blob: 0ba99cf9d84f9ac4ad333c229634aec9352ca9cf [file] [log] [blame]
* 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.
#include <memory>
#include "webrtc/common_audio/resampler/include/push_resampler.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_processing/typing_detection.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/rtc_base/criticalsection.h"
#include "webrtc/voice_engine/audio_level.h"
#include "webrtc/voice_engine/file_player.h"
#include "webrtc/voice_engine/file_recorder.h"
#include "webrtc/voice_engine/include/voe_base.h"
#include "webrtc/voice_engine/monitor_module.h"
#include "webrtc/voice_engine/voice_engine_defines.h"
#if !defined(WEBRTC_ANDROID) && !defined(WEBRTC_IOS)
namespace webrtc {
class AudioProcessing;
class ProcessThread;
namespace voe {
class ChannelManager;
class MixedAudio;
class Statistics;
class TransmitMixer : public FileCallback {
static int32_t Create(TransmitMixer*& mixer, uint32_t instanceId);
static void Destroy(TransmitMixer*& mixer);
int32_t SetEngineInformation(ProcessThread& processThread,
Statistics& engineStatistics,
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:
// 'virtual' to allow mocking.
virtual double GetTotalInputEnergy() const;
// 'virtual' to allow mocking.
virtual double GetTotalInputDuration() const;
bool IsRecordingCall();
bool IsRecordingMic();
int StartPlayingFileAsMicrophone(const char* fileName,
bool loop,
FileFormats format,
int startPosition,
float volumeScaling,
int stopPosition,
const CodecInst* codecInst);
int StartPlayingFileAsMicrophone(InStream* stream,
FileFormats format,
int startPosition,
float volumeScaling,
int stopPosition,
const CodecInst* codecInst);
int StopPlayingFileAsMicrophone();
int IsPlayingFileAsMicrophone() const;
int StartRecordingMicrophone(const char* fileName,
const CodecInst* codecInst);
int StartRecordingMicrophone(OutStream* stream,
const CodecInst* codecInst);
int StopRecordingMicrophone();
int StartRecordingCall(const char* fileName, const CodecInst* codecInst);
int StartRecordingCall(OutStream* stream, const CodecInst* codecInst);
int StopRecordingCall();
void SetMixWithMicStatus(bool mix);
int32_t RegisterVoiceEngineObserver(VoiceEngineObserver& observer);
virtual ~TransmitMixer();
// Periodic callback from the MonitorModule.
void OnPeriodicProcess();
// FileCallback
void PlayNotification(const int32_t id,
const uint32_t durationMs);
void RecordNotification(const int32_t id,
const uint32_t durationMs);
void PlayFileEnded(const int32_t id);
void RecordFileEnded(const int32_t id);
// Virtual to allow mocking.
virtual void EnableStereoChannelSwapping(bool enable);
bool IsStereoChannelSwappingEnabled();
TransmitMixer() : _monitorModule(this) {}
TransmitMixer() = default;
TransmitMixer(uint32_t instanceId);
// 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);
int32_t RecordAudioToFile(uint32_t mixingFrequency);
int32_t MixOrReplaceAudioWithFile(
int mixingFrequency);
void ProcessAudio(int delay_ms, int clock_drift, int current_mic_level,
bool key_pressed);
void TypingDetection(bool keyPressed);
// uses
Statistics* _engineStatisticsPtr = nullptr;
ChannelManager* _channelManagerPtr = nullptr;
AudioProcessing* audioproc_ = nullptr;
VoiceEngineObserver* _voiceEngineObserverPtr = nullptr;
ProcessThread* _processThreadPtr = nullptr;
// owns
AudioFrame _audioFrame;
PushResampler<int16_t> resampler_; // ADM sample rate -> mixing rate
std::unique_ptr<FilePlayer> file_player_;
std::unique_ptr<FileRecorder> file_recorder_;
std::unique_ptr<FileRecorder> file_call_recorder_;
int _filePlayerId = 0;
int _fileRecorderId = 0;
int _fileCallRecorderId = 0;
bool _filePlaying = false;
bool _fileRecording = false;
bool _fileCallRecording = false;
voe::AudioLevel _audioLevel;
// protect file instances and their variables in MixedParticipants()
rtc::CriticalSection _critSect;
rtc::CriticalSection _callbackCritSect;
MonitorModule<TransmitMixer> _monitorModule;
webrtc::TypingDetection _typingDetection;
bool _typingNoiseWarningPending = false;
bool _typingNoiseDetected = false;
int _instanceId = 0;
bool _mixFileWithMicrophone = false;
uint32_t _captureLevel = 0;
bool stereo_codec_ = false;
bool swap_stereo_channels_ = false;
} // namespace voe
} // namespace webrtc