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/*
* Copyright (c) 2012 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 VOICE_ENGINE_TRANSMIT_MIXER_H_
#define VOICE_ENGINE_TRANSMIT_MIXER_H_
#include <memory>
#include "common_audio/resampler/include/push_resampler.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/audio_processing/typing_detection.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/criticalsection.h"
#include "voice_engine/audio_level.h"
#include "voice_engine/include/voe_base.h"
#include "voice_engine/voice_engine_defines.h"
#if !defined(WEBRTC_ANDROID) && !defined(WEBRTC_IOS)
#define WEBRTC_VOICE_ENGINE_TYPING_DETECTION 1
#else
#define WEBRTC_VOICE_ENGINE_TYPING_DETECTION 0
#endif
namespace webrtc {
class AudioProcessing;
class ProcessThread;
namespace voe {
class ChannelManager;
class MixedAudio;
class TransmitMixer {
public:
static int32_t Create(TransmitMixer*& mixer);
static void Destroy(TransmitMixer*& mixer);
void SetEngineInformation(ChannelManager* channelManager);
int32_t SetAudioProcessingModule(AudioProcessing* audioProcessingModule);
int32_t PrepareDemux(const void* audioSamples,
size_t nSamples,
size_t nChannels,
uint32_t samplesPerSec,
uint16_t totalDelayMS,
int32_t clockDrift,
uint16_t currentMicLevel,
bool keyPressed);
void ProcessAndEncodeAudio();
// Must be called on the same thread as PrepareDemux().
uint32_t CaptureLevel() const;
int32_t StopSend();
// TODO(solenberg): Remove, once AudioMonitor is gone.
int8_t AudioLevel() const;
// 'virtual' to allow mocking.
virtual int16_t AudioLevelFullRange() const;
// See description of "totalAudioEnergy" in the WebRTC stats spec:
// https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy
// 'virtual' to allow mocking.
virtual double GetTotalInputEnergy() const;
// 'virtual' to allow mocking.
virtual double GetTotalInputDuration() const;
virtual ~TransmitMixer();
// Virtual to allow mocking.
virtual void EnableStereoChannelSwapping(bool enable);
bool IsStereoChannelSwappingEnabled();
// Virtual to allow mocking.
virtual bool typing_noise_detected() const;
protected:
TransmitMixer() = default;
private:
// Gets the maximum sample rate and number of channels over all currently
// sending codecs.
void GetSendCodecInfo(int* max_sample_rate, size_t* max_channels);
void GenerateAudioFrame(const int16_t audioSamples[],
size_t nSamples,
size_t nChannels,
int samplesPerSec);
void ProcessAudio(int delay_ms, int clock_drift, int current_mic_level,
bool key_pressed);
#if WEBRTC_VOICE_ENGINE_TYPING_DETECTION
void TypingDetection(bool key_pressed);
#endif
// uses
ChannelManager* _channelManagerPtr = nullptr;
AudioProcessing* audioproc_ = nullptr;
// owns
AudioFrame _audioFrame;
PushResampler<int16_t> resampler_; // ADM sample rate -> mixing rate
voe::AudioLevel _audioLevel;
#if WEBRTC_VOICE_ENGINE_TYPING_DETECTION
webrtc::TypingDetection typing_detection_;
#endif
rtc::CriticalSection lock_;
bool typing_noise_detected_ RTC_GUARDED_BY(lock_) = false;
uint32_t _captureLevel = 0;
bool stereo_codec_ = false;
bool swap_stereo_channels_ = false;
};
} // namespace voe
} // namespace webrtc
#endif // VOICE_ENGINE_TRANSMIT_MIXER_H_