#include <config.h>
◆ RtpExtension() [1/4]
webrtc::RtpExtension::RtpExtension |
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◆ RtpExtension() [2/4]
webrtc::RtpExtension::RtpExtension |
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const std::string & |
uri, |
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int |
id |
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◆ RtpExtension() [3/4]
webrtc::RtpExtension::RtpExtension |
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◆ RtpExtension() [4/4]
webrtc::RtpExtension::RtpExtension |
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const std::string & |
uri, |
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int |
id |
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inline |
◆ IsSupportedForAudio() [1/2]
bool webrtc::RtpExtension::IsSupportedForAudio |
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const std::string & |
uri | ) |
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◆ IsSupportedForAudio() [2/2]
static bool webrtc::RtpExtension::IsSupportedForAudio |
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const std::string & |
uri | ) |
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◆ IsSupportedForVideo() [1/2]
bool webrtc::RtpExtension::IsSupportedForVideo |
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const std::string & |
uri | ) |
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◆ IsSupportedForVideo() [2/2]
static bool webrtc::RtpExtension::IsSupportedForVideo |
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const std::string & |
uri | ) |
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◆ operator==() [1/2]
bool webrtc::RtpExtension::operator== |
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const RtpExtension & |
rhs | ) |
const |
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inline |
◆ operator==() [2/2]
bool webrtc::RtpExtension::operator== |
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const RtpExtension & |
rhs | ) |
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inline |
◆ ToString() [1/2]
◆ ToString() [2/2]
◆ id
int webrtc::RtpExtension::id |
◆ kAbsSendTimeDefaultId
static const int webrtc::RtpExtension::kAbsSendTimeDefaultId = 3 |
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◆ kAbsSendTimeUri
static const char * webrtc::RtpExtension::kAbsSendTimeUri |
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Initial value:=
"http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time"
◆ kAudioLevelDefaultId
static const int webrtc::RtpExtension::kAudioLevelDefaultId = 1 |
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◆ kAudioLevelUri
static const char * webrtc::RtpExtension::kAudioLevelUri |
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Initial value:=
"urn:ietf:params:rtp-hdrext:ssrc-audio-level"
◆ kMaxId
static const int webrtc::RtpExtension::kMaxId = 14 |
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◆ kMinId
static const int webrtc::RtpExtension::kMinId = 1 |
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◆ kPlayoutDelayDefaultId
static const int webrtc::RtpExtension::kPlayoutDelayDefaultId = 6 |
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◆ kPlayoutDelayUri
static const char * webrtc::RtpExtension::kPlayoutDelayUri |
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Initial value:=
"http://www.webrtc.org/experiments/rtp-hdrext/playout-delay"
◆ kTimestampOffsetDefaultId
static const int webrtc::RtpExtension::kTimestampOffsetDefaultId = 2 |
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◆ kTimestampOffsetUri
static const char * webrtc::RtpExtension::kTimestampOffsetUri |
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Initial value:=
"urn:ietf:params:rtp-hdrext:toffset"
◆ kTransportSequenceNumberDefaultId
static const int webrtc::RtpExtension::kTransportSequenceNumberDefaultId = 5 |
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◆ kTransportSequenceNumberUri
static const char * webrtc::RtpExtension::kTransportSequenceNumberUri |
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Initial value:=
"http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01"
◆ kVideoRotationDefaultId
static const int webrtc::RtpExtension::kVideoRotationDefaultId = 4 |
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◆ kVideoRotationUri
static const char * webrtc::RtpExtension::kVideoRotationUri = "urn:3gpp:video-orientation" |
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◆ uri
The documentation for this struct was generated from the following files:
- DerivedData/WebKit/Build/Products/Debug/usr/local/include/webrtc/config.h
- Source/ThirdParty/libwebrtc/Source/webrtc/config.cc