diff --git a/api/rtp_parameters.cc b/api/rtp_parameters.cc index 2b1abe0eff..6177dd3eb9 100644 --- a/api/rtp_parameters.cc +++ b/api/rtp_parameters.cc @@ -93,24 +93,19 @@ std::string RtpExtension::ToString() const { const char RtpExtension::kAudioLevelUri[] = "urn:ietf:params:rtp-hdrext:ssrc-audio-level"; -const int RtpExtension::kAudioLevelDefaultId = 1; const char RtpExtension::kTimestampOffsetUri[] = "urn:ietf:params:rtp-hdrext:toffset"; -const int RtpExtension::kTimestampOffsetDefaultId = 2; const char RtpExtension::kAbsSendTimeUri[] = "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time"; -const int RtpExtension::kAbsSendTimeDefaultId = 3; const char RtpExtension::kVideoRotationUri[] = "urn:3gpp:video-orientation"; -const int RtpExtension::kVideoRotationDefaultId = 4; const char RtpExtension::kTransportSequenceNumberUri[] = "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01"; const char RtpExtension::kTransportSequenceNumberV2Uri[] = "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-02"; -const int RtpExtension::kTransportSequenceNumberDefaultId = 5; // This extension allows applications to adaptively limit the playout delay // on frames as per the current needs. For example, a gaming application @@ -118,22 +113,17 @@ const int RtpExtension::kTransportSequenceNumberDefaultId = 5; // application. const char RtpExtension::kPlayoutDelayUri[] = "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay"; -const int RtpExtension::kPlayoutDelayDefaultId = 6; const char RtpExtension::kVideoContentTypeUri[] = "http://www.webrtc.org/experiments/rtp-hdrext/video-content-type"; -const int RtpExtension::kVideoContentTypeDefaultId = 7; const char RtpExtension::kVideoTimingUri[] = "http://www.webrtc.org/experiments/rtp-hdrext/video-timing"; -const int RtpExtension::kVideoTimingDefaultId = 8; const char RtpExtension::kMidUri[] = "urn:ietf:params:rtp-hdrext:sdes:mid"; -const int RtpExtension::kMidDefaultId = 9; const char RtpExtension::kFrameMarkingUri[] = "http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07"; -const int RtpExtension::kFrameMarkingDefaultId = 10; const char RtpExtension::kGenericFrameDescriptorUri00[] = "http://www.webrtc.org/experiments/rtp-hdrext/generic-frame-descriptor-00"; @@ -141,22 +131,18 @@ const char RtpExtension::kGenericFrameDescriptorUri01[] = "http://www.webrtc.org/experiments/rtp-hdrext/generic-frame-descriptor-01"; const char RtpExtension::kGenericFrameDescriptorUri[] = "http://www.webrtc.org/experiments/rtp-hdrext/generic-frame-descriptor-00"; -const int RtpExtension::kGenericFrameDescriptorDefaultId = 11; const char RtpExtension::kEncryptHeaderExtensionsUri[] = "urn:ietf:params:rtp-hdrext:encrypt"; const char RtpExtension::kColorSpaceUri[] = "http://www.webrtc.org/experiments/rtp-hdrext/color-space"; -const int RtpExtension::kColorSpaceDefaultId = 12; const char RtpExtension::kRidUri[] = "urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id"; -const int RtpExtension::kRidDefaultId = 13; const char RtpExtension::kRepairedRidUri[] = "urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id"; -const int RtpExtension::kRepairedRidDefaultId = 14; constexpr int RtpExtension::kMinId; constexpr int RtpExtension::kMaxId; diff --git a/api/rtp_parameters.h b/api/rtp_parameters.h index 1cf4f36091..6bb7c394bf 100644 --- a/api/rtp_parameters.h +++ b/api/rtp_parameters.h @@ -18,7 +18,6 @@ #include "absl/types/optional.h" #include "api/media_types.h" -#include "rtc_base/deprecation.h" #include "rtc_base/system/rtc_export.h" namespace webrtc { @@ -259,67 +258,45 @@ struct RtpExtension { // Header extension for audio levels, as defined in: // http://tools.ietf.org/html/draft-ietf-avtext-client-to-mixer-audio-level-03 static const char kAudioLevelUri[]; - // TODO(bugs.webrtc.org/10288): Remove once dependencies have been updated. - RTC_DEPRECATED static const int kAudioLevelDefaultId; // Header