1 files changed, 47 insertions, 0 deletions
diff --git a/third_party/libwebrtc/common_audio/signal_processing/filter_ar_fast_q12.c b/third_party/libwebrtc/common_audio/signal_processing/filter_ar_fast_q12.c
new file mode 100644
index 0000000000..9010f1ce82
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/filter_ar_fast_q12.c
@@ -0,0 +1,47 @@
+/*
+ *  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 <stddef.h>
+
+#include "rtc_base/checks.h"
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+
+// TODO(bjornv): Change the return type to report errors.
+
+void WebRtcSpl_FilterARFastQ12(const int16_t* data_in,
+                               int16_t* data_out,
+                               const int16_t* __restrict coefficients,
+                               size_t coefficients_length,
+                               size_t data_length) {
+  size_t i = 0;
+  size_t j = 0;
+
+  RTC_DCHECK_GT(data_length, 0);
+  RTC_DCHECK_GT(coefficients_length, 1);
+
+  for (i = 0; i < data_length; i++) {
+    int64_t output = 0;
+    int64_t sum = 0;
+
+    for (j = coefficients_length - 1; j > 0; j--) {
+      // Negative overflow is permitted here, because this is
+      // auto-regressive filters, and the state for each batch run is
+      // stored in the "negative" positions of the output vector.
+      sum += coefficients[j] * data_out[(ptrdiff_t) i - (ptrdiff_t) j];
+    }
+
+    output = coefficients[0] * data_in[i];
+    output -= sum;
+
+    // Saturate and store the output.
+    output = WEBRTC_SPL_SAT(134215679, output, -134217728);
+    data_out[i] = (int16_t)((output + 2048) >> 12);
+  }
+}