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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" |
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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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