| 6955870806624479723addfae6dcf5d13968796c |
|
13-Jan-2016 |
Peter Kasting <pkasting@google.com> |
Convert channel counts to size_t.
IIRC, this was originally requested by ajm during review of the other size_t conversions I did over the past year, and I agreed it made sense, but wanted to do it separately since those changes were already gargantuan.
BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, henrika@webrtc.org, kjellander@webrtc.org, minyue@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org
Review URL: https://codereview.webrtc.org/1316523002 .
Cr-Commit-Position: refs/heads/master@{#11229}
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| 3f7219be700df3fea85193e8d541e7f90a1c3ce6 |
|
29-Dec-2015 |
deadbeef <deadbeef@webrtc.org> |
Fixing issue where description contains empty ICE ufrag/pwd.
The issue occurred when deserializing and then serializing a rejected
content description, which doesn't have the ICE ufrag/pwd in the first
place.
BUG=webrtc:5105
Review URL: https://codereview.webrtc.org/1534363002
Cr-Commit-Position: refs/heads/master@{#11134}
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| a54a0801121e05f797e514731cc5c9bad2f5e597 |
|
17-Dec-2015 |
honghaiz <honghaiz@webrtc.org> |
Add ufrag to the ICE candidate signaling.
On the receiving side, if a candidate arrives with an old ufrag, it will be dropped. If it contains a new frag that has never seen before, it will hold the ufrag and create connections, although those connections are not pingable until the ICE credentials are received.
This could avoid a bunch of ICE generation issues.
BUG=webrtc:5138,webrt:5292
Review URL: https://codereview.webrtc.org/1498993002
Cr-Commit-Position: refs/heads/master@{#11060}
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| 1387149ad1669365ac05278bf779a407bec08a4e |
|
09-Dec-2015 |
deadbeef <deadbeef@webrtc.org> |
Adding reduced size RTCP configuration down to the video stream level.
Still waiting to turn on negotiation (in mediasession.cc)
until we verify it's working as expected.
BUG=webrtc:4868
Review URL: https://codereview.webrtc.org/1418123003
Cr-Commit-Position: refs/heads/master@{#10958}
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| 5237aaf243d29732f59557361b7a993c0a18cf0e |
|
11-Nov-2015 |
tfarina <tfarina@chromium.org> |
Convert usage of ARRAY_SIZE to arraysize.
ARRAY_SIZE is the old version of arraysize and does not cover
all the cases in C++, arraysize is a copy of Chromium's
version and thus have wider coverage.
BUG=None
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1405023016
Cr-Commit-Position: refs/heads/master@{#10594}
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| c80741f8957b537e968397ac54ff5b5df8a2c318 |
|
22-Oct-2015 |
deadbeef <deadbeef@webrtc.org> |
Fixing some issues with the direction attribute of m-lines in offers.
By default, we'll now offer to receive if already receiving
(meaning that the last remote description contained a track).
Also, m-lines that are neither receiving nor sending are now correctly
marked "inactive".
Also moved some logic relating to default tracks out of webrtcsdp.cc,
such that now the direction seen by upper layers will always be
consistent with the consumed/produced SDP.
BUG=528089
Review URL: https://codereview.webrtc.org/1406803004
Cr-Commit-Position: refs/heads/master@{#10376}
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| 69f576010edae80bc83fbf51fa06c3ee611125e8 |
|
08-Oct-2015 |
lally <lally@webrtc.org> |
Added parsing of either space or colon for sctp-port.
BUG=https://code.google.com/p/webrtc/issues/detail?id=5039
Review URL: https://codereview.webrtc.org/1395523002
Cr-Commit-Position: refs/heads/master@{#10225}
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| 0c4e06b4c6107a1b94f764e279e4fb4161e905b0 |
|
07-Oct-2015 |
Peter Boström <pbos@webrtc.org> |
Use suffixed {uint,int}{8,16,32,64}_t types.
Removes the use of uint8, etc. in favor of uint8_t.
BUG=webrtc:5024
R=henrik.lundin@webrtc.org, henrikg@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org
Review URL: https://codereview.webrtc.org/1362503003 .
Cr-Commit-Position: refs/heads/master@{#10196}
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| 7cbd188c5ed7df80bb737bd4ada94422730e2d89 |
|
18-Sep-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Remove GICE (again).
R=guoweis@webrtc.org
Review URL: https://codereview.webrtc.org/1353713002 .
