Bug 797671: Import Webrtc.org code from stable branch 3.12 (rev 2820) rs=jesup
[gecko.git] / media / webrtc / trunk / src / voice_engine / include / voe_external_media.h
blob50d2d387d82d2e48a098e4a6549ef4dbf0c9ac08
1 /*
2 * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
11 // In some cases it is desirable to use an audio source or sink which may
12 // not be available to the VoiceEngine, such as a DV camera. This sub-API
13 // contains functions that allow for the use of such external recording
14 // sources and playout sinks. It also describes how recorded data, or data
15 // to be played out, can be modified outside the VoiceEngine.
17 // Usage example, omitting error checking:
19 // using namespace webrtc;
20 // VoiceEngine* voe = VoiceEngine::Create();
21 // VoEBase* base = VoEBase::GetInterface(voe);
22 // VoEMediaProcess media = VoEMediaProcess::GetInterface(voe);
23 // base->Init();
24 // ...
25 // media->SetExternalRecordingStatus(true);
26 // ...
27 // base->Terminate();
28 // base->Release();
29 // media->Release();
30 // VoiceEngine::Delete(voe);
32 #ifndef WEBRTC_VOICE_ENGINE_VOE_EXTERNAL_MEDIA_H
33 #define WEBRTC_VOICE_ENGINE_VOE_EXTERNAL_MEDIA_H
35 #include "common_types.h"
37 namespace webrtc {
39 class VoiceEngine;
41 class WEBRTC_DLLEXPORT VoEMediaProcess
43 public:
44 // The VoiceEngine user should override the Process() method in a
45 // derived class. Process() will be called when audio is ready to
46 // be processed. The audio can be accessed in several different modes
47 // given by the |type| parameter. The function should modify the
48 // original data and ensure that it is copied back to the |audio10ms|
49 // array. The number of samples in the frame cannot be changed.
50 // The sampling frequency will depend upon the codec used.
51 // If |isStereo| is true, audio10ms will contain 16-bit PCM data
52 // samples in interleaved stereo format (L0,R0,L1,R1,…):
53 virtual void Process(const int channel, const ProcessingTypes type,
54 WebRtc_Word16 audio10ms[], const int length,
55 const int samplingFreq, const bool isStereo) = 0;
57 protected:
58 virtual ~VoEMediaProcess() {}
61 class WEBRTC_DLLEXPORT VoEExternalMedia
63 public:
64 // Factory for the VoEExternalMedia sub-API. Increases an internal
65 // reference counter if successful. Returns NULL if the API is not
66 // supported or if construction fails.
67 static VoEExternalMedia* GetInterface(VoiceEngine* voiceEngine);
69 // Releases the VoEExternalMedia sub-API and decreases an internal
70 // reference counter. Returns the new reference count. This value should
71 // be zero for all sub-API:s before the VoiceEngine object can be safely
72 // deleted.
73 virtual int Release() = 0;
75 // Installs a VoEMediaProcess derived instance and activates external
76 // media for the specified |channel| and |type|.
77 virtual int RegisterExternalMediaProcessing(
78 int channel, ProcessingTypes type, VoEMediaProcess& processObject) = 0;
80 // Removes the VoEMediaProcess derived instance and deactivates external
81 // media for the specified |channel| and |type|.
82 virtual int DeRegisterExternalMediaProcessing(
83 int channel, ProcessingTypes type) = 0;
85 // Toogles state of external recording.
86 virtual int SetExternalRecordingStatus(bool enable) = 0;
88 // Toogles state of external playout.
89 virtual int SetExternalPlayoutStatus(bool enable) = 0;
91 // This function accepts externally recorded audio. During transmission,
92 // this method should be called at as regular an interval as possible
93 // with frames of corresponding size.
94 virtual int ExternalRecordingInsertData(
95 const WebRtc_Word16 speechData10ms[], int lengthSamples,
96 int samplingFreqHz, int current_delay_ms) = 0;
98 // This function gets audio for an external playout sink.
99 // During transmission, this function should be called every ~10 ms
100 // to obtain a new 10 ms frame of audio. The length of the block will
101 // be 160, 320, 440 or 480 samples (for 16, 32, 44 or 48 kHz sampling
102 // rates respectively).
103 virtual int ExternalPlayoutGetData(
104 WebRtc_Word16 speechData10ms[], int samplingFreqHz,
105 int current_delay_ms, int& lengthSamples) = 0;
107 protected:
108 VoEExternalMedia() {}
109 virtual ~VoEExternalMedia() {}
112 } // namespace webrtc
114 #endif // WEBRTC_VOICE_ENGINE_VOE_EXTERNAL_MEDIA_H