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Side by Side Diff: webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h

Issue 1841453004: RtpPacket class introduced. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: added empty lines Created 4 years, 8 months ago
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1 /*
2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
3 *
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
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_TO_SEND_H_
11 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_TO_SEND_H_
12
13 #include "webrtc/modules/rtp_rtcp/source/rtp_packet.h"
14
15 namespace webrtc {
16 // Class to hold rtp packet with metadata for sender side.
17 class RtpPacketToSend : public rtp::Packet {
18 public:
19 explicit RtpPacketToSend(const ExtensionManager* extensions)
20 : Packet(extensions) {}
21 RtpPacketToSend(const ExtensionManager* extensions, size_t capacity)
22 : Packet(extensions, capacity) {}
23
24 // Time in local time base as close as it can to frame capture time.
25 int64_t capture_time_ms() const { return capture_time_ms_; }
26 void set_capture_time_ms(int64_t time) { capture_time_ms_ = time; }
27
28 private:
29 int64_t capture_time_ms_ = 0;
30 };
31
32 } // namespace webrtc
33 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_TO_SEND_H_
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