#include <rtp_format_h264.h>
◆ RtpDepacketizerH264() [1/2]
webrtc::RtpDepacketizerH264::RtpDepacketizerH264 |
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◆ ~RtpDepacketizerH264() [1/2]
webrtc::RtpDepacketizerH264::~RtpDepacketizerH264 |
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virtual |
◆ RtpDepacketizerH264() [2/2]
webrtc::RtpDepacketizerH264::RtpDepacketizerH264 |
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◆ ~RtpDepacketizerH264() [2/2]
virtual webrtc::RtpDepacketizerH264::~RtpDepacketizerH264 |
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virtual |
◆ Parse() [1/2]
bool webrtc::RtpDepacketizerH264::Parse |
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ParsedPayload * |
parsed_payload, |
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const uint8_t * |
payload_data, |
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size_t |
payload_data_length |
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) |
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overridevirtual |
◆ Parse() [2/2]
bool webrtc::RtpDepacketizerH264::Parse |
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ParsedPayload * |
parsed_payload, |
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const uint8_t * |
payload_data, |
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size_t |
payload_data_length |
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) |
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overridevirtual |
The documentation for this class was generated from the following files:
- DerivedData/WebKit/Build/Products/Debug/usr/local/include/webrtc/modules/rtp_rtcp/source/rtp_format_h264.h
- Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc