/* * 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. */ #ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_ #define MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_ #include #include #include #include "absl/strings/string_view.h" #include "absl/types/optional.h" #include "modules/audio_coding/neteq/tools/packet_source.h" #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" namespace webrtc { namespace test { class RtpFileReader; class RtpFileSource : public PacketSource { public: // Creates an RtpFileSource reading from `file_name`. If the file cannot be // opened, or has the wrong format, NULL will be returned. static RtpFileSource* Create( absl::string_view file_name, absl::optional ssrc_filter = absl::nullopt); // Checks whether a files is a valid RTP dump or PCAP (Wireshark) file. static bool ValidRtpDump(absl::string_view file_name); static bool ValidPcap(absl::string_view file_name); ~RtpFileSource() override; RtpFileSource(const RtpFileSource&) = delete; RtpFileSource& operator=(const RtpFileSource&) = delete; // Registers an RTP header extension and binds it to `id`. virtual bool RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id); std::unique_ptr NextPacket() override; private: static const int kFirstLineLength = 40; static const int kRtpFileHeaderSize = 4 + 4 + 4 + 2 + 2; static const size_t kPacketHeaderSize = 8; explicit RtpFileSource(absl::optional ssrc_filter); bool OpenFile(absl::string_view file_name); std::unique_ptr rtp_reader_; const absl::optional ssrc_filter_; RtpHeaderExtensionMap rtp_header_extension_map_; }; } // namespace test } // namespace webrtc #endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_