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

Issue 1571283002: Fixes a bug which incorrectly logs incoming RTCP as outgoing. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebase Created 4 years, 11 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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36 bandwidth_callback(nullptr), 36 bandwidth_callback(nullptr),
37 transport_feedback_callback(nullptr), 37 transport_feedback_callback(nullptr),
38 rtt_stats(nullptr), 38 rtt_stats(nullptr),
39 rtcp_packet_type_counter_observer(nullptr), 39 rtcp_packet_type_counter_observer(nullptr),
40 audio_messages(NullObjectRtpAudioFeedback()), 40 audio_messages(NullObjectRtpAudioFeedback()),
41 remote_bitrate_estimator(nullptr), 41 remote_bitrate_estimator(nullptr),
42 paced_sender(nullptr), 42 paced_sender(nullptr),
43 transport_sequence_number_allocator(nullptr), 43 transport_sequence_number_allocator(nullptr),
44 send_bitrate_observer(nullptr), 44 send_bitrate_observer(nullptr),
45 send_frame_count_observer(nullptr), 45 send_frame_count_observer(nullptr),
46 send_side_delay_observer(nullptr) {} 46 send_side_delay_observer(nullptr),
47 event_log(nullptr) {}
47 48
48 RtpRtcp* RtpRtcp::CreateRtpRtcp(const RtpRtcp::Configuration& configuration) { 49 RtpRtcp* RtpRtcp::CreateRtpRtcp(const RtpRtcp::Configuration& configuration) {
49 if (configuration.clock) { 50 if (configuration.clock) {
50 return new ModuleRtpRtcpImpl(configuration); 51 return new ModuleRtpRtcpImpl(configuration);
51 } else { 52 } else {
52 // No clock implementation provided, use default clock. 53 // No clock implementation provided, use default clock.
53 RtpRtcp::Configuration configuration_copy; 54 RtpRtcp::Configuration configuration_copy;
54 memcpy(&configuration_copy, &configuration, 55 memcpy(&configuration_copy, &configuration,
55 sizeof(RtpRtcp::Configuration)); 56 sizeof(RtpRtcp::Configuration));
56 configuration_copy.clock = Clock::GetRealTimeClock(); 57 configuration_copy.clock = Clock::GetRealTimeClock();
57 return new ModuleRtpRtcpImpl(configuration_copy); 58 return new ModuleRtpRtcpImpl(configuration_copy);
58 } 59 }
59 } 60 }
60 61
61 ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration) 62 ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
62 : rtp_sender_(configuration.audio, 63 : rtp_sender_(configuration.audio,
63 configuration.clock, 64 configuration.clock,
64 configuration.outgoing_transport, 65 configuration.outgoing_transport,
65 configuration.audio_messages, 66 configuration.audio_messages,
66 configuration.paced_sender, 67 configuration.paced_sender,
67 configuration.transport_sequence_number_allocator, 68 configuration.transport_sequence_number_allocator,
68 configuration.transport_feedback_callback, 69 configuration.transport_feedback_callback,
69 configuration.send_bitrate_observer, 70 configuration.send_bitrate_observer,
70 configuration.send_frame_count_observer, 71 configuration.send_frame_count_observer,
71 configuration.send_side_delay_observer), 72 configuration.send_side_delay_observer,
73 configuration.event_log),
72 rtcp_sender_(configuration.audio, 74 rtcp_sender_(configuration.audio,
73 configuration.clock, 75 configuration.clock,
74 configuration.receive_statistics, 76 configuration.receive_statistics,
75 configuration.rtcp_packet_type_counter_observer, 77 configuration.rtcp_packet_type_counter_observer,
78 configuration.event_log,
76 configuration.outgoing_transport), 79 configuration.outgoing_transport),
77 rtcp_receiver_(configuration.clock, 80 rtcp_receiver_(configuration.clock,
78 configuration.receiver_only, 81 configuration.receiver_only,
79 configuration.rtcp_packet_type_counter_observer, 82 configuration.rtcp_packet_type_counter_observer,
80 configuration.bandwidth_callback, 83 configuration.bandwidth_callback,
81 configuration.intra_frame_callback, 84 configuration.intra_frame_callback,
82 configuration.transport_feedback_callback, 85 configuration.transport_feedback_callback,
83 this), 86 this),
84 clock_(configuration.clock), 87 clock_(configuration.clock),
85 audio_(configuration.audio), 88 audio_(configuration.audio),
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989 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback( 992 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback(
990 StreamDataCountersCallback* callback) { 993 StreamDataCountersCallback* callback) {
991 rtp_sender_.RegisterRtpStatisticsCallback(callback); 994 rtp_sender_.RegisterRtpStatisticsCallback(callback);
992 } 995 }
993 996
994 StreamDataCountersCallback* 997 StreamDataCountersCallback*
995 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const { 998 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
996 return rtp_sender_.GetRtpStatisticsCallback(); 999 return rtp_sender_.GetRtpStatisticsCallback();
997 } 1000 }
998 } // namespace webrtc 1001 } // namespace webrtc
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