Index: webrtc/modules/audio_processing/gain_control_impl.h |
diff --git a/webrtc/modules/audio_processing/gain_control_impl.h b/webrtc/modules/audio_processing/gain_control_impl.h |
index 812b88cb8d2dac6afc17c4a016c6e8525e495726..bd56ed452d2e6e196c6876df09d45825b5ec11e2 100644 |
--- a/webrtc/modules/audio_processing/gain_control_impl.h |
+++ b/webrtc/modules/audio_processing/gain_control_impl.h |
@@ -23,6 +23,7 @@ |
namespace webrtc { |
+class ApmDataDumper; |
class AudioBuffer; |
class GainControlImpl : public GainControl { |
@@ -69,6 +70,8 @@ class GainControlImpl : public GainControl { |
rtc::CriticalSection* const crit_render_ ACQUIRED_BEFORE(crit_capture_); |
rtc::CriticalSection* const crit_capture_; |
+ std::unique_ptr<ApmDataDumper> data_dumper_; |
+ |
bool enabled_ = false; |
Mode mode_ GUARDED_BY(crit_capture_); |
@@ -86,6 +89,7 @@ class GainControlImpl : public GainControl { |
rtc::Optional<size_t> num_proc_channels_ GUARDED_BY(crit_capture_); |
rtc::Optional<int> sample_rate_hz_ GUARDED_BY(crit_capture_); |
+ static int instance_counter_; |
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(GainControlImpl); |
}; |
} // namespace webrtc |