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/*
* 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 <stdio.h>
#include <memory>
#include <string>
#include "common_types.h" // NOLINT(build/include)
#include "modules/audio_coding/neteq/tools/packet_source.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "rtc_base/constructormagic.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);
// Checks whether a files is a valid RTP dump or PCAP (Wireshark) file.
static bool ValidRtpDump(const std::string& file_name);
static bool ValidPcap(const std::string& file_name);
~RtpFileSource() override;
// Registers an RTP header extension and binds it to |id|.
virtual bool RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id);
std::unique_ptr<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);
std::unique_ptr<RtpFileReader> rtp_reader_;
std::unique_ptr<RtpHeaderParser> parser_;
RTC_DISALLOW_COPY_AND_ASSIGN(RtpFileSource);
};
} // namespace test
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_
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