| f475d365a25036725c3f545f57de59d2cc902d17 |
|
09-Jan-2016 |
Taylor Brandstetter <deadbeef@webrtc.org> |
Properly handle different transports having different SSL roles.
This meant splitting "transport_options" into audio/video/data options,
for when creating the answer, and giving "GetSslRole" a "transport_name"
parameter so we can retrieve the current role on a per-transport basis.
BUG=webrtc:4525
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1516993002 .
Cr-Commit-Position: refs/heads/master@{#11192}
/external/webrtc/talk/session/media/mediasession.cc
|
| 44f0819978c2ba1f765835bca91e3243eb9f638b |
|
16-Dec-2015 |
deadbeef <deadbeef@webrtc.org> |
Fixing bug where "mid" wasn't preserved across re-offers.
Review URL: https://codereview.webrtc.org/1529673002
Cr-Commit-Position: refs/heads/master@{#11039}
/external/webrtc/talk/session/media/mediasession.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/session/media/mediasession.cc
|
| b5cb19b37c361a263a9cec2e2fb356d16520afd1 |
|
24-Nov-2015 |
deadbeef <deadbeef@webrtc.org> |
Fixing direction attribute in answer for non-RTP protocols.
"non-RTP protocols" refers to SCTP data channels. Because
there are no streams for SCTP data channels, the answer was being
set to RECVONLY.
BUG=webrtc:5228
Review URL: https://codereview.webrtc.org/1473013002
Cr-Commit-Position: refs/heads/master@{#10762}
/external/webrtc/talk/session/media/mediasession.cc
|
| 521ed7bf022c4e30574d7970c2be5be46567f4cd |
|
19-Nov-2015 |
Guo-wei Shieh <guoweis@webrtc.org> |
Reland Convert internal representation of Srtp cryptos from string to int
TBR=pthatcher@webrtc.org
BUG=webrtc:5043
Review URL: https://codereview.webrtc.org/1458023002 .
Cr-Commit-Position: refs/heads/master@{#10703}
/external/webrtc/talk/session/media/mediasession.cc
|
| 318166bed75dcbc00a7b79f715f9953aff9ffbc7 |
|
19-Nov-2015 |
guoweis <guoweis@webrtc.org> |
Revert of Convert internal representation of Srtp cryptos from string to int. (patchset #10 id:180001 of https://codereview.webrtc.org/1416673006/ )
Reason for revert:
Broke chromium fyi build.
Original issue's description:
> Convert internal representation of Srtp cryptos from string to int.
>
> Note that the coversion from int to string happens in 3 places
> 1) SDP layer from int to external names. mediasession.cc GetSupportedSuiteNames.
> 2) for SSL_CTX_set_tlsext_use_srtp(), converting from int to internal names.
> 3) stats collection also needs external names.
>
> External names are like AES_CM_128_HMAC_SHA1_80, specified in sslstreamadapter.cc.
> Internal names are like SRTP_AES128_CM_SHA1_80, specified in opensslstreamadapter.cc.
>
> The conversion from string to int happens in one place only, SDP layer, SrtpFilter::ApplyParams().
>
> BUG=webrtc:5043
>
> Committed: https://crrev.com/2764e1027a08a5543e04b854a27a520801faf6eb
> Cr-Commit-Position: refs/heads/master@{#10701}
TBR=juberti@webrtc.org,pthatcher@webrtc.org,juberti@google.com
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5043
Review URL: https://codereview.webrtc.org/1455233005
Cr-Commit-Position: refs/heads/master@{#10702}
/external/webrtc/talk/session/media/mediasession.cc
|
| 2764e1027a08a5543e04b854a27a520801faf6eb |
|
19-Nov-2015 |
guoweis <guoweis@webrtc.org> |
Convert internal representation of Srtp cryptos from string to int.
Note that the coversion from int to string happens in 3 places
1) SDP layer from int to external names. mediasession.cc GetSupportedSuiteNames.
2) for SSL_CTX_set_tlsext_use_srtp(), converting from int to internal names.
3) stats collection also needs external names.
External names are like AES_CM_128_HMAC_SHA1_80, specified in sslstreamadapter.cc.
