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| 1 /* | 1 /* |
| 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2015 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 |
| 11 #ifndef WEBRTC_VOICE_ENGINE_CHANNEL_PROXY_H_ | 11 #ifndef WEBRTC_VOICE_ENGINE_CHANNEL_PROXY_H_ |
| 12 #define WEBRTC_VOICE_ENGINE_CHANNEL_PROXY_H_ | 12 #define WEBRTC_VOICE_ENGINE_CHANNEL_PROXY_H_ |
| 13 | 13 |
| 14 #include "webrtc/base/scoped_ref_ptr.h" | |
| 15 #include "webrtc/base/thread_checker.h" | 14 #include "webrtc/base/thread_checker.h" |
| 16 #include "webrtc/voice_engine/channel_manager.h" | 15 #include "webrtc/voice_engine/channel_manager.h" |
| 17 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h" | 16 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h" |
| 18 | 17 |
| 19 #include <string> | 18 #include <string> |
| 20 #include <vector> | 19 #include <vector> |
| 21 | 20 |
| 22 namespace webrtc { | 21 namespace webrtc { |
| 23 | 22 |
| 24 class AudioSinkInterface; | 23 class AudioSinkInterface; |
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| 59 virtual CallStatistics GetRTCPStatistics() const; | 58 virtual CallStatistics GetRTCPStatistics() const; |
| 60 virtual std::vector<ReportBlock> GetRemoteRTCPReportBlocks() const; | 59 virtual std::vector<ReportBlock> GetRemoteRTCPReportBlocks() const; |
| 61 virtual NetworkStatistics GetNetworkStatistics() const; | 60 virtual NetworkStatistics GetNetworkStatistics() const; |
| 62 virtual AudioDecodingCallStats GetDecodingCallStatistics() const; | 61 virtual AudioDecodingCallStats GetDecodingCallStatistics() const; |
| 63 virtual int32_t GetSpeechOutputLevelFullRange() const; | 62 virtual int32_t GetSpeechOutputLevelFullRange() const; |
| 64 virtual uint32_t GetDelayEstimate() const; | 63 virtual uint32_t GetDelayEstimate() const; |
| 65 | 64 |
| 66 virtual bool SetSendTelephoneEventPayloadType(int payload_type); | 65 virtual bool SetSendTelephoneEventPayloadType(int payload_type); |
| 67 virtual bool SendTelephoneEventOutband(uint8_t event, uint32_t duration_ms); | 66 virtual bool SendTelephoneEventOutband(uint8_t event, uint32_t duration_ms); |
| 68 | 67 |
| 69 virtual void SetSink(const rtc::scoped_refptr<AudioSinkInterface>& sink); | 68 virtual void SetSink(rtc::scoped_ptr<AudioSinkInterface> sink); |
| 70 | 69 |
| 71 private: | 70 private: |
| 72 Channel* channel() const; | 71 Channel* channel() const; |
| 73 | 72 |
| 74 rtc::ThreadChecker thread_checker_; | 73 rtc::ThreadChecker thread_checker_; |
| 75 ChannelOwner channel_owner_; | 74 ChannelOwner channel_owner_; |
| 76 }; | 75 }; |
| 77 } // namespace voe | 76 } // namespace voe |
| 78 } // namespace webrtc | 77 } // namespace webrtc |
| 79 | 78 |
| 80 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_PROXY_H_ | 79 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_PROXY_H_ |
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