| /* | |
| * Copyright 2018 The WebRTC Project Authors. All rights reserved. | |
| * | |
| * Use of this source code is governed by a BSD-style license | |
| * that can be found in the LICENSE file in the root of the source | |
| * tree. An additional intellectual property rights grant can be found | |
| * in the file PATENTS. All contributing project authors may | |
| * be found in the AUTHORS file in the root of the source tree. | |
| */ | |
| #include "rtc_base/socket.h" | |
| #include <cstdint> | |
| #include "api/units/timestamp.h" | |
| #include "rtc_base/buffer.h" | |
| namespace webrtc { | |
| int Socket::RecvFrom(ReceiveBuffer& buffer) { | |
| static constexpr int BUF_SIZE = 64 * 1024; | |
| int64_t timestamp = -1; | |
| buffer.payload.EnsureCapacity(BUF_SIZE); | |
| int len = RecvFrom(buffer.payload.data(), buffer.payload.capacity(), | |
| &buffer.source_address, ×tamp); | |
| buffer.payload.SetSize(len > 0 ? len : 0); | |
| if (len > 0 && timestamp != -1) { | |
| buffer.arrival_time = Timestamp::Micros(timestamp); | |
| } | |
| return len; | |
| } | |
| } // namespace webrtc |