ffmpeg source update for sep 09

source/patched-ffmpeg-mt/libavcodec/amrnbdec.c

 /*
  * AMR narrowband decoder
  * Copyright (c) 2006-2007 Robert Swain
  * Copyright (c) 2009 Colin McQuillan
  *
  * This file is part of FFmpeg.
  *
  * FFmpeg is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Lesser General Public
  * License as published by the Free Software Foundation; either
@@ skipping 204 matching lines @@
     }
 
     return mode;
 }
 
 
 /// @defgroup amr_lpc_decoding AMR pitch LPC coefficient decoding functions
 /// @{
 
 /**
- * Convert an lsf vector into an lsp vector.
- *
- * @param lsf               input lsf vector
- * @param lsp               output lsp vector
- */
-static void lsf2lsp(const float *lsf, double *lsp)
-{
-    int i;
-
-    for (i = 0; i < LP_FILTER_ORDER; i++)
-        lsp[i] = cos(2.0 * M_PI * lsf[i]);
-}
-
-/**
  * Interpolate the LSF vector (used for fixed gain smoothing).
  * The interpolation is done over all four subframes even in MODE_12k2.
  *
  * @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)
 {
     int i;
 
@@ skipping 37 matching lines @@
         memcpy(p->prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(float));
 
     for (i = 0; i < LP_FILTER_ORDER; i++)
         lsf_q[i] = lsf_r[i] * (LSF_R_FAC / 8000.0) + lsf_no_r[i] * (1.0 / 8000.0);
 
     ff_set_min_dist_lsf(lsf_q, MIN_LSF_SPACING, LP_FILTER_ORDER);
 
     if (update)
         interpolate_lsf(p->lsf_q, lsf_q);
 
-    lsf2lsp(lsf_q, lsp);
+    ff_acelp_lsf2lspd(lsp, lsf_q, LP_FILTER_ORDER);
 }
 
 /**
  * Decode a set of 5 split-matrix quantized lsf indexes into 2 lsp vectors.
  *
  * @param p                 pointer to the AMRContext
  */
 static void lsf2lsp_5(AMRContext *p)
 {
     const uint16_t *lsf_param = p->frame.lsf;
@@ skipping 43 matching lines @@
     // calculate mean-removed LSF vector and add mean
     for (i = 0; i < LP_FILTER_ORDER; i++)
         lsf_q[i] = (lsf_r[i] + p->prev_lsf_r[i] * pred_fac[i]) * (LSF_R_FAC / 8000.0) + lsf_3_mean[i] * (1.0 / 8000.0);
 
     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);
     memcpy(p->prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(*lsf_r));
 
-    lsf2lsp(lsf_q, p->lsp[3]);
+    ff_acelp_lsf2lspd(p->lsp[3], lsf_q, LP_FILTER_ORDER);
 
     // interpolate LSP vectors at subframes 1, 2 and 3
     for (i = 1; i <= 3; i++)
         for(j = 0; j < LP_FILTER_ORDER; j++)
             p->lsp[i-1][j] = p->prev_lsp_sub4[j] +
                 (p->lsp[3][j] - p->prev_lsp_sub4[j]) * 0.25 * i;
 }
 
 /// @}
 
@@ skipping 696 matching lines @@
 AVCodec amrnb_decoder = {
     .name           = "amrnb",
     .type           = AVMEDIA_TYPE_AUDIO,
     .id             = CODEC_ID_AMR_NB,
     .priv_data_size = sizeof(AMRContext),
     .init           = amrnb_decode_init,
     .decode         = amrnb_decode_frame,
     .long_name      = NULL_IF_CONFIG_SMALL("Adaptive Multi-Rate NarrowBand"),
     .sample_fmts    = (enum SampleFormat[]){SAMPLE_FMT_FLT,SAMPLE_FMT_NONE},
 };