Internal names are like SRTP_AES128_CM_SHA1_80, specified in opensslstreamadapter.cc.
The conversion from string to int happens in one place only, SDP layer, SrtpFilter::ApplyParams().
BUG=webrtc:5043
Review URL: https://codereview.webrtc.org/1416673006
Cr-Commit-Position: refs/heads/master@{#10701}
/external/webrtc/talk/session/media/mediasession.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/session/media/mediasession.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/session/media/mediasession.cc
|
| 456696a9c1bbd586701dcca3e4b2695e419a10ba |
|
01-Oct-2015 |
Guo-wei Shieh <guoweis@webrtc.org> |
Reland Change WebRTC SslCipher to be exposed as number only
This is to revert the change of https://codereview.webrtc.org/1380603005/
TBR=pthatcher@webrtc.org
BUG=523033
Review URL: https://codereview.webrtc.org/1375543003 .
Cr-Commit-Position: refs/heads/master@{#10126}
/external/webrtc/talk/session/media/mediasession.cc
|
| 27dc29b0df23eed5034f28d4d5f66ea0bb425d6c |
|
01-Oct-2015 |
guoweis <guoweis@webrtc.org> |
Revert of Change WebRTC SslCipher to be exposed as number only. (patchset #20 id:750001 of https://codereview.webrtc.org/1337673002/ )
Reason for revert:
This broke chromium.fyi bot.
Original issue's description:
> Change WebRTC SslCipher to be exposed as number only.
>
> This makes the SSL exposed as uint16_t which is the IANA value. GetRfcSslCipherName is introduced to handle the conversion to names from ID. IANA value will be used for UMA reporting. Names will still be used for WebRTC stats reporting.
>
> For SRTP, currently it's still string internally but is reported as IANA number.
>
> This is used by the ongoing CL https://codereview.chromium.org/1335023002.
>
> BUG=523033
>
> Committed: https://crrev.com/4fe3c9a77386598db9abd1f0d6983aefee9cc943
> Cr-Commit-Position: refs/heads/master@{#10124}
TBR=juberti@webrtc.org,rsleevi@chromium.org,pthatcher@webrtc.org,davidben@chromium.org,juberti@google.com,davidben@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=523033
Review URL: https://codereview.webrtc.org/1380603005
Cr-Commit-Position: refs/heads/master@{#10125}
/external/webrtc/talk/session/media/mediasession.cc
|
| 4fe3c9a77386598db9abd1f0d6983aefee9cc943 |
|
01-Oct-2015 |
guoweis <guoweis@webrtc.org> |
Change WebRTC SslCipher to be exposed as number only.
This makes the SSL exposed as uint16_t which is the IANA value. GetRfcSslCipherName is introduced to handle the conversion to names from ID. IANA value will be used for UMA reporting. Names will still be used for WebRTC stats reporting.
For SRTP, currently it's still string internally but is reported as IANA number.
This is used by the ongoing CL https://codereview.chromium.org/1335023002.
BUG=523033
Review URL: https://codereview.webrtc.org/1337673002
Cr-Commit-Position: refs/heads/master@{#10124}
/external/webrtc/talk/session/media/mediasession.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/session/media/mediasession.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/session/media/mediasession.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/session/media/mediasession.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/session/media/mediasession.cc
|
| a5b273a635b9876f88430934de19a883a1fb5728 |
|
21-Aug-2015 |
deadbeef <deadbeef@webrtc.org> |
Fixing problems with RTP extension ID conflict resolution
If the same extension URI is used for both audio and video (such as
abs-send-time), we should be able to re-use the same ID. A conflict
only exists if two different URIs are attempting to use the same ID.
Review URL: https://codereview.webrtc.org/1286273003
Cr-Commit-Position: refs/heads/master@{#9749}
/external/webrtc/talk/session/media/mediasession.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/session/media/mediasession.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/session/media/mediasession.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/session/media/mediasession.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/session/media/mediasession.cc
|
| f39382943449b7e44ac563e05a14203534591acf |
|
15-Jul-2015 |
deadbeef <deadbeef@webrtc.org> |
Use "UDP/TLS/RTP/SAVPF" profile in offer when DTLS-SRTP is used.
