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Side by Side Diff: webrtc/call/rtc_event_log_parser.cc

Issue 1966423002: Remove webrtc/base/scoped_ptr.h (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: remove new scoped_ptr uses Created 4 years, 7 months ago
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1 /* 1 /*
2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2016 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/call/rtc_event_log_parser.h" 11 #include "webrtc/call/rtc_event_log_parser.h"
12 12
13 #include <string.h> 13 #include <string.h>
14 14
15 #include <fstream> 15 #include <fstream>
16 16
17 #include "webrtc/base/checks.h" 17 #include "webrtc/base/checks.h"
18 #include "webrtc/base/logging.h" 18 #include "webrtc/base/logging.h"
19 #include "webrtc/base/scoped_ptr.h"
20 #include "webrtc/call.h" 19 #include "webrtc/call.h"
21 #include "webrtc/call/rtc_event_log.h" 20 #include "webrtc/call/rtc_event_log.h"
22 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" 21 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
23 #include "webrtc/system_wrappers/include/file_wrapper.h" 22 #include "webrtc/system_wrappers/include/file_wrapper.h"
24 23
25 namespace webrtc { 24 namespace webrtc {
26 25
27 namespace { 26 namespace {
28 MediaType GetRuntimeMediaType(rtclog::MediaType media_type) { 27 MediaType GetRuntimeMediaType(rtclog::MediaType media_type) {
29 switch (media_type) { 28 switch (media_type) {
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386 if (fraction_loss != nullptr) { 385 if (fraction_loss != nullptr) {
387 *fraction_loss = loss_event.fraction_loss(); 386 *fraction_loss = loss_event.fraction_loss();
388 } 387 }
389 RTC_CHECK(loss_event.has_total_packets()); 388 RTC_CHECK(loss_event.has_total_packets());
390 if (total_packets != nullptr) { 389 if (total_packets != nullptr) {
391 *total_packets = loss_event.total_packets(); 390 *total_packets = loss_event.total_packets();
392 } 391 }
393 } 392 }
394 393
395 } // namespace webrtc 394 } // namespace webrtc
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