OLD | NEW |
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 11 matching lines...) Expand all Loading... |
22 #include "webrtc/base/logging.h" | 22 #include "webrtc/base/logging.h" |
23 #include "webrtc/base/trace_event.h" | 23 #include "webrtc/base/trace_event.h" |
24 #include "webrtc/base/weak_ptr.h" | 24 #include "webrtc/base/weak_ptr.h" |
25 #include "webrtc/call/rtp_transport_controller_send_interface.h" | 25 #include "webrtc/call/rtp_transport_controller_send_interface.h" |
26 #include "webrtc/common_types.h" | 26 #include "webrtc/common_types.h" |
27 #include "webrtc/common_video/include/video_bitrate_allocator.h" | 27 #include "webrtc/common_video/include/video_bitrate_allocator.h" |
28 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h" | 28 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h" |
29 #include "webrtc/modules/congestion_controller/include/send_side_congestion_cont
roller.h" | 29 #include "webrtc/modules/congestion_controller/include/send_side_congestion_cont
roller.h" |
30 #include "webrtc/modules/pacing/packet_router.h" | 30 #include "webrtc/modules/pacing/packet_router.h" |
31 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" | 31 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" |
| 32 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h" |
32 #include "webrtc/modules/utility/include/process_thread.h" | 33 #include "webrtc/modules/utility/include/process_thread.h" |
33 #include "webrtc/modules/video_coding/utility/ivf_file_writer.h" | 34 #include "webrtc/modules/video_coding/utility/ivf_file_writer.h" |
34 #include "webrtc/system_wrappers/include/field_trial.h" | 35 #include "webrtc/system_wrappers/include/field_trial.h" |
35 #include "webrtc/video/call_stats.h" | 36 #include "webrtc/video/call_stats.h" |
36 #include "webrtc/video/payload_router.h" | 37 #include "webrtc/video/payload_router.h" |
37 #include "webrtc/video_send_stream.h" | 38 #include "webrtc/video_send_stream.h" |
38 | 39 |
39 namespace webrtc { | 40 namespace webrtc { |
40 | 41 |
41 static const int kMinSendSidePacketHistorySize = 600; | 42 static const int kMinSendSidePacketHistorySize = 600; |
(...skipping 82 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
124 "media streams, but our implementation currently only " | 125 "media streams, but our implementation currently only " |
125 "supports protecting a single media stream. " | 126 "supports protecting a single media stream. " |
126 "To avoid confusion, disabling FlexFEC completely."; | 127 "To avoid confusion, disabling FlexFEC completely."; |
127 return nullptr; | 128 return nullptr; |
128 } | 129 } |
129 | 130 |
130 RTC_DCHECK_EQ(1U, config.rtp.flexfec.protected_media_ssrcs.size()); | 131 RTC_DCHECK_EQ(1U, config.rtp.flexfec.protected_media_ssrcs.size()); |
131 return std::unique_ptr<FlexfecSender>(new FlexfecSender( | 132 return std::unique_ptr<FlexfecSender>(new FlexfecSender( |
132 config.rtp.flexfec.payload_type, config.rtp.flexfec.ssrc, | 133 config.rtp.flexfec.payload_type, config.rtp.flexfec.ssrc, |
133 config.rtp.flexfec.protected_media_ssrcs[0], config.rtp.extensions, | 134 config.rtp.flexfec.protected_media_ssrcs[0], config.rtp.extensions, |
134 Clock::GetRealTimeClock())); | 135 RTPSender::FecExtensionSizes(), Clock::GetRealTimeClock())); |
135 } | 136 } |
136 | 137 |
137 } // namespace | 138 } // namespace |
138 | 139 |
139 std::string | 140 std::string |
140 VideoSendStream::Config::EncoderSettings::ToString() const { | 141 VideoSendStream::Config::EncoderSettings::ToString() const { |
141 std::stringstream ss; | 142 std::stringstream ss; |
142 ss << "{payload_name: " << payload_name; | 143 ss << "{payload_name: " << payload_name; |
143 ss << ", payload_type: " << payload_type; | 144 ss << ", payload_type: " << payload_type; |
144 ss << ", encoder: " << (encoder ? "(VideoEncoder)" : "nullptr"); | 145 ss << ", encoder: " << (encoder ? "(VideoEncoder)" : "nullptr"); |
(...skipping 1177 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
1322 std::min(config_->rtp.max_packet_size, | 1323 std::min(config_->rtp.max_packet_size, |
1323 kPathMTU - transport_overhead_bytes_per_packet_); | 1324 kPathMTU - transport_overhead_bytes_per_packet_); |
1324 | 1325 |
1325 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { | 1326 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
1326 rtp_rtcp->SetMaxRtpPacketSize(rtp_packet_size); | 1327 rtp_rtcp->SetMaxRtpPacketSize(rtp_packet_size); |
1327 } | 1328 } |
1328 } | 1329 } |
1329 | 1330 |
1330 } // namespace internal | 1331 } // namespace internal |
1331 } // namespace webrtc | 1332 } // namespace webrtc |
OLD | NEW |