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317a1f09ed
WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
101 lines
3.2 KiB
C++
101 lines
3.2 KiB
C++
/*
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* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/audio_coding/neteq/tools/rtp_file_source.h"
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#include <assert.h>
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#include <string.h>
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#ifndef WIN32
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#include <netinet/in.h>
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#endif
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#include <memory>
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#include "modules/audio_coding/neteq/tools/packet.h"
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#include "rtc_base/checks.h"
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#include "test/rtp_file_reader.h"
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namespace webrtc {
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namespace test {
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RtpFileSource* RtpFileSource::Create(const std::string& file_name,
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absl::optional<uint32_t> ssrc_filter) {
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RtpFileSource* source = new RtpFileSource(ssrc_filter);
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RTC_CHECK(source->OpenFile(file_name));
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return source;
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}
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bool RtpFileSource::ValidRtpDump(const std::string& file_name) {
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std::unique_ptr<RtpFileReader> temp_file(
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RtpFileReader::Create(RtpFileReader::kRtpDump, file_name));
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return !!temp_file;
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}
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bool RtpFileSource::ValidPcap(const std::string& file_name) {
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std::unique_ptr<RtpFileReader> temp_file(
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RtpFileReader::Create(RtpFileReader::kPcap, file_name));
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return !!temp_file;
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}
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RtpFileSource::~RtpFileSource() {}
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bool RtpFileSource::RegisterRtpHeaderExtension(RTPExtensionType type,
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uint8_t id) {
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return rtp_header_extension_map_.RegisterByType(id, type);
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}
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std::unique_ptr<Packet> RtpFileSource::NextPacket() {
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while (true) {
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RtpPacket temp_packet;
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if (!rtp_reader_->NextPacket(&temp_packet)) {
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return NULL;
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}
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if (temp_packet.original_length == 0) {
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// May be an RTCP packet.
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// Read the next one.
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continue;
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}
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std::unique_ptr<uint8_t[]> packet_memory(new uint8_t[temp_packet.length]);
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memcpy(packet_memory.get(), temp_packet.data, temp_packet.length);
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RtpUtility::RtpHeaderParser parser(packet_memory.get(), temp_packet.length);
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auto packet = std::make_unique<Packet>(
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packet_memory.release(), temp_packet.length,
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temp_packet.original_length, temp_packet.time_ms, parser,
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&rtp_header_extension_map_);
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if (!packet->valid_header()) {
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continue;
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}
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if (filter_.test(packet->header().payloadType) ||
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(ssrc_filter_ && packet->header().ssrc != *ssrc_filter_)) {
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// This payload type should be filtered out. Continue to the next packet.
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continue;
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}
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return packet;
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}
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}
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RtpFileSource::RtpFileSource(absl::optional<uint32_t> ssrc_filter)
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: PacketSource(),
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ssrc_filter_(ssrc_filter) {}
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bool RtpFileSource::OpenFile(const std::string& file_name) {
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rtp_reader_.reset(RtpFileReader::Create(RtpFileReader::kRtpDump, file_name));
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if (rtp_reader_)
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return true;
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rtp_reader_.reset(RtpFileReader::Create(RtpFileReader::kPcap, file_name));
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if (!rtp_reader_) {
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FATAL() << "Couldn't open input file as either a rtpdump or .pcap. Note "
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"that .pcapng is not supported.";
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}
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return true;
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}
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} // namespace test
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} // namespace webrtc
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