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/types/optional.h" 20 #include "modules/audio_coding/neteq/tools/packet_source.h" 21 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" 22 #include "modules/rtp_rtcp/source/rtp_utility.h" 23 #include "rtc_base/constructor_magic.h" 24 25 namespace webrtc { 26 27 namespace test { 28 29 class RtpFileReader; 30 31 class RtpFileSource : public PacketSource { 32 public: 33 // Creates an RtpFileSource reading from |file_name|. If the file cannot be 34 // opened, or has the wrong format, NULL will be returned. 35 static RtpFileSource* Create( 36 const std::string& file_name, 37 absl::optional<uint32_t> ssrc_filter = absl::nullopt); 38 39 // Checks whether a files is a valid RTP dump or PCAP (Wireshark) file. 40 static bool ValidRtpDump(const std::string& file_name); 41 static bool ValidPcap(const std::string& file_name); 42 43 ~RtpFileSource() override; 44 45 // Registers an RTP header extension and binds it to |id|. 46 virtual bool RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id); 47 48 std::unique_ptr<Packet> NextPacket() override; 49 50 private: 51 static const int kFirstLineLength = 40; 52 static const int kRtpFileHeaderSize = 4 + 4 + 4 + 2 + 2; 53 static const size_t kPacketHeaderSize = 8; 54 55 explicit RtpFileSource(absl::optional<uint32_t> ssrc_filter); 56 57 bool OpenFile(const std::string& file_name); 58 59 std::unique_ptr<RtpFileReader> rtp_reader_; 60 const absl::optional<uint32_t> ssrc_filter_; 61 RtpHeaderExtensionMap rtp_header_extension_map_; 62 63 RTC_DISALLOW_COPY_AND_ASSIGN(RtpFileSource); 64 }; 65 66 } // namespace test 67 } // namespace webrtc 68 #endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_ 69