mirror of
https://github.com/Swiftgram/Telegram-iOS.git
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5caaaafa42
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
77 lines
1.7 KiB
C++
Executable File
77 lines
1.7 KiB
C++
Executable File
//
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// libtgvoip is free and unencumbered public domain software.
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// For more information, see http://unlicense.org or the UNLICENSE file
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// you should have received with this source code distribution.
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//
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#ifndef LIBTGVOIP_ECHOCANCELLER_H
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#define LIBTGVOIP_ECHOCANCELLER_H
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#include "threading.h"
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#include "Buffers.h"
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#include "BlockingQueue.h"
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#include "MediaStreamItf.h"
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#include "utils.h"
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namespace webrtc{
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class AudioProcessing;
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class AudioFrame;
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}
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namespace tgvoip{
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class EchoCanceller{
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public:
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TGVOIP_DISALLOW_COPY_AND_ASSIGN(EchoCanceller);
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EchoCanceller(bool enableAEC, bool enableNS, bool enableAGC);
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virtual ~EchoCanceller();
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virtual void Start();
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virtual void Stop();
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void SpeakerOutCallback(unsigned char* data, size_t len);
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void Enable(bool enabled);
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void ProcessInput(int16_t* inOut, size_t numSamples);
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void SetAECStrength(int strength);
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private:
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bool enableAEC;
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bool enableAGC;
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bool enableNS;
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bool isOn;
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#ifndef TGVOIP_NO_DSP
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webrtc::AudioProcessing* apm=NULL;
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webrtc::AudioFrame* audioFrame=NULL;
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void RunBufferFarendThread();
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bool didBufferFarend;
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Thread* bufferFarendThread;
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BlockingQueue<int16_t*>* farendQueue;
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BufferPool* farendBufferPool;
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bool running;
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#endif
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};
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class AudioEffect{
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public:
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virtual ~AudioEffect()=0;
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virtual void Process(int16_t* inOut, size_t numSamples)=0;
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virtual void SetPassThrough(bool passThrough);
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protected:
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bool passThrough;
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};
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class AutomaticGainControl : public AudioEffect{
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public:
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AutomaticGainControl();
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virtual ~AutomaticGainControl();
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virtual void Process(int16_t* inOut, size_t numSamples);
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private:
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void* agc;
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void* splittingFilter;
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void* splittingFilterIn;
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void* splittingFilterOut;
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int32_t agcMicLevel;
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};
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};
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#endif //LIBTGVOIP_ECHOCANCELLER_H
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