Wraparound distortion in Opus

This CL solves the wraparound distortion in Opus.

In the Opus decoder-wrapper we are downsampling the signal from 48 kHz to 32 kHz. This is done in two steps, using the following functions from the signal processing library:
WebRtcSpl_Resample48khzTo32khz() and WebRtcSpl_VectorBitShiftW32ToW16

The latter does not have a saturation check, and the signal can suffer from wraparound. I've added saturation control to the function.

BUG=issue1089

Review URL: https://webrtc-codereview.appspot.com/967004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3103 4adac7df-926f-26a2-2b94-8c16560cd09d

3 files changed, 66 insertions, 26 deletions

diff --git a/common_audio/signal_processing/include/signal_processing_library.h b/common_audio/signal_processing/include/signal_processing_library.h
index 1738e8e9..50a89c37 100644
--- a/common_audio/signal_processing/include/signal_processing_library.h
+++ b/common_audio/signal_processing/include/signal_processing_library.h
@@ -377,11 +377,10 @@ void WebRtcSpl_VectorBitShiftW32(WebRtc_Word32* out_vector,
                                  WebRtc_Word16 vector_length,
                                  G_CONST WebRtc_Word32* in_vector,
                                  WebRtc_Word16 right_shifts);
-void WebRtcSpl_VectorBitShiftW32ToW16(WebRtc_Word16* out_vector,
-                                      WebRtc_Word16 vector_length,
-                                      G_CONST WebRtc_Word32* in_vector,
-                                      WebRtc_Word16 right_shifts);
-
+void WebRtcSpl_VectorBitShiftW32ToW16(int16_t* out_vector,
+                                      int vector_length,
+                                      const int32_t* in_vector,
+                                      int right_shifts);
 void WebRtcSpl_ScaleVector(G_CONST WebRtc_Word16* in_vector,
                            WebRtc_Word16* out_vector,
                            WebRtc_Word16 gain,
@@ -1180,7 +1179,8 @@ void WebRtcSpl_SynthesisQMF(const WebRtc_Word16* low_band,
 // WebRtcSpl_VectorBitShiftW32ToW16(...)
 //
 // Bit shifts all the values in a WebRtc_Word32 vector up or downwards and
-// stores the result as a WebRtc_Word16 vector
+// stores the result as an int16_t vector. The function will saturate the
+// signal if needed, before storing in the output vector.
 //
 // Input:
 //      - vector_length : Length of vector
diff --git a/common_audio/signal_processing/signal_processing_unittest.cc b/common_audio/signal_processing/signal_processing_unittest.cc
index d5026fb8..cf7cb27f 100644
--- a/common_audio/signal_processing/signal_processing_unittest.cc
+++ b/common_audio/signal_processing/signal_processing_unittest.cc
@@ -615,3 +615,47 @@ TEST_F(SplTest, FFTTest) {
         //EXPECT_EQ(A[kk], B[kk]);
     }
 }
+
+TEST_F(SplTest, Resample48WithSaturationTest) {
+  // The test resamples 3*kBlockSize number of samples to 2*kBlockSize number
+  // of samples.
+  const int kBlockSize = 16;
+
+  // Saturated input vector of 48 samples.
+  const int32_t kVectorSaturated[3 * kBlockSize + 7] = {
+     -32768, -32768, -32768, -32768, -32768, -32768, -32768, -32768,
+     -32768, -32768, -32768, -32768, -32768, -32768, -32768, -32768,
+     -32768, -32768, -32768, -32768, -32768, -32768, -32768, -32768,
+     32767, 32767, 32767, 32767, 32767, 32767, 32767, 32767,
+     32767, 32767, 32767, 32767, 32767, 32767, 32767, 32767,
+     32767, 32767, 32767, 32767, 32767, 32767, 32767, 32767,
+     32767, 32767, 32767, 32767, 32767, 32767, 32767
+  };
+
+  // All values in |out_vector| should be |kRefValue32kHz|.
+  const int32_t kRefValue32kHz1 = -1077493760;
+  const int32_t kRefValue32kHz2 = 1077493645;
+
+  // After bit shift with saturation, |out_vector_w16| is saturated.
+
+  const int16_t kRefValue16kHz1 = -32768;
+  const int16_t kRefValue16kHz2 = 32767;
+  // Vector for storing output.
+  int32_t out_vector[2 * kBlockSize];
+  int16_t out_vector_w16[2 * kBlockSize];
+
+  WebRtcSpl_Resample48khzTo32khz(kVectorSaturated, out_vector, kBlockSize);
+  WebRtcSpl_VectorBitShiftW32ToW16(out_vector_w16, 2 * kBlockSize, out_vector,
+                                   15);
+
+  // Comparing output values against references. The values at position
+  // 12-15 are skipped to account for the filter lag.
+  for (int i = 0; i < 12; ++i) {
+    EXPECT_EQ(kRefValue32kHz1, out_vector[i]);
+    EXPECT_EQ(kRefValue16kHz1, out_vector_w16[i]);
+  }
+  for (int i = 16; i < 2 * kBlockSize; ++i) {
+    EXPECT_EQ(kRefValue32kHz2, out_vector[i]);
+    EXPECT_EQ(kRefValue16kHz2, out_vector_w16[i]);
+  }
+}
diff --git a/common_audio/signal_processing/vector_scaling_operations.c b/common_audio/signal_processing/vector_scaling_operations.c
index 242955cc..c4b1df9c 100644
--- a/common_audio/signal_processing/vector_scaling_operations.c
+++ b/common_audio/signal_processing/vector_scaling_operations.c
@@ -66,27 +66,23 @@ void WebRtcSpl_VectorBitShiftW32(WebRtc_Word32 *out_vector,
     }
 }
 
-void WebRtcSpl_VectorBitShiftW32ToW16(WebRtc_Word16 *res,
-                                  WebRtc_Word16 length,
-                                  G_CONST WebRtc_Word32 *in,
-                                  WebRtc_Word16 right_shifts)
-{
-    int i;
-
-    if (right_shifts >= 0)
-    {
-        for (i = length; i > 0; i--)
-        {
-            (*res++) = (WebRtc_Word16)((*in++) >> right_shifts);
-        }
-    } else
-    {
-        WebRtc_Word16 left_shifts = -right_shifts;
-        for (i = length; i > 0; i--)
-        {
-            (*res++) = (WebRtc_Word16)((*in++) << left_shifts);
-        }
+void WebRtcSpl_VectorBitShiftW32ToW16(int16_t* out, int length,
+                                      const int32_t* in, int right_shifts) {
+  int i;
+  int32_t tmp_w32;
+
+  if (right_shifts >= 0) {
+    for (i = length; i > 0; i--) {
+      tmp_w32 = (*in++) >> right_shifts;
+      (*out++) = WebRtcSpl_SatW32ToW16(tmp_w32);
+    }
+  } else {
+    int16_t left_shifts = -right_shifts;
+    for (i = length; i > 0; i--) {
+      tmp_w32 = (*in++) << left_shifts;
+      (*out++) = WebRtcSpl_SatW32ToW16(tmp_w32);
     }
+  }
 }
 
 void WebRtcSpl_ScaleVector(G_CONST WebRtc_Word16 *in_vector, WebRtc_Word16 *out_vector,