Tested that this doesn't break compatibility with Firefox or older
versions of Chrome, no matter which side generates the initial offer.
BUG=webrtc:2796
Review URL: https://codereview.webrtc.org/1219333002
Cr-Commit-Position: refs/heads/master@{#9589}
/external/webrtc/talk/session/media/mediasession.cc
|
| 2e7a09800595d4d82f67acfd7de04794642cef7d |
|
18-May-2015 |
Noah Richards <noahric@chromium.org> |
Ensure mediasession generated offers with RTX contain an RTX ssrc for each video ssrc.
BUG=
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49989004
Cr-Commit-Position: refs/heads/master@{#9210}
/external/webrtc/talk/session/media/mediasession.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/session/media/mediasession.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/session/media/mediasession.cc
|
| ec97c6516f0f2540a8e040d08de86709be6ab5b4 |
|
24-Feb-2015 |
lally@webrtc.org <lally@webrtc.org> |
Attempt on read-only acceptance of -12.
Cr-Commit-Position: refs/heads/master@{#8477}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8477 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/mediasession.cc
|
| 586f2eda0d90b84ffefdf2c3662073f22af73bdb |
|
23-Jan-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
Change GetStreamBySsrc to not copy StreamParams.
This is something I stumbled upon while looking at string copying we do (in spades) and did a simple change to not be constantly copying things around needlessly. There's a lot more that can be done in these files of course so this is sort of a reminder for future code edits that it's possible to design interfaces/function in a way that's more performance aware and avoid forcing creation of copies, while still being very simple. Also, we can use lambdas now :)
BUG=
R=perkj@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8131 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/mediasession.cc
|
| 5ad4178137ac869f1e057e07c3a171e11763d9df |
|
23-Dec-2014 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Move the Jingle-specific network code into webrtc/libjingle.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29319004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7977 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/mediasession.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/session/media/mediasession.cc
|
| f15dee6980152cded2f10c26748d7d88ab9501ae |
|
27-Oct-2014 |
tommi@webrtc.org <tommi@webrtc.org> |
Check if a datachannel in the current local description is an sctp channel before assuming rtp.
When generating an offer from a local description when 'sctp' is not explicitly set in the
media session options, we were generating an offer with an RTP datachannel even though the
channel in the local description was already sctp.
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7539 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/mediasession.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/session/media/mediasession.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/session/media/mediasession.cc
|
| 742922b313baaebfbacf735287f9729a8bc6f8e0 |
|
07-Oct-2014 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Make the media content send only if offerToReceive is false while local streams exist.
We previously do not add the media content if offerToReceive is false.
BUG=3833
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26609004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7390 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/mediasession.cc
|
| 7d4891d3f18861bdd5ec5d27409110cf3d110fa1 |
|
09-Sep-2014 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Fixes two issues in how we handle OfferToReceiveX for CreateOffer:
1. the options set in the first CreateOffer call should not affect the result of a second CreateOffer call, if SetLocalDescription is not called after the first CreateOffer. So the member var options_ of MediaStreamSignaling is removed to make each CreateOffer independent.
Instead, MediaSession is responsible to make sure that an m-line in the current local description is never removed from the newly created offer.
2. OfferToReceiveAudio used to default to true, i.e. always having m=audio line even if no audio track. This is changed so that by default m=audio is only added if there are any audio tracks.
BUG=2108
R=pthatcher@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=7068
Review URL: https://webrtc-codereview.appspot.com/16309004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7124 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/mediasession.cc
|
| c172320bd22311a0cf8c7c51c5c782e321622de1 |
|
08-Sep-2014 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Revert "Fixes two issues in how we handle OfferToReceiveX for CreateOffer:" because it broke content_browsertests on Android.
This reverts commit r7068.
