Index: webrtc/modules/audio_processing/agc2/gain_controller2.h |
diff --git a/webrtc/modules/audio_processing/agc2/gain_controller2.h b/webrtc/modules/audio_processing/agc2/gain_controller2.h |
index 1a8bb7f39c28be3753793a09af83f8154e201121..07525843c16b902a7922011514078cf7fe275a9f 100644 |
--- a/webrtc/modules/audio_processing/agc2/gain_controller2.h |
+++ b/webrtc/modules/audio_processing/agc2/gain_controller2.h |
@@ -14,7 +14,6 @@ |
#include <memory> |
#include <string> |
-#include "webrtc/modules/audio_processing/agc2/digital_gain_applier.h" |
#include "webrtc/modules/audio_processing/include/audio_processing.h" |
#include "webrtc/rtc_base/constructormagic.h" |
@@ -26,14 +25,15 @@ class AudioBuffer; |
// Gain Controller 2 aims to automatically adjust levels by acting on the |
// microphone gain and/or applying digital gain. |
// |
-// It temporarily implements a hard-coded gain mode only. |
+// It temporarily implements a fixed gain mode with hard-clipping. |
class GainController2 { |
public: |
- explicit GainController2(int sample_rate_hz); |
+ explicit GainController2(const float fixed_gain_db); |
peah-webrtc
2017/08/18 04:51:06
I think it would make sense to pass the parameters
|
~GainController2(); |
int sample_rate_hz() { return sample_rate_hz_; } |
+ void Initialize(int sample_rate_hz); |
void Process(AudioBuffer* audio); |
static bool Validate(const AudioProcessing::Config::GainController2& config); |
@@ -43,11 +43,9 @@ class GainController2 { |
private: |
int sample_rate_hz_; |
std::unique_ptr<ApmDataDumper> data_dumper_; |
- DigitalGainApplier digital_gain_applier_; |
static int instance_count_; |
- // TODO(alessiob): Remove once a meaningful gain controller mode is |
- // implemented. |
- const float gain_; |
+ const float fixed_gain_; |
+ |
RTC_DISALLOW_COPY_AND_ASSIGN(GainController2); |
}; |