Move src/ -> webrtc/

TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/915006

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@2963 4adac7df-926f-26a2-2b94-8c16560cd09d

1 files changed, 148 insertions, 0 deletions

diff --git a/voice_engine/voe_base_impl.h b/voice_engine/voe_base_impl.h
new file mode 100644
index 00000000..0eb44fa3
--- /dev/null
+++ b/voice_engine/voe_base_impl.h
@@ -0,0 +1,148 @@
+/*
+ *  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 "voe_base.h"
+
+#include "module_common_types.h"
+#include "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);
+
+    virtual int Terminate();
+
+    virtual int MaxNumOfChannels();
+
+    virtual int CreateChannel();
+
+    virtual int DeleteChannel(int channel);
+
+    virtual int SetLocalReceiver(int channel, int port,
+                                 int RTCPport = kVoEDefault,
+                                 const char ipAddr[64] = NULL,
+                                 const char multiCastAddr[64] = NULL);
+
+    virtual int GetLocalReceiver(int channel, int& port, int& RTCPport,
+                                 char ipAddr[64]);
+
+    virtual int SetSendDestination(int channel, int port,
+                                   const char ipAddr[64],
+                                   int sourcePort = kVoEDefault,
+                                   int RTCPport = kVoEDefault);
+
+    virtual int GetSendDestination(int channel,
+                                   int& port,
+                                   char ipAddr[64],
+                                   int& sourcePort,
+                                   int& RTCPport);
+
+    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 SetNetEQBGNMode(int channel, NetEqBgnModes mode);
+
+    virtual int GetNetEQBGNMode(int channel, NetEqBgnModes& 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();
+
+    // AudioTransport
+    virtual WebRtc_Word32
+        RecordedDataIsAvailable(const void* audioSamples,
+                                const WebRtc_UWord32 nSamples,
+                                const WebRtc_UWord8 nBytesPerSample,
+                                const WebRtc_UWord8 nChannels,
+                                const WebRtc_UWord32 samplesPerSec,
+                                const WebRtc_UWord32 totalDelayMS,
+                                const WebRtc_Word32 clockDrift,
+                                const WebRtc_UWord32 currentMicLevel,
+                                WebRtc_UWord32& newMicLevel);
+
+    virtual WebRtc_Word32 NeedMorePlayData(const WebRtc_UWord32 nSamples,
+                                           const WebRtc_UWord8 nBytesPerSample,
+                                           const WebRtc_UWord8 nChannels,
+                                           const WebRtc_UWord32 samplesPerSec,
+                                           void* audioSamples,
+                                           WebRtc_UWord32& nSamplesOut);
+
+    // AudioDeviceObserver
+    virtual void OnErrorIsReported(const ErrorCode error);
+    virtual void OnWarningIsReported(const WarningCode warning);
+
+protected:
+    VoEBaseImpl(voe::SharedData* shared);
+    virtual ~VoEBaseImpl();
+
+private:
+    WebRtc_Word32 StartPlayout();
+    WebRtc_Word32 StopPlayout();
+    WebRtc_Word32 StartSend();
+    WebRtc_Word32 StopSend();
+    WebRtc_Word32 TerminateInternal();
+
+    WebRtc_Word32 AddBuildInfo(char* str) const;
+    WebRtc_Word32 AddVoEVersion(char* str) const;
+#ifdef WEBRTC_EXTERNAL_TRANSPORT
+    WebRtc_Word32 AddExternalTransportBuild(char* str) const;
+#endif
+#ifdef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
+    WebRtc_Word32 AddExternalRecAndPlayoutBuild(char* str) const;
+#endif
+    VoiceEngineObserver* _voiceEngineObserverPtr;
+    CriticalSectionWrapper& _callbackCritSect;
+
+    bool _voiceEngineObserver;
+    WebRtc_UWord32 _oldVoEMicLevel;
+    WebRtc_UWord32 _oldMicLevel;
+    AudioFrame _audioFrame;
+    voe::SharedData* _shared;
+
+};
+
+} // namespace webrtc
+
+#endif  // WEBRTC_VOICE_ENGINE_VOE_BASE_IMPL_H