/*
 *  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.
 */

// This sub-API supports the following functionalities:
//
//  - Enables full duplex VoIP sessions via RTP using G.711 (mu-Law or A-Law).
//  - Initialization and termination.
//  - Trace information on text files or via callbacks.
//  - Multi-channel support (mixing, sending to multiple destinations etc.).
//
// To support other codecs than G.711, the VoECodec sub-API must be utilized.
//
// Usage example, omitting error checking:
//
//  using namespace webrtc;
//  VoiceEngine* voe = VoiceEngine::Create();
//  VoEBase* base = VoEBase::GetInterface(voe);
//  base->Init();
//  int ch = base->CreateChannel();
//  base->StartPlayout(ch);
//  ...
//  base->DeleteChannel(ch);
//  base->Terminate();
//  base->Release();
//  VoiceEngine::Delete(voe);
//
#ifndef WEBRTC_VOICE_ENGINE_VOE_BASE_H
#define WEBRTC_VOICE_ENGINE_VOE_BASE_H

#include "webrtc/common_types.h"

namespace webrtc {

class AudioDeviceModule;
class AudioProcessing;
class AudioTransport;
class Config;

const int kVoEDefault = -1;

// VoiceEngineObserver
class WEBRTC_DLLEXPORT VoiceEngineObserver {
 public:
  // This method will be called after the occurrence of any runtime error
  // code, or warning notification, when the observer interface has been
  // installed using VoEBase::RegisterVoiceEngineObserver().
  virtual void CallbackOnError(int channel, int errCode) = 0;

 protected:
  virtual ~VoiceEngineObserver() {}
};

// VoiceEngine
class WEBRTC_DLLEXPORT VoiceEngine {
 public:
  // Creates a VoiceEngine object, which can then be used to acquire
  // sub-APIs. Returns NULL on failure.
  static VoiceEngine* Create();
  static VoiceEngine* Create(const Config& config);

  // Deletes a created VoiceEngine object and releases the utilized resources.
  // Note that if there are outstanding references held via other interfaces,
  // the voice engine instance will not actually be deleted until those
  // references have been released.
  static bool Delete(VoiceEngine*& voiceEngine);

  // Specifies the amount and type of trace information which will be
  // created by the VoiceEngine.
  static int SetTraceFilter(unsigned int filter);

  // Sets the name of the trace file and enables non-encrypted trace messages.
  static int SetTraceFile(const char* fileNameUTF8,
                          bool addFileCounter = false);

  // Installs the TraceCallback implementation to ensure that the user
  // receives callbacks for generated trace messages.
  static int SetTraceCallback(TraceCallback* callback);

#if !defined(WEBRTC_CHROMIUM_BUILD)
  static int SetAndroidObjects(void* javaVM, void* context);
#endif

 protected:
  VoiceEngine() {}
  ~VoiceEngine() {}
};

// VoEBase
class WEBRTC_DLLEXPORT VoEBase {
 public:
  // Factory for the VoEBase sub-API. Increases an internal reference
  // counter if successful. Returns NULL if the API is not supported or if
  // construction fails.
  static VoEBase* GetInterface(VoiceEngine* voiceEngine);

  // Releases the VoEBase sub-API and decreases an internal reference
  // counter. Returns the new reference count. This value should be zero
  // for all sub-APIs before the VoiceEngine object can be safely deleted.
  virtual int Release() = 0;

  // Installs the observer class to enable runtime error control and
  // warning notifications. Returns -1 in case of an error, 0 otherwise.
  virtual int RegisterVoiceEngineObserver(VoiceEngineObserver& observer) = 0;

  // Removes and disables the observer class for runtime error control
  // and warning notifications. Returns 0.
  virtual int DeRegisterVoiceEngineObserver() = 0;

  // Initializes all common parts of the VoiceEngine; e.g. all
  // encoders/decoders, the sound card and core receiving components.
  // This method also makes it possible to install some user-defined external
  // modules:
  // - The Audio Device Module (ADM) which implements all the audio layer
  // functionality in a separate (reference counted) module.
  // - The AudioProcessing module handles capture-side processing. VoiceEngine
  // takes ownership of this object.
  // If NULL is passed for any of these, VoiceEngine will create its own.
  // Returns -1 in case of an error, 0 otherwise.
  // TODO(ajm): Remove default NULLs.
  virtual int Init(AudioDeviceModule* external_adm = NULL,
                   AudioProcessing* audioproc = NULL) = 0;

  // Returns NULL before Init() is called.
  virtual AudioProcessing* audio_processing() = 0;

  // Terminates all VoiceEngine functions and releases allocated resources.
  // Returns 0.
  virtual int Terminate() = 0;

  // Creates a new channel and allocates the required resources for it.
  // One can use |config| to configure the channel. Currently that is used for
  // choosing between ACM1 and ACM2, when creating Audio Coding Module.
  // Returns channel ID or -1 in case of an error.
  virtual int CreateChannel() = 0;
  virtual int CreateChannel(const Config& config) = 0;

  // Deletes an existing channel and releases the utilized resources.
  // Returns -1 in case of an error, 0 otherwise.
  virtual int DeleteChannel(int channel) = 0;

  // Prepares and initiates the VoiceEngine for reception of
  // incoming RTP/RTCP packets on the specified |channel|.
  virtual int StartReceive(int channel) = 0;

  // Stops receiving incoming RTP/RTCP packets on the specified |channel|.
  virtual int StopReceive(int channel) = 0;

  // Starts forwarding the packets to the mixer/soundcard for a
  // specified |channel|.
  virtual int StartPlayout(int channel) = 0;

  // Stops forwarding the packets to the mixer/soundcard for a
  // specified |channel|.
  virtual int StopPlayout(int channel) = 0;

  // Starts sending packets to an already specified IP address and
  // port number for a specified |channel|.
  virtual int StartSend(int channel) = 0;

  // Stops sending packets from a specified |channel|.
  virtual int StopSend(int channel) = 0;

  // Gets the version information for VoiceEngine and its components.
  virtual int GetVersion(char version[1024]) = 0;

  // Gets the last VoiceEngine error code.
  virtual int LastError() = 0;

  // TODO(xians): Make the interface pure virtual after libjingle
  // implements the interface in its FakeWebRtcVoiceEngine.
  virtual AudioTransport* audio_transport() { return NULL; }

  // Associate a send channel to a receive channel.
  // Used for obtaining RTT for a receive-only channel.
  // One should be careful not to crate a circular association, e.g.,
  // 1 <- 2 <- 1.
  virtual int AssociateSendChannel(int channel, int accociate_send_channel) = 0;

 protected:
  VoEBase() {}
  virtual ~VoEBase() {}
};

}  // namespace webrtc

#endif  //  WEBRTC_VOICE_ENGINE_VOE_BASE_H