Index: webrtc/tools/agc/agc_manager.cc |
diff --git a/webrtc/tools/agc/agc_manager.cc b/webrtc/tools/agc/agc_manager.cc |
index 3d7f624c9fb9757f2cb33924be07165a8b7da7cb..36290f713cf2c6d11cf71f3de99fd10da6262aa5 100644 |
--- a/webrtc/tools/agc/agc_manager.cc |
+++ b/webrtc/tools/agc/agc_manager.cc |
@@ -66,7 +66,7 @@ class MediaCallback : public VoEMediaProcess { |
protected: |
virtual void Process(const int channel, const ProcessingTypes type, |
- int16_t audio[], const int samples_per_channel, |
+ int16_t audio[], const size_t samples_per_channel, |
const int sample_rate_hz, const bool is_stereo) { |
CriticalSectionScoped cs(crit_); |
if (direct_->capture_muted()) { |
@@ -81,7 +81,7 @@ class MediaCallback : public VoEMediaProcess { |
int16_t mono[kMaxSamplesPerChannel]; |
int16_t* mono_ptr = audio; |
if (is_stereo) { |
- for (int n = 0; n < samples_per_channel; n++) { |
+ for (size_t n = 0; n < samples_per_channel; n++) { |
mono[n] = audio[n * 2]; |
} |
mono_ptr = mono; |
@@ -94,7 +94,7 @@ class MediaCallback : public VoEMediaProcess { |
frame_.num_channels_ = is_stereo ? 2 : 1; |
frame_.samples_per_channel_ = samples_per_channel; |
frame_.sample_rate_hz_ = sample_rate_hz; |
- const int length_samples = frame_.num_channels_ * samples_per_channel; |
+ const size_t length_samples = frame_.num_channels_ * samples_per_channel; |
memcpy(frame_.data_, audio, length_samples * sizeof(int16_t)); |
// Apply compression to the audio. |
@@ -122,7 +122,7 @@ class PreprocCallback : public VoEMediaProcess { |
protected: |
virtual void Process(const int channel, const ProcessingTypes type, |
- int16_t audio[], const int samples_per_channel, |
+ int16_t audio[], const size_t samples_per_channel, |
const int sample_rate_hz, const bool is_stereo) { |
CriticalSectionScoped cs(crit_); |
if (direct_->capture_muted()) { |