Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(31)

Side by Side Diff: webrtc/modules/rtp_rtcp/include/rtp_receiver.h

Issue 2809613002: Revert of Implemented the GetSources() in native code. (Closed)
Patch Set: Created 3 years, 8 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
« no previous file with comments | « webrtc/modules/rtp_rtcp/BUILD.gn ('k') | webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
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
11 #ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_ 11 #ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_
12 #define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_ 12 #define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_
13 13
14 #include <vector>
15
16 #include "webrtc/api/rtpreceiverinterface.h"
17 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" 14 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
18 #include "webrtc/typedefs.h" 15 #include "webrtc/typedefs.h"
19 16
20 namespace webrtc { 17 namespace webrtc {
21 18
22 struct CodecInst; 19 struct CodecInst;
23 class RTPPayloadRegistry; 20 class RTPPayloadRegistry;
24 class VideoCodec; 21 class VideoCodec;
25 22
26 class TelephoneEventHandler { 23 class TelephoneEventHandler {
(...skipping 58 matching lines...) Expand 10 before | Expand all | Expand 10 after
85 virtual bool LastReceivedTimeMs(int64_t* receive_time_ms) const = 0; 82 virtual bool LastReceivedTimeMs(int64_t* receive_time_ms) const = 0;
86 83
87 // Returns the remote SSRC of the currently received RTP stream. 84 // Returns the remote SSRC of the currently received RTP stream.
88 virtual uint32_t SSRC() const = 0; 85 virtual uint32_t SSRC() const = 0;
89 86
90 // Returns the current remote CSRCs. 87 // Returns the current remote CSRCs.
91 virtual int32_t CSRCs(uint32_t array_of_csrc[kRtpCsrcSize]) const = 0; 88 virtual int32_t CSRCs(uint32_t array_of_csrc[kRtpCsrcSize]) const = 0;
92 89
93 // Returns the current energy of the RTP stream received. 90 // Returns the current energy of the RTP stream received.
94 virtual int32_t Energy(uint8_t array_of_energy[kRtpCsrcSize]) const = 0; 91 virtual int32_t Energy(uint8_t array_of_energy[kRtpCsrcSize]) const = 0;
95
96 virtual std::vector<RtpSource> GetSources() const = 0;
97 }; 92 };
98 } // namespace webrtc 93 } // namespace webrtc
99 94
100 #endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_ 95 #endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_
OLDNEW
« no previous file with comments | « webrtc/modules/rtp_rtcp/BUILD.gn ('k') | webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698