webrtc_m130/api/jsepicecandidate.h

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/*
* Copyright 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
// TODO(deadbeef): Move this out of api/; it's an implementation detail and
// shouldn't be used externally.
#ifndef WEBRTC_API_JSEPICECANDIDATE_H_
#define WEBRTC_API_JSEPICECANDIDATE_H_
#include <string>
#include <utility>
#include <vector>
#include "webrtc/api/jsep.h"
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
#include "webrtc/p2p/base/candidate.h"
#include "webrtc/rtc_base/constructormagic.h"
namespace webrtc {
// Implementation of IceCandidateInterface.
class JsepIceCandidate : public IceCandidateInterface {
public:
JsepIceCandidate(const std::string& sdp_mid, int sdp_mline_index);
JsepIceCandidate(const std::string& sdp_mid, int sdp_mline_index,
const cricket::Candidate& candidate);
~JsepIceCandidate();
// |err| may be null.
bool Initialize(const std::string& sdp, SdpParseError* err);
void SetCandidate(const cricket::Candidate& candidate) {
candidate_ = candidate;
}
virtual std::string sdp_mid() const { return sdp_mid_; }
virtual int sdp_mline_index() const { return sdp_mline_index_; }
virtual const cricket::Candidate& candidate() const {
return candidate_;
}
virtual std::string server_url() const { return candidate_.url(); }
virtual bool ToString(std::string* out) const;
private:
std::string sdp_mid_;
int sdp_mline_index_;
cricket::Candidate candidate_;
RTC_DISALLOW_COPY_AND_ASSIGN(JsepIceCandidate);
};
// Implementation of IceCandidateCollection which stores JsepIceCandidates.
class JsepCandidateCollection : public IceCandidateCollection {
public:
JsepCandidateCollection() {}
// Move constructor is defined so that a vector of JsepCandidateCollections
// can be resized.
JsepCandidateCollection(JsepCandidateCollection&& o)
: candidates_(std::move(o.candidates_)) {}
~JsepCandidateCollection();
virtual size_t count() const {
return candidates_.size();
}
virtual bool HasCandidate(const IceCandidateInterface* candidate) const;
// Adds and takes ownership of the JsepIceCandidate.
// TODO(deadbeef): Make this use an std::unique_ptr<>, so ownership logic is
// more clear.
virtual void add(JsepIceCandidate* candidate) {
candidates_.push_back(candidate);
}
virtual const IceCandidateInterface* at(size_t index) const {
return candidates_[index];
}
// Removes the candidate that has a matching address and protocol.
//
// Returns the number of candidates that were removed.
size_t remove(const cricket::Candidate& candidate);
private:
std::vector<JsepIceCandidate*> candidates_;
RTC_DISALLOW_COPY_AND_ASSIGN(JsepCandidateCollection);
};
} // namespace webrtc
#endif // WEBRTC_API_JSEPICECANDIDATE_H_