| 2d110be77f14cab0bb51efe8b61d9c7a967d04cb |
|
13-Jan-2016 |
deadbeef <deadbeef@webrtc.org> |
Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ )
Reason for revert:
tommi pointed out that using a refptr for the sink may cause issues. Will reland with a slightly different approach.
Original issue's description:
> Storing raw audio sink for default audio track.
>
> BUG=webrtc:5250
>
> Committed: https://crrev.com/e591f9377f33f3f725a30faecd1bef1a71fa6b99
> Cr-Commit-Position: refs/heads/master@{#11230}
TBR=pthatcher@webrtc.org,solenberg@webrtc.org,pbos@webrtc.org,tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5250
Review URL: https://codereview.webrtc.org/1588693002
Cr-Commit-Position: refs/heads/master@{#11241}
/external/webrtc/talk/session/media/channel.h
|
| e591f9377f33f3f725a30faecd1bef1a71fa6b99 |
|
13-Jan-2016 |
deadbeef <deadbeef@webrtc.org> |
Storing raw audio sink for default audio track.
BUG=webrtc:5250
Review URL: https://codereview.webrtc.org/1551813002
Cr-Commit-Position: refs/heads/master@{#11230}
/external/webrtc/talk/session/media/channel.h
|
| e6bf587259da23e96a8de0957b172fd74c36c3c6 |
|
21-Dec-2015 |
nisse <nisse@webrtc.org> |
Deleted VideoCapturer::screencast_max_pixels, together with
VideoChannel::GetScreencastMaxPixels and VideoChannel::GetScreencastFps.
Unused in webrtc, also unused in everything indexed by google and chromium code search. With the exception of the magicflute plugin, which I'm told doesn't matter.
Review URL: https://codereview.webrtc.org/1532133002
Cr-Commit-Position: refs/heads/master@{#11108}
/external/webrtc/talk/session/media/channel.h
|
| 4638331fd8857b263bb65f12dbf5e1f7005e1a9a |
|
18-Dec-2015 |
guoweis <guoweis@webrtc.org> |
DTLS-SRTP set up is bypassed when the channel has been writable.
This regression was introduced by CL 1505573002 to support remote fingerprint update. What happened is that during PrAnswer, we incorrectly do not apply bundle. However, the channel has become writable at that time. When Answer comes, we still reset the srtp_filter but since the channel has been writable, the new SRTP context has never been applied.
We're making sure that we could always apply SRTP context even when channel has been writable. We'll address the issue that bundle should apply even in PrAnswer in a different CL.
BUG=568734
Review URL: https://codereview.webrtc.org/1532543003
Cr-Commit-Position: refs/heads/master@{#11075}
/external/webrtc/talk/session/media/channel.h
|
| f888bb58da04c5095759b5ec7ce2e1fa2cd414fd |
|
12-Dec-2015 |
Tommi <tommi@webrtc.org> |
Support for unmixed remote audio into tracks.
BUG=chromium:121673
R=solenberg@webrtc.org
Review URL: https://codereview.webrtc.org/1505253004 .
Cr-Commit-Position: refs/heads/master@{#10995}
/external/webrtc/talk/session/media/channel.h
|
| 1218d7ad2fac035376914bd0649fe99e657b33d3 |
|
05-Dec-2015 |
Guo-wei Shieh <guoweis@webrtc.org> |
Allow remote fingerprint update during a call
Changes include the following
1. modify FakeDtlsIdentityStore to support alternate certificate so we could have a different fingerprint in test case.
2. dtlstransportchannel can accept a new fingerprint and trigger DTLS handshake.
3. #2 will trigger new signal on the media side to reset SRTP context. Only reset SRTP context when we are using DTLS (not SDES).
4. Test cases for caller or callee are transfees.
TBR=pthatcher@webrtc.org
BUG=webrtc:3618
This is a reland of https://codereview.webrtc.org/1453523002
Review URL: https://codereview.webrtc.org/1505573002 .
Cr-Commit-Position: refs/heads/master@{#10903}
/external/webrtc/talk/session/media/channel.h
|
| 86aaa4be8de8f49f91faeefbfd1a23f312898dd2 |
|
05-Dec-2015 |
Guo-wei Shieh <guoweis@webrtc.org> |
Revert "Allow remote fingerprint update during a call"
This reverts commit 9c38c2d33fa6d794704d53b18f39d5235439fe63.
This commit somehow is different from what I have in my local copy. Revert and will recommit.
