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Side by Side Diff: webrtc/voice_engine/channel_proxy.h

Issue 1588693002: Revert of Storing raw audio sink for default audio track. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 11 months ago
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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;
(...skipping 34 matching lines...) Expand 10 before | Expand all | Expand 10 after
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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