Bug: webrtc:437816262 Change-Id: If69e649a985600d91ca8cece1ca83d82dfeda6c3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/404040 Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#45321}
113 lines
3.7 KiB
C++
113 lines
3.7 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 <algorithm>
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#include <cstdint>
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#include <cstring>
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#include <memory>
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#include <optional>
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#include "absl/strings/string_view.h"
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#include "api/units/timestamp.h"
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#include "modules/audio_coding/neteq/tools/packet_source.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "modules/rtp_rtcp/source/rtp_packet_received.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(absl::string_view file_name,
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std::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(absl::string_view 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(absl::string_view 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<RtpPacketReceived> 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 nullptr;
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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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auto rtp_packet =
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std::make_unique<RtpPacketReceived>(&rtp_header_extension_map_);
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if (!rtp_packet->Parse(temp_packet.data, temp_packet.length)) {
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continue;
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}
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if (filter_.test(rtp_packet->PayloadType()) ||
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(ssrc_filter_ && rtp_packet->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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rtp_packet->set_arrival_time(Timestamp::Millis(temp_packet.time_ms));
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// Simulate payload if only the RTP header was written in the file.
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if (temp_packet.original_length > rtp_packet->size()) {
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size_t payload_size =
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temp_packet.original_length - rtp_packet->headers_size();
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if (rtp_packet->has_padding()) {
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// If padding bit is set in the RTP header, assume it was a pure padding
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// packet.
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rtp_packet->SetPadding(payload_size);
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} else {
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std::fill_n(rtp_packet->AllocatePayload(payload_size), payload_size, 0);
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}
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}
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return rtp_packet;
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}
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}
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RtpFileSource::RtpFileSource(std::optional<uint32_t> ssrc_filter)
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: PacketSource(), ssrc_filter_(ssrc_filter) {}
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bool RtpFileSource::OpenFile(absl::string_view 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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RTC_FATAL()
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<< "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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