| /* |
| * 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 WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_ |
| #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_ |
| |
| #include <stdio.h> |
| #include <string> |
| |
| #include "webrtc/base/constructormagic.h" |
| #include "webrtc/common_types.h" |
| #include "webrtc/modules/audio_coding/neteq/tools/packet_source.h" |
| #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" |
| #include "webrtc/system_wrappers/interface/scoped_ptr.h" |
| |
| namespace webrtc { |
| |
| class RtpHeaderParser; |
| |
| 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(const std::string& file_name); |
| |
| virtual ~RtpFileSource(); |
| |
| // Registers an RTP header extension and binds it to |id|. |
| virtual bool RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id); |
| |
| // Returns a pointer to the next packet. Returns NULL if end of file was |
| // reached, or if a the data was corrupt. |
| virtual Packet* NextPacket() OVERRIDE; |
| |
| private: |
| static const int kFirstLineLength = 40; |
| static const int kRtpFileHeaderSize = 4 + 4 + 4 + 2 + 2; |
| static const size_t kPacketHeaderSize = 8; |
| |
| RtpFileSource(); |
| |
| bool OpenFile(const std::string& file_name); |
| |
| scoped_ptr<RtpFileReader> rtp_reader_; |
| scoped_ptr<RtpHeaderParser> parser_; |
| |
| DISALLOW_COPY_AND_ASSIGN(RtpFileSource); |
| }; |
| |
| } // namespace test |
| } // namespace webrtc |
| #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_ |