TBR=pthatcher@webrtc.org
BUG=3618
Review URL: https://codereview.webrtc.org/1494373004 .
Cr-Commit-Position: refs/heads/master@{#10902}
/external/webrtc/talk/session/media/channel.h
|
| 9c38c2d33fa6d794704d53b18f39d5235439fe63 |
|
05-Dec-2015 |
Guo-wei Shieh <guoweis@webrtc.org> |
Allow remote fingerprint update during a call
Changes include the following
1. modify FakeDtlsIdentityStore to support alternate certificate so we could have a different fingerprint in test case.
2. dtlstransportchannel can accept a new fingerprint and trigger DTLS handshake.
3. #2 will trigger new signal on the media side to reset SRTP context. Only reset SRTP context when we are using DTLS (not SDES).
4. Test cases for caller or callee are transfees.
BUG=webrtc:3618
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1453523002 .
Cr-Commit-Position: refs/heads/master@{#10901}
/external/webrtc/talk/session/media/channel.h
|
| 1d63dd0eaa44d13c5ae083200937b18bce2132ae |
|
02-Dec-2015 |
solenberg <solenberg@webrtc.org> |
- Remove cricket::VoiceChannel::PressDtmf(); AFAICT unused.
- Remove the DF_PLAY/DF_SEND flags, only allow sending.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1487393002
Cr-Commit-Position: refs/heads/master@{#10872}
/external/webrtc/talk/session/media/channel.h
|
| 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/channel.h
|
| 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/channel.h
|
| 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/channel.h
|
| ec9d187f708933c75c3b48cf62296c37c7c506d9 |
|
27-Oct-2015 |
rlester <rlester@google.com> |
Added override keyword to overridden methods to stop compiler warnings.
BUG=
Review URL: https://codereview.webrtc.org/1417543002
Cr-Commit-Position: refs/heads/master@{#10433}
/external/webrtc/talk/session/media/channel.h
|
| c1aeaf0dc37d96f31c92f893f4e30e7a5f7cc2b7 |
|
15-Oct-2015 |
stefan <stefan@webrtc.org> |
Wire up packet_id / send time callbacks to webrtc via libjingle.
BUG=webrtc:4173
Review URL: https://codereview.webrtc.org/1363573002
Cr-Commit-Position: refs/heads/master@{#10289}
/external/webrtc/talk/session/media/channel.h
|
| d4cec0d8fa7913bc9dfa9137e44cca9098e16698 |
|
09-Oct-2015 |
solenberg <solenberg@webrtc.org> |
Remove MediaChannel::SetRemoteRenderer().
This is following discussion in: https://codereview.webrtc.org/1385893002/diff/60001/talk/media/webrtc/webrtcvoiceengine.cc#newcode2410
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1398823003
Cr-Commit-Position: refs/heads/master@{#10237}
/external/webrtc/talk/session/media/channel.h
|
| 4bac9c53da9988741d59753c2d789adb94de5e68 |
|
09-Oct-2015 |
solenberg <solenberg@webrtc.org> |
Change SetOutputScaling to set a single level, not left/right levels.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1397773002
Cr-Commit-Position: refs/heads/master@{#10234}
/external/webrtc/talk/session/media/channel.h
|
| 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/channel.h
|
| 5b14b42e93f17d0ea57f1f8b3e8224082c514946 |
|
01-Oct-2015 |
solenberg <solenberg@webrtc.org> |
Remove unused SignalMediaError and infrastructure.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1362913004
Cr-Commit-Position: refs/heads/master@{#10133}
/external/webrtc/talk/session/media/channel.h
|
| dfc8f4ff8731390828884a0a91b99e51f2950275 |
|
01-Oct-2015 |
solenberg <solenberg@webrtc.org> |
Change 'mute' parameter of MediaChannel::SetAudioSend()/SetVideoSend() to 'enable'.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1378513003
Cr-Commit-Position: refs/heads/master@{#10130}
/external/webrtc/talk/session/media/channel.h
|
| 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/channel.h
|
| 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/channel.h
|
| 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/channel.h
|
| cbecd358e032021eac11fb13e04ec7f070d4f407 |
|
23-Sep-2015 |
deadbeef <deadbeef@webrtc.org> |
Reland of TransportController refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/1358413003/ )
Reason for revert:
This CL just landed: https://codereview.chromium.org/1323243006/
Which fixes the FYI bots for the original CL, and breaks them for this revert.