Cr-Commit-Position: refs/heads/master@{#9979}
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| d12140a68efdcffa1c2c18f25149905e9dae1a9c |
|
10-Sep-2015 |
guoweis <guoweis@webrtc.org> |
Revert change which removes GICE.
There are still dependencies on this functionality.
TBR=pthatcher@webrtc.org
BUG=526399
Review URL: https://codereview.webrtc.org/1336553003
Cr-Commit-Position: refs/heads/master@{#9920}
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| 2159b89fa2cb55beeef38f72bd45e217f3d33d4e |
|
22-Aug-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Reland "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots.
This reverts commit 5bdafd44c86ee46bd7e040f19828324583418b33.
Original CL: https://codereview.webrtc.org/1263663002/
R=guoweis@webrtc.org
Review URL: https://codereview.webrtc.org/1303393002 .
Cr-Commit-Position: refs/heads/master@{#9761}
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| 5bdafd44c86ee46bd7e040f19828324583418b33 |
|
21-Aug-2015 |
minyuel <minyue@webrtc.org> |
Revert "Reland "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots.""
This reverts commit 081f34b564e1a26ffbbe9515eba1fef7c736fdde.
Original code review see
https://codereview.webrtc.org/1291363005
The revert is due to a suspicion of "Reland "Remove GICE..." being the cause of failure on Linux memcheck, see
https://build.chromium.org/p/client.webrtc/builders/Linux%20Memcheck/builds/4137
TBR=pthatcher@webrtc.org,
BUG=
Review URL: https://codereview.webrtc.org/1308753003 .
Cr-Commit-Position: refs/heads/master@{#9756}
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| 081f34b564e1a26ffbbe9515eba1fef7c736fdde |
|
20-Aug-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Reland "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots."
This reverts commit 475243a134be003aab30bb17294ca6c664d0ef81.
R=guoweis@webrtc.org
Review URL: https://codereview.webrtc.org/1291363005 .
Cr-Commit-Position: refs/heads/master@{#9738}
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| fa301809b698017455847f45cc7e0dfa1bdfed35 |
|
11-Aug-2015 |
pthatcher <pthatcher@webrtc.org> |
Revert "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots.
This reverts commit 3449faa553ec94c52ef2d0949867befb60992c88.
TBR=deadbeef@webrtc.org, juberti@webrtc.org
NOPRESUBMIT=true
Review URL: https://codereview.webrtc.org/1274273005
Cr-Commit-Position: refs/heads/master@{#9698}
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| 3449faa553ec94c52ef2d0949867befb60992c88 |
|
10-Aug-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever).
R=deadbeef@webrtc.org, juberti@webrtc.org
Review URL: https://codereview.webrtc.org/1263663002 .
Cr-Commit-Position: refs/heads/master@{#9692}
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| a9b4c32052fd55df7e1d02e846fbea3178bebf71 |
|
16-Jul-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Nuke buffered latency mode. It's not actually working, and it's not used. It's just dead code complexity.
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1226093010 .
Cr-Commit-Position: refs/heads/master@{#9593}
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| 083b73fb95755b78cb0b9cbe67752b7e7b7eb263 |
|
16-Jul-2015 |
jbauch <jbauch@webrtc.org> |
Use std::string references instead of copying contents.
This CL improves the memory footprint a bit by using string references
instead of creating a copy.
Review URL: https://codereview.webrtc.org/1241973002
Cr-Commit-Position: refs/heads/master@{#9592}
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| bb36fdf95f9667fb1f3fbf3073bd15007681322c |
|
09-Jul-2015 |
pbos <pbos@webrtc.org> |
Remove empty-string comparisons.
Use .empty() and !.empty() in favor of == "" or != "".
BUG=
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1228913003
Cr-Commit-Position: refs/heads/master@{#9559}
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| c0c3a865f49a6386b0815001b9856c1eee27e7c2 |
|
25-Jun-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Prevent JS from bypassing RTP data channel bandwidth limitation.
Normally the RTP data channel is capped at 30kbps, but by mangling the
SDP string, one could get around this limitation. With this fix,
SdpDeserialize will return an error if it detects this condition.
BUG=280726
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1196403004.
