| 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;
|
| }
|
|
|