Original issue's description:
> Revert of TransportController refactoring. (patchset #6 id:100001 of https://codereview.webrtc.org/1350523003/ )
>
> Reason for revert:
> This CL causes problems with the WebRTC-in-Chromium FYI bots. Presumably it needs to be done in several steps, where removed files are emptied instead of removed in the first step.
>
> Original issue's description:
> > TransportController refactoring.
> >
> > Getting rid of TransportProxy, and in its place adding a
> > TransportController class which will facilitate access to and manage
> > the lifetimes of Transports. These Transports will now be accessed
> > solely from the worker thread, simplifying their implementation.
> >
> > This refactoring also pulls Transport-related code out of BaseSession.
> > Which means that BaseChannels will now rely on the TransportController
> > interface to create channels, rather than BaseSession.
> >
> > Committed: https://crrev.com/47ee2f3b9f33e8938948c482c921d4e13a3acd83
> > Cr-Commit-Position: refs/heads/master@{#10022}
>
> TBR=pthatcher@webrtc.org,deadbeef@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
>
> Committed: https://crrev.com/a81a42f584baa0d93a4b93da9632415e8922450c
> Cr-Commit-Position: refs/heads/master@{#10024}
TBR=pthatcher@webrtc.org,torbjorng@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1361773005
Cr-Commit-Position: refs/heads/master@{#10036}
/external/webrtc/talk/session/media/channel.h
|
| a81a42f584baa0d93a4b93da9632415e8922450c |
|
23-Sep-2015 |
torbjorng <torbjorng@webrtc.org> |
Revert of TransportController refactoring. (patchset #6 id:100001 of https://codereview.webrtc.org/1350523003/ )
Reason for revert:
This CL causes problems with the WebRTC-in-Chromium FYI bots. Presumably it needs to be done in several steps, where removed files are emptied instead of removed in the first step.
Original issue's description:
> TransportController refactoring.
>
> Getting rid of TransportProxy, and in its place adding a
> TransportController class which will facilitate access to and manage
> the lifetimes of Transports. These Transports will now be accessed
> solely from the worker thread, simplifying their implementation.
>
> This refactoring also pulls Transport-related code out of BaseSession.
> Which means that BaseChannels will now rely on the TransportController
> interface to create channels, rather than BaseSession.
>
> Committed: https://crrev.com/47ee2f3b9f33e8938948c482c921d4e13a3acd83
> Cr-Commit-Position: refs/heads/master@{#10022}
TBR=pthatcher@webrtc.org,deadbeef@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1358413003
Cr-Commit-Position: refs/heads/master@{#10024}
/external/webrtc/talk/session/media/channel.h
|
| 47ee2f3b9f33e8938948c482c921d4e13a3acd83 |
|
23-Sep-2015 |
deadbeef <deadbeef@webrtc.org> |
TransportController refactoring.
Getting rid of TransportProxy, and in its place adding a
TransportController class which will facilitate access to and manage
the lifetimes of Transports. These Transports will now be accessed
solely from the worker thread, simplifying their implementation.
This refactoring also pulls Transport-related code out of BaseSession.
Which means that BaseChannels will now rely on the TransportController
interface to create channels, rather than BaseSession.
Review URL: https://codereview.webrtc.org/1350523003
Cr-Commit-Position: refs/heads/master@{#10022}
/external/webrtc/talk/session/media/channel.h
|
| 22011c1b54021ec9a2b4885519e5ce995b1300a2 |
|
22-Sep-2015 |
solenberg <solenberg@webrtc.org> |
Remove Channel::SetRingbackTone() and Channel::PlayRingbackTone(), and the code beneath it (within libjingle).
BUG=webrtc:4690
TBR=juberti
Review URL: https://codereview.webrtc.org/1325023005
Cr-Commit-Position: refs/heads/master@{#10011}
/external/webrtc/talk/session/media/channel.h
|
| 8902433a43bbc9cc0de4966774d3dbbe37ef96fb |
|
18-Sep-2015 |
Guo-wei Shieh <guoweis@webrtc.org> |
Revert "TransportController refactoring."
This reverts commit 9af63f473e1d0d6c47a741a046c41642dfc1c178.
Cr-Commit-Position: refs/heads/master@{#9994}
/external/webrtc/talk/session/media/channel.h
|
| 9af63f473e1d0d6c47a741a046c41642dfc1c178 |
|
18-Sep-2015 |
deadbeef <deadbeef@webrtc.org> |
TransportController refactoring.
