Index: webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h |
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h b/webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h |
new file mode 100644 |
index 0000000000000000000000000000000000000000..ad749ffb61ec4f422ae9b04acd2698bc9862a8dd |
--- /dev/null |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h |
@@ -0,0 +1,33 @@ |
+/* |
+ * Copyright (c) 2016 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. |
+ */ |
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_TO_SEND_H_ |
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_TO_SEND_H_ |
+ |
+#include "webrtc/modules/rtp_rtcp/source/rtp_packet.h" |
+ |
+namespace webrtc { |
+// Class to hold rtp packet with metadata for sender side. |
+class RtpPacketToSend : public rtp::Packet { |
+ public: |
+ explicit RtpPacketToSend(const ExtensionManager* extensions) |
+ : Packet(extensions) {} |
+ RtpPacketToSend(const ExtensionManager* extensions, size_t capacity) |
+ : Packet(extensions, capacity) {} |
+ |
+ // Time in local time base as close as it can to frame capture time. |
+ int64_t capture_time_ms() const { return capture_time_ms_; } |
+ void set_capture_time_ms(int64_t time) { capture_time_ms_ = time; } |
+ |
+ private: |
+ int64_t capture_time_ms_ = 0; |
+}; |
+ |
+} // namespace webrtc |
+#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_TO_SEND_H_ |