Index: webrtc/modules/audio_processing/agc/agc_manager_direct.cc |
diff --git a/webrtc/modules/audio_processing/agc/agc_manager_direct.cc b/webrtc/modules/audio_processing/agc/agc_manager_direct.cc |
index 74f55407a425605e7209fc0e601bb1bfa942f357..48ce2f877c39377abdd58f9fe2dbb7b24780ea8c 100644 |
--- a/webrtc/modules/audio_processing/agc/agc_manager_direct.cc |
+++ b/webrtc/modules/audio_processing/agc/agc_manager_direct.cc |
@@ -95,7 +95,7 @@ class DebugFile { |
~DebugFile() { |
fclose(file_); |
} |
- void Write(const int16_t* data, int length_samples) { |
+ void Write(const int16_t* data, size_t length_samples) { |
fwrite(data, 1, length_samples * sizeof(int16_t), file_); |
} |
private: |
@@ -106,7 +106,7 @@ class DebugFile { |
} |
~DebugFile() { |
} |
- void Write(const int16_t* data, int length_samples) { |
+ void Write(const int16_t* data, size_t length_samples) { |
} |
#endif // WEBRTC_AGC_DEBUG_DUMP |
}; |
@@ -188,8 +188,8 @@ int AgcManagerDirect::Initialize() { |
void AgcManagerDirect::AnalyzePreProcess(int16_t* audio, |
int num_channels, |
- int samples_per_channel) { |
- int length = num_channels * samples_per_channel; |
+ size_t samples_per_channel) { |
+ size_t length = num_channels * samples_per_channel; |
if (capture_muted_) { |
return; |
} |
@@ -230,7 +230,7 @@ void AgcManagerDirect::AnalyzePreProcess(int16_t* audio, |
} |
void AgcManagerDirect::Process(const int16_t* audio, |
- int length, |
+ size_t length, |
int sample_rate_hz) { |
if (capture_muted_) { |
return; |