Files
platform-external-webrtc/webrtc/voice_engine/voe_base_impl.h
andrew@webrtc.org 27c6980239 Move the volume quantization workaround from VoE to AGC.
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
2014-02-18 20:24:56 +00:00

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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 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