Separated WebRtcIsacfix_PitchFilterCore() out from isac-fix pitch_filter.c into its

own files for generic C and ARMv6.
Also renamed file pitchfilter_armv6.S to pitch_filter_armv6.S, to be consistent with
others.
Review URL: https://webrtc-codereview.appspot.com/722008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2608 4adac7df-926f-26a2-2b94-8c16560cd09d

src/modules/audio_coding/codecs/isac/fix/source/isacfix.gypi

@@ -46,6 +46,7 @@
         'lpc_tables.c',
         'pitch_estimator.c',
         'pitch_filter.c',
+        'pitch_filter_c.c',
         'pitch_gain_tables.c',
         'pitch_lag_tables.c',
         'spectrum_ar_model_tables.c',
@@ -73,6 +74,14 @@
         }],
         ['OS=="android"', {
           'dependencies': [ 'isac_neon', ],
+          'sources': [
+            'lattice_armv7.S',
+            'pitch_filter_armv6.S',
+          ],
+          'sources!': [
+            'lattice_c.c',
+            'pitch_filter_c.c',
+          ],
         }],
       ],
     },
@@ -96,9 +105,3 @@
     }],
   ],
 }
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:

src/modules/audio_coding/codecs/isac/fix/source/pitch_filter.c

@@ -45,19 +45,7 @@ static const WebRtc_Word16 kIntrpCoef[PITCH_FRACS][PITCH_FRACORDER] = {
   { 271, -743,  1570, -3320, 12963,  7301, -2292,  953, -325}
 };
 
-static __inline WebRtc_Word32 CalcLrIntQ(WebRtc_Word32 fixVal,
+// Function prototype for pitch filtering.
-                                         WebRtc_Word16 qDomain) {
-  WebRtc_Word32 intgr;
-  WebRtc_Word32 roundVal;
-
-  roundVal = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)1,  qDomain - 1);
-  intgr = WEBRTC_SPL_RSHIFT_W32(fixVal + roundVal, qDomain);
-
-  return intgr;
-}
-
-#ifndef WEBRTC_ARCH_ARM_V7A
-// Pitch filtering.
 // TODO(Turaj): Add descriptions of input and output parameters.
 void WebRtcIsacfix_PitchFilterCore(int loopNumber,
                                    WebRtc_Word16 gain,
@@ -68,53 +56,18 @@ void WebRtcIsacfix_PitchFilterCore(int loopNumber,
                                    const WebRtc_Word16* coefficient,
                                    WebRtc_Word16* inputBuf,
                                    WebRtc_Word16* outputBuf,
-                                   int* index2) {
+                                   int* index2);
-  int i = 0, j = 0;  // Loop counters.
-  WebRtc_Word16* ubufQQpos2 = &outputBuf2[PITCH_BUFFSIZE - (index + 2)];
-  WebRtc_Word16 tmpW16 = 0;
 
-  for (i = 0; i < loopNumber; i++) {
+static __inline WebRtc_Word32 CalcLrIntQ(WebRtc_Word32 fixVal,
-    WebRtc_Word32 tmpW32 = 0;
+                                         WebRtc_Word16 qDomain) {
+  WebRtc_Word32 intgr;
+  WebRtc_Word32 roundVal;
 
-    // Filter to get fractional pitch.
+  roundVal = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)1,  qDomain - 1);
-    for (j = 0; j < PITCH_FRACORDER; j++) {
+  intgr = WEBRTC_SPL_RSHIFT_W32(fixVal + roundVal, qDomain);
-      tmpW32 += WEBRTC_SPL_MUL_16_16(ubufQQpos2[*index2 + j], coefficient[j]);
-    }
 
