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authorRonald S. Bultje <rsbultje@gmail.com>2010-03-11 17:42:53 +0000
committerRonald S. Bultje <rsbultje@gmail.com>2010-03-11 17:42:53 +0000
commit95c6b5ebc88532e6492845e3ad3579d551ff43e6 (patch)
tree270779841070d288b1964e1f1bf23b462d66bbc1
parent34056cbba55f0131e8d2698717ab9e86e2da5178 (diff)
downloadffmpeg-streaming-95c6b5ebc88532e6492845e3ad3579d551ff43e6.zip
ffmpeg-streaming-95c6b5ebc88532e6492845e3ad3579d551ff43e6.tar.gz
Fix spelling.
Originally committed as revision 22470 to svn://svn.ffmpeg.org/ffmpeg/trunk
-rw-r--r--libavcodec/acelp_vectors.c4
-rw-r--r--libavcodec/acelp_vectors.h6
-rw-r--r--libavcodec/amrnbdec.c4
-rw-r--r--libavcodec/sipr.c6
4 files changed, 10 insertions, 10 deletions
diff --git a/libavcodec/acelp_vectors.c b/libavcodec/acelp_vectors.c
index 736987b..a25011f 100644
--- a/libavcodec/acelp_vectors.c
+++ b/libavcodec/acelp_vectors.c
@@ -207,8 +207,8 @@ void ff_weighted_vector_sumf(float *out, const float *in_a, const float *in_b,
+ weight_coeff_b * in_b[i];
}
-void ff_adaptative_gain_control(float *buf_out, float speech_energ,
- int size, float alpha, float *gain_mem)
+void ff_adaptive_gain_control(float *buf_out, float speech_energ,
+ int size, float alpha, float *gain_mem)
{
int i;
float postfilter_energ = ff_dot_productf(buf_out, buf_out, size);
diff --git a/libavcodec/acelp_vectors.h b/libavcodec/acelp_vectors.h
index 2cc5f36..3e6652f 100644
--- a/libavcodec/acelp_vectors.h
+++ b/libavcodec/acelp_vectors.h
@@ -212,7 +212,7 @@ void ff_weighted_vector_sumf(float *out, const float *in_a, const float *in_b,
int length);
/**
- * Adaptative gain control (as used in AMR postfiltering)
+ * Adaptive gain control (as used in AMR postfiltering)
*
* @param buf_out the input speech buffer
* @param speech_energ input energy
@@ -220,8 +220,8 @@ void ff_weighted_vector_sumf(float *out, const float *in_a, const float *in_b,
* @param alpha exponential filter factor
* @param gain_mem a pointer to the filter memory (single float of size)
*/
-void ff_adaptative_gain_control(float *buf_out, float speech_energ,
- int size, float alpha, float *gain_mem);
+void ff_adaptive_gain_control(float *buf_out, float speech_energ,
+ int size, float alpha, float *gain_mem);
/**
* Set the sum of squares of a signal by scaling
diff --git a/libavcodec/amrnbdec.c b/libavcodec/amrnbdec.c
index 325acfd..e5a82b4 100644
--- a/libavcodec/amrnbdec.c
+++ b/libavcodec/amrnbdec.c
@@ -943,8 +943,8 @@ static void postfilter(AMRContext *p, float *lpc, float *buf_out)
ff_tilt_compensation(&p->tilt_mem, tilt_factor(lpc_n, lpc_d), buf_out,
AMR_SUBFRAME_SIZE);
- ff_adaptative_gain_control(buf_out, speech_gain, AMR_SUBFRAME_SIZE,
- AMR_AGC_ALPHA, &p->postfilter_agc);
+ ff_adaptive_gain_control(buf_out, speech_gain, AMR_SUBFRAME_SIZE,
+ AMR_AGC_ALPHA, &p->postfilter_agc);
}
/// @}
diff --git a/libavcodec/sipr.c b/libavcodec/sipr.c
index ada2fe9..d2ce9c7 100644
--- a/libavcodec/sipr.c
+++ b/libavcodec/sipr.c
@@ -254,7 +254,7 @@ static void sipr_decode_lp(float *lsfnew, const float *lsfold, float *Az,
}
/**
- * Evaluates the adaptative impulse response.
+ * Evaluates the adaptive impulse response.
*/
static void eval_ir(const float *Az, int pitch_lag, float *freq,
float pitch_sharp_factor)
@@ -479,8 +479,8 @@ 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_adaptative_gain_control(&synth[i * SUBFR_SIZE], energy,
- SUBFR_SIZE, 0.9, &ctx->postfilter_agc);
+ ff_adaptive_gain_control(&synth[i * SUBFR_SIZE], energy,
+ SUBFR_SIZE, 0.9, &ctx->postfilter_agc);
}
memcpy(ctx->postfilter_syn5k0, ctx->postfilter_syn5k0 + frame_size,
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