2013-07-10 00:45:36 +00:00
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/*
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2016-02-12 00:05:01 -08:00
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* Copyright 2004 The WebRTC project authors. All Rights Reserved.
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2013-07-10 00:45:36 +00:00
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*
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2016-02-12 00:05:01 -08:00
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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2013-07-10 00:45:36 +00:00
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*/
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2016-04-26 05:28:11 -07:00
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#ifndef WEBRTC_PC_RTCPMUXFILTER_H_
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#define WEBRTC_PC_RTCPMUXFILTER_H_
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2013-07-10 00:45:36 +00:00
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Move talk/media to webrtc/media
I removed the 'libjingle' target in talk/libjingle.gyp and replaced
all users of it with base/base.gyp:rtc_base. It seems the jsoncpp
and expat dependencies were not used by it's previous references.
The files in talk/media/testdata were uploaded to Google Storage and
added .sha1 files in resources/media instead of simply moving them.
The previously disabled warnings that were inherited from
talk/build/common.gypi are now replaced by target-specific disabling
of only the failing warnings. Additional disabling was needed since the stricter
compilation warnings that applies to code in webrtc/.
License headers will be updated in a follow-up CL in order to not
break Git history.
Other modifications:
* Updated the header guards.
* Sorted the includes using chromium/src/tools/sort-headers.py
except for these files:
talk/app/webrtc/peerconnectionendtoend_unittest.cc
talk/app/webrtc/java/jni/androidmediadecoder_jni.cc
talk/app/webrtc/java/jni/androidmediaencoder_jni.cc
webrtc/media/devices/win32devicemanager.cc.
* Unused GYP reference to libjingle_tests_additional_deps was removed.
* Removed duplicated GYP entries of
webrtc/base/testutils.cc
webrtc/base/testutils.h
The HAVE_WEBRTC_VIDEO and HAVE_WEBRTC_VOICE defines were used by only talk/media,
so they were moved to the media.gyp.
I also checked that none of
EXPAT_RELATIVE_PATH,
FEATURE_ENABLE_VOICEMAIL,
GTEST_RELATIVE_PATH,
JSONCPP_RELATIVE_PATH,
LOGGING=1,
SRTP_RELATIVE_PATH,
FEATURE_ENABLE_SSL,
FEATURE_ENABLE_VOICEMAIL,
FEATURE_ENABLE_PSTN,
HAVE_SCTP,
HAVE_SRTP,
are used by the talk/media code.
For Chromium, the following changes will need to be applied to the roll CL that updates the
DEPS for WebRTC and libjingle: https://codereview.chromium.org/1604303002/
BUG=webrtc:5420
NOPRESUBMIT=True
TBR=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1587193006
Cr-Commit-Position: refs/heads/master@{#11495}
2016-02-04 23:52:28 -08:00
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#include "webrtc/p2p/base/sessiondescription.h"
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2013-07-10 00:45:36 +00:00
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namespace cricket {
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// RTCP Muxer, as defined in RFC 5761 (http://tools.ietf.org/html/rfc5761)
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class RtcpMuxFilter {
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public:
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RtcpMuxFilter();
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2016-08-22 16:00:30 -07:00
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// Whether RTCP mux has been negotiated with a final answer (not provisional).
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bool IsFullyActive() const;
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// Whether RTCP mux has been negotiated with a provisional answer; this means
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// a later answer could disable RTCP mux, and so the RTCP transport should
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// not be disposed yet.
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bool IsProvisionallyActive() const;
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// Whether the filter is active, i.e. has RTCP mux been properly negotiated,
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// either with a final or provisional answer.
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2013-07-10 00:45:36 +00:00
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bool IsActive() const;
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2016-08-22 16:00:30 -07:00
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// Make the filter active (fully, not provisionally) regardless of the
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// current state. This should be used when an endpoint *requires* RTCP mux.
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void SetActive();
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2013-07-10 00:45:36 +00:00
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// Specifies whether the offer indicates the use of RTCP mux.
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bool SetOffer(bool offer_enable, ContentSource src);
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// Specifies whether the provisional answer indicates the use of RTCP mux.
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bool SetProvisionalAnswer(bool answer_enable, ContentSource src);
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// Specifies whether the answer indicates the use of RTCP mux.
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bool SetAnswer(bool answer_enable, ContentSource src);
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private:
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bool ExpectOffer(bool offer_enable, ContentSource source);
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bool ExpectAnswer(ContentSource source);
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enum State {
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// RTCP mux filter unused.
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ST_INIT,
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// Offer with RTCP mux enabled received.
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// RTCP mux filter is not active.
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ST_RECEIVEDOFFER,
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// Offer with RTCP mux enabled sent.
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// RTCP mux filter can demux incoming packets but is not active.
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ST_SENTOFFER,
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// RTCP mux filter is active but the sent answer is only provisional.
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// When the final answer is set, the state transitions to ST_ACTIVE or
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// ST_INIT.
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ST_SENTPRANSWER,
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// RTCP mux filter is active but the received answer is only provisional.
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// When the final answer is set, the state transitions to ST_ACTIVE or
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// ST_INIT.
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ST_RECEIVEDPRANSWER,
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// Offer and answer set, RTCP mux enabled. It is not possible to de-activate
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// the filter.
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ST_ACTIVE
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};
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State state_;
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bool offer_enable_;
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};
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} // namespace cricket
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2016-04-26 05:28:11 -07:00
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#endif // WEBRTC_PC_RTCPMUXFILTER_H_
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