15583c19d7
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}
82 lines
3.5 KiB
C++
82 lines
3.5 KiB
C++
/*
|
|
* 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_
|