8fff1f065e
r4326 was mistakenly committed to stable, so this is to re-merge back to trunk. Fixed the AGC and interface problems on the new path. In order to make the AGC work properly, we need to cache the volume value passed by the callback, compare it with the value returned by shared->transmit_mixer()->CaptureLevel(). If they are the same, we need to return 0 to indicate no volume needs changing, otherwise return the new volume. By doing this, we avoid setting the volume all the same, which allows the users to change the volume manually. This patch also fixes some minor issues with the interfaces too: make the int channel[] const, and correct the order of the input params in channel::Demultiplex. R=tommi@webrtc.org BUG=[2134] TEST=compile && manual AGC test Review URL: https://webrtc-codereview.appspot.com/1921004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4450 4adac7df-926f-26a2-2b94-8c16560cd09d
157 lines
5.4 KiB
C++
157 lines
5.4 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 WEBRTC_VOICE_ENGINE_VOE_BASE_IMPL_H
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#define WEBRTC_VOICE_ENGINE_VOE_BASE_IMPL_H
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#include "webrtc/voice_engine/include/voe_base.h"
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#include "webrtc/modules/interface/module_common_types.h"
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#include "webrtc/voice_engine/shared_data.h"
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namespace webrtc
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{
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class ProcessThread;
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class VoEBaseImpl: public VoEBase,
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public AudioTransport,
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public AudioDeviceObserver
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{
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public:
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virtual int RegisterVoiceEngineObserver(VoiceEngineObserver& observer);
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virtual int DeRegisterVoiceEngineObserver();
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virtual int Init(AudioDeviceModule* external_adm = NULL,
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AudioProcessing* audioproc = NULL);
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virtual AudioProcessing* audio_processing() {
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return _shared->audio_processing();
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}
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virtual int Terminate();
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virtual int MaxNumOfChannels();
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virtual int CreateChannel();
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virtual int DeleteChannel(int channel);
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virtual int StartReceive(int channel);
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virtual int StartPlayout(int channel);
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virtual int StartSend(int channel);
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virtual int StopReceive(int channel);
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virtual int StopPlayout(int channel);
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virtual int StopSend(int channel);
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virtual int SetNetEQPlayoutMode(int channel, NetEqModes mode);
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virtual int GetNetEQPlayoutMode(int channel, NetEqModes& mode);
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virtual int SetOnHoldStatus(int channel,
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bool enable,
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OnHoldModes mode = kHoldSendAndPlay);
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virtual int GetOnHoldStatus(int channel, bool& enabled, OnHoldModes& mode);
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virtual int GetVersion(char version[1024]);
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virtual int LastError();
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// AudioTransport
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virtual int32_t
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RecordedDataIsAvailable(const void* audioSamples,
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uint32_t nSamples,
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uint8_t nBytesPerSample,
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uint8_t nChannels,
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uint32_t samplesPerSec,
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uint32_t totalDelayMS,
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int32_t clockDrift,
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uint32_t currentMicLevel,
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bool keyPressed,
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uint32_t& newMicLevel);
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virtual int32_t NeedMorePlayData(uint32_t nSamples,
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uint8_t nBytesPerSample,
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uint8_t nChannels,
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uint32_t samplesPerSec,
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void* audioSamples,
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uint32_t& nSamplesOut);
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virtual int OnDataAvailable(const int voe_channels[],
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int number_of_voe_channels,
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const int16_t* audio_data,
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int sample_rate,
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int number_of_channels,
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int number_of_frames,
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int audio_delay_milliseconds,
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int current_volume,
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bool key_pressed,
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bool need_audio_processing);
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// AudioDeviceObserver
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virtual void OnErrorIsReported(ErrorCode error);
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virtual void OnWarningIsReported(WarningCode warning);
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protected:
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VoEBaseImpl(voe::SharedData* shared);
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virtual ~VoEBaseImpl();
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private:
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int32_t StartPlayout();
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int32_t StopPlayout();
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int32_t StartSend();
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int32_t StopSend();
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int32_t TerminateInternal();
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// Helper function to process the recorded data with AudioProcessing Module,
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// demultiplex the data to specific voe channels, encode and send to the
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// network. When |number_of_VoE_channels| is 0, it will demultiplex the
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// data to all the existing VoE channels.
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// It returns new AGC microphone volume or 0 if no volume changes
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// should be done.
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int ProcessRecordedDataWithAPM(const int voe_channels[],
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int number_of_voe_channels,
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const void* audio_data,
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uint32_t sample_rate,
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uint8_t number_of_channels,
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uint32_t number_of_frames,
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uint32_t audio_delay_milliseconds,
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int32_t clock_drift,
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uint32_t current_volume,
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bool key_pressed);
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int32_t AddBuildInfo(char* str) const;
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int32_t AddVoEVersion(char* str) const;
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#ifdef WEBRTC_EXTERNAL_TRANSPORT
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int32_t AddExternalTransportBuild(char* str) const;
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#endif
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#ifdef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
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int32_t AddExternalRecAndPlayoutBuild(char* str) const;
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#endif
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VoiceEngineObserver* _voiceEngineObserverPtr;
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CriticalSectionWrapper& _callbackCritSect;
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bool _voiceEngineObserver;
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uint32_t _oldVoEMicLevel;
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uint32_t _oldMicLevel;
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AudioFrame _audioFrame;
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voe::SharedData* _shared;
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};
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} // namespace webrtc
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#endif // WEBRTC_VOICE_ENGINE_VOE_BASE_IMPL_H
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