Cr-Commit-Position: refs/heads/master@{#9499}
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| 144d01850bd3e07222d3f8696debec689dcdccf5 |
|
15-May-2015 |
Donald Curtis <decurtis@webrtc.org> |
fix indent on tokenize_first function signatures
R=juberti@google.com, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/52499004
Cr-Commit-Position: refs/heads/master@{#9198}
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| 0e07f92043b333acfdaed8f22da5df903a70e0e9 |
|
15-May-2015 |
Donald Curtis <decurtis@webrtc.org> |
Split fmtp on semicolons not spaces as per RFC6871
BUG=4617
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47169004
Cr-Commit-Position: refs/heads/master@{#9193}
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| 019087f5bb294d9590b4ed68347ad79d9335ad74 |
|
28-Apr-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Add safeguards against signalling peer-reflexive candidates.
BUG=4208
R=juberti@google.com
Review URL: https://webrtc-codereview.appspot.com/50799004
Cr-Commit-Position: refs/heads/master@{#9104}
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| 7100dcd3176f6522ee96be797f73a1f50da0f5d1 |
|
27-Mar-2015 |
Minyue Li <minyue@webrtc.org> |
Adding "usedtx" as Opus codec parameter.
This is according to https://tools.ietf.org/html/draft-spittka-payload-rtp-opus-03
Specifically,
usedtx: specifies if the decoder prefers the use of DTX. values are 1 and 0. If no value is specified, usedtx is assumed to be 0.
BUG=1014
R=juberti@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48499004
Cr-Commit-Position: refs/heads/master@{#8872}
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| 2d25b44f470afdd56513b75d641166f6e7cdcd04 |
|
16-Mar-2015 |
changbin.shao@webrtc.org <changbin.shao@webrtc.org> |
Check associated payload type when negotiate RTX codecs.
At the moment, only payload name is checked when match two RTX codecs.
This will cause wrong behavior of codec negotiation if multiple RTX codecs
are added.
BUG=
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34189004
Cr-Commit-Position: refs/heads/master@{#8727}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8727 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| a74709333482783cb06405626caf9555e407eba2 |
|
24-Feb-2015 |
lally@webrtc.org <lally@webrtc.org> |
After another round of reviews.
Cr-Commit-Position: refs/heads/master@{#8483}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8483 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| 9616196c38149e9a920d59da3019f47d1d61ff85 |
|
24-Feb-2015 |
lally@webrtc.org <lally@webrtc.org> |
Merging definitions of IsSctp.
Cr-Commit-Position: refs/heads/master@{#8482}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8482 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| 12aa8a68f95eb68a006fe112fabd149fab262c56 |
|
24-Feb-2015 |
lally@webrtc.org <lally@webrtc.org> |
Post-rebase.
Cr-Commit-Position: refs/heads/master@{#8481}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8481 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| 17308695963539ed3a125ba87635b81e12fac081 |
|
24-Feb-2015 |
lally@webrtc.org <lally@webrtc.org> |
Added raw SCTP to IsSctp.
Cr-Commit-Position: refs/heads/master@{#8480}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8480 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| 871b1c373ab2170056ac792bc228ba0e3c3b38b4 |
|
24-Feb-2015 |
lally@webrtc.org <lally@webrtc.org> |
Review comments -- added IsSctp()
Cr-Commit-Position: refs/heads/master@{#8479}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8479 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| d324546ced76d4e792338af4f7d02a5cd8819f92 |
|
23-Feb-2015 |
pkasting@chromium.org <pkasting@chromium.org> |
Misc. cleanup split out of https://webrtc-codereview.appspot.com/37699004/ :
* Move constants into the files/functions that use them
* Declare variables in the narrowest scope possible
* Use correct (expected, actual) order for gtest macros
* Remove unused functions
* Untabify
* 80-column limit
* Avoid C-style casts
* Prefer true typed constants to "enum hack" constants
* Print size_t using the right format macro
* Shorten and simplify code
* Other random cleanup bits and style fixes
BUG=none
TEST=none
R=henrik.lundin@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36179004
Cr-Commit-Position: refs/heads/master@{#8467}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8467 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| a744a28b92bac9a98816bc0cae0104c2ecdd0edb |
|
18-Feb-2015 |
jlmiller@webrtc.org <jlmiller@webrtc.org> |
Templatize and clean up codec wildcards.
BUG=4123
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39209004
Cr-Commit-Position: refs/heads/master@{#8422}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8422 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| e9facf8bb32a1688f2156009c755caa2904e1ac9 |
|
17-Feb-2015 |
pkasting@chromium.org <pkasting@chromium.org> |
Add range checks in a variety of places where the values will subsequently be
expected to be 0-127.
