Move talk/app/webrtc to webrtc/api
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.
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
The HAVE_SCTP define was added for the peerconnection_unittests target
in api_tests.gyp.
I also checked that none of
SRTP_RELATIVE_PATH
HAVE_SRTP
HAVE_WEBRTC_VIDEO
HAVE_WEBRTC_VOICE
were used by the talk/app/webrtc 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/1615433002
BUG=webrtc:5418
NOPRESUBMIT=True
R=deadbeef@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1610243002 .
Cr-Commit-Position: refs/heads/master@{#11545}
diff --git a/webrtc/api/webrtcsdp.h b/webrtc/api/webrtcsdp.h
new file mode 100644
index 0000000..a75f735
--- /dev/null
+++ b/webrtc/api/webrtcsdp.h
@@ -0,0 +1,81 @@
+/*
+ * libjingle
+ * Copyright 2011 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.
+ */
+
+// This file contain functions for parsing and serializing SDP messages.
+// Related RFC/draft including:
+// * RFC 4566 - SDP
+// * RFC 5245 - ICE
+// * RFC 3388 - Grouping of Media Lines in SDP
+// * RFC 4568 - SDP Security Descriptions for Media Streams
+// * draft-lennox-mmusic-sdp-source-selection-02 -
+// Mechanisms for Media Source Selection in SDP
+
+#ifndef WEBRTC_API_WEBRTCSDP_H_
+#define WEBRTC_API_WEBRTCSDP_H_
+
+#include <string>
+
+namespace webrtc {
+
+class IceCandidateInterface;
+class JsepIceCandidate;
+class JsepSessionDescription;
+struct SdpParseError;
+
+// Serializes the passed in JsepSessionDescription.
+// Serialize SessionDescription including candidates if
+// JsepSessionDescription has candidates.
+// jdesc - The JsepSessionDescription object to be serialized.
+// return - SDP string serialized from the arguments.
+std::string SdpSerialize(const JsepSessionDescription& jdesc);
+
+// Serializes the passed in IceCandidateInterface to a SDP string.
+// candidate - The candidate to be serialized.
+std::string SdpSerializeCandidate(const IceCandidateInterface& candidate);
+
+// Deserializes the passed in SDP string to a JsepSessionDescription.
+// message - SDP string to be Deserialized.
+// jdesc - The JsepSessionDescription deserialized from the SDP string.
+// error - The detail error information when parsing fails.
+// return - true on success, false on failure.
+bool SdpDeserialize(const std::string& message,
+ JsepSessionDescription* jdesc,
+ SdpParseError* error);
+
+// Deserializes the passed in SDP string to one JsepIceCandidate.
+// The first line must be a=candidate line and only the first line will be
+// parsed.
+// message - The SDP string to be Deserialized.
+// candidates - The JsepIceCandidate from the SDP string.
+// error - The detail error information when parsing fails.
+// return - true on success, false on failure.
+bool SdpDeserializeCandidate(const std::string& message,
+ JsepIceCandidate* candidate,
+ SdpParseError* error);
+} // namespace webrtc
+
+#endif // WEBRTC_API_WEBRTCSDP_H_
