Cleanup: Replacing set extension status bool with CHECK.
This was just checked in all places were it was used, moving the check into RtpRtcp reduces the boiler plate required at the call sites. Also changing to always register and unregister extensions by URI to synchronize the code in AudioSendStream with the code in RtpVideoSender. This prepares for reducing the scope of ChannelSend. Bug: webrtc:9883 Change-Id: Ia64d79f20eb98f46cbbbe8318770e4fcf9caa1ec Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155620 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29490}
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@ -37,7 +37,7 @@ class RtpHeaderExtensionMap {
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return Register(id, Extension::kId, Extension::kUri);
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}
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bool RegisterByType(int id, RTPExtensionType type);
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bool RegisterByUri(int id, const std::string& uri);
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bool RegisterByUri(int id, absl::string_view uri);
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bool IsRegistered(RTPExtensionType type) const {
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return GetId(type) != kInvalidId;
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@ -56,6 +56,7 @@ class RtpHeaderExtensionMap {
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return RegisterByType(id, type) ? 0 : -1;
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}
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int32_t Deregister(RTPExtensionType type);
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void Deregister(absl::string_view uri);
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// Corresponds to the SDP attribute extmap-allow-mixed, see RFC8285.
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// Set to true if it's allowed to mix one- and two-byte RTP header extensions
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