diff options
Diffstat (limited to 'libavcodec/amrnbdec.c')
| -rw-r--r-- | libavcodec/amrnbdec.c | 54 |
1 files changed, 35 insertions, 19 deletions
diff --git a/libavcodec/amrnbdec.c b/libavcodec/amrnbdec.c index 6b658c0..46e4856 100644 --- a/libavcodec/amrnbdec.c +++ b/libavcodec/amrnbdec.c @@ -136,6 +136,11 @@ 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() */ @@ -171,6 +176,11 @@ 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; } @@ -214,15 +224,16 @@ 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); } @@ -266,7 +277,7 @@ 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); } @@ -329,7 +340,7 @@ 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); @@ -395,7 +406,8 @@ 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); @@ -780,12 +792,12 @@ 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_dot_productf(excitation, excitation, + float energy = p->celpm_ctx.dot_productf(excitation, excitation, AMR_SUBFRAME_SIZE); float pitch_factor = p->pitch_gain[4] * @@ -800,7 +812,8 @@ 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 @@ -846,10 +859,11 @@ 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 @@ -859,11 +873,12 @@ 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_dot_productf(hf, hf, AMR_TILT_RESPONSE); - rh1 = ff_dot_productf(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. @@ -883,7 +898,7 @@ 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_dot_productf(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 @@ -904,16 +919,16 @@ 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, @@ -990,7 +1005,7 @@ static int amrnb_decode_frame(AVCodecContext *avctx, void *data, p->fixed_gain[4] = ff_amr_set_fixed_gain(fixed_gain_factor, - ff_dot_productf(p->fixed_vector, p->fixed_vector, + p->celpm_ctx.dot_productf(p->fixed_vector, p->fixed_vector, AMR_SUBFRAME_SIZE)/AMR_SUBFRAME_SIZE, p->prediction_error, energy_mean[p->cur_frame_mode], energy_pred_fac); @@ -1034,7 +1049,8 @@ 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); @@ -1045,7 +1061,7 @@ 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; |
