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Side by Side Diff: webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc

Issue 1740133002: Simplify registration of RTP-header extensions. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: header Created 4 years, 9 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
11 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h" 11 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
12 12
13 #include <string.h> 13 #include <string.h>
14 14
15 #include <set> 15 #include <set>
16 16
17 #include "webrtc/base/checks.h" 17 #include "webrtc/base/checks.h"
18 #include "webrtc/base/logging.h" 18 #include "webrtc/base/logging.h"
19 #include "webrtc/common_types.h" 19 #include "webrtc/common_types.h"
20 #include "webrtc/config.h"
20 #include "webrtc/system_wrappers/include/trace.h" 21 #include "webrtc/system_wrappers/include/trace.h"
21 22
22 #ifdef _WIN32 23 #ifdef _WIN32
23 // Disable warning C4355: 'this' : used in base member initializer list. 24 // Disable warning C4355: 'this' : used in base member initializer list.
24 #pragma warning(disable : 4355) 25 #pragma warning(disable : 4355)
25 #endif 26 #endif
26 27
27 namespace webrtc { 28 namespace webrtc {
28 29
30 RTPExtensionType StringToRtpExtensionType(const std::string& extension) {
31 if (extension == RtpExtension::kTOffset)
32 return kRtpExtensionTransmissionTimeOffset;
33 if (extension == RtpExtension::kAudioLevel)
34 return kRtpExtensionAudioLevel;
35 if (extension == RtpExtension::kAbsSendTime)
36 return kRtpExtensionAbsoluteSendTime;
37 if (extension == RtpExtension::kVideoRotation)
38 return kRtpExtensionVideoRotation;
39 if (extension == RtpExtension::kTransportSequenceNumber)
40 return kRtpExtensionTransportSequenceNumber;
41 RTC_NOTREACHED() << "Looking up unsupported RTP extension.";
42 return kRtpExtensionNone;
43 }
44
29 RtpRtcp::Configuration::Configuration() 45 RtpRtcp::Configuration::Configuration()
30 : audio(false), 46 : audio(false),
31 receiver_only(false), 47 receiver_only(false),
32 clock(nullptr), 48 clock(nullptr),
33 receive_statistics(NullObjectReceiveStatistics()), 49 receive_statistics(NullObjectReceiveStatistics()),
34 outgoing_transport(nullptr), 50 outgoing_transport(nullptr),
35 intra_frame_callback(nullptr), 51 intra_frame_callback(nullptr),
36 bandwidth_callback(nullptr), 52 bandwidth_callback(nullptr),
37 transport_feedback_callback(nullptr), 53 transport_feedback_callback(nullptr),
38 rtt_stats(nullptr), 54 rtt_stats(nullptr),
(...skipping 938 matching lines...) Expand 10 before | Expand all | Expand 10 after
977 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback( 993 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback(
978 StreamDataCountersCallback* callback) { 994 StreamDataCountersCallback* callback) {
979 rtp_sender_.RegisterRtpStatisticsCallback(callback); 995 rtp_sender_.RegisterRtpStatisticsCallback(callback);
980 } 996 }
981 997
982 StreamDataCountersCallback* 998 StreamDataCountersCallback*
983 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const { 999 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
984 return rtp_sender_.GetRtpStatisticsCallback(); 1000 return rtp_sender_.GetRtpStatisticsCallback();
985 } 1001 }
986 } // namespace webrtc 1002 } // namespace webrtc
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