#include <rtp_file_source.h>
◆ ~RtpFileSource() [1/2]
webrtc::test::RtpFileSource::~RtpFileSource |
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virtual |
◆ ~RtpFileSource() [2/2]
virtual webrtc::test::RtpFileSource::~RtpFileSource |
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virtual |
◆ Create() [1/2]
◆ Create() [2/2]
◆ NextPacket() [1/2]
std::unique_ptr< Packet > webrtc::test::RtpFileSource::NextPacket |
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overridevirtual |
◆ NextPacket() [2/2]
std::unique_ptr<Packet> webrtc::test::RtpFileSource::NextPacket |
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overridevirtual |
◆ RegisterRtpHeaderExtension() [1/2]
◆ RegisterRtpHeaderExtension() [2/2]
◆ ValidPcap() [1/2]
static bool webrtc::test::RtpFileSource::ValidPcap |
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const std::string & |
file_name | ) |
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◆ ValidPcap() [2/2]
bool webrtc::test::RtpFileSource::ValidPcap |
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const std::string & |
file_name | ) |
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static |
◆ ValidRtpDump() [1/2]
bool webrtc::test::RtpFileSource::ValidRtpDump |
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const std::string & |
file_name | ) |
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static |
◆ ValidRtpDump() [2/2]
static bool webrtc::test::RtpFileSource::ValidRtpDump |
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const std::string & |
file_name | ) |
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static |
The documentation for this class was generated from the following files:
- DerivedData/WebKit/Build/Products/Debug/usr/local/include/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h
- Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc