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Unified Diff: webrtc/modules/audio_processing/gain_control_for_experimental_agc.h

Issue 1678813002: Moved the GainControlForNewAGC class to a separate file. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Removed threadchecker and introduced lock Created 4 years, 10 months ago
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Index: webrtc/modules/audio_processing/gain_control_for_experimental_agc.h
diff --git a/webrtc/modules/audio_processing/gain_control_for_experimental_agc.h b/webrtc/modules/audio_processing/gain_control_for_experimental_agc.h
new file mode 100644
index 0000000000000000000000000000000000000000..4fbd05c6850964824efbe9c195c102cfdfd81a52
--- /dev/null
+++ b/webrtc/modules/audio_processing/gain_control_for_experimental_agc.h
@@ -0,0 +1,70 @@
+/*
+ * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_FOR_EXPERIMENTAL_AGC_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_FOR_EXPERIMENTAL_AGC_H_
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/thread_checker.h"
+#include "webrtc/modules/audio_processing/agc/agc_manager_direct.h"
+#include "webrtc/modules/audio_processing/include/audio_processing.h"
+
+namespace webrtc {
+
+// This class has two main purposes:
+//
+// 1) It is returned instead of the real GainControl after the new AGC has been
+// enabled in order to prevent an outside user from overriding compression
+// settings. It doesn't do anything in its implementation, except for
+// delegating the const methods and Enable calls to the real GainControl, so
+// AGC can still be disabled.
+//
+// 2) It is injected into AgcManagerDirect and implements volume callbacks for
+// getting and setting the volume level. It just caches this value to be used
+// in VoiceEngine later.
+class GainControlForExperimentalAgc : public GainControl,
+ public VolumeCallbacks {
+ public:
+ explicit GainControlForExperimentalAgc(GainControl* gain_control,
+ rtc::CriticalSection* crit_capture);
+
+ // GainControl implementation.
+ int Enable(bool enable) override;
+ bool is_enabled() const override;
+ int set_stream_analog_level(int level) override;
+ int stream_analog_level() override;
+ int set_mode(Mode mode) override;
+ Mode mode() const override;
+ int set_target_level_dbfs(int level) override;
+ int target_level_dbfs() const override;
+ int set_compression_gain_db(int gain) override;
+ int compression_gain_db() const override;
+ int enable_limiter(bool enable) override;
+ bool is_limiter_enabled() const override;
+ int set_analog_level_limits(int minimum, int maximum) override;
+ int analog_level_minimum() const override;
+ int analog_level_maximum() const override;
+ bool stream_is_saturated() const override;
+
+ // VolumeCallbacks implementation.
+ void SetMicVolume(int volume) override;
+ int GetMicVolume() override;
+
+ private:
+ GainControl* real_gain_control_;
+ int volume_;
+ rtc::CriticalSection* crit_capture_;
+ RTC_DISALLOW_COPY_AND_ASSIGN(GainControlForExperimentalAgc);
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_FOR_EXPERIMENTAL_AGC_H_

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