webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h - Issue 1945773002: RtpPacketHistory rewritten to use RtpPacket class.

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Unified Diff: webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h

Issue 1945773002: RtpPacketHistory rewritten to use RtpPacket class. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: rebase Created 4 years, 4 months ago

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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

index ad749ffb61ec4f422ae9b04acd2698bc9862a8dd..f2ddc8a52e467801cc3ed5e72af5fc1945e96ca2 100644

--- a/webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h

+++ b/webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h

@@ -18,9 +18,11 @@ class RtpPacketToSend : public rtp::Packet {

public:

explicit RtpPacketToSend(const ExtensionManager* extensions)

: Packet(extensions) {}

+ RtpPacketToSend(const RtpPacketToSend& packet) = default;

RtpPacketToSend(const ExtensionManager* extensions, size_t capacity)

: Packet(extensions, capacity) {}

+ RtpPacketToSend& operator=(const RtpPacketToSend& packet) = default;

// 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; }