| /* | |
| * 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 <algorithm> | |
| #include <cstdint> | |
| #include <cstring> | |
| #include <memory> | |
| #include <optional> | |
| #include "absl/strings/string_view.h" | |
| #include "api/rtp_header_extension_id.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<uint32_t> 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<RtpFileReader> temp_file( | |
| RtpFileReader::Create(RtpFileReader::kRtpDump, file_name)); | |
| return !!temp_file; | |
| } | |
| bool RtpFileSource::ValidPcap(absl::string_view file_name) { | |
| std::unique_ptr<RtpFileReader> temp_file( | |
| RtpFileReader::Create(RtpFileReader::kPcap, file_name)); | |
| return !!temp_file; | |
| } | |
| RtpFileSource::~RtpFileSource() {} | |
| bool RtpFileSource::RegisterRtpHeaderExtension(RTPExtensionType type, | |
| RtpHeaderExtensionId id) { | |
| return rtp_header_extension_map_.RegisterByType(id, type); | |
| } | |
| std::unique_ptr<RtpPacketReceived> 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<RtpPacketReceived>(&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)); | |
| // Simulate payload if only the RTP header was written in the file. | |
| if (temp_packet.original_length > rtp_packet->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<uint32_t> 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 |