BUG=none
TEST=none
R=juberti@webrtc.org
TBR=henrika
Review URL: https://webrtc-codereview.appspot.com/37759004
Cr-Commit-Position: refs/heads/master@{#8399}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8399 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| 3341b401cce2b2e8dbec55bdae4261cf0fc19012 |
|
13-Feb-2015 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Fix bug parsing media descriptions: the final field isn't a codec type for any of DTLS/SCTP, SCTP, or SCTP/DTLS.
BUG=none
TEST=none
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34029004
Cr-Commit-Position: refs/heads/master@{#8369}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8369 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| 57ac2c84dd552dd7a56c5643163b1a5ce1dbf2ba |
|
06-Feb-2015 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Default destination used by c line should be IPv4 only to avoid parsing error in legacy client.
Make sure the IP family overwrites the preference of candidates. Also,
make sure only UDP is used as default destination.
BUG=4269
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36009004
Cr-Commit-Position: refs/heads/master@{#8258}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8258 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| 5f93d0a140515e3b8cdd1b9a4c6f5871144e5dee |
|
20-Jan-2015 |
jlmiller@webrtc.org <jlmiller@webrtc.org> |
Update libjingle license statements at top of talk files for consistency
BUG=2133
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8105 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| 61c1247224e2b696b10303b0b5479b3a246f4ff0 |
|
15-Jan-2015 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Fix a case where empty candidate id is used
BUG=4161
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35749004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8071 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| 8af11042cb1157b722b55304c292f3091290da4d |
|
07-Jan-2015 |
decurtis@webrtc.org <decurtis@webrtc.org> |
Avoid reading past end of string in GetLine.
BUG=3881
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8017 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| 950c51825109c2ca352317edef0a33777d0e6678 |
|
17-Dec-2014 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Add adapter_type into Candidate object.
Expose adapter_type from Candidate such that we could add jmidata on top of this.
Created a new type of report just for Ice candidate. The candidate's id is used as part of report identifier. This code change only reports the best connection's local candidate's adapter type. There should be cleaning later to move other candidate's attributes to the new report.
This is migrated from issue 32599004
BUG=
R=juberti@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=7885
Committed: https://code.google.com/p/webrtc/source/detail?r=7906
Review URL: https://webrtc-codereview.appspot.com/36379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7925 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| e2b7585bc277e211b7d9fc1e3e8046ea41484b5d |
|
16-Dec-2014 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Move ViewRequest and MediaStreams to streamparams.h, and remove dependency on mediasessionclient.h and mediamessages.h. This is part of the effort to remove Jingle-specific code from WebRTC and into its own repository.
R=juberti@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36519004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7921 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| 55360ae402908b24757c7983c587e69ea485e9e6 |
|
16-Dec-2014 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Revert "Add adapter_type into Candidate object."
This reverts commit aaf02cc2d4f696345ce0e6d5715f2cfa22aea689.
BUG=
TBR=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35539004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7908 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| aaf02cc2d4f696345ce0e6d5715f2cfa22aea689 |
|
16-Dec-2014 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Add adapter_type into Candidate object.
Expose adapter_type from Candidate such that we could add jmidata on top of this.
Created a new type of report just for Ice candidate. The candidate's id is used as part of report identifier. This code change only reports the best connection's local candidate's adapter type. There should be cleaning later to move other candidate's attributes to the new report.
This is migrated from issue 32599004
BUG=
R=juberti@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=7885
Review URL: https://webrtc-codereview.appspot.com/36379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7906 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| fb108b5a28a538862a4157e17de795426d86af1e |
|
15-Dec-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Revert r7885.
Breaks compile step of other code where network name of
cricket::Candidate is used.
TBR=guoweis@webrtc.org,juberti@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/31229004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7892 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| 18a3896bd28b63fa35168cd6c8d41c8cebaab3dd |
|
15-Dec-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Revert r7886:7887.
Broke build steps in other code that uses securetunnelsessionclient.cc
and others.
TBR=tommi@webrtc.org,pthatcher@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/36439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7890 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| dee76f3b89b9339699e0321a3afc643ee06afa09 |
|
12-Dec-2014 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Move the obvious/easy Jingle-specific code into webrtc/libjingle.
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7886 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| 8c9d79a29d9127d4ff8aa4ae386630c72cfb1808 |
|
12-Dec-2014 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Add adapter_type into Candidate object.
Expose adapter_type from Candidate such that we could add jmidata on top of this.
