#include "avcodec.h"
#include "libavutil/common.h"
+#include "libavutil/avassert.h"
#include "celp_math.h"
#include "celp_filters.h"
#include "acelp_filters.h"
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() */
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;
}
* 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);
}
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);
}
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);
/* 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);
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 <= mode && mode <= MODE_12k2);
if (mode == MODE_12k2) {
ff_decode_10_pulses_35bits(pulses, fixed_sparse, gray_decode, 5, 3);
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] *
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
/**
* 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
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.
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
}
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,
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);
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);
* 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;
projects / ffmpeg / blobdiff