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1 /* | 1 /* |
2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2016 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 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_TO_SEND_H_ | 10 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_TO_SEND_H_ |
11 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_TO_SEND_H_ | 11 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_TO_SEND_H_ |
12 | 12 |
13 #include "webrtc/modules/rtp_rtcp/source/rtp_packet.h" | 13 #include "webrtc/modules/rtp_rtcp/source/rtp_packet.h" |
14 | 14 |
15 namespace webrtc { | 15 namespace webrtc { |
16 // Class to hold rtp packet with metadata for sender side. | 16 // Class to hold rtp packet with metadata for sender side. |
17 class RtpPacketToSend : public rtp::Packet { | 17 class RtpPacketToSend : public rtp::Packet { |
18 public: | 18 public: |
19 explicit RtpPacketToSend(const ExtensionManager* extensions) | 19 explicit RtpPacketToSend(const ExtensionManager* extensions) |
20 : Packet(extensions) {} | 20 : Packet(extensions) {} |
| 21 RtpPacketToSend(const RtpPacketToSend& packet) = default; |
21 RtpPacketToSend(const ExtensionManager* extensions, size_t capacity) | 22 RtpPacketToSend(const ExtensionManager* extensions, size_t capacity) |
22 : Packet(extensions, capacity) {} | 23 : Packet(extensions, capacity) {} |
23 | 24 |
| 25 RtpPacketToSend& operator=(const RtpPacketToSend& packet) = default; |
24 // Time in local time base as close as it can to frame capture time. | 26 // 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_; } | 27 int64_t capture_time_ms() const { return capture_time_ms_; } |
26 void set_capture_time_ms(int64_t time) { capture_time_ms_ = time; } | 28 void set_capture_time_ms(int64_t time) { capture_time_ms_ = time; } |
27 | 29 |
28 private: | 30 private: |
29 int64_t capture_time_ms_ = 0; | 31 int64_t capture_time_ms_ = 0; |
30 }; | 32 }; |
31 | 33 |
32 } // namespace webrtc | 34 } // namespace webrtc |
33 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_TO_SEND_H_ | 35 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_TO_SEND_H_ |
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