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authorMarcelo Galvăo Póvoa <marspeoplester@gmail.com>2010-09-07 20:44:41 +0000
committerVitor Sessak <vitor1001@gmail.com>2010-09-07 20:44:41 +0000
commit1d96cc08652072730e64ecb45df81b608a015a38 (patch)
treea685761cce72c9d4acde9a2e08620a7c454ff9a6 /libavcodec
parentb33451eeacaa31a7b4f15daf3f106763a4093aa0 (diff)
downloadffmpeg-1d96cc08652072730e64ecb45df81b608a015a38.tar.gz
Move AMRNB lsf2lsp() function to common code for using in future AMRWB decoder.
Patch by Marcelo Galvăo Póvoa Originally committed as revision 25061 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'libavcodec')
-rw-r--r--libavcodec/amrnbdec.c18
-rw-r--r--libavcodec/lsp.c8
-rw-r--r--libavcodec/lsp.h5
3 files changed, 15 insertions, 16 deletions
diff --git a/libavcodec/amrnbdec.c b/libavcodec/amrnbdec.c
index d3fa7653f8..616be9710e 100644
--- a/libavcodec/amrnbdec.c
+++ b/libavcodec/amrnbdec.c
@@ -222,20 +222,6 @@ static enum Mode unpack_bitstream(AMRContext *p, const uint8_t *buf,
/// @{
/**
- * 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.
*
@@ -293,7 +279,7 @@ static void lsf2lsp_for_mode12k2(AMRContext *p, double lsp[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);
}
/**
@@ -357,7 +343,7 @@ static void lsf2lsp_3(AMRContext *p)
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++)
diff --git a/libavcodec/lsp.c b/libavcodec/lsp.c
index 7112492001..fa7152737a 100644
--- a/libavcodec/lsp.c
+++ b/libavcodec/lsp.c
@@ -65,6 +65,14 @@ void ff_acelp_lsf2lsp(int16_t *lsp, const int16_t *lsf, int lp_order)
lsp[i] = ff_cos(lsf[i] * 20861 >> 15); // divide by PI and (0,13) -> (0,14)
}
+void ff_acelp_lsf2lspd(double *lsp, const float *lsf, int lp_order)
+{
+ int i;
+
+ for(i = 0; i < lp_order; i++)
+ lsp[i] = cos(2.0 * M_PI * lsf[i]);
+}
+
/**
* \brief decodes polynomial coefficients from LSP
* \param f [out] decoded polynomial coefficients (-0x20000000 <= (3.22) <= 0x1fffffff)
diff --git a/libavcodec/lsp.h b/libavcodec/lsp.h
index 5ee5c277bc..2c22058062 100644
--- a/libavcodec/lsp.h
+++ b/libavcodec/lsp.h
@@ -63,6 +63,11 @@ void ff_set_min_dist_lsf(float *lsf, double min_spacing, int size);
void ff_acelp_lsf2lsp(int16_t *lsp, const int16_t *lsf, int lp_order);
/**
+ * Floating point version of ff_acelp_lsf2lsp()
+ */
+void ff_acelp_lsf2lspd(double *lsp, const float *lsf, int lp_order);
+
+/**
* \brief LSP to LP conversion (3.2.6 of G.729)
* \param[out] lp decoded LP coefficients (-0x8000 <= (3.12) < 0x8000)
* \param lsp LSP coefficients (-0x8000 <= (0.15) < 0x8000)