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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 | 10 |
11 #include <string.h> | 11 #include <string.h> |
12 | 12 |
13 #include <map> | 13 #include <map> |
14 #include <vector> | 14 #include <vector> |
15 | 15 |
16 #include "webrtc/audio/audio_receive_stream.h" | 16 #include "webrtc/audio/audio_receive_stream.h" |
17 #include "webrtc/audio/audio_send_stream.h" | 17 #include "webrtc/audio/audio_send_stream.h" |
18 #include "webrtc/audio/audio_state.h" | 18 #include "webrtc/audio/audio_state.h" |
19 #include "webrtc/audio/scoped_voe_interface.h" | 19 #include "webrtc/audio/scoped_voe_interface.h" |
20 #include "webrtc/base/checks.h" | 20 #include "webrtc/base/checks.h" |
21 #include "webrtc/base/logging.h" | |
21 #include "webrtc/base/scoped_ptr.h" | 22 #include "webrtc/base/scoped_ptr.h" |
22 #include "webrtc/base/thread_annotations.h" | 23 #include "webrtc/base/thread_annotations.h" |
23 #include "webrtc/base/thread_checker.h" | 24 #include "webrtc/base/thread_checker.h" |
24 #include "webrtc/base/trace_event.h" | 25 #include "webrtc/base/trace_event.h" |
25 #include "webrtc/call.h" | 26 #include "webrtc/call.h" |
26 #include "webrtc/call/bitrate_allocator.h" | 27 #include "webrtc/call/bitrate_allocator.h" |
27 #include "webrtc/call/congestion_controller.h" | 28 #include "webrtc/call/congestion_controller.h" |
28 #include "webrtc/call/rtc_event_log.h" | 29 #include "webrtc/call/rtc_event_log.h" |
29 #include "webrtc/common.h" | 30 #include "webrtc/common.h" |
30 #include "webrtc/config.h" | 31 #include "webrtc/config.h" |
31 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h" | 32 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h" |
32 #include "webrtc/modules/pacing/paced_sender.h" | 33 #include "webrtc/modules/pacing/paced_sender.h" |
33 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" | 34 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" |
34 #include "webrtc/modules/rtp_rtcp/source/byte_io.h" | 35 #include "webrtc/modules/rtp_rtcp/source/byte_io.h" |
35 #include "webrtc/modules/utility/include/process_thread.h" | 36 #include "webrtc/modules/utility/include/process_thread.h" |
36 #include "webrtc/system_wrappers/include/cpu_info.h" | 37 #include "webrtc/system_wrappers/include/cpu_info.h" |
37 #include "webrtc/system_wrappers/include/critical_section_wrapper.h" | 38 #include "webrtc/system_wrappers/include/critical_section_wrapper.h" |
38 #include "webrtc/system_wrappers/include/logging.h" | |
39 #include "webrtc/system_wrappers/include/metrics.h" | 39 #include "webrtc/system_wrappers/include/metrics.h" |
40 #include "webrtc/system_wrappers/include/rw_lock_wrapper.h" | 40 #include "webrtc/system_wrappers/include/rw_lock_wrapper.h" |
41 #include "webrtc/system_wrappers/include/trace.h" | 41 #include "webrtc/system_wrappers/include/trace.h" |
42 #include "webrtc/video/video_receive_stream.h" | 42 #include "webrtc/video/video_receive_stream.h" |
43 #include "webrtc/video/video_send_stream.h" | 43 #include "webrtc/video/video_send_stream.h" |
44 #include "webrtc/video_engine/call_stats.h" | 44 #include "webrtc/video_engine/call_stats.h" |
45 #include "webrtc/voice_engine/include/voe_codec.h" | 45 #include "webrtc/voice_engine/include/voe_codec.h" |
46 | 46 |
47 namespace webrtc { | 47 namespace webrtc { |
48 | 48 |
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119 | 119 |
120 const Clock* const clock_; | 120 const Clock* const clock_; |
121 | 121 |
122 const int num_cpu_cores_; | 122 const int num_cpu_cores_; |
123 const rtc::scoped_ptr<ProcessThread> module_process_thread_; | 123 const rtc::scoped_ptr<ProcessThread> module_process_thread_; |
124 const rtc::scoped_ptr<CallStats> call_stats_; | 124 const rtc::scoped_ptr<CallStats> call_stats_; |
125 const rtc::scoped_ptr<BitrateAllocator> bitrate_allocator_; | 125 const rtc::scoped_ptr<BitrateAllocator> bitrate_allocator_; |
126 Call::Config config_; | 126 Call::Config config_; |
127 rtc::ThreadChecker configuration_thread_checker_; | 127 rtc::ThreadChecker configuration_thread_checker_; |
128 | 128 |
129 bool network_enabled_; | 129 bool network_enabled_; |
pbos-webrtc
2015/12/03 22:58:35
(part of rebase)
| |
130 | 130 |
131 rtc::scoped_ptr<RWLockWrapper> receive_crit_; | 131 rtc::scoped_ptr<RWLockWrapper> receive_crit_; |
132 // Audio and Video receive streams are owned by the client that creates them. | 132 // Audio and Video receive streams are owned by the client that creates them. |
133 std::map<uint32_t, AudioReceiveStream*> audio_receive_ssrcs_ | 133 std::map<uint32_t, AudioReceiveStream*> audio_receive_ssrcs_ |
134 GUARDED_BY(receive_crit_); | 134 GUARDED_BY(receive_crit_); |
135 std::map<uint32_t, VideoReceiveStream*> video_receive_ssrcs_ | 135 std::map<uint32_t, VideoReceiveStream*> video_receive_ssrcs_ |
136 GUARDED_BY(receive_crit_); | 136 GUARDED_BY(receive_crit_); |
137 std::set<VideoReceiveStream*> video_receive_streams_ | 137 std::set<VideoReceiveStream*> video_receive_streams_ |
138 GUARDED_BY(receive_crit_); | 138 GUARDED_BY(receive_crit_); |
139 std::map<std::string, AudioReceiveStream*> sync_stream_mapping_ | 139 std::map<std::string, AudioReceiveStream*> sync_stream_mapping_ |
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735 // thread. Then this check can be enabled. | 735 // thread. Then this check can be enabled. |
736 // RTC_DCHECK(!configuration_thread_checker_.CalledOnValidThread()); | 736 // RTC_DCHECK(!configuration_thread_checker_.CalledOnValidThread()); |
737 if (RtpHeaderParser::IsRtcp(packet, length)) | 737 if (RtpHeaderParser::IsRtcp(packet, length)) |
738 return DeliverRtcp(media_type, packet, length); | 738 return DeliverRtcp(media_type, packet, length); |
739 | 739 |
740 return DeliverRtp(media_type, packet, length, packet_time); | 740 return DeliverRtp(media_type, packet, length, packet_time); |
741 } | 741 } |
742 | 742 |
743 } // namespace internal | 743 } // namespace internal |
744 } // namespace webrtc | 744 } // namespace webrtc |
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