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Side by Side Diff: webrtc/modules/audio_coding/main/test/Channel.cc

Issue 1413333002: system_wrappers: rename interface -> include (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebased again! Created 5 years, 1 month ago
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1 /* 1 /*
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2012 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/main/test/Channel.h" 11 #include "webrtc/modules/audio_coding/main/test/Channel.h"
12 12
13 #include <assert.h> 13 #include <assert.h>
14 #include <iostream> 14 #include <iostream>
15 15
16 #include "webrtc/base/format_macros.h" 16 #include "webrtc/base/format_macros.h"
17 #include "webrtc/system_wrappers/interface/tick_util.h" 17 #include "webrtc/system_wrappers/include/tick_util.h"
18 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" 18 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
19 19
20 namespace webrtc { 20 namespace webrtc {
21 21
22 int32_t Channel::SendData(FrameType frameType, 22 int32_t Channel::SendData(FrameType frameType,
23 uint8_t payloadType, 23 uint8_t payloadType,
24 uint32_t timeStamp, 24 uint32_t timeStamp,
25 const uint8_t* payloadData, 25 const uint8_t* payloadData,
26 size_t payloadSize, 26 size_t payloadSize,
27 const RTPFragmentationHeader* fragmentation) { 27 const RTPFragmentationHeader* fragmentation) {
28 WebRtcRTPHeader rtpInfo; 28 WebRtcRTPHeader rtpInfo;
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415 double Channel::BitRate() { 415 double Channel::BitRate() {
416 double rate; 416 double rate;
417 uint64_t currTime = TickTime::MillisecondTimestamp(); 417 uint64_t currTime = TickTime::MillisecondTimestamp();
418 _channelCritSect->Enter(); 418 _channelCritSect->Enter();
419 rate = ((double) _totalBytes * 8.0) / (double) (currTime - _beginTime); 419 rate = ((double) _totalBytes * 8.0) / (double) (currTime - _beginTime);
420 _channelCritSect->Leave(); 420 _channelCritSect->Leave();
421 return rate; 421 return rate;
422 } 422 }
423 423
424 } // namespace webrtc 424 } // namespace webrtc
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