* AMR wideband decoder
*/
+#include "libavutil/common.h"
#include "libavutil/lfg.h"
#include "avcodec.h"
-#include "get_bits.h"
+#include "dsputil.h"
#include "lsp.h"
-#include "celp_math.h"
#include "celp_filters.h"
#include "acelp_filters.h"
#include "acelp_vectors.h"
*/
static int decode_mime_header(AMRWBContext *ctx, const uint8_t *buf)
{
- GetBitContext gb;
- init_get_bits(&gb, buf, 8);
-
/* Decode frame header (1st octet) */
- skip_bits(&gb, 1); // padding bit
- ctx->fr_cur_mode = get_bits(&gb, 4);
- ctx->fr_quality = get_bits1(&gb);
- skip_bits(&gb, 2); // padding bits
+ ctx->fr_cur_mode = buf[0] >> 3 & 0x0F;
+ ctx->fr_quality = (buf[0] & 0x4) == 0x4;
return 1;
}
static float voice_factor(float *p_vector, float p_gain,
float *f_vector, float f_gain)
{
- double p_ener = (double) ff_dot_productf(p_vector, p_vector,
- AMRWB_SFR_SIZE) * p_gain * p_gain;
- double f_ener = (double) ff_dot_productf(f_vector, f_vector,
- AMRWB_SFR_SIZE) * f_gain * f_gain;
+ double p_ener = (double) ff_scalarproduct_float_c(p_vector, p_vector,
+ AMRWB_SFR_SIZE) *
+ p_gain * p_gain;
+ double f_ener = (double) ff_scalarproduct_float_c(f_vector, f_vector,
+ AMRWB_SFR_SIZE) *
+ f_gain * f_gain;
return (p_ener - f_ener) / (p_ener + f_ener);
}
/* emphasize pitch vector contribution in low bitrate modes */
if (ctx->pitch_gain[0] > 0.5 && ctx->fr_cur_mode <= MODE_8k85) {
int i;
- float energy = ff_dot_productf(excitation, excitation,
- AMRWB_SFR_SIZE);
+ float energy = ff_scalarproduct_float_c(excitation, excitation,
+ AMRWB_SFR_SIZE);
// XXX: Weird part in both ref code and spec. A unknown parameter
// {beta} seems to be identical to the current pitch gain
i++;
for (k = 1; k < 5; k++) {
- out[i] = ff_dot_productf(in0 + int_part, upsample_fir[4 - frac_part],
- UPS_MEM_SIZE);
+ out[i] = ff_scalarproduct_float_c(in0 + int_part,
+ upsample_fir[4 - frac_part],
+ UPS_MEM_SIZE);
int_part++;
frac_part--;
i++;
if (ctx->fr_cur_mode == MODE_23k85)
return qua_hb_gain[hb_idx] * (1.0f / (1 << 14));
- tilt = ff_dot_productf(synth, synth + 1, AMRWB_SFR_SIZE - 1) /
- ff_dot_productf(synth, synth, AMRWB_SFR_SIZE);
+ tilt = ff_scalarproduct_float_c(synth, synth + 1, AMRWB_SFR_SIZE - 1) /
+ ff_scalarproduct_float_c(synth, synth, AMRWB_SFR_SIZE);
/* return gain bounded by [0.1, 1.0] */
return av_clipf((1.0 - FFMAX(0.0, tilt)) * (1.25 - 0.25 * wsp), 0.1, 1.0);
const float *synth_exc, float hb_gain)
{
int i;
- float energy = ff_dot_productf(synth_exc, synth_exc, AMRWB_SFR_SIZE);
+ float energy = ff_scalarproduct_float_c(synth_exc, synth_exc, AMRWB_SFR_SIZE);
/* Generate a white-noise excitation */
for (i = 0; i < AMRWB_SFR_SIZE_16k; i++)
* Extrapolate a ISF vector to the 16kHz range (20th order LP)
* used at mode 6k60 LP filter for the high frequency band.
