webrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet.cc
Erik Språng de5507d31b Updates rtcp::CompoundPacket to contain unique pointers to packets.
Currently test code passes pointer to temporary objects, while
RtcpSender passes raw pointers to objects that are then seen as owned,
and will be manually deleted by a overloaded destructor, which is scary
and fragile.

This CL moves all usage to std::unique_ptr<RtcpPacket> instead, which
may create some heap churn in unit tests but that should be fine.

Bug: webrtc:11925
Change-Id: I981bc7ccd6a74115c5a3de64b8427adbf3f16cc7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183920
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32084}
2020-09-11 14:34:07 +00:00

50 lines
1.4 KiB
C++

/*
* 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.
*/
#include "modules/rtp_rtcp/source/rtcp_packet/compound_packet.h"
#include <memory>
#include <utility>
#include "rtc_base/checks.h"
namespace webrtc {
namespace rtcp {
CompoundPacket::CompoundPacket() = default;
CompoundPacket::~CompoundPacket() = default;
void CompoundPacket::Append(std::unique_ptr<RtcpPacket> packet) {
RTC_CHECK(packet);
appended_packets_.push_back(std::move(packet));
}
bool CompoundPacket::Create(uint8_t* packet,
size_t* index,
size_t max_length,
PacketReadyCallback callback) const {
for (const auto& appended : appended_packets_) {
if (!appended->Create(packet, index, max_length, callback))
return false;
}
return true;
}
size_t CompoundPacket::BlockLength() const {
size_t block_length = 0;
for (const auto& appended : appended_packets_) {
block_length += appended->BlockLength();
}
return block_length;
}
} // namespace rtcp
} // namespace webrtc