/*
 *  Copyright (c) 2011 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.
 */

// In some cases it is desirable to use an audio source or sink which may
// not be available to the VoiceEngine, such as a DV camera. This sub-API
// contains functions that allow for the use of such external recording
// sources and playout sinks. It also describes how recorded data, or data
// to be played out, can be modified outside the VoiceEngine.
//
// Usage example, omitting error checking:
//
//  using namespace webrtc;
//  VoiceEngine* voe = VoiceEngine::Create();
//  VoEBase* base = VoEBase::GetInterface(voe);
//  VoEMediaProcess media = VoEMediaProcess::GetInterface(voe);
//  base->Init();
//  ...
//  media->SetExternalRecordingStatus(true);
//  ...
//  base->Terminate();
//  base->Release();
//  media->Release();
//  VoiceEngine::Delete(voe);
//
#ifndef WEBRTC_VOICE_ENGINE_VOE_EXTERNAL_MEDIA_H
#define WEBRTC_VOICE_ENGINE_VOE_EXTERNAL_MEDIA_H

#include "webrtc/common_types.h"

namespace webrtc {

class VoiceEngine;
class AudioFrame;

class WEBRTC_DLLEXPORT VoEMediaProcess {
 public:
  // The VoiceEngine user should override the Process() method in a
  // derived class. Process() will be called when audio is ready to
  // be processed. The audio can be accessed in several different modes
  // given by the |type| parameter. The function should modify the
  // original data and ensure that it is copied back to the |audio10ms|
  // array. The number of samples in the frame cannot be changed.
  // The sampling frequency will depend upon the codec used.
  // If |isStereo| is true, audio10ms will contain 16-bit PCM data
  // samples in interleaved stereo format (L0,R0,L1,R1,...).
  virtual void Process(int channel,
                       ProcessingTypes type,
                       int16_t audio10ms[],
                       size_t length,
                       int samplingFreq,
                       bool isStereo) = 0;

 protected:
  virtual ~VoEMediaProcess() {}
};

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

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

  // Installs a VoEMediaProcess derived instance and activates external
  // media for the specified |channel| and |type|.
  virtual int RegisterExternalMediaProcessing(
      int channel,
      ProcessingTypes type,
      VoEMediaProcess& processObject) = 0;

  // Removes the VoEMediaProcess derived instance and deactivates external
  // media for the specified |channel| and |type|.
  virtual int DeRegisterExternalMediaProcessing(int channel,
                                                ProcessingTypes type) = 0;

  // Toogles state of external recording.
  virtual int SetExternalRecordingStatus(bool enable) = 0;

  // Toogles state of external playout.
  virtual int SetExternalPlayoutStatus(bool enable) = 0;

  // This function accepts externally recorded audio. During transmission,
  // this method should be called at as regular an interval as possible
  // with frames of corresponding size.
  virtual int ExternalRecordingInsertData(
                                          const int16_t speechData10ms[], int lengthSamples,
                                          int samplingFreqHz, int current_delay_ms) = 0;


  // This function inserts audio written to the OS audio drivers for use
  // as the far-end signal for AEC processing.  The length of the block
  // must be 160, 320, 441 or 480 samples (for 16000, 32000, 44100 or
  // 48000 kHz sampling rates respectively).
  virtual int ExternalPlayoutData(
                                  int16_t speechData10ms[], int samplingFreqHz, int num_channels,
                                  int current_delay_ms, int& lengthSamples) = 0;

  // This function gets audio for an external playout sink.
  // During transmission, this function should be called every ~10 ms
  // to obtain a new 10 ms frame of audio. The length of the block will
  // be 160, 320, 441 or 480 samples (for 16000, 32000, 44100 or
  // 48000 kHz sampling rates respectively).
  virtual int ExternalPlayoutGetData(
                                     int16_t speechData10ms[], int samplingFreqHz,
                                     int current_delay_ms, int& lengthSamples) = 0;

  // Pulls an audio frame from the specified |channel| for external mixing.
  // If the |desired_sample_rate_hz| is 0, the signal will be returned with
  // its native frequency, otherwise it will be resampled. Valid frequencies
  // are 16000, 22050, 32000, 44100 or 48000 kHz.
  virtual int GetAudioFrame(int channel,
                            int desired_sample_rate_hz,
                            AudioFrame* frame) = 0;

  // Sets the state of external mixing. Cannot be changed during playback.
  virtual int SetExternalMixing(int channel, bool enable) = 0;

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

}  // namespace webrtc

#endif  //  WEBRTC_VOICE_ENGINE_VOE_EXTERNAL_MEDIA_H
