753eada3aa
Corresponds to upstream commit 524e9b043e7e86fd72353b987c9d5f6a1ebf83e1 Update notes: * Pull in third party license file * Replace .gypi files with BUILD.gn to keep track of what changes upstream * Bunch of new filse pulled in as dependencies * Won't build yet due to changes needed on top of these
109 lines
3.5 KiB
C++
109 lines
3.5 KiB
C++
/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_AGC_MANAGER_DIRECT_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_AGC_MANAGER_DIRECT_H_
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#include "webrtc/base/scoped_ptr.h"
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#include "webrtc/modules/audio_processing/agc/agc.h"
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namespace webrtc {
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class AudioFrame;
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class DebugFile;
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class GainControl;
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// Callbacks that need to be injected into AgcManagerDirect to read and control
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// the volume values. This is done to remove the VoiceEngine dependency in
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// AgcManagerDirect.
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// TODO(aluebs): Remove VolumeCallbacks.
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class VolumeCallbacks {
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public:
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virtual ~VolumeCallbacks() {}
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virtual void SetMicVolume(int volume) = 0;
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virtual int GetMicVolume() = 0;
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};
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// Direct interface to use AGC to set volume and compression values.
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// AudioProcessing uses this interface directly to integrate the callback-less
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// AGC.
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//
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// This class is not thread-safe.
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class AgcManagerDirect final {
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public:
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// AgcManagerDirect will configure GainControl internally. The user is
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// responsible for processing the audio using it after the call to Process.
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// The operating range of startup_min_level is [12, 255] and any input value
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// outside that range will be clamped.
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AgcManagerDirect(GainControl* gctrl,
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VolumeCallbacks* volume_callbacks,
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int startup_min_level);
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// Dependency injection for testing. Don't delete |agc| as the memory is owned
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// by the manager.
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AgcManagerDirect(Agc* agc,
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GainControl* gctrl,
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VolumeCallbacks* volume_callbacks,
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int startup_min_level);
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~AgcManagerDirect();
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int Initialize();
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void AnalyzePreProcess(int16_t* audio,
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int num_channels,
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size_t samples_per_channel);
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void Process(const int16_t* audio, size_t length, int sample_rate_hz);
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// Call when the capture stream has been muted/unmuted. This causes the
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// manager to disregard all incoming audio; chances are good it's background
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// noise to which we'd like to avoid adapting.
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void SetCaptureMuted(bool muted);
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bool capture_muted() { return capture_muted_; }
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float voice_probability();
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private:
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// Sets a new microphone level, after first checking that it hasn't been
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// updated by the user, in which case no action is taken.
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void SetLevel(int new_level);
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// Set the maximum level the AGC is allowed to apply. Also updates the
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// maximum compression gain to compensate. The level must be at least
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// |kClippedLevelMin|.
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void SetMaxLevel(int level);
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int CheckVolumeAndReset();
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void UpdateGain();
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void UpdateCompressor();
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rtc::scoped_ptr<Agc> agc_;
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GainControl* gctrl_;
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VolumeCallbacks* volume_callbacks_;
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int frames_since_clipped_;
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int level_;
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int max_level_;
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int max_compression_gain_;
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int target_compression_;
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int compression_;
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float compression_accumulator_;
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bool capture_muted_;
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bool check_volume_on_next_process_;
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bool startup_;
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int startup_min_level_;
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rtc::scoped_ptr<DebugFile> file_preproc_;
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rtc::scoped_ptr<DebugFile> file_postproc_;
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RTC_DISALLOW_COPY_AND_ASSIGN(AgcManagerDirect);
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_AGC_MANAGER_DIRECT_H_
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