
Voice engine shouldn't really have to manage this. Instead, have AGC keep track of the last input volume, so that it can avoid getting stuck under coarsely quantized conditions. Add a test to verify the behavior. TESTED=unittests, and observed that AGC didn't get stuck on a MacBook where this problem can actually occur. R=bjornv@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8729004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5571 4adac7df-926f-26a2-2b94-8c16560cd09d
164 lines
5.7 KiB
C++
164 lines
5.7 KiB
C++
/*
|
|
* 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 WEBRTC_VOICE_ENGINE_VOE_BASE_IMPL_H
|
|
#define WEBRTC_VOICE_ENGINE_VOE_BASE_IMPL_H
|
|
|
|
#include "webrtc/voice_engine/include/voe_base.h"
|
|
|
|
#include "webrtc/modules/interface/module_common_types.h"
|
|
#include "webrtc/voice_engine/shared_data.h"
|
|
|
|
namespace webrtc
|
|
{
|
|
|
|
class ProcessThread;
|
|
|
|
class VoEBaseImpl: public VoEBase,
|
|
public AudioTransport,
|
|
public AudioDeviceObserver
|
|
{
|
|
public:
|
|
virtual int RegisterVoiceEngineObserver(VoiceEngineObserver& observer);
|
|
|
|
virtual int DeRegisterVoiceEngineObserver();
|
|
|
|
virtual int Init(AudioDeviceModule* external_adm = NULL,
|
|
AudioProcessing* audioproc = NULL);
|
|
virtual AudioProcessing* audio_processing() {
|
|
return _shared->audio_processing();
|
|
}
|
|
|
|
virtual int Terminate();
|
|
|
|
virtual int CreateChannel();
|
|
virtual int CreateChannel(const Config& config);
|
|
|
|
virtual int DeleteChannel(int channel);
|
|
|
|
virtual int StartReceive(int channel);
|
|
|
|
virtual int StartPlayout(int channel);
|
|
|
|
virtual int StartSend(int channel);
|
|
|
|
virtual int StopReceive(int channel);
|
|
|
|
virtual int StopPlayout(int channel);
|
|
|
|
virtual int StopSend(int channel);
|
|
|
|
virtual int SetNetEQPlayoutMode(int channel, NetEqModes mode);
|
|
|
|
virtual int GetNetEQPlayoutMode(int channel, NetEqModes& mode);
|
|
|
|
virtual int SetOnHoldStatus(int channel,
|
|
bool enable,
|
|
OnHoldModes mode = kHoldSendAndPlay);
|
|
|
|
virtual int GetOnHoldStatus(int channel, bool& enabled, OnHoldModes& mode);
|
|
|
|
virtual int GetVersion(char version[1024]);
|
|
|
|
virtual int LastError();
|
|
|
|
virtual AudioTransport* audio_transport() { return this; }
|
|
|
|
// AudioTransport
|
|
virtual int32_t
|
|
RecordedDataIsAvailable(const void* audioSamples,
|
|
uint32_t nSamples,
|
|
uint8_t nBytesPerSample,
|
|
uint8_t nChannels,
|
|
uint32_t samplesPerSec,
|
|
uint32_t totalDelayMS,
|
|
int32_t clockDrift,
|
|
uint32_t micLevel,
|
|
bool keyPressed,
|
|
uint32_t& newMicLevel);
|
|
|
|
virtual int32_t NeedMorePlayData(uint32_t nSamples,
|
|
uint8_t nBytesPerSample,
|
|
uint8_t nChannels,
|
|
uint32_t samplesPerSec,
|
|
void* audioSamples,
|
|
uint32_t& nSamplesOut);
|
|
|
|
virtual int OnDataAvailable(const int voe_channels[],
|
|
int number_of_voe_channels,
|
|
const int16_t* audio_data,
|
|
int sample_rate,
|
|
int number_of_channels,
|
|
int number_of_frames,
|
|
int audio_delay_milliseconds,
|
|
int volume,
|
|
bool key_pressed,
|
|
bool need_audio_processing);
|
|
|
|
virtual void OnData(int voe_channel, const void* audio_data,
|
|
int bits_per_sample, int sample_rate,
|
|
int number_of_channels, int number_of_frames);
|
|
|
|
// AudioDeviceObserver
|
|
virtual void OnErrorIsReported(ErrorCode error);
|
|
virtual void OnWarningIsReported(WarningCode warning);
|
|
|
|
protected:
|
|
VoEBaseImpl(voe::SharedData* shared);
|
|
virtual ~VoEBaseImpl();
|
|
|
|
private:
|
|
int32_t StartPlayout();
|
|
int32_t StopPlayout();
|
|
int32_t StartSend();
|
|
int32_t StopSend();
|
|
int32_t TerminateInternal();
|
|
|
|
// Helper function to process the recorded data with AudioProcessing Module,
|
|
// demultiplex the data to specific voe channels, encode and send to the
|
|
// network. When |number_of_VoE_channels| is 0, it will demultiplex the
|
|
// data to all the existing VoE channels.
|
|
// It returns new AGC microphone volume or 0 if no volume changes
|
|
// should be done.
|
|
int ProcessRecordedDataWithAPM(const int voe_channels[],
|
|
int number_of_voe_channels,
|
|
const void* audio_data,
|
|
uint32_t sample_rate,
|
|
uint8_t number_of_channels,
|
|
uint32_t number_of_frames,
|
|
uint32_t audio_delay_milliseconds,
|
|
int32_t clock_drift,
|
|
uint32_t volume,
|
|
bool key_pressed);
|
|
|
|
int32_t AddBuildInfo(char* str) const;
|
|
int32_t AddVoEVersion(char* str) const;
|
|
|
|
// Initialize channel by setting Engine Information then initializing
|
|
// channel.
|
|
int InitializeChannel(voe::ChannelOwner* channel_owner);
|
|
#ifdef WEBRTC_EXTERNAL_TRANSPORT
|
|
int32_t AddExternalTransportBuild(char* str) const;
|
|
#endif
|
|
#ifdef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
|
|
int32_t AddExternalRecAndPlayoutBuild(char* str) const;
|
|
#endif
|
|
VoiceEngineObserver* _voiceEngineObserverPtr;
|
|
CriticalSectionWrapper& _callbackCritSect;
|
|
|
|
bool _voiceEngineObserver;
|
|
AudioFrame _audioFrame;
|
|
voe::SharedData* _shared;
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // WEBRTC_VOICE_ENGINE_VOE_BASE_IMPL_H
|