Created a new type of report just for Ice candidate. The candidate's id is used as part of report identifier. This code change only reports the best connection's local candidate's adapter type. There should be cleaning later to move other candidate's attributes to the new report.
This is migrated from issue 32599004
BUG=
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7885 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| d105cc81dc1f5792fd4165d6aec0a654f2dfc77c |
|
07-Nov-2014 |
perkj@webrtc.org <perkj@webrtc.org> |
Change dummy address to use 0.0.0.0 instead of ::
This is to not break compatiblity with FF.
https://code.google.com/p/chromium/issues/detail?id=430333
TBR=pthatcher@webrtc.org, juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31979004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7661 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| 269fb4bc90b79bebbb8311da0110ccd6803fd0a8 |
|
28-Oct-2014 |
henrike@webrtc.org <henrike@webrtc.org> |
move xmpp and p2p to webrtc
Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and
webrtc/p2p. Also makes libjingle use those version instead of the one in the talk folder.
BUG=3379
Review URL: https://webrtc-codereview.appspot.com/26999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7549 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| c9d6d140209bd2e8f44eb41fb0de17d512d39911 |
|
24-Oct-2014 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
patch from issue 25469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7517 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| 28100cb38896fe298b6df11ffd31838d9faf5b8a |
|
18-Oct-2014 |
henrike@webrtc.org <henrike@webrtc.org> |
Reverts r7459 "Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p."
BUG=N/A
TBR=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7472 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| d1ba6d9cbfc44618d2c553ff7851948c730ae37b |
|
15-Oct-2014 |
henrike@webrtc.org <henrike@webrtc.org> |
Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p.
BUG=3379
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27709005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7459 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| 5d639b3ef36c81a2330e5f0a4f7c119294400515 |
|
10-Sep-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 75141932-> 75179475
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7129 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| 0e52772aa9d3dea65e2cd30187c4ff8e86f9eee4 |
|
08-Sep-2014 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Fix a bot-breaking memory leak from early returning in ParseMediaDescription.
BUG=3791
R=henrike@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7109 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| ddb85ab85b233b4e038d7f0de093199199903a36 |
|
05-Sep-2014 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Updated SCTP SDP attributes according to draft-ietf-mmusic-sctp-sdp-07
- SDP sctpmap attribute replaced with fmtp attribute
- SDP sctp-port attribute is newly added
BUG=3592
R=jiayl@webrtc.org, juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16169004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7087 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| a09a99950ec40aef6421e4ba35eee7196b7a6e68 |
|
13-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 73222930-> 73226398
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6891 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| e999bd087bcc7307c9e9b253e78837486213d124 |
|
13-Aug-2014 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Removing ASSERT for tcp candidate for port 0 and 9, as Android clients
may not be called with set_allow_tcp_listen(false).
This CL will also sends tcp candidate in RFC 6544 format.
BUG=https://code.google.com/p/webrtc/issues/detail?id=3677
R=braveyao@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6880 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| 7ec3f9f838f83d17f7ac1a938f174152fc3767a7 |
|
09-Aug-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix a bug in parsing IceCandidate with IPV6 address.
It used to treat ":" as a candidate delimiter and got confused by the ":" in the IPV6 address.
The new logic is to check if the input has multiple lines. If so, returns error.
BUG=3669
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18079004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6859 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| 2d60c5e8bcac85e9388e093bae91ecc829eabcea |
|
09-Aug-2014 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Encoding and Decoding of TCP candidates as defined in RFC 6544.
R=juberti@chromium.org, jiayl@webrtc.org, juberti@webrtc.org
BUG=2204
Review URL: https://webrtc-codereview.appspot.com/21479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6857 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| d4e598d57aed714a599444a7eab5e8fdde52a950 |
|
29-Jul-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 72097588-> 72159069
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6799 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| 4c3e9917e7431ba1c0d20535602209310ce48ded |
|
16-Jul-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Make sure b lines appear before all the a lines. Per RFC 4566, the order of media description should be:
m= (media name and transport address)
i=* (media title)
c=* (connection information -- optional if included at
session level)
b=* (zero or more bandwidth information lines)
k=* (encryption key)
a=* (zero or more media attribute lines)
BUG=2260
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15889004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6708 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| ec9f5fb34cc612f26ec30f357ea3e3aa5d96c5c2 |
|
24-Jun-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Change SdpSerializeCandidate to output candidate line without the "a=" and without the leading \r\n", i.e. candidate-attribute as defined in section 15.1 of [ICE].
