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1 /* | 1 /* |
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2015 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 #ifdef ENABLE_RTC_EVENT_LOG | 11 #ifdef ENABLE_RTC_EVENT_LOG |
12 | 12 |
13 #include <stdio.h> | 13 #include <stdio.h> |
14 #include <string> | 14 #include <string> |
15 #include <vector> | 15 #include <vector> |
16 | 16 |
17 #include "testing/gtest/include/gtest/gtest.h" | 17 #include "testing/gtest/include/gtest/gtest.h" |
18 #include "webrtc/base/buffer.h" | 18 #include "webrtc/base/buffer.h" |
19 #include "webrtc/base/checks.h" | 19 #include "webrtc/base/checks.h" |
20 #include "webrtc/base/scoped_ptr.h" | 20 #include "webrtc/base/scoped_ptr.h" |
21 #include "webrtc/call.h" | 21 #include "webrtc/call.h" |
| 22 #include "webrtc/call/rtc_event_log.h" |
22 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h" | 23 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h" |
23 #include "webrtc/system_wrappers/interface/clock.h" | 24 #include "webrtc/system_wrappers/interface/clock.h" |
24 #include "webrtc/test/test_suite.h" | 25 #include "webrtc/test/test_suite.h" |
25 #include "webrtc/test/testsupport/fileutils.h" | 26 #include "webrtc/test/testsupport/fileutils.h" |
26 #include "webrtc/test/testsupport/gtest_disable.h" | 27 #include "webrtc/test/testsupport/gtest_disable.h" |
27 #include "webrtc/video/rtc_event_log.h" | |
28 | 28 |
29 // Files generated at build-time by the protobuf compiler. | 29 // Files generated at build-time by the protobuf compiler. |
30 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD | 30 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD |
31 #include "external/webrtc/webrtc/video/rtc_event_log.pb.h" | 31 #include "external/webrtc/webrtc/call/rtc_event_log.pb.h" |
32 #else | 32 #else |
33 #include "webrtc/video/rtc_event_log.pb.h" | 33 #include "webrtc/call/rtc_event_log.pb.h" |
34 #endif | 34 #endif |
35 | 35 |
36 namespace webrtc { | 36 namespace webrtc { |
37 | 37 |
38 namespace { | 38 namespace { |
39 | 39 |
40 const RTPExtensionType kExtensionTypes[] = { | 40 const RTPExtensionType kExtensionTypes[] = { |
41 RTPExtensionType::kRtpExtensionTransmissionTimeOffset, | 41 RTPExtensionType::kRtpExtensionTransmissionTimeOffset, |
42 RTPExtensionType::kRtpExtensionAudioLevel, | 42 RTPExtensionType::kRtpExtensionAudioLevel, |
43 RTPExtensionType::kRtpExtensionAbsoluteSendTime, | 43 RTPExtensionType::kRtpExtensionAbsoluteSendTime, |
44 RTPExtensionType::kRtpExtensionVideoRotation, | 44 RTPExtensionType::kRtpExtensionVideoRotation, |
45 RTPExtensionType::kRtpExtensionTransportSequenceNumber}; | 45 RTPExtensionType::kRtpExtensionTransportSequenceNumber}; |
46 const char* kExtensionNames[] = {RtpExtension::kTOffset, | 46 const char* kExtensionNames[] = {RtpExtension::kTOffset, |
47 RtpExtension::kAudioLevel, | 47 RtpExtension::kAudioLevel, |
48 RtpExtension::kAbsSendTime, | 48 RtpExtension::kAbsSendTime, |
49 RtpExtension::kVideoRotation, | 49 RtpExtension::kVideoRotation, |
50 RtpExtension::kTransportSequenceNumber}; | 50 RtpExtension::kTransportSequenceNumber}; |
51 const size_t kNumExtensions = 5; | 51 const size_t kNumExtensions = 5; |
52 | 52 |
53 } // namepsace | 53 } // namespace |
54 | 54 |
55 // TODO(terelius): Place this definition with other parsing functions? | 55 // TODO(terelius): Place this definition with other parsing functions? |
56 MediaType GetRuntimeMediaType(rtclog::MediaType media_type) { | 56 MediaType GetRuntimeMediaType(rtclog::MediaType media_type) { |
57 switch (media_type) { | 57 switch (media_type) { |
58 case rtclog::MediaType::ANY: | 58 case rtclog::MediaType::ANY: |
59 return MediaType::ANY; | 59 return MediaType::ANY; |
60 case rtclog::MediaType::AUDIO: | 60 case rtclog::MediaType::AUDIO: |
61 return MediaType::AUDIO; | 61 return MediaType::AUDIO; |
62 case rtclog::MediaType::VIDEO: | 62 case rtclog::MediaType::VIDEO: |
63 return MediaType::VIDEO; | 63 return MediaType::VIDEO; |
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545 extensions, // Bit vector choosing extensions | 545 extensions, // Bit vector choosing extensions |
546 csrcs_count, // Number of contributing sources | 546 csrcs_count, // Number of contributing sources |
547 rand()); | 547 rand()); |
548 } | 548 } |
549 } | 549 } |
550 } | 550 } |
551 | 551 |
552 } // namespace webrtc | 552 } // namespace webrtc |
553 | 553 |
554 #endif // ENABLE_RTC_EVENT_LOG | 554 #endif // ENABLE_RTC_EVENT_LOG |
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