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1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 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 | 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 | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
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25 // ... | 25 // ... |
26 // base->Terminate(); | 26 // base->Terminate(); |
27 // base->Release(); | 27 // base->Release(); |
28 // codec->Release(); | 28 // codec->Release(); |
29 // VoiceEngine::Delete(voe); | 29 // VoiceEngine::Delete(voe); |
30 // | 30 // |
31 #ifndef WEBRTC_VOICE_ENGINE_VOE_CODEC_H | 31 #ifndef WEBRTC_VOICE_ENGINE_VOE_CODEC_H |
32 #define WEBRTC_VOICE_ENGINE_VOE_CODEC_H | 32 #define WEBRTC_VOICE_ENGINE_VOE_CODEC_H |
33 | 33 |
34 #include "webrtc/common_types.h" | 34 #include "webrtc/common_types.h" |
35 #include "webrtc/modules/audio_coding/codecs/audio_format.h" | |
35 | 36 |
36 namespace webrtc { | 37 namespace webrtc { |
37 | 38 |
38 class VoiceEngine; | 39 class VoiceEngine; |
39 | 40 |
40 class WEBRTC_DLLEXPORT VoECodec { | 41 class WEBRTC_DLLEXPORT VoECodec { |
41 public: | 42 public: |
42 // Factory for the VoECodec sub-API. Increases an internal | 43 // Factory for the VoECodec sub-API. Increases an internal |
43 // reference counter if successful. Returns NULL if the API is not | 44 // reference counter if successful. Returns NULL if the API is not |
44 // supported or if construction fails. | 45 // supported or if construction fails. |
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72 // Gets the currently received |codec| for a specific |channel|. | 73 // Gets the currently received |codec| for a specific |channel|. |
73 virtual int GetRecCodec(int channel, CodecInst& codec) = 0; | 74 virtual int GetRecCodec(int channel, CodecInst& codec) = 0; |
74 | 75 |
75 // Sets the dynamic payload type number for a particular |codec| or | 76 // Sets the dynamic payload type number for a particular |codec| or |
76 // disables (ignores) a codec for receiving. For instance, when receiving | 77 // disables (ignores) a codec for receiving. For instance, when receiving |
77 // an invite from a SIP-based client, this function can be used to change | 78 // an invite from a SIP-based client, this function can be used to change |
78 // the dynamic payload type number to match that in the INVITE SDP- | 79 // the dynamic payload type number to match that in the INVITE SDP- |
79 // message. The utilized parameters in the |codec| structure are: | 80 // message. The utilized parameters in the |codec| structure are: |
80 // plname, plfreq, pltype and channels. | 81 // plname, plfreq, pltype and channels. |
81 virtual int SetRecPayloadType(int channel, const CodecInst& codec) = 0; | 82 virtual int SetRecPayloadType(int channel, const CodecInst& codec) = 0; |
83 virtual int SetRecPayloadType(int channel, | |
the sun
2016/11/21 08:37:05
Why do we need to add this to the legacy API?
ossu
2016/11/30 12:43:43
Should we go from AudioReceiveStream directly thro
| |
84 int payload_type, | |
85 const SdpAudioFormat& format) = 0; | |
82 | 86 |
83 // Gets the actual payload type that is set for receiving a |codec| on a | 87 // Gets the actual payload type that is set for receiving a |codec| on a |
84 // |channel|. The value it retrieves will either be the default payload | 88 // |channel|. The value it retrieves will either be the default payload |
85 // type, or a value earlier set with SetRecPayloadType(). | 89 // type, or a value earlier set with SetRecPayloadType(). |
86 virtual int GetRecPayloadType(int channel, CodecInst& codec) = 0; | 90 virtual int GetRecPayloadType(int channel, CodecInst& codec) = 0; |
87 | 91 |
88 // Sets the payload |type| for the sending of SID-frames with background | 92 // Sets the payload |type| for the sending of SID-frames with background |
89 // noise estimation during silence periods detected by the VAD. | 93 // noise estimation during silence periods detected by the VAD. |
90 virtual int SetSendCNPayloadType( | 94 virtual int SetSendCNPayloadType( |
91 int channel, | 95 int channel, |
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138 virtual int GetOpusDtxStatus(int channel, bool* enabled) { return -1; } | 142 virtual int GetOpusDtxStatus(int channel, bool* enabled) { return -1; } |
139 | 143 |
140 protected: | 144 protected: |
141 VoECodec() {} | 145 VoECodec() {} |
142 virtual ~VoECodec() {} | 146 virtual ~VoECodec() {} |
143 }; | 147 }; |
144 | 148 |
145 } // namespace webrtc | 149 } // namespace webrtc |
146 | 150 |
147 #endif // WEBRTC_VOICE_ENGINE_VOE_CODEC_H | 151 #endif // WEBRTC_VOICE_ENGINE_VOE_CODEC_H |
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