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Side by Side Diff: webrtc/video/video_send_stream.cc

Issue 2981163003: Refactor rtcp statistics: Rtcp module take narrow interface (Closed)
Patch Set: Created 3 years, 5 months ago
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1 /* 1 /*
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2013 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 #include "webrtc/video/video_send_stream.h" 10 #include "webrtc/video/video_send_stream.h"
(...skipping 44 matching lines...) Expand 10 before | Expand all | Expand 10 after
55 FlexfecSender* flexfec_sender, 55 FlexfecSender* flexfec_sender,
56 SendStatisticsProxy* stats_proxy, 56 SendStatisticsProxy* stats_proxy,
57 SendDelayStats* send_delay_stats, 57 SendDelayStats* send_delay_stats,
58 RtcEventLog* event_log, 58 RtcEventLog* event_log,
59 RateLimiter* retransmission_rate_limiter, 59 RateLimiter* retransmission_rate_limiter,
60 OverheadObserver* overhead_observer, 60 OverheadObserver* overhead_observer,
61 size_t num_modules, 61 size_t num_modules,
62 RtpKeepAliveConfig keepalive_config) { 62 RtpKeepAliveConfig keepalive_config) {
63 RTC_DCHECK_GT(num_modules, 0); 63 RTC_DCHECK_GT(num_modules, 0);
64 RtpRtcp::Configuration configuration; 64 RtpRtcp::Configuration configuration;
65 ReceiveStatistics* null_receive_statistics = configuration.receive_statistics;
66 configuration.audio = false; 65 configuration.audio = false;
67 configuration.receiver_only = false; 66 configuration.receiver_only = false;
68 configuration.flexfec_sender = flexfec_sender; 67 configuration.flexfec_sender = flexfec_sender;
69 configuration.receive_statistics = null_receive_statistics;
70 configuration.outgoing_transport = outgoing_transport; 68 configuration.outgoing_transport = outgoing_transport;
71 configuration.intra_frame_callback = intra_frame_callback; 69 configuration.intra_frame_callback = intra_frame_callback;
72 configuration.bandwidth_callback = bandwidth_callback; 70 configuration.bandwidth_callback = bandwidth_callback;
73 configuration.transport_feedback_callback = 71 configuration.transport_feedback_callback =
74 transport->transport_feedback_observer(); 72 transport->transport_feedback_observer();
75 configuration.rtt_stats = rtt_stats; 73 configuration.rtt_stats = rtt_stats;
76 configuration.rtcp_packet_type_counter_observer = stats_proxy; 74 configuration.rtcp_packet_type_counter_observer = stats_proxy;
77 configuration.paced_sender = transport->packet_sender(); 75 configuration.paced_sender = transport->packet_sender();
78 configuration.transport_sequence_number_allocator = 76 configuration.transport_sequence_number_allocator =
79 transport->packet_router(); 77 transport->packet_router();
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1370 std::min(config_->rtp.max_packet_size, 1368 std::min(config_->rtp.max_packet_size,
1371 kPathMTU - transport_overhead_bytes_per_packet_); 1369 kPathMTU - transport_overhead_bytes_per_packet_);
1372 1370
1373 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { 1371 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
1374 rtp_rtcp->SetMaxRtpPacketSize(rtp_packet_size); 1372 rtp_rtcp->SetMaxRtpPacketSize(rtp_packet_size);
1375 } 1373 }
1376 } 1374 }
1377 1375
1378 } // namespace internal 1376 } // namespace internal
1379 } // namespace webrtc 1377 } // namespace webrtc
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