Index: webrtc/modules/audio_processing/gain_control_impl.cc |
diff --git a/webrtc/modules/audio_processing/gain_control_impl.cc b/webrtc/modules/audio_processing/gain_control_impl.cc |
index 7b284e8853dfa30c5cea13c6535c235c50080aad..04a6c7ba29c264d4d8f1ccd6f7c26d29ad3e25c9 100644 |
--- a/webrtc/modules/audio_processing/gain_control_impl.cc |
+++ b/webrtc/modules/audio_processing/gain_control_impl.cc |
@@ -75,7 +75,7 @@ int GainControlImpl::ProcessRenderAudio(AudioBuffer* audio) { |
assert(audio->num_frames_per_band() <= 160); |
render_queue_buffer_.resize(0); |
- for (int i = 0; i < num_handles(); i++) { |
+ for (size_t i = 0; i < num_handles(); i++) { |
Handle* my_handle = static_cast<Handle*>(handle(i)); |
int err = |
WebRtcAgc_GetAddFarendError(my_handle, audio->num_frames_per_band()); |
@@ -114,7 +114,7 @@ void GainControlImpl::ReadQueuedRenderData() { |
size_t buffer_index = 0; |
const size_t num_frames_per_band = |
capture_queue_buffer_.size() / num_handles(); |
- for (int i = 0; i < num_handles(); i++) { |
+ for (size_t i = 0; i < num_handles(); i++) { |
Handle* my_handle = static_cast<Handle*>(handle(i)); |
WebRtcAgc_AddFarend(my_handle, &capture_queue_buffer_[buffer_index], |
num_frames_per_band); |
@@ -138,7 +138,7 @@ int GainControlImpl::AnalyzeCaptureAudio(AudioBuffer* audio) { |
if (mode_ == kAdaptiveAnalog) { |
capture_levels_.assign(num_handles(), analog_capture_level_); |
- for (int i = 0; i < num_handles(); i++) { |
+ for (size_t i = 0; i < num_handles(); i++) { |
Handle* my_handle = static_cast<Handle*>(handle(i)); |
err = WebRtcAgc_AddMic( |
my_handle, |
@@ -152,7 +152,7 @@ int GainControlImpl::AnalyzeCaptureAudio(AudioBuffer* audio) { |
} |
} else if (mode_ == kAdaptiveDigital) { |
- for (int i = 0; i < num_handles(); i++) { |
+ for (size_t i = 0; i < num_handles(); i++) { |
Handle* my_handle = static_cast<Handle*>(handle(i)); |
int32_t capture_level_out = 0; |
@@ -191,7 +191,7 @@ int GainControlImpl::ProcessCaptureAudio(AudioBuffer* audio) { |
assert(audio->num_channels() == num_handles()); |
stream_is_saturated_ = false; |
- for (int i = 0; i < num_handles(); i++) { |
+ for (size_t i = 0; i < num_handles(); i++) { |
Handle* my_handle = static_cast<Handle*>(handle(i)); |
int32_t capture_level_out = 0; |
uint8_t saturation_warning = 0; |
@@ -222,7 +222,7 @@ int GainControlImpl::ProcessCaptureAudio(AudioBuffer* audio) { |
if (mode_ == kAdaptiveAnalog) { |
// Take the analog level to be the average across the handles. |
analog_capture_level_ = 0; |
- for (int i = 0; i < num_handles(); i++) { |
+ for (size_t i = 0; i < num_handles(); i++) { |
analog_capture_level_ += capture_levels_[i]; |
} |
@@ -433,7 +433,7 @@ int GainControlImpl::ConfigureHandle(void* handle) const { |
return WebRtcAgc_set_config(static_cast<Handle*>(handle), config); |
} |
-int GainControlImpl::num_handles_required() const { |
+size_t GainControlImpl::num_handles_required() const { |
// Not locked as it only relies on APM public API which is threadsafe. |
return apm_->num_proc_channels(); |
} |