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4332d09028
This is a reland of68007e97ecOriginal change's description: > Reland "Remove WEBRTC_TRACE." > > This is a reland of2209b90449> Original change's description: > > Remove WEBRTC_TRACE. > > > > Bug: webrtc:5118 > > Change-Id: I0af0f8845ee016fa61d7cecc526e2a672ec8732d > > Reviewed-on: https://webrtc-review.googlesource.com/5382 > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > Reviewed-by: Sami Kalliomäki <sakal@webrtc.org> > > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > > Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20114} > > Bug: webrtc:5118 > Change-Id: I2d93fd40fcaa251c363bdcfb1c04b834a3a7f0e9 > Reviewed-on: https://webrtc-review.googlesource.com/6000 > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Reviewed-by: Niels Moller <nisse@webrtc.org> > Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20132} Bug: webrtc:5118 Change-Id: I3b46406899d043c3260fc3195b524138324f7313 Reviewed-on: https://webrtc-review.googlesource.com/6301 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org> Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20144}
132 lines
3.9 KiB
C++
132 lines
3.9 KiB
C++
/*
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* Copyright (c) 2012 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 VOICE_ENGINE_TRANSMIT_MIXER_H_
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#define VOICE_ENGINE_TRANSMIT_MIXER_H_
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#include <memory>
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#include "common_audio/resampler/include/push_resampler.h"
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#include "common_types.h" // NOLINT(build/include)
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#include "modules/audio_processing/typing_detection.h"
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#include "modules/include/module_common_types.h"
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#include "rtc_base/criticalsection.h"
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#include "voice_engine/audio_level.h"
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#include "voice_engine/include/voe_base.h"
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#include "voice_engine/voice_engine_defines.h"
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#if !defined(WEBRTC_ANDROID) && !defined(WEBRTC_IOS)
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#define WEBRTC_VOICE_ENGINE_TYPING_DETECTION 1
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#else
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#define WEBRTC_VOICE_ENGINE_TYPING_DETECTION 0
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#endif
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namespace webrtc {
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class AudioProcessing;
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class ProcessThread;
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namespace voe {
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class ChannelManager;
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class MixedAudio;
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class TransmitMixer {
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public:
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static int32_t Create(TransmitMixer*& mixer);
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static void Destroy(TransmitMixer*& mixer);
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void SetEngineInformation(ChannelManager* channelManager);
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int32_t SetAudioProcessingModule(AudioProcessing* audioProcessingModule);
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int32_t PrepareDemux(const void* audioSamples,
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size_t nSamples,
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size_t nChannels,
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uint32_t samplesPerSec,
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uint16_t totalDelayMS,
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int32_t clockDrift,
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uint16_t currentMicLevel,
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bool keyPressed);
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void ProcessAndEncodeAudio();
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// Must be called on the same thread as PrepareDemux().
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uint32_t CaptureLevel() const;
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int32_t StopSend();
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// TODO(solenberg): Remove, once AudioMonitor is gone.
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int8_t AudioLevel() const;
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// 'virtual' to allow mocking.
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virtual int16_t AudioLevelFullRange() const;
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// See description of "totalAudioEnergy" in the WebRTC stats spec:
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// https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy
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// 'virtual' to allow mocking.
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virtual double GetTotalInputEnergy() const;
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// 'virtual' to allow mocking.
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virtual double GetTotalInputDuration() const;
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virtual ~TransmitMixer();
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// Virtual to allow mocking.
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virtual void EnableStereoChannelSwapping(bool enable);
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bool IsStereoChannelSwappingEnabled();
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// Virtual to allow mocking.
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virtual bool typing_noise_detected() const;
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protected:
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TransmitMixer() = default;
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private:
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// Gets the maximum sample rate and number of channels over all currently
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// sending codecs.
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void GetSendCodecInfo(int* max_sample_rate, size_t* max_channels);
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void GenerateAudioFrame(const int16_t audioSamples[],
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size_t nSamples,
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size_t nChannels,
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int samplesPerSec);
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void ProcessAudio(int delay_ms, int clock_drift, int current_mic_level,
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bool key_pressed);
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#if WEBRTC_VOICE_ENGINE_TYPING_DETECTION
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void TypingDetection(bool key_pressed);
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#endif
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// uses
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ChannelManager* _channelManagerPtr = nullptr;
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AudioProcessing* audioproc_ = nullptr;
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// owns
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AudioFrame _audioFrame;
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PushResampler<int16_t> resampler_; // ADM sample rate -> mixing rate
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voe::AudioLevel _audioLevel;
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#if WEBRTC_VOICE_ENGINE_TYPING_DETECTION
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webrtc::TypingDetection typing_detection_;
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#endif
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rtc::CriticalSection lock_;
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bool typing_noise_detected_ RTC_GUARDED_BY(lock_) = false;
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uint32_t _captureLevel = 0;
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bool stereo_codec_ = false;
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bool swap_stereo_channels_ = false;
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};
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} // namespace voe
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} // namespace webrtc
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#endif // VOICE_ENGINE_TRANSMIT_MIXER_H_
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