| Index: webrtc/modules/audio_coding/neteq/merge.cc
|
| diff --git a/webrtc/modules/audio_coding/neteq/merge.cc b/webrtc/modules/audio_coding/neteq/merge.cc
|
| index 299682f60d44b260bf5a5e99fb8eb9ff5c8cb583..d50b66588a8d2e146905ea47a83a55bb4a2ada31 100644
|
| --- a/webrtc/modules/audio_coding/neteq/merge.cc
|
| +++ b/webrtc/modules/audio_coding/neteq/merge.cc
|
| @@ -16,6 +16,8 @@
|
| #include <algorithm> // min, max
|
| #include <memory>
|
|
|
| +#include "webrtc/base/safe_conversions.h"
|
| +#include "webrtc/base/safe_minmax.h"
|
| #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
|
| #include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
|
| #include "webrtc/modules/audio_coding/neteq/cross_correlation.h"
|
| @@ -209,8 +211,8 @@ size_t Merge::GetExpandedSignal(size_t* old_length, size_t* expand_period) {
|
| int16_t Merge::SignalScaling(const int16_t* input, size_t input_length,
|
| const int16_t* expanded_signal) const {
|
| // Adjust muting factor if new vector is more or less of the BGN energy.
|
| - const size_t mod_input_length =
|
| - std::min(static_cast<size_t>(64 * fs_mult_), input_length);
|
| + const auto mod_input_length = rtc::SafeMin<size_t>(
|
| + 64 * rtc::dchecked_cast<size_t>(fs_mult_), input_length);
|
| const int16_t expanded_max =
|
| WebRtcSpl_MaxAbsValueW16(expanded_signal, mod_input_length);
|
| int32_t factor = (expanded_max * expanded_max) /
|
|
|