Getting rid of TransportProxy, and in its place adding a
TransportController class which will facilitate access to and manage
the lifetimes of Transports. These Transports will now be accessed
solely from the worker thread, simplifying their implementation.
This refactoring also pulls Transport-related code out of BaseSession.
Which means that BaseChannels will now rely on the TransportController
interface to create channels, rather than BaseSession.
This CL also adds some unit tests, and does some renaming.
For example, from "CandidateReady" to "CandidateGathered".
Review URL: https://codereview.webrtc.org/1246913005
Cr-Commit-Position: refs/heads/master@{#9993}
/external/webrtc/talk/session/media/channel.h
|
| 1dd98f321920c1442dd5b3f791ea0fca133c2756 |
|
10-Sep-2015 |
solenberg <solenberg@webrtc.org> |
- Rename VoiceChannel::MuteStream() -> SetAudioSend() (incl. media channel)
- Rename VideoChannel::MuteStream() -> SetVideoSend() (incl. media channel)
- Collapse NnChannel::SetChannelOptions() into the above.
- Collapse VoiceChannel::SetLocalRenderer into SetAudioSend().
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1311533009
Cr-Commit-Position: refs/heads/master@{#9915}
/external/webrtc/talk/session/media/channel.h
|
| 8006f0759246407261b95c792f4febf3906415dc |
|
08-Sep-2015 |
solenberg <solenberg@webrtc.org> |
Remove unused TypingMonitor class.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1327033002
Cr-Commit-Position: refs/heads/master@{#9884}
/external/webrtc/talk/session/media/channel.h
|
| c2ee2c86f905991a8cd05ee1f35bea105b41e4e0 |
|
08-Aug-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Refactor the relationship between BaseChannel and MediaChannel so that we send over all the parameters in one method call rather then having them broken up into multiple method calls. This should allow future refactorings of the WebRtcVideoEngine2 to not recreate configurations so many times, and have more simple code as well.
R=deadbeef@webrtc.org, pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1229283003 .
Cr-Commit-Position: refs/heads/master@{#9690}
/external/webrtc/talk/session/media/channel.h
|
| 0c0226408dc6f42abc2cd53cab2de02d3ee610d7 |
|
05-Aug-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Get rid of media_engine_ from BaseChannel; only VoiceChannel needs it.
BUG=webrtc:4690
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1270333002 .
Cr-Commit-Position: refs/heads/master@{#9679}
/external/webrtc/talk/session/media/channel.h
|
| 3b1e647b6a6f74d8e4392e012fe2262b3d2c4334 |
|
09-Jul-2015 |
pbos <pbos@webrtc.org> |
Remove media sinks from Channel.
Allows removing MediaRecorder which isn't in use apart from channel
unittests, along with it unittests for MediaRecorder that are flaky when
run in parallel can also go.
BUG=
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1219663008
Cr-Commit-Position: refs/heads/master@{#9558}
/external/webrtc/talk/session/media/channel.h
|
| af55ccc054de9b91f6e5f5059937a91c0c91ff30 |
|
21-May-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Add RtcpMuxPolicy support to PeerConnection.
BUG=4611
R=juberti@google.com
Review URL: https://webrtc-codereview.appspot.com/46169004
Cr-Commit-Position: refs/heads/master@{#9251}
/external/webrtc/talk/session/media/channel.h
|
| 4b60c73e74d62beff484b7f54d8f3267cb66274f |
|
07-May-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Hook up libjingle WebRtcVoiceEngine to Call API for combined A/V BWE.
BUG=4574,3109
R=pbos@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49269004
Cr-Commit-Position: refs/heads/master@{#9150}
/external/webrtc/talk/session/media/channel.h
|
| 7fb711f68312f61f392b3f33b950e97cb07da71f |
|
22-Apr-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Remove unused voice channel argument from cricket::VideoChannel ctor and corresponding field in class.
BUG=4574
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50769004
Cr-Commit-Position: refs/heads/master@{#9056}
/external/webrtc/talk/session/media/channel.h
|
| 592470b4ff39d60b52c745432ec131f05f3b6aa9 |
|
16-Mar-2015 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Remove a dependency of BaseChannel on WebRtcSession by having WebRtcSession push down new media descriptions to BaseChannel rather than having BaseChannel listen to the description changes from WebRtcSession.
