1 /* 2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. 3 * 4 * Use of this source code is governed by a BSD-style license 5 * that can be found in the LICENSE file in the root of the source 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 #ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_ 12 #define MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_ 13 14 #include <stdio.h> 15 16 #include <memory> 17 #include <string> 18 19 #include "absl/strings/string_view.h" 20 #include "absl/types/optional.h" 21 #include "modules/audio_coding/neteq/tools/packet_source.h" 22 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" 23 24 namespace webrtc { 25 26 namespace test { 27 28 class RtpFileReader; 29 30 class RtpFileSource : public PacketSource { 31 public: 32 // Creates an RtpFileSource reading from `file_name`. If the file cannot be 33 // opened, or has the wrong format, NULL will be returned. 34 static RtpFileSource* Create( 35 absl::string_view file_name, 36 absl::optional<uint32_t> ssrc_filter = absl::nullopt); 37 38 // Checks whether a files is a valid RTP dump or PCAP (Wireshark) file. 39 static bool ValidRtpDump(absl::string_view file_name); 40 static bool ValidPcap(absl::string_view file_name); 41 42 ~RtpFileSource() override; 43 44 RtpFileSource(const RtpFileSource&) = delete; 45 RtpFileSource& operator=(const RtpFileSource&) = delete; 46 47 // Registers an RTP header extension and binds it to `id`. 48 virtual bool RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id); 49 50 std::unique_ptr<Packet> NextPacket() override; 51 52 private: 53 static const int kFirstLineLength = 40; 54 static const int kRtpFileHeaderSize = 4 + 4 + 4 + 2 + 2; 55 static const size_t kPacketHeaderSize = 8; 56 57 explicit RtpFileSource(absl::optional<uint32_t> ssrc_filter); 58 59 bool OpenFile(absl::string_view file_name); 60 61 std::unique_ptr<RtpFileReader> rtp_reader_; 62 const absl::optional<uint32_t> ssrc_filter_; 63 RtpHeaderExtensionMap rtp_header_extension_map_; 64 }; 65 66 } // namespace test 67 } // namespace webrtc 68 #endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_ 69