/* * Copyright (c) 2014 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/tools/rtp_file_source.h" #include #include #include #include #include #include "absl/strings/string_view.h" #include "api/units/timestamp.h" #include "modules/audio_coding/neteq/tools/packet_source.h" #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "modules/rtp_rtcp/source/rtp_packet_received.h" #include "rtc_base/checks.h" #include "test/rtp_file_reader.h" namespace webrtc { namespace test { RtpFileSource* RtpFileSource::Create(absl::string_view file_name, std::optional ssrc_filter) { RtpFileSource* source = new RtpFileSource(ssrc_filter); RTC_CHECK(source->OpenFile(file_name)); return source; } bool RtpFileSource::ValidRtpDump(absl::string_view file_name) { std::unique_ptr temp_file( RtpFileReader::Create(RtpFileReader::kRtpDump, file_name)); return !!temp_file; } bool RtpFileSource::ValidPcap(absl::string_view file_name) { std::unique_ptr temp_file( RtpFileReader::Create(RtpFileReader::kPcap, file_name)); return !!temp_file; } RtpFileSource::~RtpFileSource() {} bool RtpFileSource::RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id) { return rtp_header_extension_map_.RegisterByType(id, type); } std::unique_ptr RtpFileSource::NextPacket() { while (true) { RtpPacket temp_packet; if (!rtp_reader_->NextPacket(&temp_packet)) { return nullptr; } if (temp_packet.original_length == 0) { // May be an RTCP packet. // Read the next one. continue; } auto rtp_packet = std::make_unique(&rtp_header_extension_map_); if (!rtp_packet->Parse(temp_packet.data, temp_packet.length)) { continue; } if (filter_.test(rtp_packet->PayloadType()) || (ssrc_filter_ && rtp_packet->Ssrc() != *ssrc_filter_)) { // This payload type should be filtered out. Continue to the next packet. continue; } rtp_packet->set_arrival_time(Timestamp::Millis(temp_packet.time_ms)); if (temp_packet.original_length > rtp_packet->headers_size()) { size_t payload_size = temp_packet.original_length - rtp_packet->headers_size(); if (rtp_packet->has_padding()) { // If padding bit is set in the RTP header, assume it was a pure padding // packet. rtp_packet->SetPadding(payload_size); } else { std::fill_n(rtp_packet->AllocatePayload(payload_size), payload_size, 0); } } return rtp_packet; } } RtpFileSource::RtpFileSource(std::optional ssrc_filter) : PacketSource(), ssrc_filter_(ssrc_filter) {} bool RtpFileSource::OpenFile(absl::string_view file_name) { rtp_reader_.reset(RtpFileReader::Create(RtpFileReader::kRtpDump, file_name)); if (rtp_reader_) return true; rtp_reader_.reset(RtpFileReader::Create(RtpFileReader::kPcap, file_name)); if (!rtp_reader_) { RTC_FATAL() << "Couldn't open input file as either a rtpdump or .pcap. Note " << "that .pcapng is not supported."; } return true; } } // namespace test } // namespace webrtc