Resolved Coverity warnings.

This CL includes changes to resolve Coverity warnings 14086, 14110 and 14111.

Tested with trybots and audioproc_unittests.

BUG=None
TEST=None

Review URL: https://webrtc-codereview.appspot.com/606004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2321 4adac7df-926f-26a2-2b94-8c16560cd09d

src/modules/audio_processing/aec/aec_core.c

@@ -777,7 +777,7 @@ static void ProcessBlock(aec_t* aec) {
     memcpy(aec->xfBuf[1] + aec->xfBufBlockPos * PART_LEN1, &xf_ptr[PART_LEN1],
            sizeof(float) * PART_LEN1);
 
-    memset(yf[0], 0, sizeof(float) * (PART_LEN1 * 2));
+    memset(yf, 0, sizeof(yf));
 
     // Filter far
     WebRtcAec_FilterFar(aec, yf);

src/modules/audio_processing/aec/aec_resampler.c

@@ -1,5 +1,5 @@
 /*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
@@ -71,21 +71,24 @@ int WebRtcAec_FreeResampler(void *resampInst)
     return 0;
 }
 
-int WebRtcAec_ResampleLinear(void *resampInst,
-                             const short *inspeech,
-                             int size,
-                             float skew,
-                             short *outspeech)
+void WebRtcAec_ResampleLinear(void *resampInst,
+                              const short *inspeech,
+                              int size,
+                              float skew,
+                              short *outspeech,
+                              int *size_out)
 {
     resampler_t *obj = (resampler_t*) resampInst;
 
     short *y;
     float be, tnew, interp;
-    int tn, outsize, mm;
+    int tn, mm;
 
-    if (size < 0 || size > 2 * FRAME_LEN) {
-        return -1;
-    }
+    assert(!(size < 0 || size > 2 * FRAME_LEN));
+    assert(resampInst != NULL);
+    assert(inspeech != NULL);
+    assert(outspeech != NULL);
+    assert(size_out != NULL);
 
     // Add new frame data in lookahead
     memcpy(&obj->buffer[FRAME_LEN + kResamplingDelay],
@@ -121,15 +124,13 @@ int WebRtcAec_ResampleLinear(void *resampInst,
         tn = (int) tnew;
     }
 
-    outsize = mm;
-    obj->position += outsize * be - size;
+    *size_out = mm;
+    obj->position += (*size_out) * be - size;
 
     // Shift buffer
     memmove(obj->buffer,
             &obj->buffer[size],
             (kResamplerBufferSize - size) * sizeof(short));
-
-    return outsize;
 }
 
 int WebRtcAec_GetSkew(void *resampInst, int rawSkew, float *skewEst)

src/modules/audio_processing/aec/aec_resampler.h

@@ -1,5 +1,5 @@
 /*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
@@ -25,11 +25,11 @@ int WebRtcAec_FreeResampler(void *resampInst);
 int WebRtcAec_GetSkew(void *resampInst, int rawSkew, float *skewEst);
 
 // Resamples input using linear interpolation.
-// Returns size of resampled array.
-int WebRtcAec_ResampleLinear(void *resampInst,
-                             const short *inspeech,
-                             int size,
-                             float skew,
-                             short *outspeech);
+void WebRtcAec_ResampleLinear(void *resampInst,
+                              const short *inspeech,
+                              int size,
+                              float skew,
+                              short *outspeech,
+                              int *size_out);
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_RESAMPLER_H_

src/modules/audio_processing/aec/echo_cancellation.c

@@ -276,11 +276,8 @@ WebRtc_Word32 WebRtcAec_BufferFarend(void *aecInst, const WebRtc_Word16 *farend,
 
     if (aecpc->skewMode == kAecTrue && aecpc->resample == kAecTrue) {
         // Resample and get a new number of samples
-        newNrOfSamples = WebRtcAec_ResampleLinear(aecpc->resampler,
-                                                  farend,
-                                                  nrOfSamples,
-                                                  skew,
-                                                  newFarend);
+        WebRtcAec_ResampleLinear(aecpc->resampler, farend, nrOfSamples, skew,
+                                 newFarend, &newNrOfSamples);
         farend_ptr = (const int16_t*) newFarend;
     }
 

src/modules/audio_processing/agc/analog_agc.c

@@ -1650,9 +1650,10 @@ int WebRtcAgc_Init(void *agcInst, WebRtc_Word32 minLevel, WebRtc_Word32 maxLevel
     {
         stt->Rxx16w32_array[0][i] = 0;
     }
-    for (i = 0; i < 20; i++)
+    for (i = 0; i < 10; i++)
     {
         stt->env[0][i] = 0;
+        stt->env[1][i] = 0;
     }
     stt->inQueue = 0;