diff options
Diffstat (limited to 'libavcodec/amrnbdec.c')
| -rw-r--r-- | libavcodec/amrnbdec.c | 74 |
1 files changed, 47 insertions, 27 deletions
diff --git a/libavcodec/amrnbdec.c b/libavcodec/amrnbdec.c index 5c359a8..40bc253 100644 --- a/libavcodec/amrnbdec.c +++ b/libavcodec/amrnbdec.c @@ -3,20 +3,20 @@ * Copyright (c) 2006-2007 Robert Swain * Copyright (c) 2009 Colin McQuillan * - * This file is part of Libav. + * This file is part of FFmpeg. * - * Libav is free software; you can redistribute it and/or + * 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. * - * Libav is distributed in the hope that it will be useful, + * 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 Libav; if not, write to the Free Software + * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ @@ -47,6 +47,8 @@ #include "avcodec.h" #include "dsputil.h" #include "libavutil/common.h" +#include "libavutil/avassert.h" +#include "celp_math.h" #include "celp_filters.h" #include "acelp_filters.h" #include "acelp_vectors.h" @@ -84,7 +86,7 @@ /** Maximum sharpening factor * * The specification says 0.8, which should be 13107, but the reference C code - * uses 13017 instead. (Amusingly the same applies to SHARP_MAX in bitexact G.729.) + * uses 13017 instead. (Amusingly the same applies to SHARP_MAX in g729dec.c.) */ #define SHARP_MAX 0.79449462890625 @@ -137,6 +139,11 @@ typedef struct AMRContext { float samples_in[LP_FILTER_ORDER + AMR_SUBFRAME_SIZE]; ///< floating point samples + ACELPFContext acelpf_ctx; ///< context for filters for ACELP-based codecs + ACELPVContext acelpv_ctx; ///< context for vector operations for ACELP-based codecs + CELPFContext celpf_ctx; ///< context for filters for CELP-based codecs + CELPMContext celpm_ctx; ///< context for fixed point math operations + } AMRContext; /** Double version of ff_weighted_vector_sumf() */ @@ -163,7 +170,8 @@ static av_cold int amrnb_decode_init(AVCodecContext *avctx) avctx->channels = 1; avctx->channel_layout = AV_CH_LAYOUT_MONO; - avctx->sample_rate = 8000; + if (!avctx->sample_rate) + avctx->sample_rate = 8000; avctx->sample_fmt = AV_SAMPLE_FMT_FLT; // p->excitation always points to the same position in p->excitation_buf @@ -180,6 +188,11 @@ static av_cold int amrnb_decode_init(AVCodecContext *avctx) avcodec_get_frame_defaults(&p->avframe); avctx->coded_frame = &p->avframe; + ff_acelp_filter_init(&p->acelpf_ctx); + ff_acelp_vectors_init(&p->acelpv_ctx); + ff_celp_filter_init(&p->celpf_ctx); + ff_celp_math_init(&p->celpm_ctx); + return 0; } @@ -223,15 +236,16 @@ static enum Mode unpack_bitstream(AMRContext *p, const uint8_t *buf, * Interpolate the LSF vector (used for fixed gain smoothing). * The interpolation is done over all four subframes even in MODE_12k2. * + * @param[in] ctx The Context * @param[in,out] lsf_q LSFs in [0,1] for each subframe * @param[in] lsf_new New LSFs in [0,1] for subframe 4 */ -static void interpolate_lsf(float lsf_q[4][LP_FILTER_ORDER], float *lsf_new) +static void interpolate_lsf(ACELPVContext *ctx, float lsf_q[4][LP_FILTER_ORDER], float *lsf_new) { int i; for (i = 0; i < 4; i++) - ff_weighted_vector_sumf(lsf_q[i], lsf_q[3], lsf_new, + ctx->weighted_vector_sumf(lsf_q[i], lsf_q[3], lsf_new, 