Files
platform-external-webrtc/talk/app/webrtc/sctputils.h
deadbeef ab9b2d1516 Reland of Moving MediaStreamSignaling logic into PeerConnection. (patchset #1 id:1 of https://codereview.webrtc.org/1403633005/ )
Reason for reland:
The original CL actually didn't break browser_tests; it was
just a coincidence that it started failing.

Original issue's description:
> Revert of Moving MediaStreamSignaling logic into PeerConnection. (patchset #10 id:180001 of https://codereview.webrtc.org/1393563002/ )
>
> Reason for revert:
> Broke browser_tests on Mac. Still need to investigate the cause.
>
> Original issue's description:
> > Moving MediaStreamSignaling logic into PeerConnection.
> >
> > This needs to happen because in the future, m-lines will be offered
> > based on the set of RtpSenders/RtpReceivers, rather than the set of
> > tracks that MediaStreamSignaling knows about.
> >
> > Besides that, MediaStreamSignaling was a "glue class" without
> > a clearly defined role, so it going away is good for other
> > reasons as well.
> >
> > Committed: https://crrev.com/97c392935411398b506861601c82e31d95c591f0
> > Cr-Commit-Position: refs/heads/master@{#10268}
>
> TBR=pthatcher@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
>
> Committed: https://crrev.com/fc648b6d934e936f4d9a32c813364b331536ec3b
> Cr-Commit-Position: refs/heads/master@{#10269}

TBR=pthatcher@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review URL: https://codereview.webrtc.org/1404473005

Cr-Commit-Position: refs/heads/master@{#10277}
2015-10-14 18:33:20 +00:00

59 lines
2.3 KiB
C++

/*
* libjingle
* Copyright 2013 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.
*/
#ifndef TALK_APP_WEBRTC_SCTPUTILS_H_
#define TALK_APP_WEBRTC_SCTPUTILS_H_
#include <string>
#include "talk/app/webrtc/datachannelinterface.h"
namespace rtc {
class Buffer;
} // namespace rtc
namespace webrtc {
struct DataChannelInit;
// Read the message type and return true if it's an OPEN message.
bool IsOpenMessage(const rtc::Buffer& payload);
bool ParseDataChannelOpenMessage(const rtc::Buffer& payload,
std::string* label,
DataChannelInit* config);
bool ParseDataChannelOpenAckMessage(const rtc::Buffer& payload);
bool WriteDataChannelOpenMessage(const std::string& label,
const DataChannelInit& config,
rtc::Buffer* payload);
void WriteDataChannelOpenAckMessage(rtc::Buffer* payload);
} // namespace webrtc
#endif // TALK_APP_WEBRTC_SCTPUTILS_H_