webrtc/modules/audio_coding/neteq/packet.cc
Philipp Hancke c4fe8256f3 IWYU modules/audio_coding
using
  find modules/audio_coding -name "*.h" -o -name "*.cc" | grep -v mock_debug_dump_writer.h | xargs tools_webrtc/iwyu/apply-include-cleaner
followed by
  tools_webrtc/gn_check_autofix.py -C out/Default/
and git cl format

Manual changes:
* controller_manager.h: add api/array_view.h include
* RTPFile.cc / dtmf_buffer_unittest.cc:
  use rtc_base/ip_address.h instea of netinet/in.h
* neteq_delay_analyzer.cc: add TODO for stringstream to ostream include
* opus: fixup paths and make third_party/opus includes go via opus_interface.h by using IWYU export pragmas

The mock_debug_dump_writer has issues with protobuf includes so was
ignored for this round.

BUG=webrtc:42226242

Change-Id: I5b8613053da6cc0a2a44d1d59bd2efe11e501681
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/387640
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#44546}
2025-05-08 05:10:13 -07:00

39 lines
1003 B
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/audio_coding/neteq/packet.h"
#include "rtc_base/checks.h"
namespace webrtc {
Packet::Packet() = default;
Packet::Packet(Packet&& b) = default;
Packet::~Packet() = default;
Packet& Packet::operator=(Packet&& b) = default;
Packet Packet::Clone() const {
RTC_CHECK(!frame);
Packet clone;
clone.timestamp = timestamp;
clone.sequence_number = sequence_number;
clone.payload_type = payload_type;
clone.payload.SetData(payload.data(), payload.size());
clone.priority = priority;
clone.packet_info = packet_info;
return clone;
}
} // namespace webrtc