webrtc/modules/audio_coding/neteq/tools/rtpcat.cc
Philipp Hancke 54e924ae8c IWYU modules/audio_coding
using
  find modules/audio_coding -name "*.h" -o -name "*.cc" | | xargs tools_webrtc/iwyu/apply-include-cleaner
followed by
  tools_webrtc/gn_check_autofix.py -C out/Default/
followed by running clang-tidy with
  tools/clang/scripts/build_clang_tools_extra.py \
    --fetch out/Default clang-tidy clang-apply-replacements
  ninja -C out/Default
  gn gen out/Default --export-compile-commands
  cd out/Default
  tools/clang/third_party/llvm/clang-tools-extra/clang-tidy/tool/run-clang-tidy.py -p . \
    -clang-tidy-binary out/Default/tools/clang/third_party/llvm/build/bin/clang-tidy \
    -clang-apply-replacements-binary \
        out/Default/tools/clang/third_party/llvm/build/bin/clang-apply-replacements \
    -checks='-*,modernize-deprecated-headers' \
    -fix modules/audio_coding
to move from C style standard headers to C++ style variants followed by
  git cl format
and git cl format.
Then IWYU again because it seems to understand C++ headers better than  C ones.

Manual fixes: include order in G711 and G722 unit tests

Bug: webrtc:42226242
Change-Id: Ie56ed311933ed629622da04fe51feee32acfcd2d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/395100
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#45016}
2025-06-24 14:55:50 -07:00

46 lines
1.4 KiB
C++

/*
* Copyright (c) 2011 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 <cstdio>
#include <cstdlib>
#include <memory>
#include "rtc_base/checks.h"
#include "test/rtp_file_reader.h"
#include "test/rtp_file_writer.h"
using webrtc::test::RtpFileReader;
using webrtc::test::RtpFileWriter;
int main(int argc, char* argv[]) {
if (argc < 3) {
printf("Concatenates multiple rtpdump files into one.\n");
printf("Usage: rtpcat in1.rtp int2.rtp [...] out.rtp\n");
exit(1);
}
std::unique_ptr<RtpFileWriter> output(
RtpFileWriter::Create(RtpFileWriter::kRtpDump, argv[argc - 1]));
RTC_CHECK(output.get() != nullptr) << "Cannot open output file.";
printf("Output RTP file: %s\n", argv[argc - 1]);
for (int i = 1; i < argc - 1; i++) {
std::unique_ptr<RtpFileReader> input(
RtpFileReader::Create(RtpFileReader::kRtpDump, argv[i]));
RTC_CHECK(input.get() != nullptr) << "Cannot open input file " << argv[i];
printf("Input RTP file: %s\n", argv[i]);
webrtc::test::RtpPacket packet;
while (input->NextPacket(&packet))
RTC_CHECK(output->WritePacket(&packet));
}
return 0;
}