0.25 * (3 - i), 0.25 * (i + 1), LP_FILTER_ORDER); } @@ -275,7 +289,7 @@ static void lsf2lsp_for_mode12k2(AMRContext *p, double lsp[LP_FILTER_ORDER], ff_set_min_dist_lsf(lsf_q, MIN_LSF_SPACING, LP_FILTER_ORDER); if (update) - interpolate_lsf(p->lsf_q, lsf_q); + interpolate_lsf(&p->acelpv_ctx, p->lsf_q, lsf_q); ff_acelp_lsf2lspd(lsp, lsf_q, LP_FILTER_ORDER); } @@ -338,7 +352,7 @@ static void lsf2lsp_3(AMRContext *p) ff_set_min_dist_lsf(lsf_q, MIN_LSF_SPACING, LP_FILTER_ORDER); // store data for computing the next frame's LSFs - interpolate_lsf(p->lsf_q, lsf_q); + interpolate_lsf(&p->acelpv_ctx, p->lsf_q, lsf_q); memcpy(p->prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(*lsf_r)); ff_acelp_lsf2lspd(p->lsp[3], lsf_q, LP_FILTER_ORDER); @@ -404,7 +418,8 @@ static void decode_pitch_vector(AMRContext *p, /* Calculate the pitch vector by interpolating the past excitation at the pitch lag using a b60 hamming windowed sinc function. */ - ff_acelp_interpolatef(p->excitation, p->excitation + 1 - pitch_lag_int, + p->acelpf_ctx.acelp_interpolatef(p->excitation, + p->excitation + 1 - pitch_lag_int, ff_b60_sinc, 6, pitch_lag_frac + 6 - 6*(pitch_lag_frac > 0), 10, AMR_SUBFRAME_SIZE); @@ -488,7 +503,7 @@ static void decode_8_pulses_31bits(const int16_t *fixed_index, static void decode_fixed_sparse(AMRFixed *fixed_sparse, const uint16_t *pulses, const enum Mode mode, const int subframe) { - assert(MODE_4k75 <= mode && mode <= MODE_12k2); + av_assert1(MODE_4k75 <= (signed)mode && mode <= MODE_12k2); if (mode == MODE_12k2) { ff_decode_10_pulses_35bits(pulses, fixed_sparse, gray_decode, 5, 3); @@ -789,12 +804,12 @@ static int synthesis(AMRContext *p, float *lpc, for (i = 0; i < AMR_SUBFRAME_SIZE; i++) p->pitch_vector[i] *= 0.25; - ff_weighted_vector_sumf(excitation, p->pitch_vector, fixed_vector, + p->acelpv_ctx.weighted_vector_sumf(excitation, p->pitch_vector, fixed_vector, p->pitch_gain[4], fixed_gain, AMR_SUBFRAME_SIZE); // emphasize pitch vector contribution if (p->pitch_gain[4] > 0.5 && !overflow) { - float energy = ff_scalarproduct_float_c(excitation, excitation, + float energy = p->celpm_ctx.dot_productf(excitation, excitation, AMR_SUBFRAME_SIZE); float pitch_factor = p->pitch_gain[4] * @@ -809,7 +824,8 @@ static int synthesis(AMRContext *p, float *lpc, AMR_SUBFRAME_SIZE); } - ff_celp_lp_synthesis_filterf(samples, lpc, excitation, AMR_SUBFRAME_SIZE, + p->celpf_ctx.celp_lp_synthesis_filterf(samples, lpc, excitation, + AMR_SUBFRAME_SIZE, LP_FILTER_ORDER); // detect overflow @@ -855,10 +871,11 @@ static void update_state(AMRContext *p) /** * Get the tilt factor of a formant filter from its transfer function * + * @param p The Context * @param lpc_n LP_FILTER_ORDER coefficients of the numerator * @param lpc_d LP_FILTER_ORDER coefficients of the denominator */ -static float tilt_factor(float *lpc_n, float *lpc_d) +static float tilt_factor(AMRContext *p, float *lpc_n, float *lpc_d) { float rh0, rh1; // autocorrelation at lag 0 and 1 @@ -868,11 +885,12 @@ static float tilt_factor(float *lpc_n, float *lpc_d) hf[0] = 1.0; memcpy(hf + 