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* 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);
#include "common_types.h"
namespace webrtc {
class VoiceEngine;
class WEBRTC_DLLEXPORT VoEMediaProcess
// 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(const int channel, const ProcessingTypes type,
WebRtc_Word16 audio10ms[], const int length,
const int samplingFreq, const bool isStereo) = 0;
virtual ~VoEMediaProcess() {}
class WEBRTC_DLLEXPORT VoEExternalMedia
// 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 WebRtc_Word16 speechData10ms[], int lengthSamples,
int samplingFreqHz, int current_delay_ms) = 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, 440 or 480 samples (for 16, 32, 44 or 48 kHz sampling
// rates respectively).
virtual int ExternalPlayoutGetData(
WebRtc_Word16 speechData10ms[], int samplingFreqHz,
int current_delay_ms, int& lengthSamples) = 0;
VoEExternalMedia() {}
virtual ~VoEExternalMedia() {}
} // namespace webrtc