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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 #include "webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h" | 11 #include "webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h" |
12 | 12 |
13 #include <assert.h> | 13 #include <assert.h> |
14 #include <string.h> | 14 #include <string.h> |
15 #include <iostream> | 15 #include <iostream> |
16 #include <limits> | 16 #include <limits> |
17 | 17 |
18 #include "webrtc/base/checks.h" | 18 #include "webrtc/base/checks.h" |
| 19 #include "webrtc/call/rtc_event_log.h" |
19 #include "webrtc/modules/audio_coding/neteq/tools/packet.h" | 20 #include "webrtc/modules/audio_coding/neteq/tools/packet.h" |
20 #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" | 21 #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" |
21 #include "webrtc/video/rtc_event_log.h" | |
22 | 22 |
23 // Files generated at build-time by the protobuf compiler. | 23 // Files generated at build-time by the protobuf compiler. |
24 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD | 24 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD |
25 #include "external/webrtc/webrtc/video/rtc_event_log.pb.h" | 25 #include "external/webrtc/webrtc/call/rtc_event_log.pb.h" |
26 #else | 26 #else |
27 #include "webrtc/video/rtc_event_log.pb.h" | 27 #include "webrtc/call/rtc_event_log.pb.h" |
28 #endif | 28 #endif |
29 | 29 |
30 namespace webrtc { | 30 namespace webrtc { |
31 namespace test { | 31 namespace test { |
32 | 32 |
33 namespace { | 33 namespace { |
34 | 34 |
35 const rtclog::RtpPacket* GetRtpPacket(const rtclog::Event& event) { | 35 const rtclog::RtpPacket* GetRtpPacket(const rtclog::Event& event) { |
36 if (!event.has_type() || event.type() != rtclog::Event::RTP_EVENT) | 36 if (!event.has_type() || event.type() != rtclog::Event::RTP_EVENT) |
37 return nullptr; | 37 return nullptr; |
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118 RtcEventLogSource::RtcEventLogSource() | 118 RtcEventLogSource::RtcEventLogSource() |
119 : PacketSource(), parser_(RtpHeaderParser::Create()) {} | 119 : PacketSource(), parser_(RtpHeaderParser::Create()) {} |
120 | 120 |
121 bool RtcEventLogSource::OpenFile(const std::string& file_name) { | 121 bool RtcEventLogSource::OpenFile(const std::string& file_name) { |
122 event_log_.reset(new rtclog::EventStream()); | 122 event_log_.reset(new rtclog::EventStream()); |
123 return RtcEventLog::ParseRtcEventLog(file_name, event_log_.get()); | 123 return RtcEventLog::ParseRtcEventLog(file_name, event_log_.get()); |
124 } | 124 } |
125 | 125 |
126 } // namespace test | 126 } // namespace test |
127 } // namespace webrtc | 127 } // namespace webrtc |
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