extension for RTP timestamp offset, see RFC 5450 for details: // http://tools.ietf.org/html/rfc5450 static const char kTimestampOffsetUri[]; - // TODO(bugs.webrtc.org/10288): Remove once dependencies have been updated. - RTC_DEPRECATED static const int kTimestampOffsetDefaultId; // Header extension for absolute send time, see url for details: // http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time static const char kAbsSendTimeUri[]; - // TODO(bugs.webrtc.org/10288): Remove once dependencies have been updated. - RTC_DEPRECATED static const int kAbsSendTimeDefaultId; // Header extension for coordination of video orientation, see url for // details: // http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/ts_126114v120700p.pdf static const char kVideoRotationUri[]; - // TODO(bugs.webrtc.org/10288): Remove once dependencies have been updated. - RTC_DEPRECATED static const int kVideoRotationDefaultId; // Header extension for video content type. E.g. default or screenshare. static const char kVideoContentTypeUri[]; - // TODO(bugs.webrtc.org/10288): Remove once dependencies have been updated. - RTC_DEPRECATED static const int kVideoContentTypeDefaultId; // Header extension for video timing. static const char kVideoTimingUri[]; - // TODO(bugs.webrtc.org/10288): Remove once dependencies have been updated. - RTC_DEPRECATED static const int kVideoTimingDefaultId; // Header extension for video frame marking. static const char kFrameMarkingUri[]; - // TODO(bugs.webrtc.org/10288): Remove once dependencies have been updated. - RTC_DEPRECATED static const int kFrameMarkingDefaultId; // Experimental codec agnostic frame descriptor. static const char kGenericFrameDescriptorUri00[]; static const char kGenericFrameDescriptorUri01[]; // TODO(bugs.webrtc.org/10243): Remove once dependencies have been updated. static const char kGenericFrameDescriptorUri[]; - // TODO(bugs.webrtc.org/10288): Remove once dependencies have been updated. - RTC_DEPRECATED static const int kGenericFrameDescriptorDefaultId; // Header extension for transport sequence number, see url for details: // http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions static const char kTransportSequenceNumberUri[]; static const char kTransportSequenceNumberV2Uri[]; - // TODO(bugs.webrtc.org/10288): Remove once dependencies have been updated. - RTC_DEPRECATED static const int kTransportSequenceNumberDefaultId; static const char kPlayoutDelayUri[]; - // TODO(bugs.webrtc.org/10288): Remove once dependencies have been updated. - RTC_DEPRECATED static const int kPlayoutDelayDefaultId; // Header extension for identifying media section within a transport. // https://tools.ietf.org/html/draft-ietf-mmusic-sdp-bundle-negotiation-49#section-15 static const char kMidUri[]; - // TODO(bugs.webrtc.org/10288): Remove once dependencies have been updated. - RTC_DEPRECATED static const int kMidDefaultId; // Encryption of Header Extensions, see RFC 6904 for details: // https://tools.ietf.org/html/rfc6904 @@ -327,18 +304,12 @@ struct RtpExtension { // Header extension for color space information. static const char kColorSpaceUri[]; - // TODO(bugs.webrtc.org/10288): Remove once dependencies have been updated. - RTC_DEPRECATED static const int kColorSpaceDefaultId; // Header extension for RIDs and Repaired RIDs // https://tools.ietf.org/html/draft-ietf-avtext-rid-09 // https://tools.ietf.org/html/draft-ietf-mmusic-rid-15 static const char kRidUri[]; - // TODO(bugs.webrtc.org/10288): Remove once dependencies have been updated. - RTC_DEPRECATED static const int kRidDefaultId; static const char kRepairedRidUri[]; - // TODO(bugs.webrtc.org/10288): Remove once dependencies have been updated. - RTC_DEPRECATED static const int kRepairedRidDefaultId; // Inclusive min and max IDs for two-byte header extensions and one-byte // header extensions, per RFC8285 Section 4.2-4.3.