This is a part of the big BUNDLE implementation at https://webrtc-codereview.appspot.com/45519004/
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47599004
Cr-Commit-Position: refs/heads/master@{#8743}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8743 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/channel.h
|
| 6ad507ac35ce638beddd7ac6687d006995637253 |
|
16-Mar-2015 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Refactor how the TransportChannels are set in the BaseChannel to rely lesson Session, so that in the future we can rely on Transport instead, and also be able to change Transports on the fly for BUNDLE.
Also, remove channel_name. It's no longer needed.
This is a part of the big BUNDLE implementation at https://webrtc-codereview.appspot.com/45519004/
R=decurtis@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43719004
Cr-Commit-Position: refs/heads/master@{#8741}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8741 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/channel.h
|
| 4eeef584a7d6f44f65c28352762775e1d1ca8a2b |
|
16-Mar-2015 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Remove a hacky dependency of BaseChannel on BaseSession by moving the handling of DTLS setup failure into a signal on BaseChannel rather than a method call on BaseSession.
This is a part of the big BUNDLE implementation at https://webrtc-codereview.appspot.com/45519004/
R=decurtis@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47589004
Cr-Commit-Position: refs/heads/master@{#8740}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8740 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/channel.h
|
| b4aac13810815f77b019f9db9d0300862c8313bc |
|
13-Mar-2015 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Cleanup SocketMonitor a little so that it can handle a change in transport channel. And cleanup some names and style and such as well.
This is a part of the big BUNDLE implementation at https://webrtc-codereview.appspot.com/45519004/
R=guoweis@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49399004
Cr-Commit-Position: refs/heads/master@{#8720}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8720 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/channel.h
|
| 990a00c30a2e87972506aac3a992a93ed3c8f79a |
|
13-Mar-2015 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Remove unused transport code.
This is a part of the big BUNDLE implementation at https://webrtc-codereview.appspot.com/45519004/
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49389004
Cr-Commit-Position: refs/heads/master@{#8719}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8719 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/channel.h
|
| 4f85288e71136671ae194fcdd730e2d0f0241db9 |
|
12-Mar-2015 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Socket options are only applied when first setting TransportChannelImpl.
Also fixed the issue when we have an TransportChannelImpl, the socket
option is not preserved.
Since this is a code path that will be modified by bundle (which Peter also has a test case already), we don't need a test case here.
BUG=4374
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42699004
Cr-Commit-Position: refs/heads/master@{#8702}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8702 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/channel.h
|
| 058b1f17ac43b1fe69a8c18aaa7999ba88733dfd |
|
04-Mar-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove GetReceiveBandwidthEstimatorStats.
Removes unnecessary non-standard stats that we don't really make use of.
BUG=
R=pthatcher@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47379004
Cr-Commit-Position: refs/heads/master@{#8588}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8588 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/channel.h
|
| 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/channel.h
|
| 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/channel.h
|
| 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/channel.h
|
| 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/session/media/channel.h
|
| 65b98d12c3b6b9ca0ded669d0a0811d2bb1712b3 |
|
08-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 72839629-> 72847605
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6854 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/channel.h
|
| 5b1ebacca2c29d73a5f3ab388b4b2a0a8e114c76 |
|
07-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 72820109-> 72822008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6850 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/channel.h
|
| d509678a4e5ba4c3047d80744e103b675d8c7c88 |
|
07-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 72819313-> 72820109
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6849 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/channel.h
|
| 94b996cc181b02d986f002230497bb2b28762060 |
|
07-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 72785516-> 72819313
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6848 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/channel.h
|
| 476efa203160463dafc2d5bf9b8a675df44d2df5 |
|
07-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 72785180-> 72785516
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6842 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/channel.h
|
| e0d03f13e4cfc5b822145597d40da9b8a8f95146 |
|
02-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 72443101-> 72446860
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6815 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/channel.h
|
| 6e203d50a3ecccc0524d36867761f80c12e0c56f |
|
02-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 72442050-> 72443101
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6814 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/channel.h
|
| 52148c2f74fe455ee126d24ec57a8bfc7cc87404 |
|
02-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 72430895-> 72442050
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6813 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/channel.h
|
| 7cb60ccae137d8db99e00ed2e073a00f110ccc57 |
|
02-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 72407428-> 72430895
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6812 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/channel.h
|
| 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/session/media/channel.h
|
| 75ce92086c955d7cba7d4fc9ffaba80097ce178c |
|
20-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 69600065-> 69617317
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6507 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/channel.h
|
| 1d66be22c8f929e1170f288472aac9d4b44b7a05 |
|
30-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 68203780-> 68206793
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6277 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/channel.h
|
| 6bfd6196ff1eac56a7f3f0191d91e06f6f9ce579 |
|
15-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 67052073-> 67134648
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6174 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/channel.h
|
| 3e924683d424f82b22ff1b61edaa560ac2675112 |
|
14-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 67037200-> 67043374
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6162 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/channel.h
|
| 5ee0f05d5fbb3fbe4862a76ab75d08ae846e6141 |
|
05-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 66138442-> 66236292
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6057 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/channel.h
|
| d43aa9de7a4a2b793e5ec59c86fb0b81e4052bb0 |
|
22-Feb-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle 61901702->61966318
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5596 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/channel.h
|
| b9a088b920d1ba16e0593698d4a613bb7bb5481f |
|
14-Feb-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 61538839.
