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Diffstat (limited to 'webrtc/common_audio/signal_processing/filter_ar_fast_q12.c')
-rw-r--r--webrtc/common_audio/signal_processing/filter_ar_fast_q12.c19
1 files changed, 12 insertions, 7 deletions
diff --git a/webrtc/common_audio/signal_processing/filter_ar_fast_q12.c b/webrtc/common_audio/signal_processing/filter_ar_fast_q12.c
index 70001a0..8b8bdb1 100644
--- a/webrtc/common_audio/signal_processing/filter_ar_fast_q12.c
+++ b/webrtc/common_audio/signal_processing/filter_ar_fast_q12.c
@@ -7,9 +7,11 @@
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
-#include <assert.h>
-#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
+#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.
@@ -21,15 +23,18 @@ void WebRtcSpl_FilterARFastQ12(const int16_t* data_in,
size_t i = 0;
size_t j = 0;
- assert(data_length > 0);
- assert(coefficients_length > 1);
+ RTC_DCHECK_GT(data_length, 0);
+ RTC_DCHECK_GT(coefficients_length, 1);
for (i = 0; i < data_length; i++) {
- int32_t output = 0;
- int32_t sum = 0;
+ int64_t output = 0;
+ int64_t sum = 0;
for (j = coefficients_length - 1; j > 0; j--) {
- sum += coefficients[j] * data_out[i - 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];