Merge remote-tracking branch 'qatar/master'
[ffmpeg.git] / libavcodec / sipr.c
index fc19ecd..d6179a8 100644 (file)
@@ -23,6 +23,7 @@
 
 #include <math.h>
 #include <stdint.h>
+#include <string.h>
 
 #include "libavutil/mathematics.h"
 #include "avcodec.h"
@@ -184,7 +185,7 @@ static void pitch_sharpening(int pitch_lag_int, float beta,
 }
 
 /**
- * Extracts decoding parameters from the input bitstream.
+ * Extract decoding parameters from the input bitstream.
  * @param parms          parameters structure
  * @param pgb            pointer to initialized GetBitContext structure
  */
@@ -209,32 +210,6 @@ static void decode_parameters(SiprParameters* parms, GetBitContext *pgb,
     }
 }
 
-static void lsp2lpc_sipr(const double *lsp, float *Az)
-{
-    int lp_half_order = LP_FILTER_ORDER >> 1;
-    double buf[(LP_FILTER_ORDER >> 1) + 1];
-    double pa[(LP_FILTER_ORDER >> 1) + 1];
-    double *qa = buf + 1;
-    int i,j;
-
-    qa[-1] = 0.0;
-
-    ff_lsp2polyf(lsp    , pa, lp_half_order    );
-    ff_lsp2polyf(lsp + 1, qa, lp_half_order - 1);
-
-    for (i = 1, j = LP_FILTER_ORDER - 1; i < lp_half_order; i++, j--) {
-        double paf =  pa[i]            * (1 + lsp[LP_FILTER_ORDER - 1]);
-        double qaf = (qa[i] - qa[i-2]) * (1 - lsp[LP_FILTER_ORDER - 1]);
-        Az[i-1]  = (paf + qaf) * 0.5;
-        Az[j-1]  = (paf - qaf) * 0.5;
-    }
-
-    Az[lp_half_order - 1] = (1.0 + lsp[LP_FILTER_ORDER - 1]) *
-        pa[lp_half_order] * 0.5;
-
-    Az[LP_FILTER_ORDER - 1] = lsp[LP_FILTER_ORDER - 1];
-}
-
 static void sipr_decode_lp(float *lsfnew, const float *lsfold, float *Az,
                            int num_subfr)
 {
@@ -247,14 +222,14 @@ static void sipr_decode_lp(float *lsfnew, const float *lsfold, float *Az,
         for (j = 0; j < LP_FILTER_ORDER; j++)
             lsfint[j] = lsfold[j] * (1 - t) + t * lsfnew[j];
 
-        lsp2lpc_sipr(lsfint, Az);
+        ff_amrwb_lsp2lpc(lsfint, Az, LP_FILTER_ORDER);
         Az += LP_FILTER_ORDER;
         t += t0;
     }
 }
 
 /**
- * Evaluates the adaptive impulse response.
+ * Evaluate the adaptive impulse response.
  */
 static void eval_ir(const float *Az, int pitch_lag, float *freq,
                     float pitch_sharp_factor)
@@ -276,7 +251,7 @@ static void eval_ir(const float *Az, int pitch_lag, float *freq,
 }
 
 /**
- * Evaluates the convolution of a vector with a sparse vector.
+ * Evaluate the convolution of a vector with a sparse vector.
  */
 static void convolute_with_sparse(float *out, const AMRFixed *pulses,
                                   const float *shape, int length)
@@ -479,25 +454,23 @@ static void decode_frame(SiprContext *ctx, SiprParameters *params,
             float energy = ff_dot_productf(ctx->postfilter_syn5k0 + LP_FILTER_ORDER + i*SUBFR_SIZE,
                                            ctx->postfilter_syn5k0 + LP_FILTER_ORDER + i*SUBFR_SIZE,
                                            SUBFR_SIZE);
-            ff_adaptive_gain_control(&synth[i * SUBFR_SIZE], energy,
+            ff_adaptive_gain_control(&synth[i * SUBFR_SIZE],
+                                     &synth[i * SUBFR_SIZE], energy,
                                      SUBFR_SIZE, 0.9, &ctx->postfilter_agc);
         }
 
         memcpy(ctx->postfilter_syn5k0, ctx->postfilter_syn5k0 + frame_size,
                LP_FILTER_ORDER*sizeof(float));
     }
-    memcpy(ctx->excitation, excitation - PITCH_DELAY_MAX - L_INTERPOL,
+    memmove(ctx->excitation, excitation - PITCH_DELAY_MAX - L_INTERPOL,
            (PITCH_DELAY_MAX + L_INTERPOL) * sizeof(float));
 
-    ff_acelp_apply_order_2_transfer_function(synth,
+    ff_acelp_apply_order_2_transfer_function(out_data, synth,
                                              (const float[2]) {-1.99997   , 1.000000000},
                                              (const float[2]) {-1.93307352, 0.935891986},
                                              0.939805806,
                                              ctx->highpass_filt_mem,
                                              frame_size);
-
-    ctx->dsp.vector_clipf(out_data, synth, -1, 32767./(1<<15), frame_size);
-
 }
 
 static av_cold int sipr_decoder_init(AVCodecContext * avctx)
@@ -521,9 +494,7 @@ static av_cold int sipr_decoder_init(AVCodecContext * avctx)
     for (i = 0; i < 4; i++)
         ctx->energy_history[i] = -14;
 
-    avctx->sample_fmt = SAMPLE_FMT_FLT;
-
-    dsputil_init(&ctx->dsp, avctx);
+    avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
 
     return 0;
 }
@@ -575,16 +546,14 @@ static int sipr_decode_frame(AVCodecContext *avctx, void *datap,
         mode_par->subframe_count * sizeof(float);
 
     return mode_par->bits_per_frame >> 3;
-};
+}
 
-AVCodec sipr_decoder = {
-    "sipr",
-    AVMEDIA_TYPE_AUDIO,
-    CODEC_ID_SIPR,
-    sizeof(SiprContext),
-    sipr_decoder_init,
-    NULL,
-    NULL,
-    sipr_decode_frame,
+AVCodec ff_sipr_decoder = {
+    .name           = "sipr",
+    .type           = AVMEDIA_TYPE_AUDIO,
+    .id             = CODEC_ID_SIPR,
+    .priv_data_size = sizeof(SiprContext),
+    .init           = sipr_decoder_init,
+    .decode         = sipr_decode_frame,
     .long_name = NULL_IF_CONFIG_SMALL("RealAudio SIPR / ACELP.NET"),
 };