*
- * @param[out] out Buffer for extrapolated isf
- * @param[in] isf Input isf vector
+ * @param[out] isf Buffer for extrapolated isf; contains LP_ORDER
+ * values on input
*/
-static void extrapolate_isf(float out[LP_ORDER_16k], float isf[LP_ORDER])
+static void extrapolate_isf(float isf[LP_ORDER_16k])
{
float diff_isf[LP_ORDER - 2], diff_mean;
float *diff_hi = diff_isf - LP_ORDER + 1; // diff array for extrapolated indexes
float est, scale;
int i, i_max_corr;
- memcpy(out, isf, (LP_ORDER - 1) * sizeof(float));
- out[LP_ORDER_16k - 1] = isf[LP_ORDER - 1];
+ isf[LP_ORDER_16k - 1] = isf[LP_ORDER - 1];
/* Calculate the difference vector */
for (i = 0; i < LP_ORDER - 2; i++)
i_max_corr++;
for (i = LP_ORDER - 1; i < LP_ORDER_16k - 1; i++)
- out[i] = isf[i - 1] + isf[i - 1 - i_max_corr]
+ isf[i] = isf[i - 1] + isf[i - 1 - i_max_corr]
- isf[i - 2 - i_max_corr];
/* Calculate an estimate for ISF(18) and scale ISF based on the error */
- est = 7965 + (out[2] - out[3] - out[4]) / 6.0;
- scale = 0.5 * (FFMIN(est, 7600) - out[LP_ORDER - 2]) /
- (out[LP_ORDER_16k - 2] - out[LP_ORDER - 2]);
+ est = 7965 + (isf[2] - isf[3] - isf[4]) / 6.0;
+ scale = 0.5 * (FFMIN(est, 7600) - isf[LP_ORDER - 2]) /
+ (isf[LP_ORDER_16k - 2] - isf[LP_ORDER - 2]);
for (i = LP_ORDER - 1; i < LP_ORDER_16k - 1; i++)
- diff_hi[i] = scale * (out[i] - out[i - 1]);
+ diff_hi[i] = scale * (isf[i] - isf[i - 1]);
/* Stability insurance */
for (i = LP_ORDER; i < LP_ORDER_16k - 1; i++)
}
for (i = LP_ORDER - 1; i < LP_ORDER_16k - 1; i++)
- out[i] = out[i - 1] + diff_hi[i] * (1.0f / (1 << 15));
+ isf[i] = isf[i - 1] + diff_hi[i] * (1.0f / (1 << 15));
/* Scale the ISF vector for 16000 Hz */
for (i = 0; i < LP_ORDER_16k - 1; i++)
- out[i] *= 0.8;
+ isf[i] *= 0.8;
}
/**
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);
+ extrapolate_isf(e_isf);
e_isf[LP_ORDER_16k - 1] *= 2.0;
ff_acelp_lsf2lspd(e_isp, e_isf, LP_ORDER_16k);
buf_out = (float *)ctx->avframe.data[0];
header_size = decode_mime_header(ctx, buf);
+ if (ctx->fr_cur_mode > MODE_SID) {
+ av_log(avctx, AV_LOG_ERROR,
+ "Invalid mode %d\n", ctx->fr_cur_mode);
+ return AVERROR_INVALIDDATA;
+ }
expected_fr_size = ((cf_sizes_wb[ctx->fr_cur_mode] + 7) >> 3) + 1;
if (buf_size < expected_fr_size) {
av_log(avctx, AV_LOG_ERROR,
"Frame too small (%d bytes). Truncated file?\n", buf_size);
*got_frame_ptr = 0;
- return buf_size;
+ return AVERROR_INVALIDDATA;
}
if (!ctx->fr_quality || ctx->fr_cur_mode > MODE_SID)
av_log(avctx, AV_LOG_ERROR, "Encountered a bad or corrupted frame\n");
- if (ctx->fr_cur_mode == MODE_SID) /* Comfort noise frame */
+ if (ctx->fr_cur_mode == MODE_SID) { /* Comfort noise frame */
av_log_missing_feature(avctx, "SID mode", 1);
-
- if (ctx->fr_cur_mode >= MODE_SID)
- return -1;
+ return AVERROR_PATCHWELCOME;
+ }
ff_amr_bit_reorder((uint16_t *) &ctx->frame, sizeof(AMRWBFrame),
buf + header_size, amr_bit_orderings_by_mode[ctx->fr_cur_mode]);
ctx->fixed_gain[0] =
ff_amr_set_fixed_gain(fixed_gain_factor,
- ff_dot_productf(ctx->fixed_vector, ctx->fixed_vector,
- AMRWB_SFR_SIZE) / AMRWB_SFR_SIZE,
+ ff_scalarproduct_float_c(ctx->fixed_vector,
+ ctx->fixed_vector,
+ AMRWB_SFR_SIZE) /
+ AMRWB_SFR_SIZE,
ctx->prediction_error,
ENERGY_MEAN, energy_pred_fac);
AVCodec ff_amrwb_decoder = {
.name = "amrwb",
.type = AVMEDIA_TYPE_AUDIO,
- .id = CODEC_ID_AMR_WB,
+ .id = AV_CODEC_ID_AMR_WB,
.priv_data_size = sizeof(AMRWBContext),
.init = amrwb_decode_init,
.decode = amrwb_decode_frame,
.capabilities = CODEC_CAP_DR1,
- .long_name = NULL_IF_CONFIG_SMALL("Adaptive Multi-Rate WideBand"),
- .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE},
+ .long_name = NULL_IF_CONFIG_SMALL("AMR-WB (Adaptive Multi-Rate WideBand)"),
+ .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT,
+ AV_SAMPLE_FMT_NONE },
};
projects / ffmpeg / blobdiff