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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/base/checks.h" | 18 #include "webrtc/base/checks.h" |
19 #include "webrtc/base/scoped_ptr.h" | 19 #include "webrtc/base/scoped_ptr.h" |
20 #include "webrtc/base/thread_annotations.h" | 20 #include "webrtc/base/thread_annotations.h" |
21 #include "webrtc/base/thread_checker.h" | 21 #include "webrtc/base/thread_checker.h" |
| 22 #include "webrtc/base/trace_event.h" |
22 #include "webrtc/call.h" | 23 #include "webrtc/call.h" |
23 #include "webrtc/call/rtc_event_log.h" | 24 #include "webrtc/call/rtc_event_log.h" |
24 #include "webrtc/common.h" | 25 #include "webrtc/common.h" |
25 #include "webrtc/config.h" | 26 #include "webrtc/config.h" |
26 #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" | 27 #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" |
27 #include "webrtc/modules/rtp_rtcp/source/byte_io.h" | 28 #include "webrtc/modules/rtp_rtcp/source/byte_io.h" |
28 #include "webrtc/modules/utility/interface/process_thread.h" | 29 #include "webrtc/modules/utility/interface/process_thread.h" |
29 #include "webrtc/system_wrappers/interface/cpu_info.h" | 30 #include "webrtc/system_wrappers/interface/cpu_info.h" |
30 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" | 31 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" |
31 #include "webrtc/system_wrappers/interface/logging.h" | 32 #include "webrtc/system_wrappers/interface/logging.h" |
32 #include "webrtc/system_wrappers/interface/rw_lock_wrapper.h" | 33 #include "webrtc/system_wrappers/interface/rw_lock_wrapper.h" |
33 #include "webrtc/system_wrappers/interface/trace.h" | 34 #include "webrtc/system_wrappers/interface/trace.h" |
34 #include "webrtc/system_wrappers/interface/trace_event.h" | |
35 #include "webrtc/video/video_receive_stream.h" | 35 #include "webrtc/video/video_receive_stream.h" |
36 #include "webrtc/video/video_send_stream.h" | 36 #include "webrtc/video/video_send_stream.h" |
37 #include "webrtc/voice_engine/include/voe_codec.h" | 37 #include "webrtc/voice_engine/include/voe_codec.h" |
38 | 38 |
39 namespace webrtc { | 39 namespace webrtc { |
40 | 40 |
41 const int Call::Config::kDefaultStartBitrateBps = 300000; | 41 const int Call::Config::kDefaultStartBitrateBps = 300000; |
42 | 42 |
43 namespace internal { | 43 namespace internal { |
44 | 44 |
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592 // thread. Then this check can be enabled. | 592 // thread. Then this check can be enabled. |
593 // RTC_DCHECK(!configuration_thread_checker_.CalledOnValidThread()); | 593 // RTC_DCHECK(!configuration_thread_checker_.CalledOnValidThread()); |
594 if (RtpHeaderParser::IsRtcp(packet, length)) | 594 if (RtpHeaderParser::IsRtcp(packet, length)) |
595 return DeliverRtcp(media_type, packet, length); | 595 return DeliverRtcp(media_type, packet, length); |
596 | 596 |
597 return DeliverRtp(media_type, packet, length, packet_time); | 597 return DeliverRtp(media_type, packet, length, packet_time); |
598 } | 598 } |
599 | 599 |
600 } // namespace internal | 600 } // namespace internal |
601 } // namespace webrtc | 601 } // namespace webrtc |
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