TBR=kjellander@webrtc.org
BUG=2108
Review URL: https://webrtc-codereview.appspot.com/23539004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7108 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/mediasession.cc
|
| 52055a276df3b0b0c3ed4c58ea74e0a4d8fe3891 |
|
04-Sep-2014 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Fixes two issues in how we handle OfferToReceiveX for CreateOffer:
1. the options set in the first CreateOffer call should not affect the result of a second CreateOffer call, if SetLocalDescription is not called after the first CreateOffer. So the member var options_ of MediaStreamSignaling is removed to make each CreateOffer independent.
Instead, MediaSession is responsible to make sure that an m-line in the current local description is never removed from the newly created offer.
2. OfferToReceiveAudio used to default to true, i.e. always having m=audio line even if no audio track. This is changed so that by default m=audio is only added if there are any audio tracks.
BUG=2108
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16309004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7068 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/mediasession.cc
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| a09a99950ec40aef6421e4ba35eee7196b7a6e68 |
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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/session/media/mediasession.cc
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| 56d8e05238a46bfc51fcb804bc1f5477dfefcc14 |
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06-Aug-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
A followup to r6828 to fix a condition check in mediasession.cc.
BUG=2395
R=juberti@chromium.org, juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17029004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6832 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/mediasession.cc
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| e7d47a1473e885a57986dcdbf06e7e1d25226ca6 |
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05-Aug-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Maintain the order of the m-lines in CreateOffer and CreateAnswer.
The order in the offer follows the order in the current local description.
The order in the answer follows the order in the current offer.
BUG=2395
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6828 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/mediasession.cc
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| d4e598d57aed714a599444a7eab5e8fdde52a950 |
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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/session/media/mediasession.cc
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| ff1b1bf0944d42700edadae68bd774835a7a13f0 |
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20-Jun-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
When creating an answer, takes the codec preference from the offer.
This change is based on RFC3264:
"Although the answerer MAY list the formats in their desired order of preference, it is RECOMMENDED that unless there is a specific reason, the answerer list formats in the same relative order they were present in the offer."
BUG=2868
TEST=unit tests and manually with munge-sdp test
R=juberti@google.com
Review URL: https://webrtc-codereview.appspot.com/14589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6514 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/mediasession.cc
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| 8dcd43c4f71da88f75ca46ed5868eb8812e1d6f7 |
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30-May-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Make MediaSessionDescriptionFactory accept offers with UDP/TLS/RTP/SAVPF.
This is the first step toward switching completely to UDP/TLS/RTP/SAVPF.
BUG=2796
R=juberti@webrtc.org, pthatcher@google.com
Review URL: https://webrtc-codereview.appspot.com/13439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6276 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/mediasession.cc
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| 9c16c39e613ebc5cdfa8ca5818a62ef5c3b18bd7 |
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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/session/media/mediasession.cc
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| 79047f99c1d39c6d3c16bd9bf0db3fb2eb1741bc |
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07-Mar-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 62691533-> 62713454
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5653 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/mediasession.cc
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| b90991dade9139e5c14c3b616a9eff07b9d6fdda |
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04-Mar-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle 62472237->62550414
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5640 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/mediasession.cc
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| 4b26e2eee3e3b2a0c22946372a38f7efa6cee146 |
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16-Jan-2014 |
sergeyu@chromium.org <sergeyu@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle to 59676287
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7229004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5390 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/mediasession.cc
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| cecfd1832dc375225da3f5f18ecac63006ed06bf |
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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/session/media/mediasession.cc
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| 97077a3ab27259164eb121034b6e0ebe9ba592df |
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25-Oct-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle to 55618622.
Update libyuv to r826.
TEST=try bots
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2889004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5038 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/mediasession.cc
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| 19f27e6a24f877fc2b0409a94b02d5f40ba3dc8c |
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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/session/media/mediasession.cc
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| 78187525665490922748d79377bcb351579e03c0 |
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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/session/media/mediasession.cc
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| a27be8e4a1f59a51ecafba71ba30ddd0bcc9f1f1 |
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28-Sep-2013 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle to CL 53398036.
Review URL: https://webrtc-codereview.appspot.com/2323004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4872 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/mediasession.cc
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| 1112c30e1e5f5c7b4b517c4954ef3f15b989a996 |
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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/session/media/mediasession.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/session/media/mediasession.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/session/media/mediasession.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/session/media/mediasession.cc
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