Adds trunk/talk folder of revision 359 from libjingles google code to
trunk/talk
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/talk/app/webrtc/jsepsessiondescription.h b/talk/app/webrtc/jsepsessiondescription.h
new file mode 100644
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+++ b/talk/app/webrtc/jsepsessiondescription.h
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+/*
+ * libjingle
+ * Copyright 2012, Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ * 1. Redistributions of source code must retain the above copyright notice,
+ * this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright notice,
+ * this list of conditions and the following disclaimer in the documentation
+ * and/or other materials provided with the distribution.
+ * 3. The name of the author may not be used to endorse or promote products
+ * derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+// Implements the SessionDescriptionInterface.
+
+#ifndef TALK_APP_WEBRTC_JSEPSESSIONDESCRIPTION_H_
+#define TALK_APP_WEBRTC_JSEPSESSIONDESCRIPTION_H_
+
+#include <string>
+#include <vector>
+
+#include "talk/app/webrtc/jsep.h"
+#include "talk/app/webrtc/jsepicecandidate.h"
+#include "talk/base/scoped_ptr.h"
+
+namespace cricket {
+class SessionDescription;
+}
+
+namespace webrtc {
+
+class JsepSessionDescription : public SessionDescriptionInterface {
+ public:
+ explicit JsepSessionDescription(const std::string& type);
+ virtual ~JsepSessionDescription();
+
+ // |error| can be NULL if don't care about the failure reason.
+ bool Initialize(const std::string& sdp, SdpParseError* error);
+
+ // Takes ownership of |description|.
+ bool Initialize(cricket::SessionDescription* description,
+ const std::string& session_id,
+ const std::string& session_version);
+
+ virtual cricket::SessionDescription* description() {
+ return description_.get();
+ }
+ virtual const cricket::SessionDescription* description() const {
+ return description_.get();
+ }
+ virtual std::string session_id() const {
+ return session_id_;
+ }
+ virtual std::string session_version() const {
+ return session_version_;
+ }
+ virtual std::string type() const {
+ return type_;
+ }
+ // Allow changing the type. Used for testing.
+ void set_type(const std::string& type) { type_ = type; }
+ virtual bool AddCandidate(const IceCandidateInterface* candidate);
+ virtual size_t number_of_mediasections() const;
+ virtual const IceCandidateCollection* candidates(
+ size_t mediasection_index) const;
+ virtual bool ToString(std::string* out) const;
+
+ // Default video encoder settings. The resolution is the max resolution.
+ // TODO(perkj): Implement proper negotiation of video resolution.
+ static const int kDefaultVideoCodecId;
+ static const int kDefaultVideoCodecFramerate;
+ static const char kDefaultVideoCodecName[];
+ static const int kMaxVideoCodecWidth;
+ static const int kMaxVideoCodecHeight;
+ static const int kDefaultVideoCodecPreference;
+
+ private:
+ talk_base::scoped_ptr<cricket::SessionDescription> description_;
+ std::string session_id_;
+ std::string session_version_;
+ std::string type_;
+ std::vector<JsepCandidateCollection> candidate_collection_;
+
+ bool GetMediasectionIndex(const IceCandidateInterface* candidate,
+ size_t* index);
+
+ DISALLOW_COPY_AND_ASSIGN(JsepSessionDescription);
+};
+
+} // namespace webrtc
+
+#endif // TALK_APP_WEBRTC_JSEPSESSIONDESCRIPTION_H_
