2013-07-10 00:45:36 +00:00
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/*
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2016-02-10 07:54:43 -08:00
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* Copyright 2012 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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2016-02-10 07:54:43 -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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2017-02-08 01:38:21 -08:00
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// TODO(deadbeef): Move this out of api/; it's an implementation detail and
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// shouldn't be used externally.
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2013-07-10 00:45:36 +00:00
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2016-02-10 10:53:12 +01:00
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#ifndef WEBRTC_API_JSEPICECANDIDATE_H_
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#define WEBRTC_API_JSEPICECANDIDATE_H_
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2013-07-10 00:45:36 +00:00
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#include <string>
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#include <utility>
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#include <vector>
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2013-07-10 00:45:36 +00:00
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2016-02-10 10:53:12 +01:00
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#include "webrtc/api/jsep.h"
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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/candidate.h"
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2017-07-06 19:44:34 +02:00
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#include "webrtc/rtc_base/constructormagic.h"
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2013-07-10 00:45:36 +00:00
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namespace webrtc {
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// Implementation of IceCandidateInterface.
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2013-07-10 00:45:36 +00:00
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class JsepIceCandidate : public IceCandidateInterface {
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public:
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JsepIceCandidate(const std::string& sdp_mid, int sdp_mline_index);
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JsepIceCandidate(const std::string& sdp_mid, int sdp_mline_index,
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const cricket::Candidate& candidate);
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~JsepIceCandidate();
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// |err| may be null.
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2013-07-10 00:45:36 +00:00
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bool Initialize(const std::string& sdp, SdpParseError* err);
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void SetCandidate(const cricket::Candidate& candidate) {
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candidate_ = candidate;
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}
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virtual std::string sdp_mid() const { return sdp_mid_; }
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virtual int sdp_mline_index() const { return sdp_mline_index_; }
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virtual const cricket::Candidate& candidate() const {
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return candidate_;
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}
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2017-02-16 11:29:39 -08:00
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virtual std::string server_url() const { return candidate_.url(); }
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2013-07-10 00:45:36 +00:00
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virtual bool ToString(std::string* out) const;
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private:
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std::string sdp_mid_;
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int sdp_mline_index_;
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cricket::Candidate candidate_;
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2015-09-16 05:37:44 -07:00
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RTC_DISALLOW_COPY_AND_ASSIGN(JsepIceCandidate);
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2013-07-10 00:45:36 +00:00
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};
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2017-02-08 01:38:21 -08:00
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// Implementation of IceCandidateCollection which stores JsepIceCandidates.
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2013-07-10 00:45:36 +00:00
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class JsepCandidateCollection : public IceCandidateCollection {
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public:
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JsepCandidateCollection() {}
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// Move constructor is defined so that a vector of JsepCandidateCollections
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// can be resized.
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JsepCandidateCollection(JsepCandidateCollection&& o)
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: candidates_(std::move(o.candidates_)) {}
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2013-07-10 00:45:36 +00:00
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~JsepCandidateCollection();
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virtual size_t count() const {
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return candidates_.size();
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}
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virtual bool HasCandidate(const IceCandidateInterface* candidate) const;
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// Adds and takes ownership of the JsepIceCandidate.
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// TODO(deadbeef): Make this use an std::unique_ptr<>, so ownership logic is
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// more clear.
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2013-07-10 00:45:36 +00:00
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virtual void add(JsepIceCandidate* candidate) {
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candidates_.push_back(candidate);
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}
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virtual const IceCandidateInterface* at(size_t index) const {
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return candidates_[index];
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}
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// Removes the candidate that has a matching address and protocol.
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//
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// Returns the number of candidates that were removed.
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size_t remove(const cricket::Candidate& candidate);
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2013-07-10 00:45:36 +00:00
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private:
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std::vector<JsepIceCandidate*> candidates_;
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RTC_DISALLOW_COPY_AND_ASSIGN(JsepCandidateCollection);
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2013-07-10 00:45:36 +00:00
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};
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} // namespace webrtc
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2016-02-10 10:53:12 +01:00
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#endif // WEBRTC_API_JSEPICECANDIDATE_H_
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