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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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