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Side by Side Diff: webrtc/voice_engine/include/voe_codec.h

Issue 2110113003: Reland of "Move RtcEventLog object from inside VoiceEngine to Call.", "Fix to make the start/stop f… (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebase. Created 4 years, 5 months ago
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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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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 35
36 namespace webrtc { 36 namespace webrtc {
37 37
38 class RtcEventLog;
39 class VoiceEngine; 38 class VoiceEngine;
40 39
41 class WEBRTC_DLLEXPORT VoECodec { 40 class WEBRTC_DLLEXPORT VoECodec {
42 public: 41 public:
43 // Factory for the VoECodec sub-API. Increases an internal 42 // Factory for the VoECodec sub-API. Increases an internal
44 // reference counter if successful. Returns NULL if the API is not 43 // reference counter if successful. Returns NULL if the API is not
45 // supported or if construction fails. 44 // supported or if construction fails.
46 static VoECodec* GetInterface(VoiceEngine* voiceEngine); 45 static VoECodec* GetInterface(VoiceEngine* voiceEngine);
47 46
48 // Releases the VoECodec sub-API and decreases an internal 47 // Releases the VoECodec sub-API and decreases an internal
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125 // TODO(minyue): Make SetOpusMaxPlaybackRate() pure virtual when 124 // TODO(minyue): Make SetOpusMaxPlaybackRate() pure virtual when
126 // fakewebrtcvoiceengine in talk is ready. 125 // fakewebrtcvoiceengine in talk is ready.
127 virtual int SetOpusMaxPlaybackRate(int channel, int frequency_hz) { 126 virtual int SetOpusMaxPlaybackRate(int channel, int frequency_hz) {
128 return -1; 127 return -1;
129 } 128 }
130 129
131 // If send codec is Opus on a specified |channel|, set its DTX. Returns 0 if 130 // If send codec is Opus on a specified |channel|, set its DTX. Returns 0 if
132 // success, and -1 if failed. 131 // success, and -1 if failed.
133 virtual int SetOpusDtx(int channel, bool enable_dtx) = 0; 132 virtual int SetOpusDtx(int channel, bool enable_dtx) = 0;
134 133
135 // Get a pointer to the event logging object associated with this Voice
136 // Engine. This pointer will remain valid until VoiceEngine is destroyed.
137 virtual RtcEventLog* GetEventLog() = 0;
138
139 protected: 134 protected:
140 VoECodec() {} 135 VoECodec() {}
141 virtual ~VoECodec() {} 136 virtual ~VoECodec() {}
142 }; 137 };
143 138
144 } // namespace webrtc 139 } // namespace webrtc
145 140
146 #endif // WEBRTC_VOICE_ENGINE_VOE_CODEC_H 141 #endif // WEBRTC_VOICE_ENGINE_VOE_CODEC_H
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