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Side by Side Diff: webrtc/audio/audio_receive_stream.cc

Issue 1227923005: Split webrtc/video into webrtc/{audio,call,video}. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 5 years, 2 months ago
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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/video/audio_receive_stream.h" 11 #include "webrtc/audio/audio_receive_stream.h"
12 12
13 #include <string> 13 #include <string>
14 14
15 #include "webrtc/base/checks.h" 15 #include "webrtc/base/checks.h"
16 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimat or.h" 16 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimat or.h"
17 #include "webrtc/system_wrappers/interface/tick_util.h" 17 #include "webrtc/system_wrappers/interface/tick_util.h"
18 18
19 namespace webrtc { 19 namespace webrtc {
20 std::string AudioReceiveStream::Config::Rtp::ToString() const { 20 std::string AudioReceiveStream::Config::Rtp::ToString() const {
21 std::stringstream ss; 21 std::stringstream ss;
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104 if (packet_time.timestamp >= 0) 104 if (packet_time.timestamp >= 0)
105 arrival_time_ms = (packet_time.timestamp + 500) / 1000; 105 arrival_time_ms = (packet_time.timestamp + 500) / 1000;
106 size_t payload_size = length - header.headerLength; 106 size_t payload_size = length - header.headerLength;
107 remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_size, 107 remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_size,
108 header, false); 108 header, false);
109 } 109 }
110 return true; 110 return true;
111 } 111 }
112 } // namespace internal 112 } // namespace internal
113 } // namespace webrtc 113 } // namespace webrtc
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