]> git.sesse.net Git - ffmpeg/blobdiff - libavcodec/amrwbdec.c
cngdec: Make the dbov variable have the right unit
[ffmpeg] / libavcodec / amrwbdec.c
index b9ae9ece6676c2e58b630f2439f37b362485d484..c9c793fb4a2db40f1678a20176f205a7e62e9755 100644 (file)
  * 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"
@@ -120,14 +120,9 @@ static av_cold int amrwb_decode_init(AVCodecContext *avctx)
  */
 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;
 }
@@ -590,10 +585,12 @@ static void pitch_sharpening(AMRWBContext *ctx, float *fixed_vector)
 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);
 }
@@ -761,8 +758,8 @@ static void synthesis(AMRWBContext *ctx, float *lpc, float *excitation,
     /* 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
@@ -821,8 +818,9 @@ static void upsample_5_4(float *out, const float *in, int o_size)
         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++;
@@ -848,8 +846,8 @@ static float find_hb_gain(AMRWBContext *ctx, const float *synth,
     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);
@@ -868,7 +866,7 @@ static void scaled_hb_excitation(AMRWBContext *ctx, float *hb_exc,
                                  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++)
@@ -904,10 +902,9 @@ static float auto_correlation(float *diff_isf, float mean, int lag)
 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 corr_lag[3];
     float est, scale;
-    int i, i_max_corr;
+    int i, j, i_max_corr;
 
     isf[LP_ORDER_16k - 1] = isf[LP_ORDER - 1];
 
@@ -938,20 +935,20 @@ static void extrapolate_isf(float isf[LP_ORDER_16k])
     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 * (isf[i] - isf[i - 1]);
+    for (i = LP_ORDER - 1, j = 0; i < LP_ORDER_16k - 1; i++, j++)
+        diff_isf[j] = scale * (isf[i] - isf[i - 1]);
 
     /* Stability insurance */
-    for (i = LP_ORDER; i < LP_ORDER_16k - 1; i++)
-        if (diff_hi[i] + diff_hi[i - 1] < 5.0) {
-            if (diff_hi[i] > diff_hi[i - 1]) {
-                diff_hi[i - 1] = 5.0 - diff_hi[i];
+    for (i = 1; i < LP_ORDER_16k - LP_ORDER; i++)
+        if (diff_isf[i] + diff_isf[i - 1] < 5.0) {
+            if (diff_isf[i] > diff_isf[i - 1]) {
+                diff_isf[i - 1] = 5.0 - diff_isf[i];
             } else
-                diff_hi[i] = 5.0 - diff_hi[i - 1];
+                diff_isf[i] = 5.0 - diff_isf[i - 1];
         }
 
-    for (i = LP_ORDER - 1; i < LP_ORDER_16k - 1; i++)
-        isf[i] = isf[i - 1] + diff_hi[i] * (1.0f / (1 << 15));
+    for (i = LP_ORDER - 1, j = 0; i < LP_ORDER_16k - 1; i++, j++)
+        isf[i] = isf[i - 1] + diff_isf[j] * (1.0f / (1 << 15));
 
     /* Scale the ISF vector for 16000 Hz */
     for (i = 0; i < LP_ORDER_16k - 1; i++)
@@ -1113,7 +1110,7 @@ static int amrwb_decode_frame(AVCodecContext *avctx, void *data,
 
     if (ctx->fr_cur_mode == MODE_SID) { /* Comfort noise frame */
         av_log_missing_feature(avctx, "SID mode", 1);
-        return -1;
+        return AVERROR_PATCHWELCOME;
     }
 
     ff_amr_bit_reorder((uint16_t *) &ctx->frame, sizeof(AMRWBFrame),
@@ -1161,8 +1158,10 @@ static int amrwb_decode_frame(AVCodecContext *avctx, void *data,
 
         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);
 
@@ -1243,11 +1242,12 @@ static int amrwb_decode_frame(AVCodecContext *avctx, void *data,
 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 },
 };