BUG=crbug/387632
R=juberti@google.com
Review URL: https://webrtc-codereview.appspot.com/17779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6533 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| 85d2794e5b57a1501a7fdade61eccd086e7a622d |
|
09-Jun-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adds support for the "apt" format parameter and turns on the RTX feature.
BUG=1811,1095
R=henrike@webrtc.org, mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12579009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6372 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| ed97bb0eb4e8b8b6a7c750d8bf5f8ad8fb5d0733 |
|
07-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 66340694-> 66388864
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6071 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| 9c16c39e613ebc5cdfa8ca5818a62ef5c3b18bd7 |
|
01-May-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Sets the SCTP port codec in the native SessionDescription.
Previously it's only set when a SDP string is parsed into SessionDescription, causing failuring for native client.
BUG=3141
R=juberti@webrtc.org, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6036 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| 36eda7cf0e16656fb4fcb7dd5e93b5555b824e56 |
|
15-Apr-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Workaround for https://bugzilla.mozilla.org/show_bug.cgi?id=996329, where the m line from firefox have a space at the end.
For example:
"m=application 38233 DTLS/SCTP 5000 "
BUG=3212
TEST=manually try to use DataChannel between FF 28 and Chrome with rtccopy.com
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12029005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5915 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| 5e760e7b94af0ecf3abbb793a793c2c551badece |
|
03-Apr-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Check the return value of the FromString call and return failure when then value is invalid. I.e. uses
bool FromString(const std::string& s, T* t)
instead of
T FromString(const std::string& str)
Before this change we will silently continue the parsing and take whatever default value returned by FromString.
TEST=new tests
BUG=2507
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11069004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5834 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| e42b8ab1293fc5c71e741f16ae920f50fe23c301 |
|
25-Mar-2014 |
sergeyu@chromium.org <sergeyu@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Cleanups in libjingle to make it compile with chromium_code=1
Fixed all warnings that show up when compiling libjingle
in chromium with compiling with chromium_code=1.
chromium_code=1 enables various warnings that are off by
default. Most changes are for unused variables and consts.
R=pthatcher@google.com, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9699004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5769 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| 704bf9ebec9c9425e1898f6c3f15eff685175b23 |
|
27-Feb-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 62063505-> 62278774
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5617 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| 571df2dca9357620e69690c562370680ddb67b6f |
|
20-Feb-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle 61759961->61834300
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5580 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| 5bc25c41fc7880545052770dbcfe67f233c9b0c0 |
|
05-Dec-2013 |
sergeyu@chromium.org <sergeyu@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle to 57692857
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5217 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| cecfd1832dc375225da3f5f18ecac63006ed06bf |
|
30-Oct-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 55821645.
TEST=try bots
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3139004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5053 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| 19f27e6a24f877fc2b0409a94b02d5f40ba3dc8c |
|
13-Oct-2013 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 54527154.
TBR=wu
Review URL: https://webrtc-codereview.appspot.com/2389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4954 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| 78187525665490922748d79377bcb351579e03c0 |
|
08-Oct-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle to 53856368.
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2366004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4941 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| 1112c30e1e5f5c7b4b517c4954ef3f15b989a996 |
|
23-Sep-2013 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle to 53057474.
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2274004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4818 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| 967bfff54d00f176a554bf9f955f14dde99f7bb9 |
|
19-Sep-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 52534915.
R=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/2251004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4786 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|
| a59696b2a5f0c138d4176249bac223ad6c4316d5 |
|
14-Sep-2013 |
sergeyu@chromium.org <sergeyu@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle to 52300956
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2213004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4744 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
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| cadf9040cbb9e7bb1b73a95e43e7d228fe6b2bdb |
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30-Aug-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 51664136.
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2148004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4649 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
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| 0be6aa0665a24ec8fd5edfdddd82a707a299508c |
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24-Aug-2013 |
sergeyu@chromium.org <sergeyu@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 51314459
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2100004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4608 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
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| 1e09a711263dd105e6f7a03812250084c64e5fd8 |
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26-Jul-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk folder to revision=49952949
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4413 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
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| 28654cbc2256230c978f41cbaf550bc2e9c2f2db |
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22-Jul-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk folder to revision=49713299.
TBR=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1848004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4380 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
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| 28e20752806a492f5a6a5d343c02f9556f39b1cd |
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10-Jul-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/talk/app/webrtc/webrtcsdp.cc
|