-    // Saturate to avoid overflow in tmpW16.
+  return intgr;
-    tmpW32 = WEBRTC_SPL_SAT(536862719, tmpW32, -536879104);
-    tmpW32 += 8192;
-    tmpW16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmpW32, 14);
-
-    // Shift low pass filter state.
-    memmove(&inputState[1], &inputState[0],
-            (PITCH_DAMPORDER - 1) * sizeof(WebRtc_Word16));
-    inputState[0] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
-                      gain, tmpW16, 12);
-
-    // Low pass filter.
-    tmpW32 = 0;
-    // TODO(kma): Define a static inline function WebRtcSpl_DotProduct()
-    // in spl_inl.h to replace this and other similar loops.
-    for (j = 0; j < PITCH_DAMPORDER; j++) {
-      tmpW32 += WEBRTC_SPL_MUL_16_16(inputState[j], kDampFilter[j]);
-    }
-
-    // Saturate to avoid overflow in tmpW16.
-    tmpW32 = WEBRTC_SPL_SAT(1073725439, tmpW32, -1073758208);
-    tmpW32 += 16384;
-    tmpW16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmpW32, 15);
-
-    // Subtract from input and update buffer.
-    tmpW32 = inputBuf[*index2] - WEBRTC_SPL_MUL_16_16(sign, tmpW16);
-    outputBuf[*index2] = WebRtcSpl_SatW32ToW16(tmpW32);
-    tmpW32 = inputBuf[*index2] + outputBuf[*index2];
-    outputBuf2[*index2 + PITCH_BUFFSIZE] = WebRtcSpl_SatW32ToW16(tmpW32);
-
-    (*index2)++;
-  }
 }
-#endif
 
 void WebRtcIsacfix_PitchFilter(WebRtc_Word16* indatQQ, // Q10 if type is 1 or 4,
                                                        // Q0 if type is 2.

src/modules/audio_coding/codecs/isac/fix/source/pitch_filter_c.c

@@ -0,0 +1,74 @@
+/*
+ *  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
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h"
+
+/* Filter coefficicients in Q15. */
+static const WebRtc_Word16 kDampFilter[PITCH_DAMPORDER] = {
+  -2294, 8192, 20972, 8192, -2294
+};
+
+void WebRtcIsacfix_PitchFilterCore(int loopNumber,
+                                   WebRtc_Word16 gain,
+                                   int index,
+                                   WebRtc_Word16 sign,
+                                   WebRtc_Word16* inputState,
+                                   WebRtc_Word16* outputBuf2,
+                                   const WebRtc_Word16* coefficient,
+                                   WebRtc_Word16* inputBuf,
+                                   WebRtc_Word16* outputBuf,
+                                   int* index2) {
+  int i = 0, j = 0;  /* Loop counters. */
+  WebRtc_Word16* ubufQQpos2 = &outputBuf2[PITCH_BUFFSIZE - (index + 2)];
+  WebRtc_Word16 tmpW16 = 0;
+
+  for (i = 0; i < loopNumber; i++) {
+    WebRtc_Word32 tmpW32 = 0;
+
+    /* Filter to get fractional pitch. */
+    for (j = 0; j < PITCH_FRACORDER; j++) {
+      tmpW32 += WEBRTC_SPL_MUL_16_16(ubufQQpos2[*index2 + j], coefficient[j]);
+    }
+
+    /* Saturate to avoid overflow in tmpW16. */
+    tmpW32 = WEBRTC_SPL_SAT(536862719, tmpW32, -536879104);
+    tmpW32 += 8192;
+    tmpW16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmpW32, 14);
+
+    /* Shift low pass filter state. */
+    memmove(&inputState[1], &inputState[0],
+            (PITCH_DAMPORDER - 1) * sizeof(WebRtc_Word16));
+    inputState[0] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
+                      gain, tmpW16, 12);
+
+    /* Low pass filter. */
+    tmpW32 = 0;
+    /* TODO(kma): Define a static inline function WebRtcSpl_DotProduct()
+       in spl_inl.h to replace this and other similar loops. */
+    for (j = 0; j < PITCH_DAMPORDER; j++) {
+      tmpW32 += WEBRTC_SPL_MUL_16_16(inputState[j], kDampFilter[j]);
+    }
+
+    /* Saturate to avoid overflow in tmpW16. */
+    tmpW32 = WEBRTC_SPL_SAT(1073725439, tmpW32, -1073758208);
+    tmpW32 += 16384;
+    tmpW16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmpW32, 15);
+
+    /* Subtract from input and update buffer. */
+    tmpW32 = inputBuf[*index2] - WEBRTC_SPL_MUL_16_16(sign, tmpW16);
+    outputBuf[*index2] = WebRtcSpl_SatW32ToW16(tmpW32);
+    tmpW32 = inputBuf[*index2] + outputBuf[*index2];
+    outputBuf2[*index2 + PITCH_BUFFSIZE] = WebRtcSpl_SatW32ToW16(tmpW32);
+
+    (*index2)++;
+  }
+}
+