Index: webrtc/modules/audio_coding/neteq/dsp_helper.cc |
diff --git a/webrtc/modules/audio_coding/neteq/dsp_helper.cc b/webrtc/modules/audio_coding/neteq/dsp_helper.cc |
index 289e66d17c86a75e847f7c960d0bbf17a6c8f13e..7451ae26f8ee997d07e6fa47a88d9dad82ba74f1 100644 |
--- a/webrtc/modules/audio_coding/neteq/dsp_helper.cc |
+++ b/webrtc/modules/audio_coding/neteq/dsp_helper.cc |
@@ -272,7 +272,7 @@ void DspHelper::CrossFade(const int16_t* input1, const int16_t* input2, |
} |
void DspHelper::UnmuteSignal(const int16_t* input, size_t length, |
- int16_t* factor, int increment, |
+ int16_t* factor, int16_t increment, |
int16_t* output) { |
uint16_t factor_16b = *factor; |
int32_t factor_32b = (static_cast<int32_t>(factor_16b) << 6) + 32; |
@@ -284,7 +284,7 @@ void DspHelper::UnmuteSignal(const int16_t* input, size_t length, |
*factor = factor_16b; |
} |
-void DspHelper::MuteSignal(int16_t* signal, int mute_slope, size_t length) { |
+void DspHelper::MuteSignal(int16_t* signal, int16_t mute_slope, size_t length) { |
int32_t factor = (16384 << 6) + 32; |
for (size_t i = 0; i < length; i++) { |
signal[i] = ((factor >> 6) * signal[i] + 8192) >> 14; |