i]; - - for (i = 0; i < LP_ORDER; i++) - isp_q[1][i] = 0.20 * isp4_past[i] + 0.80 * isp_q[3][i]; - - for (i = 0; i < LP_ORDER; i++) - isp_q[2][i] = 0.04 * isp4_past[i] + 0.96 * isp_q[3][i]; + 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]; + } } /** @@ -880,7 +878,7 @@ static void pitch_enhancer(float *fixed_vector, float voice_fac) * @param excitation [out] buffer for synthesis final excitation * @param fixed_gain [in] fixed codebook gain for synthesis * @param fixed_vector [in] algebraic codebook vector - * @param samples [out] pointer to the output speech samples + * @param samples [in/out] pointer to the output speech samples */ static void synthesis(AMRWBContext *ctx, float *lpc, float *excitation, float fixed_gain, const float *fixed_vector, @@ -985,8 +983,6 @@ static void upsample_5_4(float *out, const float *in, int o_size) } else out[i] = ff_dot_productf(in0 + int_part, upsample_fir[4 - frac_part], UPS_MEM_SIZE); - - out[i] *= 2.0 / 32768.0; // upscale output } } @@ -1064,7 +1060,7 @@ static float auto_correlation(float *diff_isf, float mean, int lag) * @param out [out] buffer for extrapolated isf * @param isf [in] input isf vector */ -static void extrapolate_isf(float *out, float *isf) +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 indices @@ -1124,6 +1120,63 @@ static void extrapolate_isf(float *out, float *isf) } /** + * Spectral expand the LP coefficients using the equation: + * y[i] = x[i] * (gamma ** i) + * + * @param out [out] output buffer (may use input array) + * @param lpc [in] LP coefficients array + * @param gamma [in] weighting factor + * @param size [in] LP array size + */ +static void lpc_weighting(float *out, const float *lpc, float gamma, int size) +{ + int i; + float fac = 1.0; + + 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 ctx [in] the context + * @param subframe [in] current subframe index (0 to 3) + * @param samples [in/out] pointer to the output speech samples + * @param exc [in] generated white-noise scaled excitation + * @param isf [in] current frame isf vector + * @param isf_past [in] 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 + 1]; + 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); + isf2isp(e_isf, e_isp); + isp2lp(e_isp, hb_lpc, LP_ORDER_16k / 2); + + lpc_weighting(hb_lpc, hb_lpc, 0.9, LP_ORDER_16k + 1); + } else { + lpc_weighting(hb_lpc, ctx->lp_coef[subframe], 0.6, LP_ORDER + 1); + } + + ff_celp_lp_synthesis_filterf(samples, hb_lpc + 1, exc, AMRWB_SFR_SIZE_OUT, + (mode == MODE_6k60) ? LP_ORDER_16k : LP_ORDER); +} + +/** * Update context state before the next subframe */ static void update_sub_state(AMRWBContext *ctx) @@ -1138,6 +1191,8 @@ static void update_sub_state(AMRWBContext *ctx) LP_ORDER * sizeof(float)); memmove(&ctx->samples_up[0], &ctx->samples_up[AMRWB_SUBFRAME_SIZE], UPS_MEM_SIZE * sizeof(float)); + memmove(&ctx->samples_hb[0], &ctx->samples_hb[AMRWB_SUBFRAME_SIZE], + LP_ORDER_16k * sizeof(float)); memset(ctx->fixed_vector, 0, AMRWB_SUBFRAME_SIZE * sizeof(float)); } @@ -1193,10 +1248,11 @@ static int amrwb_decode_frame(AVCodecContext *avctx, void *data, int *data_size, interpolate_isp(ctx->isp, ctx->isp_sub4_past); for (sub = 0; sub < 4; sub++) - isp2lp(ctx->isp[sub], ctx->lp_coef[sub], LP_ORDER/2); + isp2lp(ctx->isp[sub], ctx->lp_coef[sub], LP_ORDER / 2); for (sub = 0; sub < 4; sub++) { const AMRWBSubFrame *cur_subframe = &cf->subframe[sub]; + float *sub_buf = buf_out + sub * AMRWB_SFR_SIZE_OUT; /* Decode adaptive codebook */ decode_pitch_vector(ctx, cur_subframe, sub); @@ -1238,14 +1294,14 @@ static int amrwb_decode_frame(AVCodecContext *avctx, void *data, int *data_size, synth_fixed_vector = anti_sparseness(ctx, ctx->fixed_vector, spare_vector); - /* XXX: Tested against the ref code until here, it "succeeds" at least - * for cases in which the "opencore bug" don't interfere */ - 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]); + /* XXX: Tested against the ref code until here, it "succeeds" at least + * for cases in which the "opencore bug" don't interfere */ + /* Synthesis speech post-processing */ de_emphasis(&ctx->samples_up[UPS_MEM_SIZE], &ctx->samples_az[LP_ORDER], PREEMPH_FAC, ctx->demph_mem); @@ -1253,8 +1309,8 @@ static int amrwb_decode_frame(AVCodecContext *avctx, void *data, int *data_size, high_pass_filter(&ctx->samples_up[UPS_MEM_SIZE], hpf_31_coef, ctx->hpf_31_mem, &ctx->samples_up[UPS_MEM_SIZE]); - upsample_5_4(buf_out + sub * AMRWB_SFR_SIZE_OUT, - &ctx->samples_up[UPS_FIR_SIZE], AMRWB_SFR_SIZE_OUT); + upsample_5_4(sub_buf, &ctx->samples_up[UPS_FIR_SIZE], + AMRWB_SFR_SIZE_OUT); /* High frequency band generation */ high_pass_filter(&ctx->samples_up[UPS_MEM_SIZE], hpf_400_coef, @@ -1265,6 +1321,18 @@ static int amrwb_decode_frame(AVCodecContext *avctx, void *data, int *data_size, 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 */ + + /* Add low frequency and high frequency bands */ + for (i = 0; i < AMRWB_SFR_SIZE_OUT; i++) { + // XXX: the lower band should really be upscaled by 2.0? + sub_buf[i] = (sub_buf[i] * 2.0 + + ctx->samples_hb[i + LP_ORDER_16k]) / 32768.0; + } + /* Update buffers and history */ update_sub_state(ctx); } ----------------------------------------------------------------------- Summary of changes: libavcodec/amrwbdata.h | 3 + libavcodec/amrwbdec.c | 106 +++++++++++++++++++++++++++++++++++++++--------- 2 files changed, 90 insertions(+), 19 deletions(-) hooks/post-receive -- AMR-WB decoder