1 files changed, 74 insertions, 0 deletions
diff --git a/src/modules/audio_coding/codecs/isac/fix/source/pitch_filter_c.c b/src/modules/audio_coding/codecs/isac/fix/source/pitch_filter_c.c
new file mode 100644
index 0000000000..29b4b6af6e
--- /dev/null
+++ b/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)++;
+  }
+}
+