TBR=mallinath
Review URL: https://webrtc-codereview.appspot.com/8669005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5548 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/channel.h
|
| 9cf037b83184374230c6825e4aa407cdafaba434 |
|
07-Feb-2014 |
sergeyu@chromium.org <sergeyu@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle to 61168196
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8139004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5502 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/channel.h
|
| 4b26e2eee3e3b2a0c22946372a38f7efa6cee146 |
|
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/channel.h
|
| aebb1ade9d760841f243e380fa22b7ecff2d3ecc |
|
14-Jan-2014 |
henrika@webrtc.org <henrika@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
pRevert 5371 "Revert 5367 "Update talk to 59410372.""
> Revert 5367 "Update talk to 59410372."
>
> > Update talk to 59410372.
> >
> > R=jiayl@webrtc.org, wu@webrtc.org
> >
> > Review URL: https://webrtc-codereview.appspot.com/6929004
>
> TBR=mallinath@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/6999004
TBR=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7109004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5381 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/channel.h
|
| 44461fa5cbecd556691b0ba963f95973f6abece1 |
|
13-Jan-2014 |
henrika@webrtc.org <henrika@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 5367 "Update talk to 59410372."
> Update talk to 59410372.
>
> R=jiayl@webrtc.org, wu@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/6929004
TBR=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5371 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/channel.h
|
| 0f3356e20b70416f13e12ef596da66f6c347eea7 |
|
11-Jan-2014 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 59410372.
R=jiayl@webrtc.org, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6929004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5367 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/channel.h
|
| a9890800e078105f21f0a21358ee59a0b3736af6 |
|
13-Dec-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 58127566 together with
https://webrtc-codereview.appspot.com/5309005/.
R=mallinath@webrtc.org, niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5277 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/channel.h
|
| 2018269dc3a1c1bb01c946583ca0750ae0db68e3 |
|
12-Dec-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 5274 "Update talk to 58113193 together with https://webrt..."
> Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/.
>
> R=mallinath@webrtc.org, niklas.enbom@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/5719004
TBR=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5729004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5275 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/channel.h
|
| a129b6cd132788a931b47da3370ae473673f320d |
|
12-Dec-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/.
R=mallinath@webrtc.org, niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5274 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/channel.h
|
| 07a6fbe83d901fc9b98579ab44e8c9632f038b36 |
|
04-Nov-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 56092586.
R=jiayl@webrtc.org, mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3359004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5078 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/channel.h
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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/channel.h
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| 1d1ffc9ad267d7e6e9ec9001052fd4abf29d7622 |
|
16-Oct-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 54898858.
TEST=try bots
TBR=mallinath
Review URL: https://webrtc-codereview.appspot.com/2414004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4979 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/channel.h
|
| 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/session/media/channel.h
|
| 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/session/media/channel.h
|
| 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/session/media/channel.h
|
| cadf9040cbb9e7bb1b73a95e43e7d228fe6b2bdb |
|
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/session/media/channel.h
|
| d64719d8954262fee94e7615422f3d027dc1ae6b |
|
01-Aug-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle to 50191337.
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1885005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4461 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/channel.h
|
| 1e09a711263dd105e6f7a03812250084c64e5fd8 |
|
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/channel.h
|
| 9de257d00f1f805af28f15fd814a8a84460028e5 |
|
17-Jul-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk folder to revision=49470012. Same as 375 in libjingle's google code repository.
TBR=wu@webrtc.org
BUG=N/A
Review URL: https://webrtc-codereview.appspot.com/1824004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4364 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/channel.h
|
| 28e20752806a492f5a6a5d343c02f9556f39b1cd |
|
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/channel.h
|