Index: webrtc/common_audio/signal_processing/downsample_fast.c |
diff --git a/webrtc/common_audio/signal_processing/downsample_fast.c b/webrtc/common_audio/signal_processing/downsample_fast.c |
index 726a88819ac01e47e3dbfb4e41ccd2739d2f3dcd..3cbc3c111a67fae19b6bc5dccec76882aeaa2d06 100644 |
--- a/webrtc/common_audio/signal_processing/downsample_fast.c |
+++ b/webrtc/common_audio/signal_processing/downsample_fast.c |
@@ -10,6 +10,9 @@ |
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h" |
+#include "webrtc/base/checks.h" |
+#include "webrtc/base/sanitizer.h" |
+ |
// TODO(Bjornv): Change the function parameter order to WebRTC code style. |
// C version of WebRtcSpl_DownsampleFast() for generic platforms. |
int WebRtcSpl_DownsampleFastC(const int16_t* data_in, |
@@ -20,6 +23,7 @@ int WebRtcSpl_DownsampleFastC(const int16_t* data_in, |
size_t coefficients_length, |
int factor, |
size_t delay) { |
+ int16_t* const original_data_out = data_out; |
size_t i = 0; |
size_t j = 0; |
int32_t out_s32 = 0; |
@@ -31,10 +35,14 @@ int WebRtcSpl_DownsampleFastC(const int16_t* data_in, |
return -1; |
} |
+ rtc_MsanCheckInitialized(coefficients, sizeof(coefficients[0]), |
+ coefficients_length); |
+ |
for (i = delay; i < endpos; i += factor) { |
out_s32 = 2048; // Round value, 0.5 in Q12. |
for (j = 0; j < coefficients_length; j++) { |
+ rtc_MsanCheckInitialized(&data_in[i - j], sizeof(data_in[0]), 1); |
out_s32 += coefficients[j] * data_in[i - j]; // Q12. |
} |
@@ -44,5 +52,9 @@ int WebRtcSpl_DownsampleFastC(const int16_t* data_in, |
*data_out++ = WebRtcSpl_SatW32ToW16(out_s32); |
} |
+ RTC_DCHECK_EQ(original_data_out + data_out_length, data_out); |
+ rtc_MsanCheckInitialized(original_data_out, sizeof(original_data_out[0]), |
+ data_out_length); |
+ |
return 0; |
} |