1, lpc_n, sizeof(float) * LP_FILTER_ORDER); - ff_celp_lp_synthesis_filterf(hf, lpc_d, hf, AMR_TILT_RESPONSE, + p->celpf_ctx.celp_lp_synthesis_filterf(hf, lpc_d, hf, + AMR_TILT_RESPONSE, LP_FILTER_ORDER); - rh0 = ff_scalarproduct_float_c(hf, hf, AMR_TILT_RESPONSE); - rh1 = ff_scalarproduct_float_c(hf, hf + 1, AMR_TILT_RESPONSE - 1); + rh0 = p->celpm_ctx.dot_productf(hf, hf, AMR_TILT_RESPONSE); + rh1 = p->celpm_ctx.dot_productf(hf, hf + 1, AMR_TILT_RESPONSE - 1); // The spec only specifies this check for 12.2 and 10.2 kbit/s // modes. But in the ref source the tilt is always non-negative. @@ -892,7 +910,7 @@ static void postfilter(AMRContext *p, float *lpc, float *buf_out) int i; float *samples = p->samples_in + LP_FILTER_ORDER; // Start of input - float speech_gain = ff_scalarproduct_float_c(samples, samples, + float speech_gain = p->celpm_ctx.dot_productf(samples, samples, AMR_SUBFRAME_SIZE); float pole_out[AMR_SUBFRAME_SIZE + LP_FILTER_ORDER]; // Output of pole filter @@ -913,16 +931,16 @@ static void postfilter(AMRContext *p, float *lpc, float *buf_out) } memcpy(pole_out, p->postfilter_mem, sizeof(float) * LP_FILTER_ORDER); - ff_celp_lp_synthesis_filterf(pole_out + LP_FILTER_ORDER, lpc_d, samples, + p->celpf_ctx.celp_lp_synthesis_filterf(pole_out + LP_FILTER_ORDER, lpc_d, samples, AMR_SUBFRAME_SIZE, LP_FILTER_ORDER); memcpy(p->postfilter_mem, pole_out + AMR_SUBFRAME_SIZE, sizeof(float) * LP_FILTER_ORDER); - ff_celp_lp_zero_synthesis_filterf(buf_out, lpc_n, + p->celpf_ctx.celp_lp_zero_synthesis_filterf(buf_out, lpc_n, pole_out + LP_FILTER_ORDER, AMR_SUBFRAME_SIZE, LP_FILTER_ORDER); - ff_tilt_compensation(&p->tilt_mem, tilt_factor(lpc_n, lpc_d), buf_out, + ff_tilt_compensation(&p->tilt_mem, tilt_factor(p, lpc_n, lpc_d), buf_out, AMR_SUBFRAME_SIZE); ff_adaptive_gain_control(buf_out, buf_out, speech_gain, AMR_SUBFRAME_SIZE, @@ -960,7 +978,8 @@ static int amrnb_decode_frame(AVCodecContext *avctx, void *data, return AVERROR_INVALIDDATA; } if (p->cur_frame_mode == MODE_DTX) { - av_log_missing_feature(avctx, "dtx mode", 1); + av_log_missing_feature(avctx, "dtx mode", 0); + av_log(avctx, AV_LOG_INFO, "Note: libopencore_amrnb supports dtx\n"); return AVERROR_PATCHWELCOME; } @@ -998,7 +1017,7 @@ static int amrnb_decode_frame(AVCodecContext *avctx, void *data, p->fixed_gain[4] = ff_amr_set_fixed_gain(fixed_gain_factor, - ff_scalarproduct_float_c(p->fixed_vector, + p->celpm_ctx.dot_productf(p->fixed_vector, p->fixed_vector, AMR_SUBFRAME_SIZE) / AMR_SUBFRAME_SIZE, @@ -1044,7 +1063,8 @@ static int amrnb_decode_frame(AVCodecContext *avctx, void *data, update_state(p); } - ff_acelp_apply_order_2_transfer_function(buf_out, buf_out, highpass_zeros, + p->acelpf_ctx.acelp_apply_order_2_transfer_function(buf_out, + buf_out, highpass_zeros, highpass_poles, highpass_gain * AMR_SAMPLE_SCALE, p->high_pass_mem, AMR_BLOCK_SIZE); @@ -1055,7 +1075,7 @@ static int amrnb_decode_frame(AVCodecContext *avctx, void *data, * for fixed_gain_smooth. * The specification has an incorrect formula: the reference decoder uses * qbar(n-1) rather than qbar(n) in section 6.1(4) equation 71. */ - ff_weighted_vector_sumf(p->lsf_avg, p->lsf_avg, p->lsf_q[3], + p->acelpv_ctx.weighted_vector_sumf(p->lsf_avg, p->lsf_avg, p->lsf_q[3], 0.84, 0.16, LP_FILTER_ORDER); *got_frame_ptr = 1; |
